App-MHFS
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share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
David Reid - mackron@gmail.com
Website: https://miniaud.io
Documentation: https://miniaud.io/docs
GitHub: https://github.com/mackron/miniaudio
*/
/*
1. Introduction
===============
miniaudio is a single file library for audio playback and capture. To use it, do the following in
one .c file:
```c
#define MINIAUDIO_IMPLEMENTATION
#include "miniaudio.h"
```
You can do `#include "miniaudio.h"` in other parts of the program just like any other header.
miniaudio includes both low level and high level APIs. The low level API is good for those who want
to do all of their mixing themselves and only require a light weight interface to the underlying
audio device. The high level API is good for those who have complex mixing and effect requirements.
In miniaudio, objects are transparent structures. Unlike many other libraries, there are no handles
to opaque objects which means you need to allocate memory for objects yourself. In the examples
presented in this documentation you will often see objects declared on the stack. You need to be
careful when translating these examples to your own code so that you don't accidentally declare
your objects on the stack and then cause them to become invalid once the function returns. In
addition, you must ensure the memory address of your objects remain the same throughout their
lifetime. You therefore cannot be making copies of your objects.
A config/init pattern is used throughout the entire library. The idea is that you set up a config
object and pass that into the initialization routine. The advantage to this system is that the
config object can be initialized with logical defaults and new properties added to it without
breaking the API. The config object can be allocated on the stack and does not need to be
maintained after initialization of the corresponding object.
1.1. Low Level API
------------------
The low level API gives you access to the raw audio data of an audio device. It supports playback,
capture, full-duplex and loopback (WASAPI only). You can enumerate over devices to determine which
physical device(s) you want to connect to.
The low level API uses the concept of a "device" as the abstraction for physical devices. The idea
is that you choose a physical device to emit or capture audio from, and then move data to/from the
device when miniaudio tells you to. Data is delivered to and from devices asynchronously via a
callback which you specify when initializing the device.
When initializing the device you first need to configure it. The device configuration allows you to
specify things like the format of the data delivered via the callback, the size of the internal
buffer and the ID of the device you want to emit or capture audio from.
Once you have the device configuration set up you can initialize the device. When initializing a
device you need to allocate memory for the device object beforehand. This gives the application
complete control over how the memory is allocated. In the example below we initialize a playback
device on the stack, but you could allocate it on the heap if that suits your situation better.
```c
void data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
{
// In playback mode copy data to pOutput. In capture mode read data from pInput. In full-duplex mode, both
// pOutput and pInput will be valid and you can move data from pInput into pOutput. Never process more than
// frameCount frames.
}
int main()
{
ma_device_config config = ma_device_config_init(ma_device_type_playback);
config.playback.format = ma_format_f32; // Set to ma_format_unknown to use the device's native format.
config.playback.channels = 2; // Set to 0 to use the device's native channel count.
config.sampleRate = 48000; // Set to 0 to use the device's native sample rate.
config.dataCallback = data_callback; // This function will be called when miniaudio needs more data.
config.pUserData = pMyCustomData; // Can be accessed from the device object (device.pUserData).
ma_device device;
if (ma_device_init(NULL, &config, &device) != MA_SUCCESS) {
return -1; // Failed to initialize the device.
}
ma_device_start(&device); // The device is sleeping by default so you'll need to start it manually.
// Do something here. Probably your program's main loop.
ma_device_uninit(&device); // This will stop the device so no need to do that manually.
return 0;
}
```
In the example above, `data_callback()` is where audio data is written and read from the device.
The idea is in playback mode you cause sound to be emitted from the speakers by writing audio data
to the output buffer (`pOutput` in the example). In capture mode you read data from the input
buffer (`pInput`) to extract sound captured by the microphone. The `frameCount` parameter tells you
how many frames can be written to the output buffer and read from the input buffer. A "frame" is
one sample for each channel. For example, in a stereo stream (2 channels), one frame is 2
samples: one for the left, one for the right. The channel count is defined by the device config.
The size in bytes of an individual sample is defined by the sample format which is also specified
in the device config. Multi-channel audio data is always interleaved, which means the samples for
each frame are stored next to each other in memory. For example, in a stereo stream the first pair
of samples will be the left and right samples for the first frame, the second pair of samples will
be the left and right samples for the second frame, etc.
The configuration of the device is defined by the `ma_device_config` structure. The config object
is always initialized with `ma_device_config_init()`. It's important to always initialize the
config with this function as it initializes it with logical defaults and ensures your program
doesn't break when new members are added to the `ma_device_config` structure. The example above
uses a fairly simple and standard device configuration. The call to `ma_device_config_init()` takes
a single parameter, which is whether or not the device is a playback, capture, duplex or loopback
device (loopback devices are not supported on all backends). The `config.playback.format` member
sets the sample format which can be one of the following (all formats are native-endian):
+---------------+----------------------------------------+---------------------------+
| Symbol | Description | Range |
+---------------+----------------------------------------+---------------------------+
| ma_format_f32 | 32-bit floating point | [-1, 1] |
| ma_format_s16 | 16-bit signed integer | [-32768, 32767] |
| ma_format_s24 | 24-bit signed integer (tightly packed) | [-8388608, 8388607] |
| ma_format_s32 | 32-bit signed integer | [-2147483648, 2147483647] |
| ma_format_u8 | 8-bit unsigned integer | [0, 255] |
+---------------+----------------------------------------+---------------------------+
The `config.playback.channels` member sets the number of channels to use with the device. The
channel count cannot exceed MA_MAX_CHANNELS. The `config.sampleRate` member sets the sample rate
(which must be the same for both playback and capture in full-duplex configurations). This is
usually set to 44100 or 48000, but can be set to anything. It's recommended to keep this between
8000 and 384000, however.
Note that leaving the format, channel count and/or sample rate at their default values will result
in the internal device's native configuration being used which is useful if you want to avoid the
overhead of miniaudio's automatic data conversion.
In addition to the sample format, channel count and sample rate, the data callback and user data
pointer are also set via the config. The user data pointer is not passed into the callback as a
parameter, but is instead set to the `pUserData` member of `ma_device` which you can access
directly since all miniaudio structures are transparent.
Initializing the device is done with `ma_device_init()`. This will return a result code telling you
what went wrong, if anything. On success it will return `MA_SUCCESS`. After initialization is
complete the device will be in a stopped state. To start it, use `ma_device_start()`.
Uninitializing the device will stop it, which is what the example above does, but you can also stop
the device with `ma_device_stop()`. To resume the device simply call `ma_device_start()` again.
Note that it's important to never stop or start the device from inside the callback. This will
result in a deadlock. Instead you set a variable or signal an event indicating that the device
needs to stop and handle it in a different thread. The following APIs must never be called inside
the callback:
```c
ma_device_init()
ma_device_init_ex()
ma_device_uninit()
ma_device_start()
ma_device_stop()
```
You must never try uninitializing and reinitializing a device inside the callback. You must also
never try to stop and start it from inside the callback. There are a few other things you shouldn't
do in the callback depending on your requirements, however this isn't so much a thread-safety
thing, but rather a real-time processing thing which is beyond the scope of this introduction.
The example above demonstrates the initialization of a playback device, but it works exactly the
same for capture. All you need to do is change the device type from `ma_device_type_playback` to
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ma_engine* pEngine = malloc(sizeof(*pEngine));
```
The `ma_engine` API uses the same config/init pattern used all throughout miniaudio. To configure
an engine, you can fill out a `ma_engine_config` object and pass it into the first parameter of
`ma_engine_init()`:
```c
ma_result result;
ma_engine engine;
ma_engine_config engineConfig;
engineConfig = ma_engine_config_init();
engineConfig.pResourceManager = &myCustomResourceManager; // <-- Initialized as some earlier stage.
result = ma_engine_init(&engineConfig, &engine);
if (result != MA_SUCCESS) {
return result;
}
```
This creates an engine instance using a custom config. In this particular example it's showing how
you can specify a custom resource manager rather than having the engine initialize one internally.
This is particularly useful if you want to have multiple engine's share the same resource manager.
The engine must be uninitialized with `ma_engine_uninit()` when it's no longer needed.
By default the engine will be started, but nothing will be playing because no sounds have been
initialized. The easiest but least flexible way of playing a sound is like so:
```c
ma_engine_play_sound(&engine, "my_sound.wav", NULL);
```
This plays what miniaudio calls an "inline" sound. It plays the sound once, and then puts the
internal sound up for recycling. The last parameter is used to specify which sound group the sound
should be associated with which will be explained later. This particular way of playing a sound is
simple, but lacks flexibility and features. A more flexible way of playing a sound is to first
initialize a sound:
```c
ma_result result;
ma_sound sound;
result = ma_sound_init_from_file(&engine, "my_sound.wav", 0, NULL, NULL, &sound);
if (result != MA_SUCCESS) {
return result;
}
ma_sound_start(&sound);
```
This returns a `ma_sound` object which represents a single instance of the specified sound file. If
you want to play the same file multiple times simultaneously, you need to create one sound for each
instance.
Sounds should be uninitialized with `ma_sound_uninit()`.
Sounds are not started by default. Start a sound with `ma_sound_start()` and stop it with
`ma_sound_stop()`. When a sound is stopped, it is not rewound to the start. Use
`ma_sound_seek_to_pcm_frames(&sound, 0)` to seek back to the start of a sound. By default, starting
and stopping sounds happens immediately, but sometimes it might be convenient to schedule the sound
the be started and/or stopped at a specific time. This can be done with the following functions:
```c
ma_sound_set_start_time_in_pcm_frames()
ma_sound_set_start_time_in_milliseconds()
ma_sound_set_stop_time_in_pcm_frames()
ma_sound_set_stop_time_in_milliseconds()
```
The start/stop time needs to be specified based on the absolute timer which is controlled by the
engine. The current global time time in PCM frames can be retrieved with `ma_engine_get_time()`.
The engine's global time can be changed with `ma_engine_set_time()` for synchronization purposes if
required. Note that scheduling a start time still requires an explicit call to `ma_sound_start()`
before anything will play:
```c
ma_sound_set_start_time_in_pcm_frames(&sound, ma_engine_get_time(&engine) + (ma_engine_get_sample_rate(&engine) * 2);
ma_sound_start(&sound);
```
The third parameter of `ma_sound_init_from_file()` is a set of flags that control how the sound be
loaded and a few options on which features should be enabled for that sound. By default, the sound
is synchronously loaded fully into memory straight from the file system without any kind of
decoding. If you want to decode the sound before storing it in memory, you need to specify the
`MA_SOUND_FLAG_DECODE` flag. This is useful if you want to incur the cost of decoding at an earlier
stage, such as a loading stage. Without this option, decoding will happen dynamically at mixing
time which might be too expensive on the audio thread.
If you want to load the sound asynchronously, you can specify the `MA_SOUND_FLAG_ASYNC` flag. This
will result in `ma_sound_init_from_file()` returning quickly, but the sound will not start playing
until the sound has had some audio decoded.
The fourth parameter is a pointer to sound group. A sound group is used as a mechanism to organise
sounds into groups which have their own effect processing and volume control. An example is a game
which might have separate groups for sfx, voice and music. Each of these groups have their own
independent volume control. Use `ma_sound_group_init()` or `ma_sound_group_init_ex()` to initialize
a sound group.
Sounds and sound groups are nodes in the engine's node graph and can be plugged into any `ma_node`
API. This makes it possible to connect sounds and sound groups to effect nodes to produce complex
effect chains.
A sound can have it's volume changed with `ma_sound_set_volume()`. If you prefer decibel volume
control you can use `ma_volume_db_to_linear()` to convert from decibel representation to linear.
Panning and pitching is supported with `ma_sound_set_pan()` and `ma_sound_set_pitch()`. If you know
a sound will never have it's pitch changed with `ma_sound_set_pitch()` or via the doppler effect,
you can specify the `MA_SOUND_FLAG_NO_PITCH` flag when initializing the sound for an optimization.
By default, sounds and sound groups have spatialization enabled. If you don't ever want to
spatialize your sounds, initialize the sound with the `MA_SOUND_FLAG_NO_SPATIALIZATION` flag. The
spatialization model is fairly simple and is roughly on feature parity with OpenAL. HRTF and
environmental occlusion are not currently supported, but planned for the future. The supported
features include:
* Sound and listener positioning and orientation with cones
* Attenuation models: none, inverse, linear and exponential
* Doppler effect
Sounds can be faded in and out with `ma_sound_set_fade_in_pcm_frames()`.
To check if a sound is currently playing, you can use `ma_sound_is_playing()`. To check if a sound
is at the end, use `ma_sound_at_end()`. Looping of a sound can be controlled with
`ma_sound_set_looping()`. Use `ma_sound_is_looping()` to check whether or not the sound is looping.
2. Building
===========
miniaudio should work cleanly out of the box without the need to download or install any
dependencies. See below for platform-specific details.
2.1. Windows
------------
The Windows build should compile cleanly on all popular compilers without the need to configure any
include paths nor link to any libraries.
The UWP build may require linking to mmdevapi.lib if you get errors about an unresolved external
symbol for `ActivateAudioInterfaceAsync()`.
2.2. macOS and iOS
------------------
The macOS build should compile cleanly without the need to download any dependencies nor link to
any libraries or frameworks. The iOS build needs to be compiled as Objective-C and will need to
link the relevant frameworks but should compile cleanly out of the box with Xcode. Compiling
through the command line requires linking to `-lpthread` and `-lm`.
Due to the way miniaudio links to frameworks at runtime, your application may not pass Apple's
notarization process. To fix this there are two options. The first is to use the
`MA_NO_RUNTIME_LINKING` option, like so:
```c
#ifdef __APPLE__
#define MA_NO_RUNTIME_LINKING
#endif
#define MINIAUDIO_IMPLEMENTATION
#include "miniaudio.h"
```
This will require linking with `-framework CoreFoundation -framework CoreAudio -framework AudioUnit`.
Alternatively, if you would rather keep using runtime linking you can add the following to your
entitlements.xcent file:
```
<key>com.apple.security.cs.allow-dyld-environment-variables</key>
<true/>
<key>com.apple.security.cs.allow-unsigned-executable-memory</key>
<true/>
```
See this discussion for more info: https://github.com/mackron/miniaudio/issues/203.
2.3. Linux
----------
The Linux build only requires linking to `-ldl`, `-lpthread` and `-lm`. You do not need any
development packages. You may need to link with `-latomic` if you're compiling for 32-bit ARM.
2.4. BSD
--------
The BSD build only requires linking to `-lpthread` and `-lm`. NetBSD uses audio(4), OpenBSD uses
sndio and FreeBSD uses OSS. You may need to link with `-latomic` if you're compiling for 32-bit
ARM.
2.5. Android
------------
AAudio is the highest priority backend on Android. This should work out of the box without needing
any kind of compiler configuration. Support for AAudio starts with Android 8 which means older
versions will fall back to OpenSL|ES which requires API level 16+.
There have been reports that the OpenSL|ES backend fails to initialize on some Android based
devices due to `dlopen()` failing to open "libOpenSLES.so". If this happens on your platform
you'll need to disable run-time linking with `MA_NO_RUNTIME_LINKING` and link with -lOpenSLES.
2.6. Emscripten
---------------
The Emscripten build emits Web Audio JavaScript directly and should compile cleanly out of the box.
You cannot use `-std=c*` compiler flags, nor `-ansi`.
2.7. Build Options
------------------
`#define` these options before including miniaudio.h.
+----------------------------------+--------------------------------------------------------------------+
| Option | Description |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_WASAPI | Disables the WASAPI backend. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_DSOUND | Disables the DirectSound backend. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_WINMM | Disables the WinMM backend. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_ALSA | Disables the ALSA backend. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_PULSEAUDIO | Disables the PulseAudio backend. |
+----------------------------------+--------------------------------------------------------------------+
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_WAV | Disables the built-in WAV decoder and encoder. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_FLAC | Disables the built-in FLAC decoder. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_MP3 | Disables the built-in MP3 decoder. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_DEVICE_IO | Disables playback and recording. This will disable `ma_context` |
| | and `ma_device` APIs. This is useful if you only want to use |
| | miniaudio's data conversion and/or decoding APIs. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_THREADING | Disables the `ma_thread`, `ma_mutex`, `ma_semaphore` and |
| | `ma_event` APIs. This option is useful if you only need to use |
| | miniaudio for data conversion, decoding and/or encoding. Some |
| | families of APIsrequire threading which means the following |
| | options must also be set: |
| | |
| | ``` |
| | MA_NO_DEVICE_IO |
| | ``` |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_GENERATION | Disables generation APIs such a `ma_waveform` and `ma_noise`. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_SSE2 | Disables SSE2 optimizations. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_AVX2 | Disables AVX2 optimizations. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_NEON | Disables NEON optimizations. |
+----------------------------------+--------------------------------------------------------------------+
| MA_NO_RUNTIME_LINKING | Disables runtime linking. This is useful for passing Apple's |
| | notarization process. When enabling this, you may need to avoid |
| | using `-std=c89` or `-std=c99` on Linux builds or else you may end |
| | up with compilation errors due to conflicts with `timespec` and |
| | `timeval` data types. |
| | |
| | You may need to enable this if your target platform does not allow |
| | runtime linking via `dlopen()`. |
+----------------------------------+--------------------------------------------------------------------+
| MA_DEBUG_OUTPUT | Enable `printf()` output of debug logs (`MA_LOG_LEVEL_DEBUG`). |
+----------------------------------+--------------------------------------------------------------------+
| MA_COINIT_VALUE | Windows only. The value to pass to internal calls to |
| | `CoInitializeEx()`. Defaults to `COINIT_MULTITHREADED`. |
+----------------------------------+--------------------------------------------------------------------+
| MA_API | Controls how public APIs should be decorated. Default is `extern`. |
+----------------------------------+--------------------------------------------------------------------+
3. Definitions
==============
This section defines common terms used throughout miniaudio. Unfortunately there is often ambiguity
in the use of terms throughout the audio space, so this section is intended to clarify how miniaudio
uses each term.
3.1. Sample
-----------
A sample is a single unit of audio data. If the sample format is f32, then one sample is one 32-bit
floating point number.
3.2. Frame / PCM Frame
----------------------
A frame is a group of samples equal to the number of channels. For a stereo stream a frame is 2
samples, a mono frame is 1 sample, a 5.1 surround sound frame is 6 samples, etc. The terms "frame"
and "PCM frame" are the same thing in miniaudio. Note that this is different to a compressed frame.
If ever miniaudio needs to refer to a compressed frame, such as a FLAC frame, it will always
clarify what it's referring to with something like "FLAC frame".
3.3. Channel
------------
A stream of monaural audio that is emitted from an individual speaker in a speaker system, or
received from an individual microphone in a microphone system. A stereo stream has two channels (a
left channel, and a right channel), a 5.1 surround sound system has 6 channels, etc. Some audio
systems refer to a channel as a complex audio stream that's mixed with other channels to produce
the final mix - this is completely different to miniaudio's use of the term "channel" and should
not be confused.
3.4. Sample Rate
----------------
The sample rate in miniaudio is always expressed in Hz, such as 44100, 48000, etc. It's the number
of PCM frames that are processed per second.
3.5. Formats
------------
Throughout miniaudio you will see references to different sample formats:
+---------------+----------------------------------------+---------------------------+
| Symbol | Description | Range |
+---------------+----------------------------------------+---------------------------+
| ma_format_f32 | 32-bit floating point | [-1, 1] |
| ma_format_s16 | 16-bit signed integer | [-32768, 32767] |
| ma_format_s24 | 24-bit signed integer (tightly packed) | [-8388608, 8388607] |
| ma_format_s32 | 32-bit signed integer | [-2147483648, 2147483647] |
| ma_format_u8 | 8-bit unsigned integer | [0, 255] |
+---------------+----------------------------------------+---------------------------+
All formats are native-endian.
4. Data Sources
===============
The data source abstraction in miniaudio is used for retrieving audio data from some source. A few
examples include `ma_decoder`, `ma_noise` and `ma_waveform`. You will need to be familiar with data
sources in order to make sense of some of the higher level concepts in miniaudio.
The `ma_data_source` API is a generic interface for reading from a data source. Any object that
implements the data source interface can be plugged into any `ma_data_source` function.
To read data from a data source:
```c
ma_result result;
ma_uint64 framesRead;
result = ma_data_source_read_pcm_frames(pDataSource, pFramesOut, frameCount, &framesRead, loop);
if (result != MA_SUCCESS) {
return result; // Failed to read data from the data source.
}
```
If you don't need the number of frames that were successfully read you can pass in `NULL` to the
`pFramesRead` parameter. If this returns a value less than the number of frames requested it means
the end of the file has been reached. `MA_AT_END` will be returned only when the number of frames
read is 0.
When calling any data source function, with the exception of `ma_data_source_init()` and
`ma_data_source_uninit()`, you can pass in any object that implements a data source. For example,
you could plug in a decoder like so:
```c
ma_result result;
ma_uint64 framesRead;
ma_decoder decoder; // <-- This would be initialized with `ma_decoder_init_*()`.
result = ma_data_source_read_pcm_frames(&decoder, pFramesOut, frameCount, &framesRead, loop);
if (result != MA_SUCCESS) {
return result; // Failed to read data from the decoder.
}
```
If you want to seek forward you can pass in `NULL` to the `pFramesOut` parameter. Alternatively you
can use `ma_data_source_seek_pcm_frames()`.
To seek to a specific PCM frame:
```c
result = ma_data_source_seek_to_pcm_frame(pDataSource, frameIndex);
if (result != MA_SUCCESS) {
return result; // Failed to seek to PCM frame.
}
```
You can retrieve the total length of a data source in PCM frames, but note that some data sources
may not have the notion of a length, such as noise and waveforms, and others may just not have a
way of determining the length such as some decoders. To retrieve the length:
```c
ma_uint64 length;
result = ma_data_source_get_length_in_pcm_frames(pDataSource, &length);
if (result != MA_SUCCESS) {
return result; // Failed to retrieve the length.
}
```
Care should be taken when retrieving the length of a data source where the underlying decoder is
pulling data from a data stream with an undefined length, such as internet radio or some kind of
broadcast. If you do this, `ma_data_source_get_length_in_pcm_frames()` may never return.
The current position of the cursor in PCM frames can also be retrieved:
```c
ma_uint64 cursor;
result = ma_data_source_get_cursor_in_pcm_frames(pDataSource, &cursor);
if (result != MA_SUCCESS) {
return result; // Failed to retrieve the cursor.
}
```
You will often need to know the data format that will be returned after reading. This can be
retrieved like so:
```c
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
ma_channel channelMap[MA_MAX_CHANNELS];
result = ma_data_source_get_data_format(pDataSource, &format, &channels, &sampleRate, channelMap, MA_MAX_CHANNELS);
if (result != MA_SUCCESS) {
return result; // Failed to retrieve data format.
}
```
If you do not need a specific data format property, just pass in NULL to the respective parameter.
There may be cases where you want to implement something like a sound bank where you only want to
read data within a certain range of the underlying data. To do this you can use a range:
```c
result = ma_data_source_set_range_in_pcm_frames(pDataSource, rangeBegInFrames, rangeEndInFrames);
if (result != MA_SUCCESS) {
return result; // Failed to set the range.
}
```
This is useful if you have a sound bank where many sounds are stored in the same file and you want
the data source to only play one of those sub-sounds.
Custom loop points can also be used with data sources. By default, data sources will loop after
they reach the end of the data source, but if you need to loop at a specific location, you can do
the following:
```c
result = ma_data_set_loop_point_in_pcm_frames(pDataSource, loopBegInFrames, loopEndInFrames);
if (result != MA_SUCCESS) {
return result; // Failed to set the loop point.
}
```
The loop point is relative to the current range.
It's sometimes useful to chain data sources together so that a seamless transition can be achieved.
To do this, you can use chaining:
```c
ma_decoder decoder1;
ma_decoder decoder2;
// ... initialize decoders with ma_decoder_init_*() ...
result = ma_data_source_set_next(&decoder1, &decoder2);
if (result != MA_SUCCESS) {
return result; // Failed to set the next data source.
}
result = ma_data_source_read_pcm_frames(&decoder1, pFramesOut, frameCount, pFramesRead, MA_FALSE);
if (result != MA_SUCCESS) {
return result; // Failed to read from the decoder.
}
```
In the example above we're using decoders. When reading from a chain, you always want to read from
the top level data source in the chain. In the example above, `decoder1` is the top level data
source in the chain. When `decoder1` reaches the end, `decoder2` will start seamlessly without any
gaps.
Note that the `loop` parameter is set to false in the example above. When this is set to true, only
the current data source will be looped. You can loop the entire chain by linking in a loop like so:
```c
ma_data_source_set_next(&decoder1, &decoder2); // decoder1 -> decoder2
ma_data_source_set_next(&decoder2, &decoder1); // decoder2 -> decoder1 (loop back to the start).
```
Note that setting up chaining is not thread safe, so care needs to be taken if you're dynamically
changing links while the audio thread is in the middle of reading.
Do not use `ma_decoder_seek_to_pcm_frame()` as a means to reuse a data source to play multiple
instances of the same sound simultaneously. Instead, initialize multiple data sources for each
instance. This can be extremely inefficient depending on the data source and can result in
glitching due to subtle changes to the state of internal filters.
4.1. Custom Data Sources
------------------------
You can implement a custom data source by implementing the functions in `ma_data_source_vtable`.
Your custom object must have `ma_data_source_base` as it's first member:
```c
struct my_data_source
{
ma_data_source_base base;
...
};
```
In your initialization routine, you need to call `ma_data_source_init()` in order to set up the
base object (`ma_data_source_base`):
```c
static ma_result my_data_source_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
// Read data here. Output in the same format returned by my_data_source_get_data_format().
}
static ma_result my_data_source_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
// Seek to a specific PCM frame here. Return MA_NOT_IMPLEMENTED if seeking is not supported.
}
static ma_result my_data_source_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
// Return the format of the data here.
}
static ma_result my_data_source_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
// Retrieve the current position of the cursor here. Return MA_NOT_IMPLEMENTED and set *pCursor to 0 if there is no notion of a cursor.
}
static ma_result my_data_source_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
{
// Retrieve the length in PCM frames here. Return MA_NOT_IMPLEMENTED and set *pLength to 0 if there is no notion of a length or if the length is unknown.
}
static g_my_data_source_vtable =
{
my_data_source_read,
my_data_source_seek,
my_data_source_get_data_format,
my_data_source_get_cursor,
my_data_source_get_length
};
ma_result my_data_source_init(my_data_source* pMyDataSource)
{
ma_result result;
ma_data_source_config baseConfig;
baseConfig = ma_data_source_config_init();
baseConfig.vtable = &g_my_data_source_vtable;
result = ma_data_source_init(&baseConfig, &pMyDataSource->base);
if (result != MA_SUCCESS) {
return result;
}
// ... do the initialization of your custom data source here ...
return MA_SUCCESS;
}
void my_data_source_uninit(my_data_source* pMyDataSource)
{
// ... do the uninitialization of your custom data source here ...
// You must uninitialize the base data source.
ma_data_source_uninit(&pMyDataSource->base);
}
```
Note that `ma_data_source_init()` and `ma_data_source_uninit()` are never called directly outside
of the custom data source. It's up to the custom data source itself to call these within their own
init/uninit functions.
5. Engine
=========
The `ma_engine` API is a high level API for managing and mixing sounds and effect processing. The
`ma_engine` object encapsulates a resource manager and a node graph, both of which will be
explained in more detail later.
Sounds are called `ma_sound` and are created from an engine. Sounds can be associated with a mixing
group called `ma_sound_group` which are also created from the engine. Both `ma_sound` and
`ma_sound_group` objects are nodes within the engine's node graph.
When the engine is initialized, it will normally create a device internally. If you would rather
manage the device yourself, you can do so and just pass a pointer to it via the engine config when
you initialize the engine. You can also just use the engine without a device, which again can be
configured via the engine config.
The most basic way to initialize the engine is with a default config, like so:
```c
ma_result result;
ma_engine engine;
result = ma_engine_init(NULL, &engine);
if (result != MA_SUCCESS) {
return result; // Failed to initialize the engine.
}
```
This will result in the engine initializing a playback device using the operating system's default
device. This will be sufficient for many use cases, but if you need more flexibility you'll want to
configure the engine with an engine config:
```c
ma_result result;
ma_engine engine;
ma_engine_config engineConfig;
engineConfig = ma_engine_config_init();
engineConfig.pPlaybackDevice = &myDevice;
result = ma_engine_init(&engineConfig, &engine);
if (result != MA_SUCCESS) {
return result; // Failed to initialize the engine.
}
```
In the example above we're passing in a pre-initialized device. Since the caller is the one in
control of the device's data callback, it's their responsibility to manually call
`ma_engine_read_pcm_frames()` from inside their data callback:
```c
void playback_data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
{
ma_engine_read_pcm_frames(&g_Engine, pOutput, frameCount, NULL);
}
```
You can also use the engine independent of a device entirely:
```c
ma_result result;
ma_engine engine;
ma_engine_config engineConfig;
engineConfig = ma_engine_config_init();
engineConfig.noDevice = MA_TRUE;
engineConfig.channels = 2; // Must be set when not using a device.
engineConfig.sampleRate = 48000; // Must be set when not using a device.
result = ma_engine_init(&engineConfig, &engine);
if (result != MA_SUCCESS) {
return result; // Failed to initialize the engine.
}
```
Note that when you're not using a device, you must set the channel count and sample rate in the
config or else miniaudio won't know what to use (miniaudio will use the device to determine this
normally). When not using a device, you need to use `ma_engine_read_pcm_frames()` to process audio
data from the engine. This kind of setup is useful if you want to do something like offline
processing.
When a sound is loaded it goes through a resource manager. By default the engine will initialize a
resource manager internally, but you can also specify a pre-initialized resource manager:
```c
ma_result result;
ma_engine engine1;
ma_engine engine2;
ma_engine_config engineConfig;
engineConfig = ma_engine_config_init();
engineConfig.pResourceManager = &myResourceManager;
ma_engine_init(&engineConfig, &engine1);
ma_engine_init(&engineConfig, &engine2);
```
In this example we are initializing two engines, both of which are sharing the same resource
manager. This is especially useful for saving memory when loading the same file across multiple
engines. If you were not to use a shared resource manager, each engine instance would use their own
which would result in any sounds that are used between both engine's being loaded twice. By using
a shared resource manager, it would only be loaded once. Using multiple engine's is useful when you
need to output to multiple playback devices, such as in a local multiplayer game where each player
is using their own set of headphones.
By default an engine will be in a started state. To make it so the engine is not automatically
started you can configure it as such:
```c
engineConfig.noAutoStart = MA_TRUE;
// The engine will need to be started manually.
ma_engine_start(&engine);
// Later on the engine can be stopped with ma_engine_stop().
ma_engine_stop(&engine);
```
The concept of starting or stopping an engine is only relevant when using the engine with a
device. Attempting to start or stop an engine that is not associated with a device will result in
`MA_INVALID_OPERATION`.
The master volume of the engine can be controlled with `ma_engine_set_volume()` which takes a
linear scale, with 0 resulting in silence and anything above 1 resulting in amplification. If you
prefer decibel based volume control, use `ma_volume_db_to_linear()` to convert from dB to linear.
When a sound is spatialized, it is done so relative to a listener. An engine can be configured to
have multiple listeners which can be configured via the config:
```c
engineConfig.listenerCount = 2;
```
The maximum number of listeners is restricted to `MA_ENGINE_MAX_LISTENERS`. By default, when a
sound is spatialized, it will be done so relative to the closest listener. You can also pin a sound
to a specific listener which will be explained later. Listener's have a position, direction, cone,
and velocity (for doppler effect). A listener is referenced by an index, the meaning of which is up
to the caller (the index is 0 based and cannot go beyond the listener count, minus 1). The
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ma_sound_set_cone(&sound, innerAngleInRadians, outerAngleInRadians, outerGain);
```
The velocity of a sound is used for doppler effect and can be set as such:
```c
ma_sound_set_velocity(&sound, velocityX, velocityY, velocityZ);
```
The engine supports different attenuation models which can be configured on a per-sound basis. By
default the attenuation model is set to `ma_attenuation_model_inverse` which is the equivalent to
OpenAL's `AL_INVERSE_DISTANCE_CLAMPED`. Configure the attenuation model like so:
```c
ma_sound_set_attenuation_model(&sound, ma_attenuation_model_inverse);
```
The supported attenuation models include the following:
+----------------------------------+----------------------------------------------+
| ma_attenuation_model_none | No distance attenuation. |
+----------------------------------+----------------------------------------------+
| ma_attenuation_model_inverse | Equivalent to `AL_INVERSE_DISTANCE_CLAMPED`. |
+----------------------------------+----------------------------------------------+
| ma_attenuation_model_linear | Linear attenuation. |
+----------------------------------+----------------------------------------------+
| ma_attenuation_model_exponential | Exponential attenuation. |
+----------------------------------+----------------------------------------------+
To control how quickly a sound rolls off as it moves away from the listener, you need to configure
the rolloff:
```c
ma_sound_set_rolloff(&sound, rolloff);
```
You can control the minimum and maximum gain to apply from spatialization:
```c
ma_sound_set_min_gain(&sound, minGain);
ma_sound_set_max_gain(&sound, maxGain);
```
Likewise, in the calculation of attenuation, you can control the minimum and maximum distances for
the attenuation calculation. This is useful if you want to ensure sounds don't drop below a certain
volume after the listener moves further away and to have sounds play a maximum volume when the
listener is within a certain distance:
```c
ma_sound_set_min_distance(&sound, minDistance);
ma_sound_set_max_distance(&sound, maxDistance);
```
The engine's spatialization system supports doppler effect. The doppler factor can be configure on
a per-sound basis like so:
```c
ma_sound_set_doppler_factor(&sound, dopplerFactor);
```
You can fade sounds in and out with `ma_sound_set_fade_in_pcm_frames()` and
`ma_sound_set_fade_in_milliseconds()`. Set the volume to -1 to use the current volume as the
starting volume:
```c
// Fade in over 1 second.
ma_sound_set_fade_in_milliseconds(&sound, 0, 1, 1000);
// ... sometime later ...
// Fade out over 1 second, starting from the current volume.
ma_sound_set_fade_in_milliseconds(&sound, -1, 0, 1000);
```
By default sounds will start immediately, but sometimes for timing and synchronization purposes it
can be useful to schedule a sound to start or stop:
```c
// Start the sound in 1 second from now.
ma_sound_set_start_time_in_pcm_frames(&sound, ma_engine_get_time(&engine) + (ma_engine_get_sample_rate(&engine) * 1));
// Stop the sound in 2 seconds from now.
ma_sound_set_stop_time_in_pcm_frames(&sound, ma_engine_get_time(&engine) + (ma_engine_get_sample_rate(&engine) * 2));
```
Note that scheduling a start time still requires an explicit call to `ma_sound_start()` before
anything will play.
The time is specified in global time which is controlled by the engine. You can get the engine's
current time with `ma_engine_get_time()`. The engine's global time is incremented automatically as
audio data is read, but it can be reset with `ma_engine_set_time()` in case it needs to be
resynchronized for some reason.
To determine whether or not a sound is currently playing, use `ma_sound_is_playing()`. This will
take the scheduled start and stop times into account.
Whether or not a sound should loop can be controlled with `ma_sound_set_looping()`. Sounds will not
be looping by default. Use `ma_sound_is_looping()` to determine whether or not a sound is looping.
Use `ma_sound_at_end()` to determine whether or not a sound is currently at the end. For a looping
sound this should never return true.
Internally a sound wraps around a data source. Some APIs exist to control the underlying data
source, mainly for convenience:
```c
ma_sound_seek_to_pcm_frame(&sound, frameIndex);
ma_sound_get_data_format(&sound, &format, &channels, &sampleRate, pChannelMap, channelMapCapacity);
ma_sound_get_cursor_in_pcm_frames(&sound, &cursor);
ma_sound_get_length_in_pcm_frames(&sound, &length);
```
Sound groups have the same API as sounds, only they are called `ma_sound_group`, and since they do
not have any notion of a data source, anything relating to a data source is unavailable.
Internally, sound data is loaded via the `ma_decoder` API which means by default in only supports
file formats that have built-in support in miniaudio. You can extend this to support any kind of
file format through the use of custom decoders. To do this you'll need to use a self-managed
resource manager and configure it appropriately. See the "Resource Management" section below for
details on how to set this up.
6. Resource Management
======================
Many programs will want to manage sound resources for things such as reference counting and
streaming. This is supported by miniaudio via the `ma_resource_manager` API.
The resource manager is mainly responsible for the following:
* Loading of sound files into memory with reference counting.
* Streaming of sound data
When loading a sound file, the resource manager will give you back a `ma_data_source` compatible
object called `ma_resource_manager_data_source`. This object can be passed into any
`ma_data_source` API which is how you can read and seek audio data. When loading a sound file, you
specify whether or not you want the sound to be fully loaded into memory (and optionally
pre-decoded) or streamed. When loading into memory, you can also specify whether or not you want
the data to be loaded asynchronously.
The example below is how you can initialize a resource manager using it's default configuration:
```c
ma_resource_manager_config config;
ma_resource_manager resourceManager;
config = ma_resource_manager_config_init();
result = ma_resource_manager_init(&config, &resourceManager);
if (result != MA_SUCCESS) {
ma_device_uninit(&device);
printf("Failed to initialize the resource manager.");
return -1;
}
```
You can configure the format, channels and sample rate of the decoded audio data. By default it
will use the file's native data format, but you can configure it to use a consistent format. This
is useful for offloading the cost of data conversion to load time rather than dynamically
converting at mixing time. To do this, you configure the decoded format, channels and sample rate
like the code below:
```c
config = ma_resource_manager_config_init();
config.decodedFormat = device.playback.format;
config.decodedChannels = device.playback.channels;
config.decodedSampleRate = device.sampleRate;
```
In the code above, the resource manager will be configured so that any decoded audio data will be
pre-converted at load time to the device's native data format. If instead you used defaults and
the data format of the file did not match the device's data format, you would need to convert the
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ma_job job;
ma_result result = ma_resource_manager_next_job(pMyResourceManager, &job);
if (result != MA_SUCCESS) {
if (result == MA_NOT_DATA_AVAILABLE) {
// No jobs are available. Keep going. Will only get this if the resource manager was initialized
// with MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING.
continue;
} else if (result == MA_CANCELLED) {
// MA_JOB_TYPE_QUIT was posted. Exit.
break;
} else {
// Some other error occurred.
break;
}
}
ma_job_process(&job);
}
}
```
In the example above, the `MA_JOB_TYPE_QUIT` event is the used as the termination
indicator, but you can use whatever you would like to terminate the thread. The call to
`ma_resource_manager_next_job()` is blocking by default, but can be configured to be non-blocking
by initializing the resource manager with the `MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING` configuration
flag. Note that the `MA_JOB_TYPE_QUIT` will never be removed from the job queue. This
is to give every thread the opportunity to catch the event and terminate naturally.
When loading a file, it's sometimes convenient to be able to customize how files are opened and
read instead of using standard `fopen()`, `fclose()`, etc. which is what miniaudio will use by
default. This can be done by setting `pVFS` member of the resource manager's config:
```c
// Initialize your custom VFS object. See documentation for VFS for information on how to do this.
my_custom_vfs vfs = my_custom_vfs_init();
config = ma_resource_manager_config_init();
config.pVFS = &vfs;
```
This is particularly useful in programs like games where you want to read straight from an archive
rather than the normal file system. If you do not specify a custom VFS, the resource manager will
use the operating system's normal file operations. This is default.
To load a sound file and create a data source, call `ma_resource_manager_data_source_init()`. When
loading a sound you need to specify the file path and options for how the sounds should be loaded.
By default a sound will be loaded synchronously. The returned data source is owned by the caller
which means the caller is responsible for the allocation and freeing of the data source. Below is
an example for initializing a data source:
```c
ma_resource_manager_data_source dataSource;
ma_result result = ma_resource_manager_data_source_init(pResourceManager, pFilePath, flags, &dataSource);
if (result != MA_SUCCESS) {
// Error.
}
// ...
// A ma_resource_manager_data_source object is compatible with the `ma_data_source` API. To read data, just call
// the `ma_data_source_read_pcm_frames()` like you would with any normal data source.
result = ma_data_source_read_pcm_frames(&dataSource, pDecodedData, frameCount, &framesRead);
if (result != MA_SUCCESS) {
// Failed to read PCM frames.
}
// ...
ma_resource_manager_data_source_uninit(pResourceManager, &dataSource);
```
The `flags` parameter specifies how you want to perform loading of the sound file. It can be a
combination of the following flags:
```
MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM
MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE
MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC
MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT
```
When no flags are specified (set to 0), the sound will be fully loaded into memory, but not
decoded, meaning the raw file data will be stored in memory, and then dynamically decoded when
`ma_data_source_read_pcm_frames()` is called. To instead decode the audio data before storing it in
memory, use the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE` flag. By default, the sound file will
be loaded synchronously, meaning `ma_resource_manager_data_source_init()` will only return after
the entire file has been loaded. This is good for simplicity, but can be prohibitively slow. You
can instead load the sound asynchronously using the `MA_RESOURCE_MANAGER_DATA_SOURCE_ASYNC` flag.
This will result in `ma_resource_manager_data_source_init()` returning quickly, but no data will be
returned by `ma_data_source_read_pcm_frames()` until some data is available. When no data is
available because the asynchronous decoding hasn't caught up, `MA_BUSY` will be returned by
`ma_data_source_read_pcm_frames()`.
For large sounds, it's often prohibitive to store the entire file in memory. To mitigate this, you
can instead stream audio data which you can do by specifying the
`MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag. When streaming, data will be decoded in 1
second pages. When a new page needs to be decoded, a job will be posted to the job queue and then
subsequently processed in a job thread.
For in-memory sounds, reference counting is used to ensure the data is loaded only once. This means
multiple calls to `ma_resource_manager_data_source_init()` with the same file path will result in
the file data only being loaded once. Each call to `ma_resource_manager_data_source_init()` must be
matched up with a call to `ma_resource_manager_data_source_uninit()`. Sometimes it can be useful
for a program to register self-managed raw audio data and associate it with a file path. Use the
`ma_resource_manager_register_*()` and `ma_resource_manager_unregister_*()` APIs to do this.
`ma_resource_manager_register_decoded_data()` is used to associate a pointer to raw, self-managed
decoded audio data in the specified data format with the specified name. Likewise,
`ma_resource_manager_register_encoded_data()` is used to associate a pointer to raw self-managed
encoded audio data (the raw file data) with the specified name. Note that these names need not be
actual file paths. When `ma_resource_manager_data_source_init()` is called (without the
`MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag), the resource manager will look for these
explicitly registered data buffers and, if found, will use it as the backing data for the data
source. Note that the resource manager does *not* make a copy of this data so it is up to the
caller to ensure the pointer stays valid for it's lifetime. Use
`ma_resource_manager_unregister_data()` to unregister the self-managed data. You can also use
`ma_resource_manager_register_file()` and `ma_resource_manager_unregister_file()` to register and
unregister a file. It does not make sense to use the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM`
flag with a self-managed data pointer.
6.1. Asynchronous Loading and Synchronization
---------------------------------------------
When loading asynchronously, it can be useful to poll whether or not loading has finished. Use
`ma_resource_manager_data_source_result()` to determine this. For in-memory sounds, this will
return `MA_SUCCESS` when the file has been *entirely* decoded. If the sound is still being decoded,
`MA_BUSY` will be returned. Otherwise, some other error code will be returned if the sound failed
to load. For streaming data sources, `MA_SUCCESS` will be returned when the first page has been
decoded and the sound is ready to be played. If the first page is still being decoded, `MA_BUSY`
will be returned. Otherwise, some other error code will be returned if the sound failed to load.
In addition to polling, you can also use a simple synchronization object called a "fence" to wait
for asynchronously loaded sounds to finish. This is called `ma_fence`. The advantage to using a
fence is that it can be used to wait for a group of sounds to finish loading rather than waiting
for sounds on an individual basis. There are two stages to loading a sound:
* Initialization of the internal decoder; and
* Completion of decoding of the file (the file is fully decoded)
You can specify separate fences for each of the different stages. Waiting for the initialization
of the internal decoder is important for when you need to know the sample format, channels and
sample rate of the file.
The example below shows how you could use a fence when loading a number of sounds:
```c
// This fence will be released when all sounds are finished loading entirely.
ma_fence fence;
ma_fence_init(&fence);
// This will be passed into the initialization routine for each sound.
ma_resource_manager_pipeline_notifications notifications = ma_resource_manager_pipeline_notifications_init();
notifications.done.pFence = &fence;
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void my_notification_callback(ma_async_notification* pNotification)
{
my_notification* pMyNotification = (my_notification*)pNotification;
// Do something in response to the sound finishing loading.
}
...
my_notification myCallback;
myCallback.cb.onSignal = my_notification_callback;
myCallback.pMyData = pMyData;
ma_resource_manager_pipeline_notifications notifications = ma_resource_manager_pipeline_notifications_init();
notifications.done.pNotification = &myCallback;
ma_resource_manager_data_source_init(pResourceManager, "my_sound.wav", flags, ¬ifications, &mySound);
```
In the example above we just extend the `ma_async_notification_callbacks` object and pass an
instantiation into the `ma_resource_manager_pipeline_notifications` in the same way as we did with
the fence, only we set `pNotification` instead of `pFence`. You can set both of these at the same
time and they should both work as expected. If using the `pNotification` system, you need to ensure
your `ma_async_notification_callbacks` object stays valid.
6.2. Resource Manager Implementation Details
--------------------------------------------
Resources are managed in two main ways:
* By storing the entire sound inside an in-memory buffer (referred to as a data buffer)
* By streaming audio data on the fly (referred to as a data stream)
A resource managed data source (`ma_resource_manager_data_source`) encapsulates a data buffer or
data stream, depending on whether or not the data source was initialized with the
`MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag. If so, it will make use of a
`ma_resource_manager_data_stream` object. Otherwise it will use a `ma_resource_manager_data_buffer`
object. Both of these objects are data sources which means they can be used with any
`ma_data_source_*()` API.
Another major feature of the resource manager is the ability to asynchronously decode audio files.
This relieves the audio thread of time-consuming decoding which can negatively affect scalability
due to the audio thread needing to complete it's work extremely quickly to avoid glitching.
Asynchronous decoding is achieved through a job system. There is a central multi-producer,
multi-consumer, fixed-capacity job queue. When some asynchronous work needs to be done, a job is
posted to the queue which is then read by a job thread. The number of job threads can be
configured for improved scalability, and job threads can all run in parallel without needing to
worry about the order of execution (how this is achieved is explained below).
When a sound is being loaded asynchronously, playback can begin before the sound has been fully
decoded. This enables the application to start playback of the sound quickly, while at the same
time allowing to resource manager to keep loading in the background. Since there may be less
threads than the number of sounds being loaded at a given time, a simple scheduling system is used
to keep decoding time balanced and fair. The resource manager solves this by splitting decoding
into chunks called pages. By default, each page is 1 second long. When a page has been decoded, a
new job will be posted to start decoding the next page. By dividing up decoding into pages, an
individual sound shouldn't ever delay every other sound from having their first page decoded. Of
course, when loading many sounds at the same time, there will always be an amount of time required
to process jobs in the queue so in heavy load situations there will still be some delay. To
determine if a data source is ready to have some frames read, use
`ma_resource_manager_data_source_get_available_frames()`. This will return the number of frames
available starting from the current position.
6.2.1. Job Queue
----------------
The resource manager uses a job queue which is multi-producer, multi-consumer, and fixed-capacity.
This job queue is not currently lock-free, and instead uses a spinlock to achieve thread-safety.
Only a fixed number of jobs can be allocated and inserted into the queue which is done through a
lock-free data structure for allocating an index into a fixed sized array, with reference counting
for mitigation of the ABA problem. The reference count is 32-bit.
For many types of jobs it's important that they execute in a specific order. In these cases, jobs
are executed serially. For the resource manager, serial execution of jobs is only required on a
per-object basis (per data buffer or per data stream). Each of these objects stores an execution
counter. When a job is posted it is associated with an execution counter. When the job is
processed, it checks if the execution counter of the job equals the execution counter of the
owning object and if so, processes the job. If the counters are not equal, the job will be posted
back onto the job queue for later processing. When the job finishes processing the execution order
of the main object is incremented. This system means the no matter how many job threads are
executing, decoding of an individual sound will always get processed serially. The advantage to
having multiple threads comes into play when loading multiple sounds at the same time.
The resource manager's job queue is not 100% lock-free and will use a spinlock to achieve
thread-safety for a very small section of code. This is only relevant when the resource manager
uses more than one job thread. If only using a single job thread, which is the default, the
lock should never actually wait in practice. The amount of time spent locking should be quite
short, but it's something to be aware of for those who have pedantic lock-free requirements and
need to use more than one job thread. There are plans to remove this lock in a future version.
In addition, posting a job will release a semaphore, which on Win32 is implemented with
`ReleaseSemaphore` and on POSIX platforms via a condition variable:
```c
pthread_mutex_lock(&pSemaphore->lock);
{
pSemaphore->value += 1;
pthread_cond_signal(&pSemaphore->cond);
}
pthread_mutex_unlock(&pSemaphore->lock);
```
Again, this is relevant for those with strict lock-free requirements in the audio thread. To avoid
this, you can use non-blocking mode (via the `MA_JOB_QUEUE_FLAG_NON_BLOCKING`
flag) and implement your own job processing routine (see the "Resource Manager" section above for
details on how to do this).
6.2.2. Data Buffers
-------------------
When the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM` flag is excluded at initialization time, the
resource manager will try to load the data into an in-memory data buffer. Before doing so, however,
it will first check if the specified file is already loaded. If so, it will increment a reference
counter and just use the already loaded data. This saves both time and memory. When the data buffer
is uninitialized, the reference counter will be decremented. If the counter hits zero, the file
will be unloaded. This is a detail to keep in mind because it could result in excessive loading and
unloading of a sound. For example, the following sequence will result in a file be loaded twice,
once after the other:
```c
ma_resource_manager_data_source_init(pResourceManager, "my_file", ..., &myDataBuffer0); // Refcount = 1. Initial load.
ma_resource_manager_data_source_uninit(pResourceManager, &myDataBuffer0); // Refcount = 0. Unloaded.
ma_resource_manager_data_source_init(pResourceManager, "my_file", ..., &myDataBuffer1); // Refcount = 1. Reloaded because previous uninit() unloaded it.
ma_resource_manager_data_source_uninit(pResourceManager, &myDataBuffer1); // Refcount = 0. Unloaded.
```
A binary search tree (BST) is used for storing data buffers as it has good balance between
efficiency and simplicity. The key of the BST is a 64-bit hash of the file path that was passed
into `ma_resource_manager_data_source_init()`. The advantage of using a hash is that it saves
memory over storing the entire path, has faster comparisons, and results in a mostly balanced BST
due to the random nature of the hash. The disadvantage is that file names are case-sensitive. If
this is an issue, you should normalize your file names to upper- or lower-case before initializing
your data sources.
When a sound file has not already been loaded and the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC`
flag is excluded, the file will be decoded synchronously by the calling thread. There are two
options for controlling how the audio is stored in the data buffer - encoded or decoded. When the
`MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE` option is excluded, the raw file data will be stored
in memory. Otherwise the sound will be decoded before storing it in memory. Synchronous loading is
a very simple and standard process of simply adding an item to the BST, allocating a block of
memory and then decoding (if `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE` is specified).
When the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC` flag is specified, loading of the data buffer
is done asynchronously. In this case, a job is posted to the queue to start loading and then the
function immediately returns, setting an internal result code to `MA_BUSY`. This result code is
returned when the program calls `ma_resource_manager_data_source_result()`. When decoding has fully
completed `MA_SUCCESS` will be returned. This can be used to know if loading has fully completed.
When loading asynchronously, a single job is posted to the queue of the type
`MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER_NODE`. This involves making a copy of the file path and
associating it with job. When the job is processed by the job thread, it will first load the file
using the VFS associated with the resource manager. When using a custom VFS, it's important that it
be completely thread-safe because it will be used from one or more job threads at the same time.
Individual files should only ever be accessed by one thread at a time, however. After opening the
file via the VFS, the job will determine whether or not the file is being decoded. If not, it
simply allocates a block of memory and loads the raw file contents into it and returns. On the
other hand, when the file is being decoded, it will first allocate a decoder on the heap and
initialize it. Then it will check if the length of the file is known. If so it will allocate a
block of memory to store the decoded output and initialize it to silence. If the size is unknown,
it will allocate room for one page. After memory has been allocated, the first page will be
decoded. If the sound is shorter than a page, the result code will be set to `MA_SUCCESS` and the
completion event will be signalled and loading is now complete. If, however, there is more to
decode, a job with the code `MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE` is posted. This job
will decode the next page and perform the same process if it reaches the end. If there is more to
decode, the job will post another `MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE` job which will
keep on happening until the sound has been fully decoded. For sounds of an unknown length, each
page will be linked together as a linked list. Internally this is implemented via the
`ma_paged_audio_buffer` object.
6.2.3. Data Streams
-------------------
Data streams only ever store two pages worth of data for each instance. They are most useful for
large sounds like music tracks in games that would consume too much memory if fully decoded in
memory. After every frame from a page has been read, a job will be posted to load the next page
which is done from the VFS.
For data streams, the `MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC` flag will determine whether or
not initialization of the data source waits until the two pages have been decoded. When unset,
`ma_resource_manager_data_source_init()` will wait until the two pages have been loaded, otherwise
it will return immediately.
When frames are read from a data stream using `ma_resource_manager_data_source_read_pcm_frames()`,
`MA_BUSY` will be returned if there are no frames available. If there are some frames available,
but less than the number requested, `MA_SUCCESS` will be returned, but the actual number of frames
read will be less than the number requested. Due to the asynchronous nature of data streams,
seeking is also asynchronous. If the data stream is in the middle of a seek, `MA_BUSY` will be
returned when trying to read frames.
When `ma_resource_manager_data_source_read_pcm_frames()` results in a page getting fully consumed
a job is posted to load the next page. This will be posted from the same thread that called
`ma_resource_manager_data_source_read_pcm_frames()`.
Data streams are uninitialized by posting a job to the queue, but the function won't return until
that job has been processed. The reason for this is that the caller owns the data stream object and
therefore miniaudio needs to ensure everything completes before handing back control to the caller.
Also, if the data stream is uninitialized while pages are in the middle of decoding, they must
complete before destroying any underlying object and the job system handles this cleanly.
Note that when a new page needs to be loaded, a job will be posted to the resource manager's job
thread from the audio thread. You must keep in mind the details mentioned in the "Job Queue"
section above regarding locking when posting an event if you require a strictly lock-free audio
thread.
7. Node Graph
=============
miniaudio's routing infrastructure follows a node graph paradigm. The idea is that you create a
node whose outputs are attached to inputs of another node, thereby creating a graph. There are
different types of nodes, with each node in the graph processing input data to produce output,
which is then fed through the chain. Each node in the graph can apply their own custom effects. At
the start of the graph will usually be one or more data source nodes which have no inputs, but
instead pull their data from a data source. At the end of the graph is an endpoint which represents
the end of the chain and is where the final output is ultimately extracted from.
Each node has a number of input buses and a number of output buses. An output bus from a node is
attached to an input bus of another. Multiple nodes can connect their output buses to another
node's input bus, in which case their outputs will be mixed before processing by the node. Below is
a diagram that illustrates a hypothetical node graph setup:
```
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Data flows left to right >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
+---------------+ +-----------------+
| Data Source 1 =----+ +----------+ +----= Low Pass Filter =----+
+---------------+ | | =----+ +-----------------+ | +----------+
+----= Splitter | +----= ENDPOINT |
+---------------+ | | =----+ +-----------------+ | +----------+
| Data Source 2 =----+ +----------+ +----= Echo / Delay =----+
+---------------+ +-----------------+
```
In the above graph, it starts with two data sources whose outputs are attached to the input of a
splitter node. It's at this point that the two data sources are mixed. After mixing, the splitter
performs it's processing routine and produces two outputs which is simply a duplication of the
input stream. One output is attached to a low pass filter, whereas the other output is attached to
a echo/delay. The outputs of the the low pass filter and the echo are attached to the endpoint, and
since they're both connected to the same input but, they'll be mixed.
Each input bus must be configured to accept the same number of channels, but the number of channels
used by input buses can be different to the number of channels for output buses in which case
miniaudio will automatically convert the input data to the output channel count before processing.
The number of channels of an output bus of one node must match the channel count of the input bus
it's attached to. The channel counts cannot be changed after the node has been initialized. If you
attempt to attach an output bus to an input bus with a different channel count, attachment will
fail.
To use a node graph, you first need to initialize a `ma_node_graph` object. This is essentially a
container around the entire graph. The `ma_node_graph` object is required for some thread-safety
issues which will be explained later. A `ma_node_graph` object is initialized using miniaudio's
standard config/init system:
```c
ma_node_graph_config nodeGraphConfig = ma_node_graph_config_init(myChannelCount);
result = ma_node_graph_init(&nodeGraphConfig, NULL, &nodeGraph); // Second parameter is a pointer to allocation callbacks.
if (result != MA_SUCCESS) {
// Failed to initialize node graph.
}
```
When you initialize the node graph, you're specifying the channel count of the endpoint. The
endpoint is a special node which has one input bus and one output bus, both of which have the
same channel count, which is specified in the config. Any nodes that connect directly to the
endpoint must be configured such that their output buses have the same channel count. When you read
audio data from the node graph, it'll have the channel count you specified in the config. To read
data from the graph:
```c
ma_uint32 framesRead;
result = ma_node_graph_read_pcm_frames(&nodeGraph, pFramesOut, frameCount, &framesRead);
if (result != MA_SUCCESS) {
// Failed to read data from the node graph.
}
```
When you read audio data, miniaudio starts at the node graph's endpoint node which then pulls in
data from it's input attachments, which in turn recusively pull in data from their inputs, and so
on. At the start of the graph there will be some kind of data source node which will have zero
inputs and will instead read directly from a data source. The base nodes don't literally need to
read from a `ma_data_source` object, but they will always have some kind of underlying object that
sources some kind of audio. The `ma_data_source_node` node can be used to read from a
`ma_data_source`. Data is always in floating-point format and in the number of channels you
specified when the graph was initialized. The sample rate is defined by the underlying data sources.
It's up to you to ensure they use a consistent and appropraite sample rate.
The `ma_node` API is designed to allow custom nodes to be implemented with relative ease, but
miniaudio includes a few stock nodes for common functionality. This is how you would initialize a
node which reads directly from a data source (`ma_data_source_node`) which is an example of one
of the stock nodes that comes with miniaudio:
```c
ma_data_source_node_config config = ma_data_source_node_config_init(pMyDataSource);
ma_data_source_node dataSourceNode;
result = ma_data_source_node_init(&nodeGraph, &config, NULL, &dataSourceNode);
if (result != MA_SUCCESS) {
// Failed to create data source node.
}
```
The data source node will use the output channel count to determine the channel count of the output
bus. There will be 1 output bus and 0 input buses (data will be drawn directly from the data
source). The data source must output to floating-point (`ma_format_f32`) or else an error will be
returned from `ma_data_source_node_init()`.
By default the node will not be attached to the graph. To do so, use `ma_node_attach_output_bus()`:
```c
result = ma_node_attach_output_bus(&dataSourceNode, 0, ma_node_graph_get_endpoint(&nodeGraph), 0);
if (result != MA_SUCCESS) {
// Failed to attach node.
}
```
The code above connects the data source node directly to the endpoint. Since the data source node
has only a single output bus, the index will always be 0. Likewise, the endpoint only has a single
input bus which means the input bus index will also always be 0.
To detach a specific output bus, use `ma_node_detach_output_bus()`. To detach all output buses, use
`ma_node_detach_all_output_buses()`. If you want to just move the output bus from one attachment to
another, you do not need to detach first. You can just call `ma_node_attach_output_bus()` and it'll
deal with it for you.
Less frequently you may want to create a specialized node. This will be a node where you implement
your own processing callback to apply a custom effect of some kind. This is similar to initalizing
one of the stock node types, only this time you need to specify a pointer to a vtable containing a
pointer to the processing function and the number of input and output buses. Example:
```c
static void my_custom_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
// Do some processing of ppFramesIn (one stream of audio data per input bus)
const float* pFramesIn_0 = ppFramesIn[0]; // Input bus @ index 0.
const float* pFramesIn_1 = ppFramesIn[1]; // Input bus @ index 1.
float* pFramesOut_0 = ppFramesOut[0]; // Output bus @ index 0.
// Do some processing. On input, `pFrameCountIn` will be the number of input frames in each
// buffer in `ppFramesIn` and `pFrameCountOut` will be the capacity of each of the buffers
// in `ppFramesOut`. On output, `pFrameCountIn` should be set to the number of input frames
// your node consumed and `pFrameCountOut` should be set the number of output frames that
// were produced.
//
// You should process as many frames as you can. If your effect consumes input frames at the
// same rate as output frames (always the case, unless you're doing resampling), you need
// only look at `ppFramesOut` and process that exact number of frames. If you're doing
// resampling, you'll need to be sure to set both `pFrameCountIn` and `pFrameCountOut`
// properly.
}
static ma_node_vtable my_custom_node_vtable =
{
my_custom_node_process_pcm_frames, // The function that will be called process your custom node. This is where you'd implement your effect processing.
NULL, // Optional. A callback for calculating the number of input frames that are required to process a specified number of output frames.
2, // 2 input buses.
1, // 1 output bus.
0 // Default flags.
};
...
// Each bus needs to have a channel count specified. To do this you need to specify the channel
// counts in an array and then pass that into the node config.
ma_uint32 inputChannels[2]; // Equal in size to the number of input channels specified in the vtable.
ma_uint32 outputChannels[1]; // Equal in size to the number of output channels specicied in the vtable.
inputChannels[0] = channelsIn;
inputChannels[1] = channelsIn;
outputChannels[0] = channelsOut;
ma_node_config nodeConfig = ma_node_config_init();
nodeConfig.vtable = &my_custom_node_vtable;
nodeConfig.pInputChannels = inputChannels;
nodeConfig.pOutputChannels = outputChannels;
ma_node_base node;
result = ma_node_init(&nodeGraph, &nodeConfig, NULL, &node);
if (result != MA_SUCCESS) {
// Failed to initialize node.
}
```
When initializing a custom node, as in the code above, you'll normally just place your vtable in
static space. The number of input and output buses are specified as part of the vtable. If you need
a variable number of buses on a per-node bases, the vtable should have the relevant bus count set
to `MA_NODE_BUS_COUNT_UNKNOWN`. In this case, the bus count should be set in the node config:
```c
static ma_node_vtable my_custom_node_vtable =
{
my_custom_node_process_pcm_frames, // The function that will be called process your custom node. This is where you'd implement your effect processing.
NULL, // Optional. A callback for calculating the number of input frames that are required to process a specified number of output frames.
MA_NODE_BUS_COUNT_UNKNOWN, // The number of input buses is determined on a per-node basis.
1, // 1 output bus.
0 // Default flags.
};
...
ma_node_config nodeConfig = ma_node_config_init();
nodeConfig.vtable = &my_custom_node_vtable;
nodeConfig.inputBusCount = myBusCount; // <-- Since the vtable specifies MA_NODE_BUS_COUNT_UNKNOWN, the input bus count should be set here.
nodeConfig.pInputChannels = inputChannels; // <-- Make sure there are nodeConfig.inputBusCount elements in this array.
nodeConfig.pOutputChannels = outputChannels; // <-- The vtable specifies 1 output bus, so there must be 1 element in this array.
```
In the above example it's important to never set the `inputBusCount` and `outputBusCount` members
to anything other than their defaults if the vtable specifies an explicit count. They can only be
set if the vtable specifies MA_NODE_BUS_COUNT_UNKNOWN in the relevant bus count.
Most often you'll want to create a structure to encapsulate your node with some extra data. You
need to make sure the `ma_node_base` object is your first member of the structure:
```c
typedef struct
{
ma_node_base base; // <-- Make sure this is always the first member.
float someCustomData;
} my_custom_node;
```
By doing this, your object will be compatible with all `ma_node` APIs and you can attach it to the
graph just like any other node.
In the custom processing callback (`my_custom_node_process_pcm_frames()` in the example above), the
number of channels for each bus is what was specified by the config when the node was initialized
with `ma_node_init()`. In addition, all attachments to each of the input buses will have been
pre-mixed by miniaudio. The config allows you to specify different channel counts for each
individual input and output bus. It's up to the effect to handle it appropriate, and if it can't,
return an error in it's initialization routine.
Custom nodes can be assigned some flags to describe their behaviour. These are set via the vtable
and include the following:
+-----------------------------------------+---------------------------------------------------+
| Flag Name | Description |
+-----------------------------------------+---------------------------------------------------+
| MA_NODE_FLAG_PASSTHROUGH | Useful for nodes that do not do any kind of audio |
| | processing, but are instead used for tracking |
| | time, handling events, etc. Also used by the |
| | internal endpoint node. It reads directly from |
| | the input bus to the output bus. Nodes with this |
| | flag must have exactly 1 input bus and 1 output |
| | bus, and both buses must have the same channel |
| | counts. |
+-----------------------------------------+---------------------------------------------------+
| MA_NODE_FLAG_CONTINUOUS_PROCESSING | Causes the processing callback to be called even |
| | when no data is available to be read from input |
| | attachments. This is useful for effects like |
| | echos where there will be a tail of audio data |
| | that still needs to be processed even when the |
| | original data sources have reached their ends. |
+-----------------------------------------+---------------------------------------------------+
| MA_NODE_FLAG_ALLOW_NULL_INPUT | Used in conjunction with |
| | `MA_NODE_FLAG_CONTINUOUS_PROCESSING`. When this |
| | is set, the `ppFramesIn` parameter of the |
| | processing callback will be set to NULL when |
| | there are no input frames are available. When |
| | this is unset, silence will be posted to the |
| | processing callback. |
+-----------------------------------------+---------------------------------------------------+
| MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES | Used to tell miniaudio that input and output |
| | frames are processed at different rates. You |
| | should set this for any nodes that perform |
| | resampling. |
+-----------------------------------------+---------------------------------------------------+
| MA_NODE_FLAG_SILENT_OUTPUT | Used to tell miniaudio that a node produces only |
| | silent output. This is useful for nodes where you |
| | don't want the output to contribute to the final |
| | mix. An example might be if you want split your |
| | stream and have one branch be output to a file. |
| | When using this flag, you should avoid writing to |
| | the output buffer of the node's processing |
| | callback because miniaudio will ignore it anyway. |
+-----------------------------------------+---------------------------------------------------+
If you need to make a copy of an audio stream for effect processing you can use a splitter node
called `ma_splitter_node`. This takes has 1 input bus and splits the stream into 2 output buses.
You can use it like this:
```c
ma_splitter_node_config splitterNodeConfig = ma_splitter_node_config_init(channelsIn, channelsOut);
ma_splitter_node splitterNode;
result = ma_splitter_node_init(&nodeGraph, &splitterNodeConfig, NULL, &splitterNode);
if (result != MA_SUCCESS) {
// Failed to create node.
}
// Attach your output buses to two different input buses (can be on two different nodes).
ma_node_attach_output_bus(&splitterNode, 0, ma_node_graph_get_endpoint(&nodeGraph), 0); // Attach directly to the endpoint.
ma_node_attach_output_bus(&splitterNode, 1, &myEffectNode, 0); // Attach to input bus 0 of some effect node.
```
The volume of an output bus can be configured on a per-bus basis:
```c
ma_node_set_output_bus_volume(&splitterNode, 0, 0.5f);
ma_node_set_output_bus_volume(&splitterNode, 1, 0.5f);
```
In the code above we're using the splitter node from before and changing the volume of each of the
copied streams.
You can start and stop a node with the following:
```c
ma_node_set_state(&splitterNode, ma_node_state_started); // The default state.
ma_node_set_state(&splitterNode, ma_node_state_stopped);
```
By default the node is in a started state, but since it won't be connected to anything won't
actually be invoked by the node graph until it's connected. When you stop a node, data will not be
read from any of it's input connections. You can use this property to stop a group of sounds
atomically.
You can configure the initial state of a node in it's config:
```c
nodeConfig.initialState = ma_node_state_stopped;
```
Note that for the stock specialized nodes, all of their configs will have a `nodeConfig` member
which is the config to use with the base node. This is where the initial state can be configured
for specialized nodes:
```c
dataSourceNodeConfig.nodeConfig.initialState = ma_node_state_stopped;
```
When using a specialized node like `ma_data_source_node` or `ma_splitter_node`, be sure to not
modify the `vtable` member of the `nodeConfig` object.
7.1. Timing
-----------
The node graph supports starting and stopping nodes at scheduled times. This is especially useful
for data source nodes where you want to get the node set up, but only start playback at a specific
time. There are two clocks: local and global.
A local clock is per-node, whereas the global clock is per graph. Scheduling starts and stops can
only be done based on the global clock because the local clock will not be running while the node
is stopped. The global clocks advances whenever `ma_node_graph_read_pcm_frames()` is called. On the
other hand, the local clock only advances when the node's processing callback is fired, and is
advanced based on the output frame count.
To retrieve the global time, use `ma_node_graph_get_time()`. The global time can be set with
`ma_node_graph_set_time()` which might be useful if you want to do seeking on a global timeline.
Getting and setting the local time is similar. Use `ma_node_get_time()` to retrieve the local time,
and `ma_node_set_time()` to set the local time. The global and local times will be advanced by the
audio thread, so care should be taken to avoid data races. Ideally you should avoid calling these
outside of the node processing callbacks which are always run on the audio thread.
There is basic support for scheduling the starting and stopping of nodes. You can only schedule one
start and one stop at a time. This is mainly intended for putting nodes into a started or stopped
state in a frame-exact manner. Without this mechanism, starting and stopping of a node is limited
to the resolution of a call to `ma_node_graph_read_pcm_frames()` which would typically be in blocks
of several milliseconds. The following APIs can be used for scheduling node states:
```c
ma_node_set_state_time()
ma_node_get_state_time()
```
The time is absolute and must be based on the global clock. An example is below:
```c
ma_node_set_state_time(&myNode, ma_node_state_started, sampleRate*1); // Delay starting to 1 second.
ma_node_set_state_time(&myNode, ma_node_state_stopped, sampleRate*5); // Delay stopping to 5 seconds.
```
An example for changing the state using a relative time.
```c
ma_node_set_state_time(&myNode, ma_node_state_started, sampleRate*1 + ma_node_graph_get_time(&myNodeGraph));
ma_node_set_state_time(&myNode, ma_node_state_stopped, sampleRate*5 + ma_node_graph_get_time(&myNodeGraph));
```
Note that due to the nature of multi-threading the times may not be 100% exact. If this is an
issue, consider scheduling state changes from within a processing callback. An idea might be to
have some kind of passthrough trigger node that is used specifically for tracking time and handling
events.
7.2. Thread Safety and Locking
------------------------------
When processing audio, it's ideal not to have any kind of locking in the audio thread. Since it's
expected that `ma_node_graph_read_pcm_frames()` would be run on the audio thread, it does so
without the use of any locks. This section discusses the implementation used by miniaudio and goes
over some of the compromises employed by miniaudio to achieve this goal. Note that the current
implementation may not be ideal - feedback and critiques are most welcome.
The node graph API is not *entirely* lock-free. Only `ma_node_graph_read_pcm_frames()` is expected
to be lock-free. Attachment, detachment and uninitialization of nodes use locks to simplify the
implementation, but are crafted in a way such that such locking is not required when reading audio
data from the graph. Locking in these areas are achieved by means of spinlocks.
The main complication with keeping `ma_node_graph_read_pcm_frames()` lock-free stems from the fact
that a node can be uninitialized, and it's memory potentially freed, while in the middle of being
processed on the audio thread. There are times when the audio thread will be referencing a node,
which means the uninitialization process of a node needs to make sure it delays returning until the
audio thread is finished so that control is not handed back to the caller thereby giving them a
chance to free the node's memory.
When the audio thread is processing a node, it does so by reading from each of the output buses of
the node. In order for a node to process data for one of it's output buses, it needs to read from
each of it's input buses, and so on an so forth. It follows that once all output buses of a node
are detached, the node as a whole will be disconnected and no further processing will occur unless
it's output buses are reattached, which won't be happening when the node is being uninitialized.
By having `ma_node_detach_output_bus()` wait until the audio thread is finished with it, we can
simplify a few things, at the expense of making `ma_node_detach_output_bus()` a bit slower. By
doing this, the implementation of `ma_node_uninit()` becomes trivial - just detach all output
nodes, followed by each of the attachments to each of it's input nodes, and then do any final clean
up.
With the above design, the worst-case scenario is `ma_node_detach_output_bus()` taking as long as
it takes to process the output bus being detached. This will happen if it's called at just the
wrong moment where the audio thread has just iterated it and has just started processing. The
caller of `ma_node_detach_output_bus()` will stall until the audio thread is finished, which
includes the cost of recursively processing it's inputs. This is the biggest compromise made with
the approach taken by miniaudio for it's lock-free processing system. The cost of detaching nodes
earlier in the pipeline (data sources, for example) will be cheaper than the cost of detaching
higher level nodes, such as some kind of final post-processing endpoint. If you need to do mass
detachments, detach starting from the lowest level nodes and work your way towards the final
endpoint node (but don't try detaching the node graph's endpoint). If the audio thread is not
running, detachment will be fast and detachment in any order will be the same. The reason nodes
need to wait for their input attachments to complete is due to the potential for desyncs between
data sources. If the node was to terminate processing mid way through processing it's inputs,
there's a chance that some of the underlying data sources will have been read, but then others not.
That will then result in a potential desynchronization when detaching and reattaching higher-level
nodes. A possible solution to this is to have an option when detaching to terminate processing
before processing all input attachments which should be fairly simple.
Another compromise, albeit less significant, is locking when attaching and detaching nodes. This
locking is achieved by means of a spinlock in order to reduce memory overhead. A lock is present
for each input bus and output bus. When an output bus is connected to an input bus, both the output
bus and input bus is locked. This locking is specifically for attaching and detaching across
different threads and does not affect `ma_node_graph_read_pcm_frames()` in any way. The locking and
unlocking is mostly self-explanatory, but a slightly less intuitive aspect comes into it when
considering that iterating over attachments must not break as a result of attaching or detaching a
node while iteration is occuring.
Attaching and detaching are both quite simple. When an output bus of a node is attached to an input
bus of another node, it's added to a linked list. Basically, an input bus is a linked list, where
each item in the list is and output bus. We have some intentional (and convenient) restrictions on
what can done with the linked list in order to simplify the implementation. First of all, whenever
something needs to iterate over the list, it must do so in a forward direction. Backwards iteration
is not supported. Also, items can only be added to the start of the list.
The linked list is a doubly-linked list where each item in the list (an output bus) holds a pointer
to the next item in the list, and another to the previous item. A pointer to the previous item is
only required for fast detachment of the node - it is never used in iteration. This is an
important property because it means from the perspective of iteration, attaching and detaching of
an item can be done with a single atomic assignment. This is exploited by both the attachment and
detachment process. When attaching the node, the first thing that is done is the setting of the
local "next" and "previous" pointers of the node. After that, the item is "attached" to the list
by simply performing an atomic exchange with the head pointer. After that, the node is "attached"
to the list from the perspective of iteration. Even though the "previous" pointer of the next item
hasn't yet been set, from the perspective of iteration it's been attached because iteration will
only be happening in a forward direction which means the "previous" pointer won't actually ever get
used. The same general process applies to detachment. See `ma_node_attach_output_bus()` and
`ma_node_detach_output_bus()` for the implementation of this mechanism.
8. Decoding
===========
The `ma_decoder` API is used for reading audio files. Decoders are completely decoupled from
devices and can be used independently. The following formats are supported:
+---------+------------------+----------+
| Format | Decoding Backend | Built-In |
+---------+------------------+----------+
| WAV | dr_wav | Yes |
| MP3 | dr_mp3 | Yes |
| FLAC | dr_flac | Yes |
| Vorbis | stb_vorbis | No |
+---------+------------------+----------+
Vorbis is supported via stb_vorbis which can be enabled by including the header section before the
implementation of miniaudio, like the following:
```c
#define STB_VORBIS_HEADER_ONLY
#include "extras/stb_vorbis.c" // Enables Vorbis decoding.
#define MINIAUDIO_IMPLEMENTATION
#include "miniaudio.h"
// The stb_vorbis implementation must come after the implementation of miniaudio.
#undef STB_VORBIS_HEADER_ONLY
#include "extras/stb_vorbis.c"
```
A copy of stb_vorbis is included in the "extras" folder in the miniaudio repository (https://github.com/mackron/miniaudio).
Built-in decoders are amalgamated into the implementation section of miniaudio. You can disable the
built-in decoders by specifying one or more of the following options before the miniaudio
implementation:
```c
#define MA_NO_WAV
#define MA_NO_MP3
#define MA_NO_FLAC
```
Disabling built-in decoding libraries is useful if you use these libraries independantly of the
`ma_decoder` API.
A decoder can be initialized from a file with `ma_decoder_init_file()`, a block of memory with
`ma_decoder_init_memory()`, or from data delivered via callbacks with `ma_decoder_init()`. Here is
an example for loading a decoder from a file:
```c
ma_decoder decoder;
ma_result result = ma_decoder_init_file("MySong.mp3", NULL, &decoder);
if (result != MA_SUCCESS) {
return false; // An error occurred.
}
...
ma_decoder_uninit(&decoder);
```
When initializing a decoder, you can optionally pass in a pointer to a `ma_decoder_config` object
(the `NULL` argument in the example above) which allows you to configure the output format, channel
count, sample rate and channel map:
```c
ma_decoder_config config = ma_decoder_config_init(ma_format_f32, 2, 48000);
```
When passing in `NULL` for decoder config in `ma_decoder_init*()`, the output format will be the
same as that defined by the decoding backend.
Data is read from the decoder as PCM frames. This will output the number of PCM frames actually
read. If this is less than the requested number of PCM frames it means you've reached the end. The
return value will be `MA_AT_END` if no samples have been read and the end has been reached.
```c
ma_result result = ma_decoder_read_pcm_frames(pDecoder, pFrames, framesToRead, &framesRead);
if (framesRead < framesToRead) {
// Reached the end.
}
```
You can also seek to a specific frame like so:
```c
ma_result result = ma_decoder_seek_to_pcm_frame(pDecoder, targetFrame);
if (result != MA_SUCCESS) {
return false; // An error occurred.
}
```
If you want to loop back to the start, you can simply seek back to the first PCM frame:
```c
ma_decoder_seek_to_pcm_frame(pDecoder, 0);
```
When loading a decoder, miniaudio uses a trial and error technique to find the appropriate decoding
backend. This can be unnecessarily inefficient if the type is already known. In this case you can
use `encodingFormat` variable in the device config to specify a specific encoding format you want
to decode:
```c
decoderConfig.encodingFormat = ma_encoding_format_wav;
```
See the `ma_encoding_format` enum for possible encoding formats.
The `ma_decoder_init_file()` API will try using the file extension to determine which decoding
backend to prefer.
8.1. Custom Decoders
--------------------
It's possible to implement a custom decoder and plug it into miniaudio. This is extremely useful
when you want to use the `ma_decoder` API, but need to support an encoding format that's not one of
the stock formats supported by miniaudio. This can be put to particularly good use when using the
`ma_engine` and/or `ma_resource_manager` APIs because they use `ma_decoder` internally. If, for
example, you wanted to support Opus, you can do so with a custom decoder (there if a reference
Opus decoder in the "extras" folder of the miniaudio repository which uses libopus + libopusfile).
A custom decoder must implement a data source. A vtable called `ma_decoding_backend_vtable` needs
to be implemented which is then passed into the decoder config:
```c
ma_decoding_backend_vtable* pCustomBackendVTables[] =
{
&g_ma_decoding_backend_vtable_libvorbis,
&g_ma_decoding_backend_vtable_libopus
};
...
decoderConfig = ma_decoder_config_init_default();
decoderConfig.pCustomBackendUserData = NULL;
decoderConfig.ppCustomBackendVTables = pCustomBackendVTables;
decoderConfig.customBackendCount = sizeof(pCustomBackendVTables) / sizeof(pCustomBackendVTables[0]);
```
The `ma_decoding_backend_vtable` vtable has the following functions:
```
onInit
onInitFile
onInitFileW
onInitMemory
onUninit
```
There are only two functions that must be implemented - `onInit` and `onUninit`. The other
functions can be implemented for a small optimization for loading from a file path or memory. If
these are not specified, miniaudio will deal with it for you via a generic implementation.
When you initialize a custom data source (by implementing the `onInit` function in the vtable) you
will need to output a pointer to a `ma_data_source` which implements your custom decoder. See the
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
decoding. When you call them, you will pass in the `pReadSeekTellUserData` pointer to the relevant
parameter.
The `pConfig` parameter in `onInit` can be used to configure the backend if appropriate. It's only
used as a hint and can be ignored. However, if any of the properties are relevant to your decoder,
an optimal implementation will handle the relevant properties appropriately.
If memory allocation is required, it should be done so via the specified allocation callbacks if
possible (the `pAllocationCallbacks` parameter).
If an error occurs when initializing the decoder, you should leave `ppBackend` unset, or set to
NULL, and make sure everything is cleaned up appropriately and an appropriate result code returned.
When multiple custom backends are specified, miniaudio will cycle through the vtables in the order
they're listed in the array that's passed into the decoder config so it's important that your
initialization routine is clean.
When a decoder is uninitialized, the `onUninit` callback will be fired which will give you an
opportunity to clean up and internal data.
9. Encoding
===========
The `ma_encoding` API is used for writing audio files. The only supported output format is WAV
which is achieved via dr_wav which is amalgamated into the implementation section of miniaudio.
This can be disabled by specifying the following option before the implementation of miniaudio:
```c
#define MA_NO_WAV
```
An encoder can be initialized to write to a file with `ma_encoder_init_file()` or from data
delivered via callbacks with `ma_encoder_init()`. Below is an example for initializing an encoder
to output to a file.
```c
ma_encoder_config config = ma_encoder_config_init(ma_encoding_format_wav, FORMAT, CHANNELS, SAMPLE_RATE);
ma_encoder encoder;
ma_result result = ma_encoder_init_file("my_file.wav", &config, &encoder);
if (result != MA_SUCCESS) {
// Error
}
...
ma_encoder_uninit(&encoder);
```
When initializing an encoder you must specify a config which is initialized with
`ma_encoder_config_init()`. Here you must specify the file type, the output sample format, output
channel count and output sample rate. The following file types are supported:
+------------------------+-------------+
| Enum | Description |
+------------------------+-------------+
| ma_encoding_format_wav | WAV |
+------------------------+-------------+
If the format, channel count or sample rate is not supported by the output file type an error will
be returned. The encoder will not perform data conversion so you will need to convert it before
outputting any audio data. To output audio data, use `ma_encoder_write_pcm_frames()`, like in the
example below:
```c
framesWritten = ma_encoder_write_pcm_frames(&encoder, pPCMFramesToWrite, framesToWrite);
```
Encoders must be uninitialized with `ma_encoder_uninit()`.
10. Data Conversion
===================
A data conversion API is included with miniaudio which supports the majority of data conversion
requirements. This supports conversion between sample formats, channel counts (with channel
mapping) and sample rates.
10.1. Sample Format Conversion
------------------------------
Conversion between sample formats is achieved with the `ma_pcm_*_to_*()`, `ma_pcm_convert()` and
`ma_convert_pcm_frames_format()` APIs. Use `ma_pcm_*_to_*()` to convert between two specific
formats. Use `ma_pcm_convert()` to convert based on a `ma_format` variable. Use
`ma_convert_pcm_frames_format()` to convert PCM frames where you want to specify the frame count
and channel count as a variable instead of the total sample count.
10.1.1. Dithering
-----------------
Dithering can be set using the ditherMode parameter.
The different dithering modes include the following, in order of efficiency:
+-----------+--------------------------+
| Type | Enum Token |
+-----------+--------------------------+
| None | ma_dither_mode_none |
| Rectangle | ma_dither_mode_rectangle |
| Triangle | ma_dither_mode_triangle |
+-----------+--------------------------+
Note that even if the dither mode is set to something other than `ma_dither_mode_none`, it will be
ignored for conversions where dithering is not needed. Dithering is available for the following
conversions:
```
s16 -> u8
s24 -> u8
s32 -> u8
f32 -> u8
s24 -> s16
s32 -> s16
f32 -> s16
```
Note that it is not an error to pass something other than ma_dither_mode_none for conversions where
dither is not used. It will just be ignored.
10.2. Channel Conversion
------------------------
Channel conversion is used for channel rearrangement and conversion from one channel count to
another. The `ma_channel_converter` API is used for channel conversion. Below is an example of
initializing a simple channel converter which converts from mono to stereo.
```c
ma_channel_converter_config config = ma_channel_converter_config_init(
ma_format, // Sample format
1, // Input channels
NULL, // Input channel map
2, // Output channels
NULL, // Output channel map
ma_channel_mix_mode_default); // The mixing algorithm to use when combining channels.
result = ma_channel_converter_init(&config, NULL, &converter);
if (result != MA_SUCCESS) {
// Error.
}
```
To perform the conversion simply call `ma_channel_converter_process_pcm_frames()` like so:
```c
ma_result result = ma_channel_converter_process_pcm_frames(&converter, pFramesOut, pFramesIn, frameCount);
if (result != MA_SUCCESS) {
// Error.
}
```
It is up to the caller to ensure the output buffer is large enough to accomodate the new PCM
frames.
Input and output PCM frames are always interleaved. Deinterleaved layouts are not supported.
10.2.1. Channel Mapping
-----------------------
In addition to converting from one channel count to another, like the example above, the channel
converter can also be used to rearrange channels. When initializing the channel converter, you can
optionally pass in channel maps for both the input and output frames. If the channel counts are the
same, and each channel map contains the same channel positions with the exception that they're in
a different order, a simple shuffling of the channels will be performed. If, however, there is not
a 1:1 mapping of channel positions, or the channel counts differ, the input channels will be mixed
based on a mixing mode which is specified when initializing the `ma_channel_converter_config`
object.
When converting from mono to multi-channel, the mono channel is simply copied to each output
channel. When going the other way around, the audio of each output channel is simply averaged and
copied to the mono channel.
In more complicated cases blending is used. The `ma_channel_mix_mode_simple` mode will drop excess
channels and silence extra channels. For example, converting from 4 to 2 channels, the 3rd and 4th
channels will be dropped, whereas converting from 2 to 4 channels will put silence into the 3rd and
4th channels.
The `ma_channel_mix_mode_rectangle` mode uses spacial locality based on a rectangle to compute a
simple distribution between input and output. Imagine sitting in the middle of a room, with
speakers on the walls representing channel positions. The `MA_CHANNEL_FRONT_LEFT` position can be
thought of as being in the corner of the front and left walls.
Finally, the `ma_channel_mix_mode_custom_weights` mode can be used to use custom user-defined
weights. Custom weights can be passed in as the last parameter of
`ma_channel_converter_config_init()`.
Predefined channel maps can be retrieved with `ma_channel_map_init_standard()`. This takes a
`ma_standard_channel_map` enum as it's first parameter, which can be one of the following:
+-----------------------------------+-----------------------------------------------------------+
| Name | Description |
+-----------------------------------+-----------------------------------------------------------+
| ma_standard_channel_map_default | Default channel map used by miniaudio. See below. |
| ma_standard_channel_map_microsoft | Channel map used by Microsoft's bitfield channel maps. |
| ma_standard_channel_map_alsa | Default ALSA channel map. |
| ma_standard_channel_map_rfc3551 | RFC 3551. Based on AIFF. |
| ma_standard_channel_map_flac | FLAC channel map. |
| ma_standard_channel_map_vorbis | Vorbis channel map. |
| ma_standard_channel_map_sound4 | FreeBSD's sound(4). |
| ma_standard_channel_map_sndio | sndio channel map. http://www.sndio.org/tips.html. |
| ma_standard_channel_map_webaudio | https://webaudio.github.io/web-audio-api/#ChannelOrdering |
+-----------------------------------+-----------------------------------------------------------+
Below are the channel maps used by default in miniaudio (`ma_standard_channel_map_default`):
+---------------+---------------------------------+
| Channel Count | Mapping |
+---------------+---------------------------------+
| 1 (Mono) | 0: MA_CHANNEL_MONO |
+---------------+---------------------------------+
| 2 (Stereo) | 0: MA_CHANNEL_FRONT_LEFT <br> |
| | 1: MA_CHANNEL_FRONT_RIGHT |
+---------------+---------------------------------+
| 3 | 0: MA_CHANNEL_FRONT_LEFT <br> |
| | 1: MA_CHANNEL_FRONT_RIGHT <br> |
| | 2: MA_CHANNEL_FRONT_CENTER |
+---------------+---------------------------------+
| 4 (Surround) | 0: MA_CHANNEL_FRONT_LEFT <br> |
| | 1: MA_CHANNEL_FRONT_RIGHT <br> |
| | 2: MA_CHANNEL_FRONT_CENTER <br> |
| | 3: MA_CHANNEL_BACK_CENTER |
+---------------+---------------------------------+
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
| 6 (5.1) | 0: MA_CHANNEL_FRONT_LEFT <br> |
| | 1: MA_CHANNEL_FRONT_RIGHT <br> |
| | 2: MA_CHANNEL_FRONT_CENTER <br> |
| | 3: MA_CHANNEL_LFE <br> |
| | 4: MA_CHANNEL_SIDE_LEFT <br> |
| | 5: MA_CHANNEL_SIDE_RIGHT |
+---------------+---------------------------------+
| 7 | 0: MA_CHANNEL_FRONT_LEFT <br> |
| | 1: MA_CHANNEL_FRONT_RIGHT <br> |
| | 2: MA_CHANNEL_FRONT_CENTER <br> |
| | 3: MA_CHANNEL_LFE <br> |
| | 4: MA_CHANNEL_BACK_CENTER <br> |
| | 4: MA_CHANNEL_SIDE_LEFT <br> |
| | 5: MA_CHANNEL_SIDE_RIGHT |
+---------------+---------------------------------+
| 8 (7.1) | 0: MA_CHANNEL_FRONT_LEFT <br> |
| | 1: MA_CHANNEL_FRONT_RIGHT <br> |
| | 2: MA_CHANNEL_FRONT_CENTER <br> |
| | 3: MA_CHANNEL_LFE <br> |
| | 4: MA_CHANNEL_BACK_LEFT <br> |
| | 5: MA_CHANNEL_BACK_RIGHT <br> |
| | 6: MA_CHANNEL_SIDE_LEFT <br> |
| | 7: MA_CHANNEL_SIDE_RIGHT |
+---------------+---------------------------------+
| Other | All channels set to 0. This |
| | is equivalent to the same |
| | mapping as the device. |
+---------------+---------------------------------+
10.3. Resampling
----------------
Resampling is achieved with the `ma_resampler` object. To create a resampler object, do something
like the following:
```c
ma_resampler_config config = ma_resampler_config_init(
ma_format_s16,
channels,
sampleRateIn,
sampleRateOut,
ma_resample_algorithm_linear);
ma_resampler resampler;
ma_result result = ma_resampler_init(&config, &resampler);
if (result != MA_SUCCESS) {
// An error occurred...
}
```
Do the following to uninitialize the resampler:
```c
ma_resampler_uninit(&resampler);
```
The following example shows how data can be processed
```c
ma_uint64 frameCountIn = 1000;
ma_uint64 frameCountOut = 2000;
ma_result result = ma_resampler_process_pcm_frames(&resampler, pFramesIn, &frameCountIn, pFramesOut, &frameCountOut);
if (result != MA_SUCCESS) {
// An error occurred...
}
// At this point, frameCountIn contains the number of input frames that were consumed and frameCountOut contains the
// number of output frames written.
```
To initialize the resampler you first need to set up a config (`ma_resampler_config`) with
`ma_resampler_config_init()`. You need to specify the sample format you want to use, the number of
channels, the input and output sample rate, and the algorithm.
The sample format can be either `ma_format_s16` or `ma_format_f32`. If you need a different format
you will need to perform pre- and post-conversions yourself where necessary. Note that the format
is the same for both input and output. The format cannot be changed after initialization.
The resampler supports multiple channels and is always interleaved (both input and output). The
channel count cannot be changed after initialization.
The sample rates can be anything other than zero, and are always specified in hertz. They should be
set to something like 44100, etc. The sample rate is the only configuration property that can be
changed after initialization.
The miniaudio resampler has built-in support for the following algorithms:
+-----------+------------------------------+
| Algorithm | Enum Token |
+-----------+------------------------------+
| Linear | ma_resample_algorithm_linear |
| Custom | ma_resample_algorithm_custom |
+-----------+------------------------------+
The algorithm cannot be changed after initialization.
Processing always happens on a per PCM frame basis and always assumes interleaved input and output.
De-interleaved processing is not supported. To process frames, use
`ma_resampler_process_pcm_frames()`. On input, this function takes the number of output frames you
can fit in the output buffer and the number of input frames contained in the input buffer. On
output these variables contain the number of output frames that were written to the output buffer
and the number of input frames that were consumed in the process. You can pass in NULL for the
input buffer in which case it will be treated as an infinitely large buffer of zeros. The output
buffer can also be NULL, in which case the processing will be treated as seek.
The sample rate can be changed dynamically on the fly. You can change this with explicit sample
rates with `ma_resampler_set_rate()` and also with a decimal ratio with
`ma_resampler_set_rate_ratio()`. The ratio is in/out.
Sometimes it's useful to know exactly how many input frames will be required to output a specific
number of frames. You can calculate this with `ma_resampler_get_required_input_frame_count()`.
Likewise, it's sometimes useful to know exactly how many frames would be output given a certain
number of input frames. You can do this with `ma_resampler_get_expected_output_frame_count()`.
Due to the nature of how resampling works, the resampler introduces some latency. This can be
retrieved in terms of both the input rate and the output rate with
`ma_resampler_get_input_latency()` and `ma_resampler_get_output_latency()`.
10.3.1. Resampling Algorithms
-----------------------------
The choice of resampling algorithm depends on your situation and requirements.
10.3.1.1. Linear Resampling
---------------------------
The linear resampler is the fastest, but comes at the expense of poorer quality. There is, however,
some control over the quality of the linear resampler which may make it a suitable option depending
on your requirements.
The linear resampler performs low-pass filtering before or after downsampling or upsampling,
depending on the sample rates you're converting between. When decreasing the sample rate, the
low-pass filter will be applied before downsampling. When increasing the rate it will be performed
after upsampling. By default a fourth order low-pass filter will be applied. This can be configured
via the `lpfOrder` configuration variable. Setting this to 0 will disable filtering.
The low-pass filter has a cutoff frequency which defaults to half the sample rate of the lowest of
the input and output sample rates (Nyquist Frequency).
The API for the linear resampler is the same as the main resampler API, only it's called
`ma_linear_resampler`.
10.3.2. Custom Resamplers
-------------------------
You can implement a custom resampler by using the `ma_resample_algorithm_custom` resampling
algorithm and setting a vtable in the resampler config:
```c
ma_resampler_config config = ma_resampler_config_init(..., ma_resample_algorithm_custom);
config.pBackendVTable = &g_customResamplerVTable;
```
Custom resamplers are useful if the stock algorithms are not appropriate for your use case. You
need to implement the required functions in `ma_resampling_backend_vtable`. Note that not all
functions in the vtable need to be implemented, but if it's possible to implement, they should be.
You can use the `ma_linear_resampler` object for an example on how to implement the vtable. The
`onGetHeapSize` callback is used to calculate the size of any internal heap allocation the custom
resampler will need to make given the supplied config. When you initialize the resampler via the
`onInit` callback, you'll be given a pointer to a heap allocation which is where you should store
the heap allocated data. You should not free this data in `onUninit` because miniaudio will manage
it for you.
The `onProcess` callback is where the actual resampling takes place. On input, `pFrameCountIn`
points to a variable containing the number of frames in the `pFramesIn` buffer and
`pFrameCountOut` points to a variable containing the capacity in frames of the `pFramesOut` buffer.
On output, `pFrameCountIn` should be set to the number of input frames that were fully consumed,
whereas `pFrameCountOut` should be set to the number of frames that were written to `pFramesOut`.
The `onSetRate` callback is optional and is used for dynamically changing the sample rate. If
dynamic rate changes are not supported, you can set this callback to NULL.
The `onGetInputLatency` and `onGetOutputLatency` functions are used for retrieving the latency in
input and output rates respectively. These can be NULL in which case latency calculations will be
assumed to be NULL.
The `onGetRequiredInputFrameCount` callback is used to give miniaudio a hint as to how many input
frames are required to be available to produce the given number of output frames. Likewise, the
`onGetExpectedOutputFrameCount` callback is used to determine how many output frames will be
produced given the specified number of input frames. miniaudio will use these as a hint, but they
are optional and can be set to NULL if you're unable to implement them.
10.4. General Data Conversion
-----------------------------
The `ma_data_converter` API can be used to wrap sample format conversion, channel conversion and
resampling into one operation. This is what miniaudio uses internally to convert between the format
requested when the device was initialized and the format of the backend's native device. The API
for general data conversion is very similar to the resampling API. Create a `ma_data_converter`
object like this:
```c
ma_data_converter_config config = ma_data_converter_config_init(
inputFormat,
outputFormat,
inputChannels,
outputChannels,
inputSampleRate,
outputSampleRate
);
ma_data_converter converter;
ma_result result = ma_data_converter_init(&config, NULL, &converter);
if (result != MA_SUCCESS) {
// An error occurred...
}
```
In the example above we use `ma_data_converter_config_init()` to initialize the config, however
there's many more properties that can be configured, such as channel maps and resampling quality.
Something like the following may be more suitable depending on your requirements:
```c
ma_data_converter_config config = ma_data_converter_config_init_default();
config.formatIn = inputFormat;
config.formatOut = outputFormat;
config.channelsIn = inputChannels;
config.channelsOut = outputChannels;
config.sampleRateIn = inputSampleRate;
config.sampleRateOut = outputSampleRate;
ma_channel_map_init_standard(ma_standard_channel_map_flac, config.channelMapIn, sizeof(config.channelMapIn)/sizeof(config.channelMapIn[0]), config.channelCountIn);
config.resampling.linear.lpfOrder = MA_MAX_FILTER_ORDER;
```
Do the following to uninitialize the data converter:
```c
ma_data_converter_uninit(&converter, NULL);
```
The following example shows how data can be processed
```c
ma_uint64 frameCountIn = 1000;
ma_uint64 frameCountOut = 2000;
ma_result result = ma_data_converter_process_pcm_frames(&converter, pFramesIn, &frameCountIn, pFramesOut, &frameCountOut);
if (result != MA_SUCCESS) {
// An error occurred...
}
// At this point, frameCountIn contains the number of input frames that were consumed and frameCountOut contains the number
// of output frames written.
```
The data converter supports multiple channels and is always interleaved (both input and output).
The channel count cannot be changed after initialization.
Sample rates can be anything other than zero, and are always specified in hertz. They should be set
to something like 44100, etc. The sample rate is the only configuration property that can be
changed after initialization, but only if the `resampling.allowDynamicSampleRate` member of
`ma_data_converter_config` is set to `MA_TRUE`. To change the sample rate, use
`ma_data_converter_set_rate()` or `ma_data_converter_set_rate_ratio()`. The ratio must be in/out.
The resampling algorithm cannot be changed after initialization.
Processing always happens on a per PCM frame basis and always assumes interleaved input and output.
De-interleaved processing is not supported. To process frames, use
`ma_data_converter_process_pcm_frames()`. On input, this function takes the number of output frames
you can fit in the output buffer and the number of input frames contained in the input buffer. On
output these variables contain the number of output frames that were written to the output buffer
and the number of input frames that were consumed in the process. You can pass in NULL for the
input buffer in which case it will be treated as an infinitely large
buffer of zeros. The output buffer can also be NULL, in which case the processing will be treated
as seek.
Sometimes it's useful to know exactly how many input frames will be required to output a specific
number of frames. You can calculate this with `ma_data_converter_get_required_input_frame_count()`.
Likewise, it's sometimes useful to know exactly how many frames would be output given a certain
number of input frames. You can do this with `ma_data_converter_get_expected_output_frame_count()`.
Due to the nature of how resampling works, the data converter introduces some latency if resampling
is required. This can be retrieved in terms of both the input rate and the output rate with
`ma_data_converter_get_input_latency()` and `ma_data_converter_get_output_latency()`.
11. Filtering
=============
11.1. Biquad Filtering
----------------------
Biquad filtering is achieved with the `ma_biquad` API. Example:
```c
ma_biquad_config config = ma_biquad_config_init(ma_format_f32, channels, b0, b1, b2, a0, a1, a2);
ma_result result = ma_biquad_init(&config, &biquad);
if (result != MA_SUCCESS) {
// Error.
}
...
ma_biquad_process_pcm_frames(&biquad, pFramesOut, pFramesIn, frameCount);
```
Biquad filtering is implemented using transposed direct form 2. The numerator coefficients are b0,
b1 and b2, and the denominator coefficients are a0, a1 and a2. The a0 coefficient is required and
coefficients must not be pre-normalized.
Supported formats are `ma_format_s16` and `ma_format_f32`. If you need to use a different format
you need to convert it yourself beforehand. When using `ma_format_s16` the biquad filter will use
fixed point arithmetic. When using `ma_format_f32`, floating point arithmetic will be used.
Input and output frames are always interleaved.
Filtering can be applied in-place by passing in the same pointer for both the input and output
buffers, like so:
```c
ma_biquad_process_pcm_frames(&biquad, pMyData, pMyData, frameCount);
```
If you need to change the values of the coefficients, but maintain the values in the registers you
can do so with `ma_biquad_reinit()`. This is useful if you need to change the properties of the
filter while keeping the values of registers valid to avoid glitching. Do not use
`ma_biquad_init()` for this as it will do a full initialization which involves clearing the
registers to 0. Note that changing the format or channel count after initialization is invalid and
will result in an error.
11.2. Low-Pass Filtering
------------------------
Low-pass filtering is achieved with the following APIs:
+---------+------------------------------------------+
| API | Description |
+---------+------------------------------------------+
| ma_lpf1 | First order low-pass filter |
| ma_lpf2 | Second order low-pass filter |
| ma_lpf | High order low-pass filter (Butterworth) |
+---------+------------------------------------------+
Low-pass filter example:
```c
ma_lpf_config config = ma_lpf_config_init(ma_format_f32, channels, sampleRate, cutoffFrequency, order);
ma_result result = ma_lpf_init(&config, &lpf);
if (result != MA_SUCCESS) {
// Error.
}
...
ma_lpf_process_pcm_frames(&lpf, pFramesOut, pFramesIn, frameCount);
```
Supported formats are `ma_format_s16` and` ma_format_f32`. If you need to use a different format
you need to convert it yourself beforehand. Input and output frames are always interleaved.
Filtering can be applied in-place by passing in the same pointer for both the input and output
buffers, like so:
```c
ma_lpf_process_pcm_frames(&lpf, pMyData, pMyData, frameCount);
```
The maximum filter order is limited to `MA_MAX_FILTER_ORDER` which is set to 8. If you need more,
you can chain first and second order filters together.
```c
for (iFilter = 0; iFilter < filterCount; iFilter += 1) {
ma_lpf2_process_pcm_frames(&lpf2[iFilter], pMyData, pMyData, frameCount);
}
```
If you need to change the configuration of the filter, but need to maintain the state of internal
registers you can do so with `ma_lpf_reinit()`. This may be useful if you need to change the sample
rate and/or cutoff frequency dynamically while maintaing smooth transitions. Note that changing the
format or channel count after initialization is invalid and will result in an error.
The `ma_lpf` object supports a configurable order, but if you only need a first order filter you
may want to consider using `ma_lpf1`. Likewise, if you only need a second order filter you can use
`ma_lpf2`. The advantage of this is that they're lighter weight and a bit more efficient.
If an even filter order is specified, a series of second order filters will be processed in a
chain. If an odd filter order is specified, a first order filter will be applied, followed by a
series of second order filters in a chain.
11.3. High-Pass Filtering
-------------------------
High-pass filtering is achieved with the following APIs:
+---------+-------------------------------------------+
| API | Description |
+---------+-------------------------------------------+
| ma_hpf1 | First order high-pass filter |
| ma_hpf2 | Second order high-pass filter |
| ma_hpf | High order high-pass filter (Butterworth) |
+---------+-------------------------------------------+
High-pass filters work exactly the same as low-pass filters, only the APIs are called `ma_hpf1`,
`ma_hpf2` and `ma_hpf`. See example code for low-pass filters for example usage.
11.4. Band-Pass Filtering
-------------------------
Band-pass filtering is achieved with the following APIs:
+---------+-------------------------------+
| API | Description |
+---------+-------------------------------+
| ma_bpf2 | Second order band-pass filter |
| ma_bpf | High order band-pass filter |
+---------+-------------------------------+
Band-pass filters work exactly the same as low-pass filters, only the APIs are called `ma_bpf2` and
`ma_hpf`. See example code for low-pass filters for example usage. Note that the order for
band-pass filters must be an even number which means there is no first order band-pass filter,
unlike low-pass and high-pass filters.
11.5. Notch Filtering
---------------------
Notch filtering is achieved with the following APIs:
+-----------+------------------------------------------+
| API | Description |
+-----------+------------------------------------------+
| ma_notch2 | Second order notching filter |
+-----------+------------------------------------------+
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
| ma_peak2 | Second order peaking filter |
+----------+------------------------------------------+
11.7. Low Shelf Filtering
-------------------------
Low shelf filtering is achieved with the following APIs:
+-------------+------------------------------------------+
| API | Description |
+-------------+------------------------------------------+
| ma_loshelf2 | Second order low shelf filter |
+-------------+------------------------------------------+
Where a high-pass filter is used to eliminate lower frequencies, a low shelf filter can be used to
just turn them down rather than eliminate them entirely.
11.8. High Shelf Filtering
--------------------------
High shelf filtering is achieved with the following APIs:
+-------------+------------------------------------------+
| API | Description |
+-------------+------------------------------------------+
| ma_hishelf2 | Second order high shelf filter |
+-------------+------------------------------------------+
The high shelf filter has the same API as the low shelf filter, only you would use `ma_hishelf`
instead of `ma_loshelf`. Where a low shelf filter is used to adjust the volume of low frequencies,
the high shelf filter does the same thing for high frequencies.
12. Waveform and Noise Generation
=================================
12.1. Waveforms
---------------
miniaudio supports generation of sine, square, triangle and sawtooth waveforms. This is achieved
with the `ma_waveform` API. Example:
```c
ma_waveform_config config = ma_waveform_config_init(
FORMAT,
CHANNELS,
SAMPLE_RATE,
ma_waveform_type_sine,
amplitude,
frequency);
ma_waveform waveform;
ma_result result = ma_waveform_init(&config, &waveform);
if (result != MA_SUCCESS) {
// Error.
}
...
ma_waveform_read_pcm_frames(&waveform, pOutput, frameCount);
```
The amplitude, frequency, type, and sample rate can be changed dynamically with
`ma_waveform_set_amplitude()`, `ma_waveform_set_frequency()`, `ma_waveform_set_type()`, and
`ma_waveform_set_sample_rate()` respectively.
You can invert the waveform by setting the amplitude to a negative value. You can use this to
control whether or not a sawtooth has a positive or negative ramp, for example.
Below are the supported waveform types:
+---------------------------+
| Enum Name |
+---------------------------+
| ma_waveform_type_sine |
| ma_waveform_type_square |
| ma_waveform_type_triangle |
| ma_waveform_type_sawtooth |
+---------------------------+
12.2. Noise
-----------
miniaudio supports generation of white, pink and Brownian noise via the `ma_noise` API. Example:
```c
ma_noise_config config = ma_noise_config_init(
FORMAT,
CHANNELS,
ma_noise_type_white,
SEED,
amplitude);
ma_noise noise;
ma_result result = ma_noise_init(&config, &noise);
if (result != MA_SUCCESS) {
// Error.
}
...
ma_noise_read_pcm_frames(&noise, pOutput, frameCount);
```
The noise API uses simple LCG random number generation. It supports a custom seed which is useful
for things like automated testing requiring reproducibility. Setting the seed to zero will default
to `MA_DEFAULT_LCG_SEED`.
The amplitude, seed, and type can be changed dynamically with `ma_noise_set_amplitude()`,
`ma_noise_set_seed()`, and `ma_noise_set_type()` respectively.
By default, the noise API will use different values for different channels. So, for example, the
left side in a stereo stream will be different to the right side. To instead have each channel use
the same random value, set the `duplicateChannels` member of the noise config to true, like so:
```c
config.duplicateChannels = MA_TRUE;
```
Below are the supported noise types.
+------------------------+
| Enum Name |
+------------------------+
| ma_noise_type_white |
| ma_noise_type_pink |
| ma_noise_type_brownian |
+------------------------+
13. Audio Buffers
=================
miniaudio supports reading from a buffer of raw audio data via the `ma_audio_buffer` API. This can
read from memory that's managed by the application, but can also handle the memory management for
you internally. Memory management is flexible and should support most use cases.
Audio buffers are initialised using the standard configuration system used everywhere in miniaudio:
```c
ma_audio_buffer_config config = ma_audio_buffer_config_init(
format,
channels,
sizeInFrames,
pExistingData,
&allocationCallbacks);
ma_audio_buffer buffer;
result = ma_audio_buffer_init(&config, &buffer);
if (result != MA_SUCCESS) {
// Error.
}
...
ma_audio_buffer_uninit(&buffer);
```
In the example above, the memory pointed to by `pExistingData` will *not* be copied and is how an
application can do self-managed memory allocation. If you would rather make a copy of the data, use
`ma_audio_buffer_init_copy()`. To uninitialize the buffer, use `ma_audio_buffer_uninit()`.
Sometimes it can be convenient to allocate the memory for the `ma_audio_buffer` structure and the
raw audio data in a contiguous block of memory. That is, the raw audio data will be located
immediately after the `ma_audio_buffer` structure. To do this, use
`ma_audio_buffer_alloc_and_init()`:
```c
ma_audio_buffer_config config = ma_audio_buffer_config_init(
format,
channels,
sizeInFrames,
pExistingData,
&allocationCallbacks);
ma_audio_buffer* pBuffer
result = ma_audio_buffer_alloc_and_init(&config, &pBuffer);
if (result != MA_SUCCESS) {
// Error
}
...
ma_audio_buffer_uninit_and_free(&buffer);
```
If you initialize the buffer with `ma_audio_buffer_alloc_and_init()` you should uninitialize it
with `ma_audio_buffer_uninit_and_free()`. In the example above, the memory pointed to by
`pExistingData` will be copied into the buffer, which is contrary to the behavior of
`ma_audio_buffer_init()`.
An audio buffer has a playback cursor just like a decoder. As you read frames from the buffer, the
cursor moves forward. The last parameter (`loop`) can be used to determine if the buffer should
loop. The return value is the number of frames actually read. If this is less than the number of
frames requested it means the end has been reached. This should never happen if the `loop`
parameter is set to true. If you want to manually loop back to the start, you can do so with with
`ma_audio_buffer_seek_to_pcm_frame(pAudioBuffer, 0)`. Below is an example for reading data from an
audio buffer.
```c
ma_uint64 framesRead = ma_audio_buffer_read_pcm_frames(pAudioBuffer, pFramesOut, desiredFrameCount, isLooping);
if (framesRead < desiredFrameCount) {
// If not looping, this means the end has been reached. This should never happen in looping mode with valid input.
}
```
Sometimes you may want to avoid the cost of data movement between the internal buffer and the
output buffer. Instead you can use memory mapping to retrieve a pointer to a segment of data:
```c
void* pMappedFrames;
ma_uint64 frameCount = frameCountToTryMapping;
ma_result result = ma_audio_buffer_map(pAudioBuffer, &pMappedFrames, &frameCount);
if (result == MA_SUCCESS) {
// Map was successful. The value in frameCount will be how many frames were _actually_ mapped, which may be
// less due to the end of the buffer being reached.
ma_copy_pcm_frames(pFramesOut, pMappedFrames, frameCount, pAudioBuffer->format, pAudioBuffer->channels);
// You must unmap the buffer.
ma_audio_buffer_unmap(pAudioBuffer, frameCount);
}
```
When you use memory mapping, the read cursor is increment by the frame count passed in to
`ma_audio_buffer_unmap()`. If you decide not to process every frame you can pass in a value smaller
than the value returned by `ma_audio_buffer_map()`. The disadvantage to using memory mapping is
that it does not handle looping for you. You can determine if the buffer is at the end for the
purpose of looping with `ma_audio_buffer_at_end()` or by inspecting the return value of
`ma_audio_buffer_unmap()` and checking if it equals `MA_AT_END`. You should not treat `MA_AT_END`
as an error when returned by `ma_audio_buffer_unmap()`.
14. Ring Buffers
================
miniaudio supports lock free (single producer, single consumer) ring buffers which are exposed via
the `ma_rb` and `ma_pcm_rb` APIs. The `ma_rb` API operates on bytes, whereas the `ma_pcm_rb`
operates on PCM frames. They are otherwise identical as `ma_pcm_rb` is just a wrapper around
`ma_rb`.
Unlike most other APIs in miniaudio, ring buffers support both interleaved and deinterleaved
streams. The caller can also allocate their own backing memory for the ring buffer to use
internally for added flexibility. Otherwise the ring buffer will manage it's internal memory for
you.
The examples below use the PCM frame variant of the ring buffer since that's most likely the one
you will want to use. To initialize a ring buffer, do something like the following:
```c
ma_pcm_rb rb;
ma_result result = ma_pcm_rb_init(FORMAT, CHANNELS, BUFFER_SIZE_IN_FRAMES, NULL, NULL, &rb);
if (result != MA_SUCCESS) {
// Error
}
```
The `ma_pcm_rb_init()` function takes the sample format and channel count as parameters because
it's the PCM varient of the ring buffer API. For the regular ring buffer that operates on bytes you
would call `ma_rb_init()` which leaves these out and just takes the size of the buffer in bytes
instead of frames. The fourth parameter is an optional pre-allocated buffer and the fifth parameter
is a pointer to a `ma_allocation_callbacks` structure for custom memory allocation routines.
Passing in `NULL` for this results in `MA_MALLOC()` and `MA_FREE()` being used.
Use `ma_pcm_rb_init_ex()` if you need a deinterleaved buffer. The data for each sub-buffer is
offset from each other based on the stride. To manage your sub-buffers you can use
`ma_pcm_rb_get_subbuffer_stride()`, `ma_pcm_rb_get_subbuffer_offset()` and
`ma_pcm_rb_get_subbuffer_ptr()`.
Use `ma_pcm_rb_acquire_read()` and `ma_pcm_rb_acquire_write()` to retrieve a pointer to a section
of the ring buffer. You specify the number of frames you need, and on output it will set to what
was actually acquired. If the read or write pointer is positioned such that the number of frames
requested will require a loop, it will be clamped to the end of the buffer. Therefore, the number
of frames you're given may be less than the number you requested.
After calling `ma_pcm_rb_acquire_read()` or `ma_pcm_rb_acquire_write()`, you do your work on the
buffer and then "commit" it with `ma_pcm_rb_commit_read()` or `ma_pcm_rb_commit_write()`. This is
where the read/write pointers are updated. When you commit you need to pass in the buffer that was
returned by the earlier call to `ma_pcm_rb_acquire_read()` or `ma_pcm_rb_acquire_write()` and is
only used for validation. The number of frames passed to `ma_pcm_rb_commit_read()` and
`ma_pcm_rb_commit_write()` is what's used to increment the pointers, and can be less that what was
originally requested.
If you want to correct for drift between the write pointer and the read pointer you can use a
combination of `ma_pcm_rb_pointer_distance()`, `ma_pcm_rb_seek_read()` and
`ma_pcm_rb_seek_write()`. Note that you can only move the pointers forward, and you should only
move the read pointer forward via the consumer thread, and the write pointer forward by the
producer thread. If there is too much space between the pointers, move the read pointer forward. If
there is too little space between the pointers, move the write pointer forward.
You can use a ring buffer at the byte level instead of the PCM frame level by using the `ma_rb`
API. This is exactly the same, only you will use the `ma_rb` functions instead of `ma_pcm_rb` and
instead of frame counts you will pass around byte counts.
The maximum size of the buffer in bytes is `0x7FFFFFFF-(MA_SIMD_ALIGNMENT-1)` due to the most
significant bit being used to encode a loop flag and the internally managed buffers always being
aligned to `MA_SIMD_ALIGNMENT`.
Note that the ring buffer is only thread safe when used by a single consumer thread and single
producer thread.
15. Backends
============
The following backends are supported by miniaudio.
+-------------+-----------------------+--------------------------------------------------------+
| Name | Enum Name | Supported Operating Systems |
+-------------+-----------------------+--------------------------------------------------------+
| WASAPI | ma_backend_wasapi | Windows Vista+ |
| DirectSound | ma_backend_dsound | Windows XP+ |
| WinMM | ma_backend_winmm | Windows XP+ (may work on older versions, but untested) |
| Core Audio | ma_backend_coreaudio | macOS, iOS |
| ALSA | ma_backend_alsa | Linux |
| PulseAudio | ma_backend_pulseaudio | Cross Platform (disabled on Windows, BSD and Android) |
| JACK | ma_backend_jack | Cross Platform (disabled on BSD and Android) |
| sndio | ma_backend_sndio | OpenBSD |
| audio(4) | ma_backend_audio4 | NetBSD, OpenBSD |
| OSS | ma_backend_oss | FreeBSD |
| AAudio | ma_backend_aaudio | Android 8+ |
| OpenSL ES | ma_backend_opensl | Android (API level 16+) |
| Web Audio | ma_backend_webaudio | Web (via Emscripten) |
| Custom | ma_backend_custom | Cross Platform |
| Null | ma_backend_null | Cross Platform (not used on Web) |
+-------------+-----------------------+--------------------------------------------------------+
Some backends have some nuance details you may want to be aware of.
15.1. WASAPI
------------
- Low-latency shared mode will be disabled when using an application-defined sample rate which is
different to the device's native sample rate. To work around this, set `wasapi.noAutoConvertSRC`
to true in the device config. This is due to IAudioClient3_InitializeSharedAudioStream() failing
when the `AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM` flag is specified. Setting wasapi.noAutoConvertSRC
will result in miniaudio's internal resampler being used instead which will in turn enable the
use of low-latency shared mode.
15.2. PulseAudio
----------------
- If you experience bad glitching/noise on Arch Linux, consider this fix from the Arch wiki:
https://wiki.archlinux.org/index.php/PulseAudio/Troubleshooting#Glitches,_skips_or_crackling.
Alternatively, consider using a different backend such as ALSA.
15.3. Android
-------------
- To capture audio on Android, remember to add the RECORD_AUDIO permission to your manifest:
`<uses-permission android:name="android.permission.RECORD_AUDIO" />`
- With OpenSL|ES, only a single ma_context can be active at any given time. This is due to a
limitation with OpenSL|ES.
- With AAudio, only default devices are enumerated. This is due to AAudio not having an enumeration
API (devices are enumerated through Java). You can however perform your own device enumeration
through Java and then set the ID in the ma_device_id structure (ma_device_id.aaudio) and pass it
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
#endif
} ma_log;
MA_API ma_result ma_log_init(const ma_allocation_callbacks* pAllocationCallbacks, ma_log* pLog);
MA_API void ma_log_uninit(ma_log* pLog);
MA_API ma_result ma_log_register_callback(ma_log* pLog, ma_log_callback callback);
MA_API ma_result ma_log_unregister_callback(ma_log* pLog, ma_log_callback callback);
MA_API ma_result ma_log_post(ma_log* pLog, ma_uint32 level, const char* pMessage);
MA_API ma_result ma_log_postv(ma_log* pLog, ma_uint32 level, const char* pFormat, va_list args);
MA_API ma_result ma_log_postf(ma_log* pLog, ma_uint32 level, const char* pFormat, ...) MA_ATTRIBUTE_FORMAT(3, 4);
/**************************************************************************************************************************************************************
Biquad Filtering
**************************************************************************************************************************************************************/
typedef union
{
float f32;
ma_int32 s32;
} ma_biquad_coefficient;
typedef struct
{
ma_format format;
ma_uint32 channels;
double b0;
double b1;
double b2;
double a0;
double a1;
double a2;
} ma_biquad_config;
MA_API ma_biquad_config ma_biquad_config_init(ma_format format, ma_uint32 channels, double b0, double b1, double b2, double a0, double a1, double a2);
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_biquad_coefficient b0;
ma_biquad_coefficient b1;
ma_biquad_coefficient b2;
ma_biquad_coefficient a1;
ma_biquad_coefficient a2;
ma_biquad_coefficient* pR1;
ma_biquad_coefficient* pR2;
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
} ma_biquad;
MA_API ma_result ma_biquad_get_heap_size(const ma_biquad_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_biquad_init_preallocated(const ma_biquad_config* pConfig, void* pHeap, ma_biquad* pBQ);
MA_API ma_result ma_biquad_init(const ma_biquad_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_biquad* pBQ);
MA_API void ma_biquad_uninit(ma_biquad* pBQ, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_biquad_reinit(const ma_biquad_config* pConfig, ma_biquad* pBQ);
MA_API ma_result ma_biquad_clear_cache(ma_biquad* pBQ);
MA_API ma_result ma_biquad_process_pcm_frames(ma_biquad* pBQ, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_biquad_get_latency(const ma_biquad* pBQ);
/**************************************************************************************************************************************************************
Low-Pass Filtering
**************************************************************************************************************************************************************/
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
double cutoffFrequency;
double q;
} ma_lpf1_config, ma_lpf2_config;
MA_API ma_lpf1_config ma_lpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency);
MA_API ma_lpf2_config ma_lpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q);
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_biquad_coefficient a;
ma_biquad_coefficient* pR1;
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
} ma_lpf1;
MA_API ma_result ma_lpf1_get_heap_size(const ma_lpf1_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_lpf1_init_preallocated(const ma_lpf1_config* pConfig, void* pHeap, ma_lpf1* pLPF);
MA_API ma_result ma_lpf1_init(const ma_lpf1_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf1* pLPF);
MA_API void ma_lpf1_uninit(ma_lpf1* pLPF, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_lpf1_reinit(const ma_lpf1_config* pConfig, ma_lpf1* pLPF);
MA_API ma_result ma_lpf1_clear_cache(ma_lpf1* pLPF);
MA_API ma_result ma_lpf1_process_pcm_frames(ma_lpf1* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_lpf1_get_latency(const ma_lpf1* pLPF);
typedef struct
{
ma_biquad bq; /* The second order low-pass filter is implemented as a biquad filter. */
} ma_lpf2;
MA_API ma_result ma_lpf2_get_heap_size(const ma_lpf2_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_lpf2_init_preallocated(const ma_lpf2_config* pConfig, void* pHeap, ma_lpf2* pHPF);
MA_API ma_result ma_lpf2_init(const ma_lpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf2* pLPF);
MA_API void ma_lpf2_uninit(ma_lpf2* pLPF, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_lpf2_reinit(const ma_lpf2_config* pConfig, ma_lpf2* pLPF);
MA_API ma_result ma_lpf2_clear_cache(ma_lpf2* pLPF);
MA_API ma_result ma_lpf2_process_pcm_frames(ma_lpf2* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_lpf2_get_latency(const ma_lpf2* pLPF);
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
double cutoffFrequency;
ma_uint32 order; /* If set to 0, will be treated as a passthrough (no filtering will be applied). */
} ma_lpf_config;
MA_API ma_lpf_config ma_lpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order);
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
ma_uint32 lpf1Count;
ma_uint32 lpf2Count;
ma_lpf1* pLPF1;
ma_lpf2* pLPF2;
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
} ma_lpf;
MA_API ma_result ma_lpf_get_heap_size(const ma_lpf_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_lpf_init_preallocated(const ma_lpf_config* pConfig, void* pHeap, ma_lpf* pLPF);
MA_API ma_result ma_lpf_init(const ma_lpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf* pLPF);
MA_API void ma_lpf_uninit(ma_lpf* pLPF, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_lpf_reinit(const ma_lpf_config* pConfig, ma_lpf* pLPF);
MA_API ma_result ma_lpf_clear_cache(ma_lpf* pLPF);
MA_API ma_result ma_lpf_process_pcm_frames(ma_lpf* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_lpf_get_latency(const ma_lpf* pLPF);
/**************************************************************************************************************************************************************
High-Pass Filtering
**************************************************************************************************************************************************************/
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
double cutoffFrequency;
double q;
} ma_hpf1_config, ma_hpf2_config;
MA_API ma_hpf1_config ma_hpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency);
MA_API ma_hpf2_config ma_hpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q);
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_biquad_coefficient a;
ma_biquad_coefficient* pR1;
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
} ma_hpf1;
MA_API ma_result ma_hpf1_get_heap_size(const ma_hpf1_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_hpf1_init_preallocated(const ma_hpf1_config* pConfig, void* pHeap, ma_hpf1* pLPF);
MA_API ma_result ma_hpf1_init(const ma_hpf1_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf1* pHPF);
MA_API void ma_hpf1_uninit(ma_hpf1* pHPF, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_hpf1_reinit(const ma_hpf1_config* pConfig, ma_hpf1* pHPF);
MA_API ma_result ma_hpf1_process_pcm_frames(ma_hpf1* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_hpf1_get_latency(const ma_hpf1* pHPF);
typedef struct
{
ma_biquad bq; /* The second order high-pass filter is implemented as a biquad filter. */
} ma_hpf2;
MA_API ma_result ma_hpf2_get_heap_size(const ma_hpf2_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_hpf2_init_preallocated(const ma_hpf2_config* pConfig, void* pHeap, ma_hpf2* pHPF);
MA_API ma_result ma_hpf2_init(const ma_hpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf2* pHPF);
MA_API void ma_hpf2_uninit(ma_hpf2* pHPF, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_hpf2_reinit(const ma_hpf2_config* pConfig, ma_hpf2* pHPF);
MA_API ma_result ma_hpf2_process_pcm_frames(ma_hpf2* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_hpf2_get_latency(const ma_hpf2* pHPF);
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
double cutoffFrequency;
ma_uint32 order; /* If set to 0, will be treated as a passthrough (no filtering will be applied). */
} ma_hpf_config;
MA_API ma_hpf_config ma_hpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order);
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
ma_uint32 hpf1Count;
ma_uint32 hpf2Count;
ma_hpf1* pHPF1;
ma_hpf2* pHPF2;
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
} ma_hpf;
MA_API ma_result ma_hpf_get_heap_size(const ma_hpf_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_hpf_init_preallocated(const ma_hpf_config* pConfig, void* pHeap, ma_hpf* pLPF);
MA_API ma_result ma_hpf_init(const ma_hpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf* pHPF);
MA_API void ma_hpf_uninit(ma_hpf* pHPF, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_hpf_reinit(const ma_hpf_config* pConfig, ma_hpf* pHPF);
MA_API ma_result ma_hpf_process_pcm_frames(ma_hpf* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_hpf_get_latency(const ma_hpf* pHPF);
/**************************************************************************************************************************************************************
Band-Pass Filtering
**************************************************************************************************************************************************************/
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
double cutoffFrequency;
double q;
} ma_bpf2_config;
MA_API ma_bpf2_config ma_bpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q);
typedef struct
{
ma_biquad bq; /* The second order band-pass filter is implemented as a biquad filter. */
} ma_bpf2;
MA_API ma_result ma_bpf2_get_heap_size(const ma_bpf2_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_bpf2_init_preallocated(const ma_bpf2_config* pConfig, void* pHeap, ma_bpf2* pBPF);
MA_API ma_result ma_bpf2_init(const ma_bpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf2* pBPF);
MA_API void ma_bpf2_uninit(ma_bpf2* pBPF, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_bpf2_reinit(const ma_bpf2_config* pConfig, ma_bpf2* pBPF);
MA_API ma_result ma_bpf2_process_pcm_frames(ma_bpf2* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_bpf2_get_latency(const ma_bpf2* pBPF);
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
double cutoffFrequency;
ma_uint32 order; /* If set to 0, will be treated as a passthrough (no filtering will be applied). */
} ma_bpf_config;
MA_API ma_bpf_config ma_bpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order);
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 bpf2Count;
ma_bpf2* pBPF2;
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
} ma_bpf;
MA_API ma_result ma_bpf_get_heap_size(const ma_bpf_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_bpf_init_preallocated(const ma_bpf_config* pConfig, void* pHeap, ma_bpf* pBPF);
MA_API ma_result ma_bpf_init(const ma_bpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf* pBPF);
MA_API void ma_bpf_uninit(ma_bpf* pBPF, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_bpf_reinit(const ma_bpf_config* pConfig, ma_bpf* pBPF);
MA_API ma_result ma_bpf_process_pcm_frames(ma_bpf* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_bpf_get_latency(const ma_bpf* pBPF);
/**************************************************************************************************************************************************************
Notching Filter
**************************************************************************************************************************************************************/
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
double q;
double frequency;
} ma_notch2_config, ma_notch_config;
MA_API ma_notch2_config ma_notch2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double q, double frequency);
typedef struct
{
ma_biquad bq;
} ma_notch2;
MA_API ma_result ma_notch2_get_heap_size(const ma_notch2_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_notch2_init_preallocated(const ma_notch2_config* pConfig, void* pHeap, ma_notch2* pFilter);
MA_API ma_result ma_notch2_init(const ma_notch2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_notch2* pFilter);
MA_API void ma_notch2_uninit(ma_notch2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_notch2_reinit(const ma_notch2_config* pConfig, ma_notch2* pFilter);
MA_API ma_result ma_notch2_process_pcm_frames(ma_notch2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_notch2_get_latency(const ma_notch2* pFilter);
/**************************************************************************************************************************************************************
Peaking EQ Filter
**************************************************************************************************************************************************************/
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
double gainDB;
double q;
double frequency;
} ma_peak2_config, ma_peak_config;
MA_API ma_peak2_config ma_peak2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency);
typedef struct
{
ma_biquad bq;
} ma_peak2;
MA_API ma_result ma_peak2_get_heap_size(const ma_peak2_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_peak2_init_preallocated(const ma_peak2_config* pConfig, void* pHeap, ma_peak2* pFilter);
MA_API ma_result ma_peak2_init(const ma_peak2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_peak2* pFilter);
MA_API void ma_peak2_uninit(ma_peak2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_peak2_reinit(const ma_peak2_config* pConfig, ma_peak2* pFilter);
MA_API ma_result ma_peak2_process_pcm_frames(ma_peak2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_peak2_get_latency(const ma_peak2* pFilter);
/**************************************************************************************************************************************************************
Low Shelf Filter
**************************************************************************************************************************************************************/
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
double gainDB;
double shelfSlope;
double frequency;
} ma_loshelf2_config, ma_loshelf_config;
MA_API ma_loshelf2_config ma_loshelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency);
typedef struct
{
ma_biquad bq;
} ma_loshelf2;
MA_API ma_result ma_loshelf2_get_heap_size(const ma_loshelf2_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_loshelf2_init_preallocated(const ma_loshelf2_config* pConfig, void* pHeap, ma_loshelf2* pFilter);
MA_API ma_result ma_loshelf2_init(const ma_loshelf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_loshelf2* pFilter);
MA_API void ma_loshelf2_uninit(ma_loshelf2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_loshelf2_reinit(const ma_loshelf2_config* pConfig, ma_loshelf2* pFilter);
MA_API ma_result ma_loshelf2_process_pcm_frames(ma_loshelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_loshelf2_get_latency(const ma_loshelf2* pFilter);
/**************************************************************************************************************************************************************
High Shelf Filter
**************************************************************************************************************************************************************/
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
double gainDB;
double shelfSlope;
double frequency;
} ma_hishelf2_config, ma_hishelf_config;
MA_API ma_hishelf2_config ma_hishelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency);
typedef struct
{
ma_biquad bq;
} ma_hishelf2;
MA_API ma_result ma_hishelf2_get_heap_size(const ma_hishelf2_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_hishelf2_init_preallocated(const ma_hishelf2_config* pConfig, void* pHeap, ma_hishelf2* pFilter);
MA_API ma_result ma_hishelf2_init(const ma_hishelf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hishelf2* pFilter);
MA_API void ma_hishelf2_uninit(ma_hishelf2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_hishelf2_reinit(const ma_hishelf2_config* pConfig, ma_hishelf2* pFilter);
MA_API ma_result ma_hishelf2_process_pcm_frames(ma_hishelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_hishelf2_get_latency(const ma_hishelf2* pFilter);
/*
Delay
*/
typedef struct
{
ma_uint32 channels;
ma_uint32 sampleRate;
ma_uint32 delayInFrames;
ma_bool32 delayStart; /* Set to true to delay the start of the output; false otherwise. */
float wet; /* 0..1. Default = 1. */
float dry; /* 0..1. Default = 1. */
float decay; /* 0..1. Default = 0 (no feedback). Feedback decay. Use this for echo. */
} ma_delay_config;
MA_API ma_delay_config ma_delay_config_init(ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 delayInFrames, float decay);
typedef struct
{
ma_delay_config config;
ma_uint32 cursor; /* Feedback is written to this cursor. Always equal or in front of the read cursor. */
ma_uint32 bufferSizeInFrames; /* The maximum of config.startDelayInFrames and config.feedbackDelayInFrames. */
float* pBuffer;
} ma_delay;
MA_API ma_result ma_delay_init(const ma_delay_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_delay* pDelay);
MA_API void ma_delay_uninit(ma_delay* pDelay, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_delay_process_pcm_frames(ma_delay* pDelay, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount);
MA_API void ma_delay_set_wet(ma_delay* pDelay, float value);
MA_API float ma_delay_get_wet(const ma_delay* pDelay);
MA_API void ma_delay_set_dry(ma_delay* pDelay, float value);
MA_API float ma_delay_get_dry(const ma_delay* pDelay);
MA_API void ma_delay_set_decay(ma_delay* pDelay, float value);
MA_API float ma_delay_get_decay(const ma_delay* pDelay);
/* Gainer for smooth volume changes. */
typedef struct
{
ma_uint32 channels;
ma_uint32 smoothTimeInFrames;
} ma_gainer_config;
MA_API ma_gainer_config ma_gainer_config_init(ma_uint32 channels, ma_uint32 smoothTimeInFrames);
typedef struct
{
ma_gainer_config config;
ma_uint32 t;
float* pOldGains;
float* pNewGains;
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
} ma_gainer;
MA_API ma_result ma_gainer_get_heap_size(const ma_gainer_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_gainer_init_preallocated(const ma_gainer_config* pConfig, void* pHeap, ma_gainer* pGainer);
MA_API ma_result ma_gainer_init(const ma_gainer_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_gainer* pGainer);
MA_API void ma_gainer_uninit(ma_gainer* pGainer, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_gainer_process_pcm_frames(ma_gainer* pGainer, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_result ma_gainer_set_gain(ma_gainer* pGainer, float newGain);
MA_API ma_result ma_gainer_set_gains(ma_gainer* pGainer, float* pNewGains);
/* Stereo panner. */
typedef enum
{
ma_pan_mode_balance = 0, /* Does not blend one side with the other. Technically just a balance. Compatible with other popular audio engines and therefore the default. */
ma_pan_mode_pan /* A true pan. The sound from one side will "move" to the other side and blend with it. */
} ma_pan_mode;
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_pan_mode mode;
float pan;
} ma_panner_config;
MA_API ma_panner_config ma_panner_config_init(ma_format format, ma_uint32 channels);
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_pan_mode mode;
float pan; /* -1..1 where 0 is no pan, -1 is left side, +1 is right side. Defaults to 0. */
} ma_panner;
MA_API ma_result ma_panner_init(const ma_panner_config* pConfig, ma_panner* pPanner);
MA_API ma_result ma_panner_process_pcm_frames(ma_panner* pPanner, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API void ma_panner_set_mode(ma_panner* pPanner, ma_pan_mode mode);
MA_API ma_pan_mode ma_panner_get_mode(const ma_panner* pPanner);
MA_API void ma_panner_set_pan(ma_panner* pPanner, float pan);
MA_API float ma_panner_get_pan(const ma_panner* pPanner);
/* Fader. */
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
} ma_fader_config;
MA_API ma_fader_config ma_fader_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate);
typedef struct
{
ma_fader_config config;
float volumeBeg; /* If volumeBeg and volumeEnd is equal to 1, no fading happens (ma_fader_process_pcm_frames() will run as a passthrough). */
float volumeEnd;
ma_uint64 lengthInFrames; /* The total length of the fade. */
ma_uint64 cursorInFrames; /* The current time in frames. Incremented by ma_fader_process_pcm_frames(). */
} ma_fader;
MA_API ma_result ma_fader_init(const ma_fader_config* pConfig, ma_fader* pFader);
MA_API ma_result ma_fader_process_pcm_frames(ma_fader* pFader, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API void ma_fader_get_data_format(const ma_fader* pFader, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate);
MA_API void ma_fader_set_fade(ma_fader* pFader, float volumeBeg, float volumeEnd, ma_uint64 lengthInFrames);
MA_API float ma_fader_get_current_volume(ma_fader* pFader);
/* Spatializer. */
typedef struct
{
float x;
float y;
float z;
} ma_vec3f;
typedef enum
{
ma_attenuation_model_none, /* No distance attenuation and no spatialization. */
ma_attenuation_model_inverse, /* Equivalent to OpenAL's AL_INVERSE_DISTANCE_CLAMPED. */
ma_attenuation_model_linear, /* Linear attenuation. Equivalent to OpenAL's AL_LINEAR_DISTANCE_CLAMPED. */
ma_attenuation_model_exponential /* Exponential attenuation. Equivalent to OpenAL's AL_EXPONENT_DISTANCE_CLAMPED. */
} ma_attenuation_model;
typedef enum
{
ma_positioning_absolute,
ma_positioning_relative
} ma_positioning;
typedef enum
{
ma_handedness_right,
ma_handedness_left
} ma_handedness;
typedef struct
{
ma_uint32 channelsOut;
ma_channel* pChannelMapOut;
ma_handedness handedness; /* Defaults to right. Forward is -1 on the Z axis. In a left handed system, forward is +1 on the Z axis. */
float coneInnerAngleInRadians;
float coneOuterAngleInRadians;
float coneOuterGain;
float speedOfSound;
ma_vec3f worldUp;
} ma_spatializer_listener_config;
MA_API ma_spatializer_listener_config ma_spatializer_listener_config_init(ma_uint32 channelsOut);
typedef struct
{
ma_spatializer_listener_config config;
ma_vec3f position; /* The absolute position of the listener. */
ma_vec3f direction; /* The direction the listener is facing. The world up vector is config.worldUp. */
ma_vec3f velocity;
ma_bool32 isEnabled;
/* Memory management. */
ma_bool32 _ownsHeap;
void* _pHeap;
} ma_spatializer_listener;
MA_API ma_result ma_spatializer_listener_get_heap_size(const ma_spatializer_listener_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_spatializer_listener_init_preallocated(const ma_spatializer_listener_config* pConfig, void* pHeap, ma_spatializer_listener* pListener);
MA_API ma_result ma_spatializer_listener_init(const ma_spatializer_listener_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_spatializer_listener* pListener);
MA_API void ma_spatializer_listener_uninit(ma_spatializer_listener* pListener, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_channel* ma_spatializer_listener_get_channel_map(ma_spatializer_listener* pListener);
MA_API void ma_spatializer_listener_set_cone(ma_spatializer_listener* pListener, float innerAngleInRadians, float outerAngleInRadians, float outerGain);
MA_API void ma_spatializer_listener_get_cone(const ma_spatializer_listener* pListener, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain);
MA_API void ma_spatializer_listener_set_position(ma_spatializer_listener* pListener, float x, float y, float z);
MA_API ma_vec3f ma_spatializer_listener_get_position(const ma_spatializer_listener* pListener);
MA_API void ma_spatializer_listener_set_direction(ma_spatializer_listener* pListener, float x, float y, float z);
MA_API ma_vec3f ma_spatializer_listener_get_direction(const ma_spatializer_listener* pListener);
MA_API void ma_spatializer_listener_set_velocity(ma_spatializer_listener* pListener, float x, float y, float z);
MA_API ma_vec3f ma_spatializer_listener_get_velocity(const ma_spatializer_listener* pListener);
MA_API void ma_spatializer_listener_set_speed_of_sound(ma_spatializer_listener* pListener, float speedOfSound);
MA_API float ma_spatializer_listener_get_speed_of_sound(const ma_spatializer_listener* pListener);
MA_API void ma_spatializer_listener_set_world_up(ma_spatializer_listener* pListener, float x, float y, float z);
MA_API ma_vec3f ma_spatializer_listener_get_world_up(const ma_spatializer_listener* pListener);
MA_API void ma_spatializer_listener_set_enabled(ma_spatializer_listener* pListener, ma_bool32 isEnabled);
MA_API ma_bool32 ma_spatializer_listener_is_enabled(const ma_spatializer_listener* pListener);
typedef struct
{
ma_uint32 channelsIn;
ma_uint32 channelsOut;
ma_channel* pChannelMapIn;
ma_attenuation_model attenuationModel;
ma_positioning positioning;
ma_handedness handedness; /* Defaults to right. Forward is -1 on the Z axis. In a left handed system, forward is +1 on the Z axis. */
float minGain;
float maxGain;
float minDistance;
float maxDistance;
float rolloff;
float coneInnerAngleInRadians;
float coneOuterAngleInRadians;
float coneOuterGain;
float dopplerFactor; /* Set to 0 to disable doppler effect. */
float directionalAttenuationFactor; /* Set to 0 to disable directional attenuation. */
ma_uint32 gainSmoothTimeInFrames; /* When the gain of a channel changes during spatialization, the transition will be linearly interpolated over this number of frames. */
} ma_spatializer_config;
MA_API ma_spatializer_config ma_spatializer_config_init(ma_uint32 channelsIn, ma_uint32 channelsOut);
typedef struct
{
ma_uint32 channelsIn;
ma_uint32 channelsOut;
ma_channel* pChannelMapIn;
ma_attenuation_model attenuationModel;
ma_positioning positioning;
ma_handedness handedness; /* Defaults to right. Forward is -1 on the Z axis. In a left handed system, forward is +1 on the Z axis. */
float minGain;
float maxGain;
float minDistance;
float maxDistance;
float rolloff;
float coneInnerAngleInRadians;
float coneOuterAngleInRadians;
float coneOuterGain;
float dopplerFactor; /* Set to 0 to disable doppler effect. */
float directionalAttenuationFactor; /* Set to 0 to disable directional attenuation. */
ma_uint32 gainSmoothTimeInFrames; /* When the gain of a channel changes during spatialization, the transition will be linearly interpolated over this number of frames. */
ma_vec3f position;
ma_vec3f direction;
ma_vec3f velocity; /* For doppler effect. */
float dopplerPitch; /* Will be updated by ma_spatializer_process_pcm_frames() and can be used by higher level functions to apply a pitch shift for doppler effect. */
ma_gainer gainer; /* For smooth gain transitions. */
float* pNewChannelGainsOut; /* An offset of _pHeap. Used by ma_spatializer_process_pcm_frames() to store new channel gains. The number of elements in this array is equal to config.channelsOut. */
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
} ma_spatializer;
MA_API ma_result ma_spatializer_get_heap_size(const ma_spatializer_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_spatializer_init_preallocated(const ma_spatializer_config* pConfig, void* pHeap, ma_spatializer* pSpatializer);
MA_API ma_result ma_spatializer_init(const ma_spatializer_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_spatializer* pSpatializer);
MA_API void ma_spatializer_uninit(ma_spatializer* pSpatializer, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_spatializer_process_pcm_frames(ma_spatializer* pSpatializer, ma_spatializer_listener* pListener, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_uint32 ma_spatializer_get_input_channels(const ma_spatializer* pSpatializer);
MA_API ma_uint32 ma_spatializer_get_output_channels(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_set_attenuation_model(ma_spatializer* pSpatializer, ma_attenuation_model attenuationModel);
MA_API ma_attenuation_model ma_spatializer_get_attenuation_model(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_set_positioning(ma_spatializer* pSpatializer, ma_positioning positioning);
MA_API ma_positioning ma_spatializer_get_positioning(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_set_rolloff(ma_spatializer* pSpatializer, float rolloff);
MA_API float ma_spatializer_get_rolloff(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_set_min_gain(ma_spatializer* pSpatializer, float minGain);
MA_API float ma_spatializer_get_min_gain(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_set_max_gain(ma_spatializer* pSpatializer, float maxGain);
MA_API float ma_spatializer_get_max_gain(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_set_min_distance(ma_spatializer* pSpatializer, float minDistance);
MA_API float ma_spatializer_get_min_distance(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_set_max_distance(ma_spatializer* pSpatializer, float maxDistance);
MA_API float ma_spatializer_get_max_distance(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_set_cone(ma_spatializer* pSpatializer, float innerAngleInRadians, float outerAngleInRadians, float outerGain);
MA_API void ma_spatializer_get_cone(const ma_spatializer* pSpatializer, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain);
MA_API void ma_spatializer_set_doppler_factor(ma_spatializer* pSpatializer, float dopplerFactor);
MA_API float ma_spatializer_get_doppler_factor(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_set_directional_attenuation_factor(ma_spatializer* pSpatializer, float directionalAttenuationFactor);
MA_API float ma_spatializer_get_directional_attenuation_factor(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_set_position(ma_spatializer* pSpatializer, float x, float y, float z);
MA_API ma_vec3f ma_spatializer_get_position(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_set_direction(ma_spatializer* pSpatializer, float x, float y, float z);
MA_API ma_vec3f ma_spatializer_get_direction(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_set_velocity(ma_spatializer* pSpatializer, float x, float y, float z);
MA_API ma_vec3f ma_spatializer_get_velocity(const ma_spatializer* pSpatializer);
MA_API void ma_spatializer_get_relative_position_and_direction(const ma_spatializer* pSpatializer, const ma_spatializer_listener* pListener, ma_vec3f* pRelativePos, ma_vec3f* pRelativeDir);
/************************************************************************************************************************************************************
*************************************************************************************************************************************************************
DATA CONVERSION
===============
This section contains the APIs for data conversion. You will find everything here for channel mapping, sample format conversion, resampling, etc.
*************************************************************************************************************************************************************
************************************************************************************************************************************************************/
/**************************************************************************************************************************************************************
Resampling
**************************************************************************************************************************************************************/
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRateIn;
ma_uint32 sampleRateOut;
ma_uint32 lpfOrder; /* The low-pass filter order. Setting this to 0 will disable low-pass filtering. */
double lpfNyquistFactor; /* 0..1. Defaults to 1. 1 = Half the sampling frequency (Nyquist Frequency), 0.5 = Quarter the sampling frequency (half Nyquest Frequency), etc. */
} ma_linear_resampler_config;
MA_API ma_linear_resampler_config ma_linear_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
typedef struct
{
ma_linear_resampler_config config;
ma_uint32 inAdvanceInt;
ma_uint32 inAdvanceFrac;
ma_uint32 inTimeInt;
ma_uint32 inTimeFrac;
union
{
float* f32;
ma_int16* s16;
} x0; /* The previous input frame. */
union
{
float* f32;
ma_int16* s16;
} x1; /* The next input frame. */
ma_lpf lpf;
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
} ma_linear_resampler;
MA_API ma_result ma_linear_resampler_get_heap_size(const ma_linear_resampler_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_linear_resampler_init_preallocated(const ma_linear_resampler_config* pConfig, void* pHeap, ma_linear_resampler* pResampler);
MA_API ma_result ma_linear_resampler_init(const ma_linear_resampler_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_linear_resampler* pResampler);
MA_API void ma_linear_resampler_uninit(ma_linear_resampler* pResampler, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_linear_resampler_process_pcm_frames(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut);
MA_API ma_result ma_linear_resampler_set_rate(ma_linear_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
MA_API ma_result ma_linear_resampler_set_rate_ratio(ma_linear_resampler* pResampler, float ratioInOut);
MA_API ma_uint64 ma_linear_resampler_get_input_latency(const ma_linear_resampler* pResampler);
MA_API ma_uint64 ma_linear_resampler_get_output_latency(const ma_linear_resampler* pResampler);
MA_API ma_result ma_linear_resampler_get_required_input_frame_count(const ma_linear_resampler* pResampler, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount);
MA_API ma_result ma_linear_resampler_get_expected_output_frame_count(const ma_linear_resampler* pResampler, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount);
MA_API ma_result ma_linear_resampler_reset(ma_linear_resampler* pResampler);
typedef struct ma_resampler_config ma_resampler_config;
typedef void ma_resampling_backend;
typedef struct
{
ma_result (* onGetHeapSize )(void* pUserData, const ma_resampler_config* pConfig, size_t* pHeapSizeInBytes);
ma_result (* onInit )(void* pUserData, const ma_resampler_config* pConfig, void* pHeap, ma_resampling_backend** ppBackend);
void (* onUninit )(void* pUserData, ma_resampling_backend* pBackend, const ma_allocation_callbacks* pAllocationCallbacks);
ma_result (* onProcess )(void* pUserData, ma_resampling_backend* pBackend, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut);
ma_result (* onSetRate )(void* pUserData, ma_resampling_backend* pBackend, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut); /* Optional. Rate changes will be disabled. */
ma_uint64 (* onGetInputLatency )(void* pUserData, const ma_resampling_backend* pBackend); /* Optional. Latency will be reported as 0. */
ma_uint64 (* onGetOutputLatency )(void* pUserData, const ma_resampling_backend* pBackend); /* Optional. Latency will be reported as 0. */
ma_result (* onGetRequiredInputFrameCount )(void* pUserData, const ma_resampling_backend* pBackend, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount); /* Optional. Latency mitigation will be disabled. */
ma_result (* onGetExpectedOutputFrameCount)(void* pUserData, const ma_resampling_backend* pBackend, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount); /* Optional. Latency mitigation will be disabled. */
ma_result (* onReset )(void* pUserData, ma_resampling_backend* pBackend);
} ma_resampling_backend_vtable;
typedef enum
{
ma_resample_algorithm_linear = 0, /* Fastest, lowest quality. Optional low-pass filtering. Default. */
ma_resample_algorithm_custom,
} ma_resample_algorithm;
struct ma_resampler_config
{
ma_format format; /* Must be either ma_format_f32 or ma_format_s16. */
ma_uint32 channels;
ma_uint32 sampleRateIn;
ma_uint32 sampleRateOut;
ma_resample_algorithm algorithm; /* When set to ma_resample_algorithm_custom, pBackendVTable will be used. */
ma_resampling_backend_vtable* pBackendVTable;
void* pBackendUserData;
struct
{
ma_uint32 lpfOrder;
} linear;
};
MA_API ma_resampler_config ma_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_resample_algorithm algorithm);
typedef struct
{
ma_resampling_backend* pBackend;
ma_resampling_backend_vtable* pBackendVTable;
void* pBackendUserData;
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRateIn;
ma_uint32 sampleRateOut;
union
{
ma_linear_resampler linear;
} state; /* State for stock resamplers so we can avoid a malloc. For stock resamplers, pBackend will point here. */
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
} ma_resampler;
MA_API ma_result ma_resampler_get_heap_size(const ma_resampler_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_resampler_init_preallocated(const ma_resampler_config* pConfig, void* pHeap, ma_resampler* pResampler);
/*
Initializes a new resampler object from a config.
*/
MA_API ma_result ma_resampler_init(const ma_resampler_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_resampler* pResampler);
/*
Uninitializes a resampler.
*/
MA_API void ma_resampler_uninit(ma_resampler* pResampler, const ma_allocation_callbacks* pAllocationCallbacks);
/*
Converts the given input data.
Both the input and output frames must be in the format specified in the config when the resampler was initilized.
On input, [pFrameCountOut] contains the number of output frames to process. On output it contains the number of output frames that
were actually processed, which may be less than the requested amount which will happen if there's not enough input data. You can use
ma_resampler_get_expected_output_frame_count() to know how many output frames will be processed for a given number of input frames.
On input, [pFrameCountIn] contains the number of input frames contained in [pFramesIn]. On output it contains the number of whole
input frames that were actually processed. You can use ma_resampler_get_required_input_frame_count() to know how many input frames
you should provide for a given number of output frames. [pFramesIn] can be NULL, in which case zeroes will be used instead.
If [pFramesOut] is NULL, a seek is performed. In this case, if [pFrameCountOut] is not NULL it will seek by the specified number of
output frames. Otherwise, if [pFramesCountOut] is NULL and [pFrameCountIn] is not NULL, it will seek by the specified number of input
frames. When seeking, [pFramesIn] is allowed to NULL, in which case the internal timing state will be updated, but no input will be
processed. In this case, any internal filter state will be updated as if zeroes were passed in.
It is an error for [pFramesOut] to be non-NULL and [pFrameCountOut] to be NULL.
It is an error for both [pFrameCountOut] and [pFrameCountIn] to be NULL.
*/
MA_API ma_result ma_resampler_process_pcm_frames(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut);
/*
Sets the input and output sample sample rate.
*/
MA_API ma_result ma_resampler_set_rate(ma_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
/*
Sets the input and output sample rate as a ratio.
The ration is in/out.
*/
MA_API ma_result ma_resampler_set_rate_ratio(ma_resampler* pResampler, float ratio);
/*
Retrieves the latency introduced by the resampler in input frames.
*/
MA_API ma_uint64 ma_resampler_get_input_latency(const ma_resampler* pResampler);
/*
Retrieves the latency introduced by the resampler in output frames.
*/
MA_API ma_uint64 ma_resampler_get_output_latency(const ma_resampler* pResampler);
/*
Calculates the number of whole input frames that would need to be read from the client in order to output the specified
number of output frames.
The returned value does not include cached input frames. It only returns the number of extra frames that would need to be
read from the input buffer in order to output the specified number of output frames.
*/
MA_API ma_result ma_resampler_get_required_input_frame_count(const ma_resampler* pResampler, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount);
/*
Calculates the number of whole output frames that would be output after fully reading and consuming the specified number of
input frames.
*/
MA_API ma_result ma_resampler_get_expected_output_frame_count(const ma_resampler* pResampler, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount);
/*
Resets the resampler's timer and clears it's internal cache.
*/
MA_API ma_result ma_resampler_reset(ma_resampler* pResampler);
/**************************************************************************************************************************************************************
Channel Conversion
**************************************************************************************************************************************************************/
typedef enum
{
ma_channel_conversion_path_unknown,
ma_channel_conversion_path_passthrough,
ma_channel_conversion_path_mono_out, /* Converting to mono. */
ma_channel_conversion_path_mono_in, /* Converting from mono. */
ma_channel_conversion_path_shuffle, /* Simple shuffle. Will use this when all channels are present in both input and output channel maps, but just in a different order. */
ma_channel_conversion_path_weights /* Blended based on weights. */
} ma_channel_conversion_path;
typedef enum
{
ma_mono_expansion_mode_duplicate = 0, /* The default. */
ma_mono_expansion_mode_average, /* Average the mono channel across all channels. */
ma_mono_expansion_mode_stereo_only, /* Duplicate to the left and right channels only and ignore the others. */
ma_mono_expansion_mode_default = ma_mono_expansion_mode_duplicate
} ma_mono_expansion_mode;
typedef struct
{
ma_format format;
ma_uint32 channelsIn;
ma_uint32 channelsOut;
const ma_channel* pChannelMapIn;
const ma_channel* pChannelMapOut;
ma_channel_mix_mode mixingMode;
float** ppWeights; /* [in][out]. Only used when mixingMode is set to ma_channel_mix_mode_custom_weights. */
} ma_channel_converter_config;
MA_API ma_channel_converter_config ma_channel_converter_config_init(ma_format format, ma_uint32 channelsIn, const ma_channel* pChannelMapIn, ma_uint32 channelsOut, const ma_channel* pChannelMapOut, ma_channel_mix_mode mixingMode);
typedef struct
{
ma_format format;
ma_uint32 channelsIn;
ma_uint32 channelsOut;
ma_channel_mix_mode mixingMode;
ma_channel_conversion_path conversionPath;
ma_channel* pChannelMapIn;
ma_channel* pChannelMapOut;
ma_uint8* pShuffleTable; /* Indexed by output channel index. */
union
{
float** f32;
ma_int32** s16;
} weights; /* [in][out] */
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
} ma_channel_converter;
MA_API ma_result ma_channel_converter_get_heap_size(const ma_channel_converter_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_channel_converter_init_preallocated(const ma_channel_converter_config* pConfig, void* pHeap, ma_channel_converter* pConverter);
MA_API ma_result ma_channel_converter_init(const ma_channel_converter_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_channel_converter* pConverter);
MA_API void ma_channel_converter_uninit(ma_channel_converter* pConverter, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_channel_converter_process_pcm_frames(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount);
MA_API ma_result ma_channel_converter_get_input_channel_map(const ma_channel_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap);
MA_API ma_result ma_channel_converter_get_output_channel_map(const ma_channel_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap);
/**************************************************************************************************************************************************************
Data Conversion
**************************************************************************************************************************************************************/
typedef struct
{
ma_format formatIn;
ma_format formatOut;
ma_uint32 channelsIn;
ma_uint32 channelsOut;
ma_uint32 sampleRateIn;
ma_uint32 sampleRateOut;
ma_channel* pChannelMapIn;
ma_channel* pChannelMapOut;
ma_dither_mode ditherMode;
ma_channel_mix_mode channelMixMode;
float** ppChannelWeights; /* [in][out]. Only used when mixingMode is set to ma_channel_mix_mode_custom_weights. */
ma_bool32 allowDynamicSampleRate;
ma_resampler_config resampling;
} ma_data_converter_config;
MA_API ma_data_converter_config ma_data_converter_config_init_default(void);
MA_API ma_data_converter_config ma_data_converter_config_init(ma_format formatIn, ma_format formatOut, ma_uint32 channelsIn, ma_uint32 channelsOut, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
typedef enum
{
ma_data_converter_execution_path_passthrough, /* No conversion. */
ma_data_converter_execution_path_format_only, /* Only format conversion. */
ma_data_converter_execution_path_channels_only, /* Only channel conversion. */
ma_data_converter_execution_path_resample_only, /* Only resampling. */
ma_data_converter_execution_path_resample_first, /* All conversions, but resample as the first step. */
ma_data_converter_execution_path_channels_first /* All conversions, but channels as the first step. */
} ma_data_converter_execution_path;
typedef struct
{
ma_format formatIn;
ma_format formatOut;
ma_uint32 channelsIn;
ma_uint32 channelsOut;
ma_uint32 sampleRateIn;
ma_uint32 sampleRateOut;
ma_dither_mode ditherMode;
ma_data_converter_execution_path executionPath; /* The execution path the data converter will follow when processing. */
ma_channel_converter channelConverter;
ma_resampler resampler;
ma_bool8 hasPreFormatConversion;
ma_bool8 hasPostFormatConversion;
ma_bool8 hasChannelConverter;
ma_bool8 hasResampler;
ma_bool8 isPassthrough;
/* Memory management. */
ma_bool8 _ownsHeap;
void* _pHeap;
} ma_data_converter;
MA_API ma_result ma_data_converter_get_heap_size(const ma_data_converter_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_data_converter_init_preallocated(const ma_data_converter_config* pConfig, void* pHeap, ma_data_converter* pConverter);
MA_API ma_result ma_data_converter_init(const ma_data_converter_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_converter* pConverter);
MA_API void ma_data_converter_uninit(ma_data_converter* pConverter, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_data_converter_process_pcm_frames(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut);
MA_API ma_result ma_data_converter_set_rate(ma_data_converter* pConverter, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut);
MA_API ma_result ma_data_converter_set_rate_ratio(ma_data_converter* pConverter, float ratioInOut);
MA_API ma_uint64 ma_data_converter_get_input_latency(const ma_data_converter* pConverter);
MA_API ma_uint64 ma_data_converter_get_output_latency(const ma_data_converter* pConverter);
MA_API ma_result ma_data_converter_get_required_input_frame_count(const ma_data_converter* pConverter, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount);
MA_API ma_result ma_data_converter_get_expected_output_frame_count(const ma_data_converter* pConverter, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount);
MA_API ma_result ma_data_converter_get_input_channel_map(const ma_data_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap);
MA_API ma_result ma_data_converter_get_output_channel_map(const ma_data_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap);
MA_API ma_result ma_data_converter_reset(ma_data_converter* pConverter);
/************************************************************************************************************************************************************
Format Conversion
************************************************************************************************************************************************************/
MA_API void ma_pcm_u8_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_u8_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_u8_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_u8_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_s16_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_s16_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_s16_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_s16_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_s24_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_s24_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_s24_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_s24_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_s32_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_s32_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_s32_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_s32_to_f32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_f32_to_u8(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_f32_to_s16(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_f32_to_s24(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_f32_to_s32(void* pOut, const void* pIn, ma_uint64 count, ma_dither_mode ditherMode);
MA_API void ma_pcm_convert(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 sampleCount, ma_dither_mode ditherMode);
MA_API void ma_convert_pcm_frames_format(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 frameCount, ma_uint32 channels, ma_dither_mode ditherMode);
/*
Deinterleaves an interleaved buffer.
*/
MA_API void ma_deinterleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void* pInterleavedPCMFrames, void** ppDeinterleavedPCMFrames);
/*
Interleaves a group of deinterleaved buffers.
*/
MA_API void ma_interleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void** ppDeinterleavedPCMFrames, void* pInterleavedPCMFrames);
/************************************************************************************************************************************************************
Channel Maps
************************************************************************************************************************************************************/
/*
This is used in the shuffle table to indicate that the channel index is undefined and should be ignored.
*/
#define MA_CHANNEL_INDEX_NULL 255
/*
Retrieves the channel position of the specified channel in the given channel map.
The pChannelMap parameter can be null, in which case miniaudio's default channel map will be assumed.
*/
MA_API ma_channel ma_channel_map_get_channel(const ma_channel* pChannelMap, ma_uint32 channelCount, ma_uint32 channelIndex);
/*
Initializes a blank channel map.
When a blank channel map is specified anywhere it indicates that the native channel map should be used.
*/
MA_API void ma_channel_map_init_blank(ma_channel* pChannelMap, ma_uint32 channels);
/*
Helper for retrieving a standard channel map.
The output channel map buffer must have a capacity of at least `channelMapCap`.
*/
MA_API void ma_channel_map_init_standard(ma_standard_channel_map standardChannelMap, ma_channel* pChannelMap, size_t channelMapCap, ma_uint32 channels);
/*
Copies a channel map.
Both input and output channel map buffers must have a capacity of at at least `channels`.
*/
MA_API void ma_channel_map_copy(ma_channel* pOut, const ma_channel* pIn, ma_uint32 channels);
/*
Copies a channel map if one is specified, otherwise copies the default channel map.
The output buffer must have a capacity of at least `channels`. If not NULL, the input channel map must also have a capacity of at least `channels`.
*/
MA_API void ma_channel_map_copy_or_default(ma_channel* pOut, size_t channelMapCapOut, const ma_channel* pIn, ma_uint32 channels);
/*
Determines whether or not a channel map is valid.
A blank channel map is valid (all channels set to MA_CHANNEL_NONE). The way a blank channel map is handled is context specific, but
is usually treated as a passthrough.
Invalid channel maps:
- A channel map with no channels
- A channel map with more than one channel and a mono channel
The channel map buffer must have a capacity of at least `channels`.
*/
MA_API ma_bool32 ma_channel_map_is_valid(const ma_channel* pChannelMap, ma_uint32 channels);
/*
Helper for comparing two channel maps for equality.
This assumes the channel count is the same between the two.
Both channels map buffers must have a capacity of at least `channels`.
*/
MA_API ma_bool32 ma_channel_map_is_equal(const ma_channel* pChannelMapA, const ma_channel* pChannelMapB, ma_uint32 channels);
/*
Helper for determining if a channel map is blank (all channels set to MA_CHANNEL_NONE).
The channel map buffer must have a capacity of at least `channels`.
*/
MA_API ma_bool32 ma_channel_map_is_blank(const ma_channel* pChannelMap, ma_uint32 channels);
/*
Helper for determining whether or not a channel is present in the given channel map.
The channel map buffer must have a capacity of at least `channels`.
*/
MA_API ma_bool32 ma_channel_map_contains_channel_position(ma_uint32 channels, const ma_channel* pChannelMap, ma_channel channelPosition);
/************************************************************************************************************************************************************
Conversion Helpers
************************************************************************************************************************************************************/
/*
High-level helper for doing a full format conversion in one go. Returns the number of output frames. Call this with pOut set to NULL to
determine the required size of the output buffer. frameCountOut should be set to the capacity of pOut. If pOut is NULL, frameCountOut is
ignored.
A return value of 0 indicates an error.
This function is useful for one-off bulk conversions, but if you're streaming data you should use the ma_data_converter APIs instead.
*/
MA_API ma_uint64 ma_convert_frames(void* pOut, ma_uint64 frameCountOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, const void* pIn, ma_uint64 frameCountIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn);
MA_API ma_uint64 ma_convert_frames_ex(void* pOut, ma_uint64 frameCountOut, const void* pIn, ma_uint64 frameCountIn, const ma_data_converter_config* pConfig);
/************************************************************************************************************************************************************
Ring Buffer
************************************************************************************************************************************************************/
typedef struct
{
void* pBuffer;
ma_uint32 subbufferSizeInBytes;
ma_uint32 subbufferCount;
ma_uint32 subbufferStrideInBytes;
MA_ATOMIC(4, ma_uint32) encodedReadOffset; /* Most significant bit is the loop flag. Lower 31 bits contains the actual offset in bytes. Must be used atomically. */
MA_ATOMIC(4, ma_uint32) encodedWriteOffset; /* Most significant bit is the loop flag. Lower 31 bits contains the actual offset in bytes. Must be used atomically. */
ma_bool8 ownsBuffer; /* Used to know whether or not miniaudio is responsible for free()-ing the buffer. */
ma_bool8 clearOnWriteAcquire; /* When set, clears the acquired write buffer before returning from ma_rb_acquire_write(). */
ma_allocation_callbacks allocationCallbacks;
} ma_rb;
MA_API ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size_t subbufferStrideInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB);
MA_API ma_result ma_rb_init(size_t bufferSizeInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB);
MA_API void ma_rb_uninit(ma_rb* pRB);
MA_API void ma_rb_reset(ma_rb* pRB);
MA_API ma_result ma_rb_acquire_read(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut);
MA_API ma_result ma_rb_commit_read(ma_rb* pRB, size_t sizeInBytes);
MA_API ma_result ma_rb_acquire_write(ma_rb* pRB, size_t* pSizeInBytes, void** ppBufferOut);
MA_API ma_result ma_rb_commit_write(ma_rb* pRB, size_t sizeInBytes);
MA_API ma_result ma_rb_seek_read(ma_rb* pRB, size_t offsetInBytes);
MA_API ma_result ma_rb_seek_write(ma_rb* pRB, size_t offsetInBytes);
MA_API ma_int32 ma_rb_pointer_distance(ma_rb* pRB); /* Returns the distance between the write pointer and the read pointer. Should never be negative for a correct program. Will return the number of bytes that can be read before the read pointer hi...
MA_API ma_uint32 ma_rb_available_read(ma_rb* pRB);
MA_API ma_uint32 ma_rb_available_write(ma_rb* pRB);
MA_API size_t ma_rb_get_subbuffer_size(ma_rb* pRB);
MA_API size_t ma_rb_get_subbuffer_stride(ma_rb* pRB);
MA_API size_t ma_rb_get_subbuffer_offset(ma_rb* pRB, size_t subbufferIndex);
MA_API void* ma_rb_get_subbuffer_ptr(ma_rb* pRB, size_t subbufferIndex, void* pBuffer);
typedef struct
{
ma_rb rb;
ma_format format;
ma_uint32 channels;
} ma_pcm_rb;
MA_API ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subbufferSizeInFrames, ma_uint32 subbufferCount, ma_uint32 subbufferStrideInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallba...
MA_API ma_result ma_pcm_rb_init(ma_format format, ma_uint32 channels, ma_uint32 bufferSizeInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB);
MA_API void ma_pcm_rb_uninit(ma_pcm_rb* pRB);
MA_API void ma_pcm_rb_reset(ma_pcm_rb* pRB);
MA_API ma_result ma_pcm_rb_acquire_read(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut);
MA_API ma_result ma_pcm_rb_commit_read(ma_pcm_rb* pRB, ma_uint32 sizeInFrames);
MA_API ma_result ma_pcm_rb_acquire_write(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut);
MA_API ma_result ma_pcm_rb_commit_write(ma_pcm_rb* pRB, ma_uint32 sizeInFrames);
MA_API ma_result ma_pcm_rb_seek_read(ma_pcm_rb* pRB, ma_uint32 offsetInFrames);
MA_API ma_result ma_pcm_rb_seek_write(ma_pcm_rb* pRB, ma_uint32 offsetInFrames);
MA_API ma_int32 ma_pcm_rb_pointer_distance(ma_pcm_rb* pRB); /* Return value is in frames. */
MA_API ma_uint32 ma_pcm_rb_available_read(ma_pcm_rb* pRB);
MA_API ma_uint32 ma_pcm_rb_available_write(ma_pcm_rb* pRB);
MA_API ma_uint32 ma_pcm_rb_get_subbuffer_size(ma_pcm_rb* pRB);
MA_API ma_uint32 ma_pcm_rb_get_subbuffer_stride(ma_pcm_rb* pRB);
MA_API ma_uint32 ma_pcm_rb_get_subbuffer_offset(ma_pcm_rb* pRB, ma_uint32 subbufferIndex);
MA_API void* ma_pcm_rb_get_subbuffer_ptr(ma_pcm_rb* pRB, ma_uint32 subbufferIndex, void* pBuffer);
/*
The idea of the duplex ring buffer is to act as the intermediary buffer when running two asynchronous devices in a duplex set up. The
capture device writes to it, and then a playback device reads from it.
At the moment this is just a simple naive implementation, but in the future I want to implement some dynamic resampling to seamlessly
handle desyncs. Note that the API is work in progress and may change at any time in any version.
The size of the buffer is based on the capture side since that's what'll be written to the buffer. It is based on the capture period size
in frames. The internal sample rate of the capture device is also needed in order to calculate the size.
*/
typedef struct
{
ma_pcm_rb rb;
} ma_duplex_rb;
MA_API ma_result ma_duplex_rb_init(ma_format captureFormat, ma_uint32 captureChannels, ma_uint32 sampleRate, ma_uint32 captureInternalSampleRate, ma_uint32 captureInternalPeriodSizeInFrames, const ma_allocation_callbacks* pAllocationCallbacks, ma_dup...
MA_API ma_result ma_duplex_rb_uninit(ma_duplex_rb* pRB);
/************************************************************************************************************************************************************
Miscellaneous Helpers
************************************************************************************************************************************************************/
/*
Retrieves a human readable description of the given result code.
*/
MA_API const char* ma_result_description(ma_result result);
/*
malloc()
*/
MA_API void* ma_malloc(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks);
/*
calloc()
*/
MA_API void* ma_calloc(size_t sz, const ma_allocation_callbacks* pAllocationCallbacks);
/*
realloc()
*/
MA_API void* ma_realloc(void* p, size_t sz, const ma_allocation_callbacks* pAllocationCallbacks);
/*
free()
*/
MA_API void ma_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks);
/*
Performs an aligned malloc, with the assumption that the alignment is a power of 2.
*/
MA_API void* ma_aligned_malloc(size_t sz, size_t alignment, const ma_allocation_callbacks* pAllocationCallbacks);
/*
Free's an aligned malloc'd buffer.
*/
MA_API void ma_aligned_free(void* p, const ma_allocation_callbacks* pAllocationCallbacks);
/*
Retrieves a friendly name for a format.
*/
MA_API const char* ma_get_format_name(ma_format format);
/*
Blends two frames in floating point format.
*/
MA_API void ma_blend_f32(float* pOut, float* pInA, float* pInB, float factor, ma_uint32 channels);
/*
Retrieves the size of a sample in bytes for the given format.
This API is efficient and is implemented using a lookup table.
Thread Safety: SAFE
This API is pure.
*/
MA_API ma_uint32 ma_get_bytes_per_sample(ma_format format);
static MA_INLINE ma_uint32 ma_get_bytes_per_frame(ma_format format, ma_uint32 channels) { return ma_get_bytes_per_sample(format) * channels; }
/*
Converts a log level to a string.
*/
MA_API const char* ma_log_level_to_string(ma_uint32 logLevel);
/************************************************************************************************************************************************************
Synchronization
************************************************************************************************************************************************************/
/*
Locks a spinlock.
*/
MA_API ma_result ma_spinlock_lock(volatile ma_spinlock* pSpinlock);
/*
Locks a spinlock, but does not yield() when looping.
*/
MA_API ma_result ma_spinlock_lock_noyield(volatile ma_spinlock* pSpinlock);
/*
Unlocks a spinlock.
*/
MA_API ma_result ma_spinlock_unlock(volatile ma_spinlock* pSpinlock);
#ifndef MA_NO_THREADING
/*
Creates a mutex.
A mutex must be created from a valid context. A mutex is initially unlocked.
*/
MA_API ma_result ma_mutex_init(ma_mutex* pMutex);
/*
Deletes a mutex.
*/
MA_API void ma_mutex_uninit(ma_mutex* pMutex);
/*
Locks a mutex with an infinite timeout.
*/
MA_API void ma_mutex_lock(ma_mutex* pMutex);
/*
Unlocks a mutex.
*/
MA_API void ma_mutex_unlock(ma_mutex* pMutex);
/*
Initializes an auto-reset event.
*/
MA_API ma_result ma_event_init(ma_event* pEvent);
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ma_uintptr data1;
} custom;
/* Resource Manager */
union
{
struct
{
/*ma_resource_manager**/ void* pResourceManager;
/*ma_resource_manager_data_buffer_node**/ void* pDataBufferNode;
char* pFilePath;
wchar_t* pFilePathW;
ma_uint32 flags; /* Resource manager data source flags that were used when initializing the data buffer. */
ma_async_notification* pInitNotification; /* Signalled when the data buffer has been initialized and the format/channels/rate can be retrieved. */
ma_async_notification* pDoneNotification; /* Signalled when the data buffer has been fully decoded. Will be passed through to MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_BUFFER_NODE when decoding. */
ma_fence* pInitFence; /* Released when initialization of the decoder is complete. */
ma_fence* pDoneFence; /* Released if initialization of the decoder fails. Passed through to PAGE_DATA_BUFFER_NODE untouched if init is successful. */
} loadDataBufferNode;
struct
{
/*ma_resource_manager**/ void* pResourceManager;
/*ma_resource_manager_data_buffer_node**/ void* pDataBufferNode;
ma_async_notification* pDoneNotification;
ma_fence* pDoneFence;
} freeDataBufferNode;
struct
{
/*ma_resource_manager**/ void* pResourceManager;
/*ma_resource_manager_data_buffer_node**/ void* pDataBufferNode;
/*ma_decoder**/ void* pDecoder;
ma_async_notification* pDoneNotification; /* Signalled when the data buffer has been fully decoded. */
ma_fence* pDoneFence; /* Passed through from LOAD_DATA_BUFFER_NODE and released when the data buffer completes decoding or an error occurs. */
} pageDataBufferNode;
struct
{
/*ma_resource_manager_data_buffer**/ void* pDataBuffer;
ma_async_notification* pInitNotification; /* Signalled when the data buffer has been initialized and the format/channels/rate can be retrieved. */
ma_async_notification* pDoneNotification; /* Signalled when the data buffer has been fully decoded. */
ma_fence* pInitFence; /* Released when the data buffer has been initialized and the format/channels/rate can be retrieved. */
ma_fence* pDoneFence; /* Released when the data buffer has been fully decoded. */
ma_uint64 rangeBegInPCMFrames;
ma_uint64 rangeEndInPCMFrames;
ma_uint64 loopPointBegInPCMFrames;
ma_uint64 loopPointEndInPCMFrames;
ma_uint32 isLooping;
} loadDataBuffer;
struct
{
/*ma_resource_manager_data_buffer**/ void* pDataBuffer;
ma_async_notification* pDoneNotification;
ma_fence* pDoneFence;
} freeDataBuffer;
struct
{
/*ma_resource_manager_data_stream**/ void* pDataStream;
char* pFilePath; /* Allocated when the job is posted, freed by the job thread after loading. */
wchar_t* pFilePathW; /* ^ As above ^. Only used if pFilePath is NULL. */
ma_uint64 initialSeekPoint;
ma_async_notification* pInitNotification; /* Signalled after the first two pages have been decoded and frames can be read from the stream. */
ma_fence* pInitFence;
} loadDataStream;
struct
{
/*ma_resource_manager_data_stream**/ void* pDataStream;
ma_async_notification* pDoneNotification;
ma_fence* pDoneFence;
} freeDataStream;
struct
{
/*ma_resource_manager_data_stream**/ void* pDataStream;
ma_uint32 pageIndex; /* The index of the page to decode into. */
} pageDataStream;
struct
{
/*ma_resource_manager_data_stream**/ void* pDataStream;
ma_uint64 frameIndex;
} seekDataStream;
} resourceManager;
/* Device. */
union
{
union
{
struct
{
/*ma_device**/ void* pDevice;
/*ma_device_type*/ ma_uint32 deviceType;
} reroute;
} aaudio;
} device;
} data;
};
MA_API ma_job ma_job_init(ma_uint16 code);
MA_API ma_result ma_job_process(ma_job* pJob);
/*
When set, ma_job_queue_next() will not wait and no semaphore will be signaled in
ma_job_queue_post(). ma_job_queue_next() will return MA_NO_DATA_AVAILABLE if nothing is available.
This flag should always be used for platforms that do not support multithreading.
*/
typedef enum
{
MA_JOB_QUEUE_FLAG_NON_BLOCKING = 0x00000001
} ma_job_queue_flags;
typedef struct
{
ma_uint32 flags;
ma_uint32 capacity; /* The maximum number of jobs that can fit in the queue at a time. */
} ma_job_queue_config;
MA_API ma_job_queue_config ma_job_queue_config_init(ma_uint32 flags, ma_uint32 capacity);
typedef struct
{
ma_uint32 flags; /* Flags passed in at initialization time. */
ma_uint32 capacity; /* The maximum number of jobs that can fit in the queue at a time. Set by the config. */
MA_ATOMIC(8, ma_uint64) head; /* The first item in the list. Required for removing from the top of the list. */
MA_ATOMIC(8, ma_uint64) tail; /* The last item in the list. Required for appending to the end of the list. */
#ifndef MA_NO_THREADING
ma_semaphore sem; /* Only used when MA_JOB_QUEUE_FLAG_NON_BLOCKING is unset. */
#endif
ma_slot_allocator allocator;
ma_job* pJobs;
#ifndef MA_USE_EXPERIMENTAL_LOCK_FREE_JOB_QUEUE
ma_spinlock lock;
#endif
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
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int _unused;
} stopped;
struct
{
int _unused;
} rerouted;
struct
{
int _unused;
} interruption;
} data;
} ma_device_notification;
/*
The notification callback for when the application should be notified of a change to the device.
This callback is used for notifying the application of changes such as when the device has started,
stopped, rerouted or an interruption has occurred. Note that not all backends will post all
notification types. For example, some backends will perform automatic stream routing without any
kind of notification to the host program which means miniaudio will never know about it and will
never be able to fire the rerouted notification. You should keep this in mind when designing your
program.
The stopped notification will *not* get fired when a device is rerouted.
Parameters
----------
pNotification (in)
A pointer to a structure containing information about the event. Use the `pDevice` member of
this object to retrieve the relevant device. The `type` member can be used to discriminate
against each of the notification types.
Remarks
-------
Do not restart or uninitialize the device from the callback.
Not all notifications will be triggered by all backends, however the started and stopped events
should be reliable for all backends. Some backends do not have a good way to detect device
stoppages due to unplugging the device which may result in the stopped callback not getting
fired. This has been observed with at least one BSD variant.
The rerouted notification is fired *after* the reroute has occurred. The stopped notification will
*not* get fired when a device is rerouted. The following backends are known to do automatic stream
rerouting, but do not have a way to be notified of the change:
* DirectSound
The interruption notifications are used on mobile platforms for detecting when audio is interrupted
due to things like an incoming phone call. Currently this is only implemented on iOS. None of the
Android backends will report this notification.
*/
typedef void (* ma_device_notification_proc)(const ma_device_notification* pNotification);
/*
The callback for processing audio data from the device.
The data callback is fired by miniaudio whenever the device needs to have more data delivered to a playback device, or when a capture device has some data
available. This is called as soon as the backend asks for more data which means it may be called with inconsistent frame counts. You cannot assume the
callback will be fired with a consistent frame count.
Parameters
----------
pDevice (in)
A pointer to the relevant device.
pOutput (out)
A pointer to the output buffer that will receive audio data that will later be played back through the speakers. This will be non-null for a playback or
full-duplex device and null for a capture and loopback device.
pInput (in)
A pointer to the buffer containing input data from a recording device. This will be non-null for a capture, full-duplex or loopback device and null for a
playback device.
frameCount (in)
The number of PCM frames to process. Note that this will not necessarily be equal to what you requested when you initialized the device. The
`periodSizeInFrames` and `periodSizeInMilliseconds` members of the device config are just hints, and are not necessarily exactly what you'll get. You must
not assume this will always be the same value each time the callback is fired.
Remarks
-------
You cannot stop and start the device from inside the callback or else you'll get a deadlock. You must also not uninitialize the device from inside the
callback. The following APIs cannot be called from inside the callback:
ma_device_init()
ma_device_init_ex()
ma_device_uninit()
ma_device_start()
ma_device_stop()
The proper way to stop the device is to call `ma_device_stop()` from a different thread, normally the main application thread.
*/
typedef void (* ma_device_data_proc)(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
/*
DEPRECATED. Use ma_device_notification_proc instead.
The callback for when the device has been stopped.
This will be called when the device is stopped explicitly with `ma_device_stop()` and also called implicitly when the device is stopped through external forces
such as being unplugged or an internal error occuring.
Parameters
----------
pDevice (in)
A pointer to the device that has just stopped.
Remarks
-------
Do not restart or uninitialize the device from the callback.
*/
typedef void (* ma_stop_proc)(ma_device* pDevice); /* DEPRECATED. Use ma_device_notification_proc instead. */
typedef enum
{
ma_device_type_playback = 1,
ma_device_type_capture = 2,
ma_device_type_duplex = ma_device_type_playback | ma_device_type_capture, /* 3 */
ma_device_type_loopback = 4
} ma_device_type;
typedef enum
{
ma_share_mode_shared = 0,
ma_share_mode_exclusive
} ma_share_mode;
/* iOS/tvOS/watchOS session categories. */
typedef enum
{
ma_ios_session_category_default = 0, /* AVAudioSessionCategoryPlayAndRecord with AVAudioSessionCategoryOptionDefaultToSpeaker. */
ma_ios_session_category_none, /* Leave the session category unchanged. */
ma_ios_session_category_ambient, /* AVAudioSessionCategoryAmbient */
ma_ios_session_category_solo_ambient, /* AVAudioSessionCategorySoloAmbient */
ma_ios_session_category_playback, /* AVAudioSessionCategoryPlayback */
ma_ios_session_category_record, /* AVAudioSessionCategoryRecord */
ma_ios_session_category_play_and_record, /* AVAudioSessionCategoryPlayAndRecord */
ma_ios_session_category_multi_route /* AVAudioSessionCategoryMultiRoute */
} ma_ios_session_category;
/* iOS/tvOS/watchOS session category options */
typedef enum
{
ma_ios_session_category_option_mix_with_others = 0x01, /* AVAudioSessionCategoryOptionMixWithOthers */
ma_ios_session_category_option_duck_others = 0x02, /* AVAudioSessionCategoryOptionDuckOthers */
ma_ios_session_category_option_allow_bluetooth = 0x04, /* AVAudioSessionCategoryOptionAllowBluetooth */
ma_ios_session_category_option_default_to_speaker = 0x08, /* AVAudioSessionCategoryOptionDefaultToSpeaker */
ma_ios_session_category_option_interrupt_spoken_audio_and_mix_with_others = 0x11, /* AVAudioSessionCategoryOptionInterruptSpokenAudioAndMixWithOthers */
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
typedef ma_bool32 (* ma_enum_devices_callback_proc)(ma_context* pContext, ma_device_type deviceType, const ma_device_info* pInfo, void* pUserData);
/*
Describes some basic details about a playback or capture device.
*/
typedef struct
{
const ma_device_id* pDeviceID;
ma_share_mode shareMode;
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
ma_channel channelMap[MA_MAX_CHANNELS];
ma_uint32 periodSizeInFrames;
ma_uint32 periodSizeInMilliseconds;
ma_uint32 periodCount;
} ma_device_descriptor;
/*
These are the callbacks required to be implemented for a backend. These callbacks are grouped into two parts: context and device. There is one context
to many devices. A device is created from a context.
The general flow goes like this:
1) A context is created with `onContextInit()`
1a) Available devices can be enumerated with `onContextEnumerateDevices()` if required.
1b) Detailed information about a device can be queried with `onContextGetDeviceInfo()` if required.
2) A device is created from the context that was created in the first step using `onDeviceInit()`, and optionally a device ID that was
selected from device enumeration via `onContextEnumerateDevices()`.
3) A device is started or stopped with `onDeviceStart()` / `onDeviceStop()`
4) Data is delivered to and from the device by the backend. This is always done based on the native format returned by the prior call
to `onDeviceInit()`. Conversion between the device's native format and the format requested by the application will be handled by
miniaudio internally.
Initialization of the context is quite simple. You need to do any necessary initialization of internal objects and then output the
callbacks defined in this structure.
Once the context has been initialized you can initialize a device. Before doing so, however, the application may want to know which
physical devices are available. This is where `onContextEnumerateDevices()` comes in. This is fairly simple. For each device, fire the
given callback with, at a minimum, the basic information filled out in `ma_device_info`. When the callback returns `MA_FALSE`, enumeration
needs to stop and the `onContextEnumerateDevices()` function returns with a success code.
Detailed device information can be retrieved from a device ID using `onContextGetDeviceInfo()`. This takes as input the device type and ID,
and on output returns detailed information about the device in `ma_device_info`. The `onContextGetDeviceInfo()` callback must handle the
case when the device ID is NULL, in which case information about the default device needs to be retrieved.
Once the context has been created and the device ID retrieved (if using anything other than the default device), the device can be created.
This is a little bit more complicated than initialization of the context due to it's more complicated configuration. When initializing a
device, a duplex device may be requested. This means a separate data format needs to be specified for both playback and capture. On input,
the data format is set to what the application wants. On output it's set to the native format which should match as closely as possible to
the requested format. The conversion between the format requested by the application and the device's native format will be handled
internally by miniaudio.
On input, if the sample format is set to `ma_format_unknown`, the backend is free to use whatever sample format it desires, so long as it's
supported by miniaudio. When the channel count is set to 0, the backend should use the device's native channel count. The same applies for
sample rate. For the channel map, the default should be used when `ma_channel_map_is_blank()` returns true (all channels set to
`MA_CHANNEL_NONE`). On input, the `periodSizeInFrames` or `periodSizeInMilliseconds` option should always be set. The backend should
inspect both of these variables. If `periodSizeInFrames` is set, it should take priority, otherwise it needs to be derived from the period
size in milliseconds (`periodSizeInMilliseconds`) and the sample rate, keeping in mind that the sample rate may be 0, in which case the
sample rate will need to be determined before calculating the period size in frames. On output, all members of the `ma_device_data_format`
object should be set to a valid value, except for `periodSizeInMilliseconds` which is optional (`periodSizeInFrames` *must* be set).
Starting and stopping of the device is done with `onDeviceStart()` and `onDeviceStop()` and should be self-explanatory. If the backend uses
asynchronous reading and writing, `onDeviceStart()` and `onDeviceStop()` should always be implemented.
The handling of data delivery between the application and the device is the most complicated part of the process. To make this a bit
easier, some helper callbacks are available. If the backend uses a blocking read/write style of API, the `onDeviceRead()` and
`onDeviceWrite()` callbacks can optionally be implemented. These are blocking and work just like reading and writing from a file. If the
backend uses a callback for data delivery, that callback must call `ma_device_handle_backend_data_callback()` from within it's callback.
This allows miniaudio to then process any necessary data conversion and then pass it to the miniaudio data callback.
If the backend requires absolute flexibility with it's data delivery, it can optionally implement the `onDeviceDataLoop()` callback
which will allow it to implement the logic that will run on the audio thread. This is much more advanced and is completely optional.
The audio thread should run data delivery logic in a loop while `ma_device_get_state() == ma_device_state_started` and no errors have been
encounted. Do not start or stop the device here. That will be handled from outside the `onDeviceDataLoop()` callback.
The invocation of the `onDeviceDataLoop()` callback will be handled by miniaudio. When you start the device, miniaudio will fire this
callback. When the device is stopped, the `ma_device_get_state() == ma_device_state_started` condition will fail and the loop will be terminated
which will then fall through to the part that stops the device. For an example on how to implement the `onDeviceDataLoop()` callback,
look at `ma_device_audio_thread__default_read_write()`. Implement the `onDeviceDataLoopWakeup()` callback if you need a mechanism to
wake up the audio thread.
If the backend supports an optimized retrieval of device information from an initialized `ma_device` object, it should implement the
`onDeviceGetInfo()` callback. This is optional, in which case it will fall back to `onContextGetDeviceInfo()` which is less efficient.
*/
struct ma_backend_callbacks
{
ma_result (* onContextInit)(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks);
ma_result (* onContextUninit)(ma_context* pContext);
ma_result (* onContextEnumerateDevices)(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData);
ma_result (* onContextGetDeviceInfo)(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo);
ma_result (* onDeviceInit)(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture);
ma_result (* onDeviceUninit)(ma_device* pDevice);
ma_result (* onDeviceStart)(ma_device* pDevice);
ma_result (* onDeviceStop)(ma_device* pDevice);
ma_result (* onDeviceRead)(ma_device* pDevice, void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesRead);
ma_result (* onDeviceWrite)(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten);
ma_result (* onDeviceDataLoop)(ma_device* pDevice);
ma_result (* onDeviceDataLoopWakeup)(ma_device* pDevice);
ma_result (* onDeviceGetInfo)(ma_device* pDevice, ma_device_type type, ma_device_info* pDeviceInfo);
};
struct ma_context_config
{
ma_log* pLog;
ma_thread_priority threadPriority;
size_t threadStackSize;
void* pUserData;
ma_allocation_callbacks allocationCallbacks;
struct
{
ma_bool32 useVerboseDeviceEnumeration;
} alsa;
struct
{
const char* pApplicationName;
const char* pServerName;
ma_bool32 tryAutoSpawn; /* Enables autospawning of the PulseAudio daemon if necessary. */
} pulse;
struct
{
ma_ios_session_category sessionCategory;
ma_uint32 sessionCategoryOptions;
ma_bool32 noAudioSessionActivate; /* iOS only. When set to true, does not perform an explicit [[AVAudioSession sharedInstace] setActive:true] on initialization. */
ma_bool32 noAudioSessionDeactivate; /* iOS only. When set to true, does not perform an explicit [[AVAudioSession sharedInstace] setActive:false] on uninitialization. */
} coreaudio;
struct
{
const char* pClientName;
ma_bool32 tryStartServer;
} jack;
ma_backend_callbacks custom;
};
/* WASAPI specific structure for some commands which must run on a common thread due to bugs in WASAPI. */
typedef struct
{
int code;
ma_event* pEvent; /* This will be signalled when the event is complete. */
union
{
struct
{
int _unused;
} quit;
struct
{
ma_device_type deviceType;
void* pAudioClient;
void** ppAudioClientService;
ma_result* pResult; /* The result from creating the audio client service. */
} createAudioClient;
struct
{
ma_device* pDevice;
ma_device_type deviceType;
} releaseAudioClient;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
ma_proc pthread_attr_getschedparam;
ma_proc pthread_attr_setschedparam;
} posix;
#endif
int _unused;
};
};
struct ma_device
{
ma_context* pContext;
ma_device_type type;
ma_uint32 sampleRate;
MA_ATOMIC(4, ma_device_state) state; /* The state of the device is variable and can change at any time on any thread. Must be used atomically. */
ma_device_data_proc onData; /* Set once at initialization time and should not be changed after. */
ma_device_notification_proc onNotification; /* Set once at initialization time and should not be changed after. */
ma_stop_proc onStop; /* DEPRECATED. Use the notification callback instead. Set once at initialization time and should not be changed after. */
void* pUserData; /* Application defined data. */
ma_mutex startStopLock;
ma_event wakeupEvent;
ma_event startEvent;
ma_event stopEvent;
ma_thread thread;
ma_result workResult; /* This is set by the worker thread after it's finished doing a job. */
ma_bool8 isOwnerOfContext; /* When set to true, uninitializing the device will also uninitialize the context. Set to true when NULL is passed into ma_device_init(). */
ma_bool8 noPreSilencedOutputBuffer;
ma_bool8 noClip;
ma_bool8 noDisableDenormals;
ma_bool8 noFixedSizedCallback;
MA_ATOMIC(4, float) masterVolumeFactor; /* Linear 0..1. Can be read and written simultaneously by different threads. Must be used atomically. */
ma_duplex_rb duplexRB; /* Intermediary buffer for duplex device on asynchronous backends. */
struct
{
ma_resample_algorithm algorithm;
ma_resampling_backend_vtable* pBackendVTable;
void* pBackendUserData;
struct
{
ma_uint32 lpfOrder;
} linear;
} resampling;
struct
{
ma_device_id* pID; /* Set to NULL if using default ID, otherwise set to the address of "id". */
ma_device_id id; /* If using an explicit device, will be set to a copy of the ID used for initialization. Otherwise cleared to 0. */
char name[MA_MAX_DEVICE_NAME_LENGTH + 1]; /* Maybe temporary. Likely to be replaced with a query API. */
ma_share_mode shareMode; /* Set to whatever was passed in when the device was initialized. */
ma_format format;
ma_uint32 channels;
ma_channel channelMap[MA_MAX_CHANNELS];
ma_format internalFormat;
ma_uint32 internalChannels;
ma_uint32 internalSampleRate;
ma_channel internalChannelMap[MA_MAX_CHANNELS];
ma_uint32 internalPeriodSizeInFrames;
ma_uint32 internalPeriods;
ma_channel_mix_mode channelMixMode;
ma_data_converter converter;
void* pIntermediaryBuffer; /* For implementing fixed sized buffer callbacks. Will be null if using variable sized callbacks. */
ma_uint32 intermediaryBufferCap;
ma_uint32 intermediaryBufferLen; /* How many valid frames are sitting in the intermediary buffer. */
void* pInputCache; /* In external format. Can be null. */
ma_uint64 inputCacheCap;
ma_uint64 inputCacheConsumed;
ma_uint64 inputCacheRemaining;
} playback;
struct
{
ma_device_id* pID; /* Set to NULL if using default ID, otherwise set to the address of "id". */
ma_device_id id; /* If using an explicit device, will be set to a copy of the ID used for initialization. Otherwise cleared to 0. */
char name[MA_MAX_DEVICE_NAME_LENGTH + 1]; /* Maybe temporary. Likely to be replaced with a query API. */
ma_share_mode shareMode; /* Set to whatever was passed in when the device was initialized. */
ma_format format;
ma_uint32 channels;
ma_channel channelMap[MA_MAX_CHANNELS];
ma_format internalFormat;
ma_uint32 internalChannels;
ma_uint32 internalSampleRate;
ma_channel internalChannelMap[MA_MAX_CHANNELS];
ma_uint32 internalPeriodSizeInFrames;
ma_uint32 internalPeriods;
ma_channel_mix_mode channelMixMode;
ma_data_converter converter;
void* pIntermediaryBuffer; /* For implementing fixed sized buffer callbacks. Will be null if using variable sized callbacks. */
ma_uint32 intermediaryBufferCap;
ma_uint32 intermediaryBufferLen; /* How many valid frames are sitting in the intermediary buffer. */
} capture;
union
{
#ifdef MA_SUPPORT_WASAPI
struct
{
/*IAudioClient**/ ma_ptr pAudioClientPlayback;
/*IAudioClient**/ ma_ptr pAudioClientCapture;
/*IAudioRenderClient**/ ma_ptr pRenderClient;
/*IAudioCaptureClient**/ ma_ptr pCaptureClient;
/*IMMDeviceEnumerator**/ ma_ptr pDeviceEnumerator; /* Used for IMMNotificationClient notifications. Required for detecting default device changes. */
ma_IMMNotificationClient notificationClient;
/*HANDLE*/ ma_handle hEventPlayback; /* Auto reset. Initialized to signaled. */
/*HANDLE*/ ma_handle hEventCapture; /* Auto reset. Initialized to unsignaled. */
ma_uint32 actualBufferSizeInFramesPlayback; /* Value from GetBufferSize(). internalPeriodSizeInFrames is not set to the _actual_ buffer size when low-latency shared mode is being used due to the way the IAudioClient3 API works...
ma_uint32 actualBufferSizeInFramesCapture;
ma_uint32 originalPeriodSizeInFrames;
ma_uint32 originalPeriodSizeInMilliseconds;
ma_uint32 originalPeriods;
ma_performance_profile originalPerformanceProfile;
ma_uint32 periodSizeInFramesPlayback;
ma_uint32 periodSizeInFramesCapture;
void* pMappedBufferCapture;
ma_uint32 mappedBufferCaptureCap;
ma_uint32 mappedBufferCaptureLen;
void* pMappedBufferPlayback;
ma_uint32 mappedBufferPlaybackCap;
ma_uint32 mappedBufferPlaybackLen;
MA_ATOMIC(4, ma_bool32) isStartedCapture; /* Can be read and written simultaneously across different threads. Must be used atomically, and must be 32-bit. */
MA_ATOMIC(4, ma_bool32) isStartedPlayback; /* Can be read and written simultaneously across different threads. Must be used atomically, and must be 32-bit. */
ma_bool8 noAutoConvertSRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM. */
ma_bool8 noDefaultQualitySRC; /* When set to true, disables the use of AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY. */
ma_bool8 noHardwareOffloading;
ma_bool8 allowCaptureAutoStreamRouting;
ma_bool8 allowPlaybackAutoStreamRouting;
ma_bool8 isDetachedPlayback;
ma_bool8 isDetachedCapture;
} wasapi;
#endif
#ifdef MA_SUPPORT_DSOUND
struct
{
/*LPDIRECTSOUND*/ ma_ptr pPlayback;
/*LPDIRECTSOUNDBUFFER*/ ma_ptr pPlaybackPrimaryBuffer;
/*LPDIRECTSOUNDBUFFER*/ ma_ptr pPlaybackBuffer;
/*LPDIRECTSOUNDCAPTURE*/ ma_ptr pCapture;
/*LPDIRECTSOUNDCAPTUREBUFFER*/ ma_ptr pCaptureBuffer;
} dsound;
#endif
#ifdef MA_SUPPORT_WINMM
struct
{
/*HWAVEOUT*/ ma_handle hDevicePlayback;
/*HWAVEIN*/ ma_handle hDeviceCapture;
/*HANDLE*/ ma_handle hEventPlayback;
/*HANDLE*/ ma_handle hEventCapture;
ma_uint32 fragmentSizeInFrames;
ma_uint32 iNextHeaderPlayback; /* [0,periods). Used as an index into pWAVEHDRPlayback. */
ma_uint32 iNextHeaderCapture; /* [0,periods). Used as an index into pWAVEHDRCapture. */
ma_uint32 headerFramesConsumedPlayback; /* The number of PCM frames consumed in the buffer in pWAVEHEADER[iNextHeader]. */
ma_uint32 headerFramesConsumedCapture; /* ^^^ */
/*WAVEHDR**/ ma_uint8* pWAVEHDRPlayback; /* One instantiation for each period. */
/*WAVEHDR**/ ma_uint8* pWAVEHDRCapture; /* One instantiation for each period. */
ma_uint8* pIntermediaryBufferPlayback;
ma_uint8* pIntermediaryBufferCapture;
ma_uint8* _pHeapData; /* Used internally and is used for the heap allocated data for the intermediary buffer and the WAVEHDR structures. */
} winmm;
#endif
#ifdef MA_SUPPORT_ALSA
struct
{
/*snd_pcm_t**/ ma_ptr pPCMPlayback;
/*snd_pcm_t**/ ma_ptr pPCMCapture;
/*struct pollfd**/ void* pPollDescriptorsPlayback;
/*struct pollfd**/ void* pPollDescriptorsCapture;
int pollDescriptorCountPlayback;
int pollDescriptorCountCapture;
int wakeupfdPlayback; /* eventfd for waking up from poll() when the playback device is stopped. */
int wakeupfdCapture; /* eventfd for waking up from poll() when the capture device is stopped. */
ma_bool8 isUsingMMapPlayback;
ma_bool8 isUsingMMapCapture;
} alsa;
#endif
#ifdef MA_SUPPORT_PULSEAUDIO
struct
{
/*pa_mainloop**/ ma_ptr pMainLoop;
/*pa_context**/ ma_ptr pPulseContext;
/*pa_stream**/ ma_ptr pStreamPlayback;
/*pa_stream**/ ma_ptr pStreamCapture;
} pulse;
#endif
#ifdef MA_SUPPORT_JACK
struct
{
/*jack_client_t**/ ma_ptr pClient;
/*jack_port_t**/ ma_ptr* ppPortsPlayback;
/*jack_port_t**/ ma_ptr* ppPortsCapture;
float* pIntermediaryBufferPlayback; /* Typed as a float because JACK is always floating point. */
float* pIntermediaryBufferCapture;
} jack;
#endif
#ifdef MA_SUPPORT_COREAUDIO
struct
{
ma_uint32 deviceObjectIDPlayback;
ma_uint32 deviceObjectIDCapture;
/*AudioUnit*/ ma_ptr audioUnitPlayback;
/*AudioUnit*/ ma_ptr audioUnitCapture;
/*AudioBufferList**/ ma_ptr pAudioBufferList; /* Only used for input devices. */
ma_uint32 audioBufferCapInFrames; /* Only used for input devices. The capacity in frames of each buffer in pAudioBufferList. */
ma_event stopEvent;
ma_uint32 originalPeriodSizeInFrames;
ma_uint32 originalPeriodSizeInMilliseconds;
ma_uint32 originalPeriods;
ma_performance_profile originalPerformanceProfile;
ma_bool32 isDefaultPlaybackDevice;
ma_bool32 isDefaultCaptureDevice;
ma_bool32 isSwitchingPlaybackDevice; /* <-- Set to true when the default device has changed and miniaudio is in the process of switching. */
ma_bool32 isSwitchingCaptureDevice; /* <-- Set to true when the default device has changed and miniaudio is in the process of switching. */
void* pNotificationHandler; /* Only used on mobile platforms. Obj-C object for handling route changes. */
} coreaudio;
#endif
#ifdef MA_SUPPORT_SNDIO
struct
{
ma_ptr handlePlayback;
ma_ptr handleCapture;
ma_bool32 isStartedPlayback;
ma_bool32 isStartedCapture;
} sndio;
#endif
#ifdef MA_SUPPORT_AUDIO4
struct
{
int fdPlayback;
int fdCapture;
} audio4;
#endif
#ifdef MA_SUPPORT_OSS
struct
{
int fdPlayback;
int fdCapture;
} oss;
#endif
#ifdef MA_SUPPORT_AAUDIO
struct
{
/*AAudioStream**/ ma_ptr pStreamPlayback;
/*AAudioStream**/ ma_ptr pStreamCapture;
ma_aaudio_usage usage;
ma_aaudio_content_type contentType;
ma_aaudio_input_preset inputPreset;
ma_bool32 noAutoStartAfterReroute;
} aaudio;
#endif
#ifdef MA_SUPPORT_OPENSL
struct
{
/*SLObjectItf*/ ma_ptr pOutputMixObj;
/*SLOutputMixItf*/ ma_ptr pOutputMix;
/*SLObjectItf*/ ma_ptr pAudioPlayerObj;
/*SLPlayItf*/ ma_ptr pAudioPlayer;
/*SLObjectItf*/ ma_ptr pAudioRecorderObj;
/*SLRecordItf*/ ma_ptr pAudioRecorder;
/*SLAndroidSimpleBufferQueueItf*/ ma_ptr pBufferQueuePlayback;
/*SLAndroidSimpleBufferQueueItf*/ ma_ptr pBufferQueueCapture;
ma_bool32 isDrainingCapture;
ma_bool32 isDrainingPlayback;
ma_uint32 currentBufferIndexPlayback;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
| ma_device_type_loopback |
|-------------------------|
Return Value
------------
A new device config object with default settings. You will typically want to adjust the config after this function returns. See remarks.
Thread Safety
-------------
Safe.
Callback Safety
---------------
Safe, but don't try initializing a device in a callback.
Remarks
-------
The returned config will be initialized to defaults. You will normally want to customize a few variables before initializing the device. See Example 1 for a
typical configuration which sets the sample format, channel count, sample rate, data callback and user data. These are usually things you will want to change
before initializing the device.
See `ma_device_init()` for details on specific configuration options.
Example 1 - Simple Configuration
--------------------------------
The example below is what a program will typically want to configure for each device at a minimum. Notice how `ma_device_config_init()` is called first, and
then the returned object is modified directly. This is important because it ensures that your program continues to work as new configuration options are added
to the `ma_device_config` structure.
```c
ma_device_config config = ma_device_config_init(ma_device_type_playback);
config.playback.format = ma_format_f32;
config.playback.channels = 2;
config.sampleRate = 48000;
config.dataCallback = ma_data_callback;
config.pUserData = pMyUserData;
```
See Also
--------
ma_device_init()
ma_device_init_ex()
*/
MA_API ma_device_config ma_device_config_init(ma_device_type deviceType);
/*
Initializes a device.
A device represents a physical audio device. The idea is you send or receive audio data from the device to either play it back through a speaker, or capture it
from a microphone. Whether or not you should send or receive data from the device (or both) depends on the type of device you are initializing which can be
playback, capture, full-duplex or loopback. (Note that loopback mode is only supported on select backends.) Sending and receiving audio data to and from the
device is done via a callback which is fired by miniaudio at periodic time intervals.
The frequency at which data is delivered to and from a device depends on the size of it's period. The size of the period can be defined in terms of PCM frames
or milliseconds, whichever is more convenient. Generally speaking, the smaller the period, the lower the latency at the expense of higher CPU usage and
increased risk of glitching due to the more frequent and granular data deliver intervals. The size of a period will depend on your requirements, but
miniaudio's defaults should work fine for most scenarios. If you're building a game you should leave this fairly small, whereas if you're building a simple
media player you can make it larger. Note that the period size you request is actually just a hint - miniaudio will tell the backend what you want, but the
backend is ultimately responsible for what it gives you. You cannot assume you will get exactly what you ask for.
When delivering data to and from a device you need to make sure it's in the correct format which you can set through the device configuration. You just set the
format that you want to use and miniaudio will perform all of the necessary conversion for you internally. When delivering data to and from the callback you
can assume the format is the same as what you requested when you initialized the device. See Remarks for more details on miniaudio's data conversion pipeline.
Parameters
----------
pContext (in, optional)
A pointer to the context that owns the device. This can be null, in which case it creates a default context internally.
pConfig (in)
A pointer to the device configuration. Cannot be null. See remarks for details.
pDevice (out)
A pointer to the device object being initialized.
Return Value
------------
MA_SUCCESS if successful; any other error code otherwise.
Thread Safety
-------------
Unsafe. It is not safe to call this function simultaneously for different devices because some backends depend on and mutate global state. The same applies to
calling this at the same time as `ma_device_uninit()`.
Callback Safety
---------------
Unsafe. It is not safe to call this inside any callback.
Remarks
-------
Setting `pContext` to NULL will result in miniaudio creating a default context internally and is equivalent to passing in a context initialized like so:
```c
ma_context_init(NULL, 0, NULL, &context);
```
Do not set `pContext` to NULL if you are needing to open multiple devices. You can, however, use NULL when initializing the first device, and then use
device.pContext for the initialization of other devices.
The device can be configured via the `pConfig` argument. The config object is initialized with `ma_device_config_init()`. Individual configuration settings can
then be set directly on the structure. Below are the members of the `ma_device_config` object.
deviceType
Must be `ma_device_type_playback`, `ma_device_type_capture`, `ma_device_type_duplex` of `ma_device_type_loopback`.
sampleRate
The sample rate, in hertz. The most common sample rates are 48000 and 44100. Setting this to 0 will use the device's native sample rate.
periodSizeInFrames
The desired size of a period in PCM frames. If this is 0, `periodSizeInMilliseconds` will be used instead. If both are 0 the default buffer size will
be used depending on the selected performance profile. This value affects latency. See below for details.
periodSizeInMilliseconds
The desired size of a period in milliseconds. If this is 0, `periodSizeInFrames` will be used instead. If both are 0 the default buffer size will be
used depending on the selected performance profile. The value affects latency. See below for details.
periods
The number of periods making up the device's entire buffer. The total buffer size is `periodSizeInFrames` or `periodSizeInMilliseconds` multiplied by
this value. This is just a hint as backends will be the ones who ultimately decide how your periods will be configured.
performanceProfile
A hint to miniaudio as to the performance requirements of your program. Can be either `ma_performance_profile_low_latency` (default) or
`ma_performance_profile_conservative`. This mainly affects the size of default buffers and can usually be left at it's default value.
noPreSilencedOutputBuffer
When set to true, the contents of the output buffer passed into the data callback will be left undefined. When set to false (default), the contents of
the output buffer will be cleared the zero. You can use this to avoid the overhead of zeroing out the buffer if you can guarantee that your data
callback will write to every sample in the output buffer, or if you are doing your own clearing.
noClip
When set to true, the contents of the output buffer passed into the data callback will be clipped after returning. When set to false (default), the
contents of the output buffer are left alone after returning and it will be left up to the backend itself to decide whether or not the clip. This only
applies when the playback sample format is f32.
noDisableDenormals
By default, miniaudio will disable denormals when the data callback is called. Setting this to true will prevent the disabling of denormals.
noFixedSizedCallback
Allows miniaudio to fire the data callback with any frame count. When this is set to true, the data callback will be fired with a consistent frame
count as specified by `periodSizeInFrames` or `periodSizeInMilliseconds`. When set to false, miniaudio will fire the callback with whatever the
backend requests, which could be anything.
dataCallback
The callback to fire whenever data is ready to be delivered to or from the device.
notificationCallback
The callback to fire when something has changed with the device, such as whether or not it has been started or stopped.
pUserData
The user data pointer to use with the device. You can access this directly from the device object like `device.pUserData`.
resampling.algorithm
The resampling algorithm to use when miniaudio needs to perform resampling between the rate specified by `sampleRate` and the device's native rate. The
default value is `ma_resample_algorithm_linear`, and the quality can be configured with `resampling.linear.lpfOrder`.
resampling.pBackendVTable
A pointer to an optional vtable that can be used for plugging in a custom resampler.
resampling.pBackendUserData
A pointer that will passed to callbacks in pBackendVTable.
resampling.linear.lpfOrder
The linear resampler applies a low-pass filter as part of it's procesing for anti-aliasing. This setting controls the order of the filter. The higher
the value, the better the quality, in general. Setting this to 0 will disable low-pass filtering altogether. The maximum value is
`MA_MAX_FILTER_ORDER`. The default value is `min(4, MA_MAX_FILTER_ORDER)`.
playback.pDeviceID
A pointer to a `ma_device_id` structure containing the ID of the playback device to initialize. Setting this NULL (default) will use the system's
default playback device. Retrieve the device ID from the `ma_device_info` structure, which can be retrieved using device enumeration.
playback.format
The sample format to use for playback. When set to `ma_format_unknown` the device's native format will be used. This can be retrieved after
initialization from the device object directly with `device.playback.format`.
playback.channels
The number of channels to use for playback. When set to 0 the device's native channel count will be used. This can be retrieved after initialization
from the device object directly with `device.playback.channels`.
playback.pChannelMap
The channel map to use for playback. When left empty, the device's native channel map will be used. This can be retrieved after initialization from the
device object direct with `device.playback.pChannelMap`. When set, the buffer should contain `channels` items.
playback.shareMode
The preferred share mode to use for playback. Can be either `ma_share_mode_shared` (default) or `ma_share_mode_exclusive`. Note that if you specify
exclusive mode, but it's not supported by the backend, initialization will fail. You can then fall back to shared mode if desired by changing this to
ma_share_mode_shared and reinitializing.
capture.pDeviceID
A pointer to a `ma_device_id` structure containing the ID of the capture device to initialize. Setting this NULL (default) will use the system's
default capture device. Retrieve the device ID from the `ma_device_info` structure, which can be retrieved using device enumeration.
capture.format
The sample format to use for capture. When set to `ma_format_unknown` the device's native format will be used. This can be retrieved after
initialization from the device object directly with `device.capture.format`.
capture.channels
The number of channels to use for capture. When set to 0 the device's native channel count will be used. This can be retrieved after initialization
from the device object directly with `device.capture.channels`.
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
ma_device_get_master_volume_gain_db()
ma_device_set_master_volume()
ma_device_get_master_volume()
*/
MA_API ma_result ma_device_set_master_volume_db(ma_device* pDevice, float gainDB);
/*
Retrieves the master gain in decibels.
Parameters
----------
pDevice (in)
A pointer to the device whose gain is being retrieved.
pGainDB (in)
A pointer to the variable that will receive the gain in decibels. The returned value will be <= 0.
Return Value
------------
MA_SUCCESS if successful.
MA_INVALID_ARGS if pDevice is NULL.
MA_INVALID_ARGS if pGainDB is NULL.
Thread Safety
-------------
Safe. This just a simple member retrieval.
Callback Safety
---------------
Safe.
Remarks
-------
If an error occurs, `*pGainDB` will be set to 0.
See Also
--------
ma_device_set_master_volume_db()
ma_device_set_master_volume()
ma_device_get_master_volume()
*/
MA_API ma_result ma_device_get_master_volume_db(ma_device* pDevice, float* pGainDB);
/*
Called from the data callback of asynchronous backends to allow miniaudio to process the data and fire the miniaudio data callback.
Parameters
----------
pDevice (in)
A pointer to device whose processing the data callback.
pOutput (out)
A pointer to the buffer that will receive the output PCM frame data. On a playback device this must not be NULL. On a duplex device
this can be NULL, in which case pInput must not be NULL.
pInput (in)
A pointer to the buffer containing input PCM frame data. On a capture device this must not be NULL. On a duplex device this can be
NULL, in which case `pOutput` must not be NULL.
frameCount (in)
The number of frames being processed.
Return Value
------------
MA_SUCCESS if successful; any other result code otherwise.
Thread Safety
-------------
This function should only ever be called from the internal data callback of the backend. It is safe to call this simultaneously between a
playback and capture device in duplex setups.
Callback Safety
---------------
Do not call this from the miniaudio data callback. It should only ever be called from the internal data callback of the backend.
Remarks
-------
If both `pOutput` and `pInput` are NULL, and error will be returned. In duplex scenarios, both `pOutput` and `pInput` can be non-NULL, in
which case `pInput` will be processed first, followed by `pOutput`.
If you are implementing a custom backend, and that backend uses a callback for data delivery, you'll need to call this from inside that
callback.
*/
MA_API ma_result ma_device_handle_backend_data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount);
/*
Calculates an appropriate buffer size from a descriptor, native sample rate and performance profile.
This function is used by backends for helping determine an appropriately sized buffer to use with
the device depending on the values of `periodSizeInFrames` and `periodSizeInMilliseconds` in the
`pDescriptor` object. Since buffer size calculations based on time depends on the sample rate, a
best guess at the device's native sample rate is also required which is where `nativeSampleRate`
comes in. In addition, the performance profile is also needed for cases where both the period size
in frames and milliseconds are both zero.
Parameters
----------
pDescriptor (in)
A pointer to device descriptor whose `periodSizeInFrames` and `periodSizeInMilliseconds` members
will be used for the calculation of the buffer size.
nativeSampleRate (in)
The device's native sample rate. This is only ever used when the `periodSizeInFrames` member of
`pDescriptor` is zero. In this case, `periodSizeInMilliseconds` will be used instead, in which
case a sample rate is required to convert to a size in frames.
performanceProfile (in)
When both the `periodSizeInFrames` and `periodSizeInMilliseconds` members of `pDescriptor` are
zero, miniaudio will fall back to a buffer size based on the performance profile. The profile
to use for this calculation is determine by this parameter.
Return Value
------------
The calculated buffer size in frames.
Thread Safety
-------------
This is safe so long as nothing modifies `pDescriptor` at the same time. However, this function
should only ever be called from within the backend's device initialization routine and therefore
shouldn't have any multithreading concerns.
Callback Safety
---------------
This is safe to call within the data callback, but there is no reason to ever do this.
Remarks
-------
If `nativeSampleRate` is zero, this function will fall back to `pDescriptor->sampleRate`. If that
is also zero, `MA_DEFAULT_SAMPLE_RATE` will be used instead.
*/
MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_descriptor(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile);
/*
Retrieves a friendly name for a backend.
*/
MA_API const char* ma_get_backend_name(ma_backend backend);
/*
Determines whether or not the given backend is available by the compilation environment.
*/
MA_API ma_bool32 ma_is_backend_enabled(ma_backend backend);
/*
Retrieves compile-time enabled backends.
Parameters
----------
pBackends (out, optional)
A pointer to the buffer that will receive the enabled backends. Set to NULL to retrieve the backend count. Setting
the capacity of the buffer to `MA_BUFFER_COUNT` will guarantee it's large enough for all backends.
backendCap (in)
The capacity of the `pBackends` buffer.
pBackendCount (out)
A pointer to the variable that will receive the enabled backend count.
Return Value
------------
MA_SUCCESS if successful.
MA_INVALID_ARGS if `pBackendCount` is NULL.
MA_NO_SPACE if the capacity of `pBackends` is not large enough.
If `MA_NO_SPACE` is returned, the `pBackends` buffer will be filled with `*pBackendCount` values.
Thread Safety
-------------
Safe.
Callback Safety
---------------
Safe.
Remarks
-------
If you want to retrieve the number of backends so you can determine the capacity of `pBackends` buffer, you can call
this function with `pBackends` set to NULL.
This will also enumerate the null backend. If you don't want to include this you need to check for `ma_backend_null`
when you enumerate over the returned backends and handle it appropriately. Alternatively, you can disable it at
compile time with `MA_NO_NULL`.
The returned backends are determined based on compile time settings, not the platform it's currently running on. For
example, PulseAudio will be returned if it was enabled at compile time, even when the user doesn't actually have
PulseAudio installed.
Example 1
---------
The example below retrieves the enabled backend count using a fixed sized buffer allocated on the stack. The buffer is
given a capacity of `MA_BACKEND_COUNT` which will guarantee it'll be large enough to store all available backends.
Since `MA_BACKEND_COUNT` is always a relatively small value, this should be suitable for most scenarios.
```
ma_backend enabledBackends[MA_BACKEND_COUNT];
size_t enabledBackendCount;
result = ma_get_enabled_backends(enabledBackends, MA_BACKEND_COUNT, &enabledBackendCount);
if (result != MA_SUCCESS) {
// Failed to retrieve enabled backends. Should never happen in this example since all inputs are valid.
}
```
See Also
--------
ma_is_backend_enabled()
*/
MA_API ma_result ma_get_enabled_backends(ma_backend* pBackends, size_t backendCap, size_t* pBackendCount);
/*
Determines whether or not loopback mode is support by a backend.
*/
MA_API ma_bool32 ma_is_loopback_supported(ma_backend backend);
#endif /* MA_NO_DEVICE_IO */
/************************************************************************************************************************************************************
Utiltities
************************************************************************************************************************************************************/
/*
Calculates a buffer size in milliseconds from the specified number of frames and sample rate.
*/
MA_API ma_uint32 ma_calculate_buffer_size_in_milliseconds_from_frames(ma_uint32 bufferSizeInFrames, ma_uint32 sampleRate);
/*
Calculates a buffer size in frames from the specified number of milliseconds and sample rate.
*/
MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_milliseconds(ma_uint32 bufferSizeInMilliseconds, ma_uint32 sampleRate);
/*
Copies PCM frames from one buffer to another.
*/
MA_API void ma_copy_pcm_frames(void* dst, const void* src, ma_uint64 frameCount, ma_format format, ma_uint32 channels);
/*
Copies silent frames into the given buffer.
Remarks
-------
For all formats except `ma_format_u8`, the output buffer will be filled with 0. For `ma_format_u8` it will be filled with 128. The reason for this is that it
makes more sense for the purpose of mixing to initialize it to the center point.
*/
MA_API void ma_silence_pcm_frames(void* p, ma_uint64 frameCount, ma_format format, ma_uint32 channels);
/*
Offsets a pointer by the specified number of PCM frames.
*/
MA_API void* ma_offset_pcm_frames_ptr(void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels);
MA_API const void* ma_offset_pcm_frames_const_ptr(const void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels);
static MA_INLINE float* ma_offset_pcm_frames_ptr_f32(float* p, ma_uint64 offsetInFrames, ma_uint32 channels) { return (float*)ma_offset_pcm_frames_ptr((void*)p, offsetInFrames, ma_format_f32, channels); }
static MA_INLINE const float* ma_offset_pcm_frames_const_ptr_f32(const float* p, ma_uint64 offsetInFrames, ma_uint32 channels) { return (const float*)ma_offset_pcm_frames_const_ptr((const void*)p, offsetInFrames, ma_format_f32, channels); }
/*
Clips samples.
*/
MA_API void ma_clip_samples_u8(ma_uint8* pDst, const ma_int16* pSrc, ma_uint64 count);
MA_API void ma_clip_samples_s16(ma_int16* pDst, const ma_int32* pSrc, ma_uint64 count);
MA_API void ma_clip_samples_s24(ma_uint8* pDst, const ma_int64* pSrc, ma_uint64 count);
MA_API void ma_clip_samples_s32(ma_int32* pDst, const ma_int64* pSrc, ma_uint64 count);
MA_API void ma_clip_samples_f32(float* pDst, const float* pSrc, ma_uint64 count);
MA_API void ma_clip_pcm_frames(void* pDst, const void* pSrc, ma_uint64 frameCount, ma_format format, ma_uint32 channels);
/*
Helper for applying a volume factor to samples.
Note that the source and destination buffers can be the same, in which case it'll perform the operation in-place.
*/
MA_API void ma_copy_and_apply_volume_factor_u8(ma_uint8* pSamplesOut, const ma_uint8* pSamplesIn, ma_uint64 sampleCount, float factor);
MA_API void ma_copy_and_apply_volume_factor_s16(ma_int16* pSamplesOut, const ma_int16* pSamplesIn, ma_uint64 sampleCount, float factor);
MA_API void ma_copy_and_apply_volume_factor_s24(void* pSamplesOut, const void* pSamplesIn, ma_uint64 sampleCount, float factor);
MA_API void ma_copy_and_apply_volume_factor_s32(ma_int32* pSamplesOut, const ma_int32* pSamplesIn, ma_uint64 sampleCount, float factor);
MA_API void ma_copy_and_apply_volume_factor_f32(float* pSamplesOut, const float* pSamplesIn, ma_uint64 sampleCount, float factor);
MA_API void ma_apply_volume_factor_u8(ma_uint8* pSamples, ma_uint64 sampleCount, float factor);
MA_API void ma_apply_volume_factor_s16(ma_int16* pSamples, ma_uint64 sampleCount, float factor);
MA_API void ma_apply_volume_factor_s24(void* pSamples, ma_uint64 sampleCount, float factor);
MA_API void ma_apply_volume_factor_s32(ma_int32* pSamples, ma_uint64 sampleCount, float factor);
MA_API void ma_apply_volume_factor_f32(float* pSamples, ma_uint64 sampleCount, float factor);
MA_API void ma_copy_and_apply_volume_factor_pcm_frames_u8(ma_uint8* pFramesOut, const ma_uint8* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s16(ma_int16* pFramesOut, const ma_int16* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s24(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s32(ma_int32* pFramesOut, const ma_int32* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
MA_API void ma_copy_and_apply_volume_factor_pcm_frames_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor);
MA_API void ma_copy_and_apply_volume_factor_pcm_frames(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor);
MA_API void ma_apply_volume_factor_pcm_frames_u8(ma_uint8* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
MA_API void ma_apply_volume_factor_pcm_frames_s16(ma_int16* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
MA_API void ma_apply_volume_factor_pcm_frames_s24(void* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
MA_API void ma_apply_volume_factor_pcm_frames_s32(ma_int32* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
MA_API void ma_apply_volume_factor_pcm_frames_f32(float* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor);
MA_API void ma_apply_volume_factor_pcm_frames(void* pFrames, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor);
MA_API void ma_copy_and_apply_volume_factor_per_channel_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float* pChannelGains);
MA_API void ma_copy_and_apply_volume_and_clip_samples_u8(ma_uint8* pDst, const ma_int16* pSrc, ma_uint64 count, float volume);
MA_API void ma_copy_and_apply_volume_and_clip_samples_s16(ma_int16* pDst, const ma_int32* pSrc, ma_uint64 count, float volume);
MA_API void ma_copy_and_apply_volume_and_clip_samples_s24(ma_uint8* pDst, const ma_int64* pSrc, ma_uint64 count, float volume);
MA_API void ma_copy_and_apply_volume_and_clip_samples_s32(ma_int32* pDst, const ma_int64* pSrc, ma_uint64 count, float volume);
MA_API void ma_copy_and_apply_volume_and_clip_samples_f32(float* pDst, const float* pSrc, ma_uint64 count, float volume);
MA_API void ma_copy_and_apply_volume_and_clip_pcm_frames(void* pDst, const void* pSrc, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float volume);
/*
Helper for converting a linear factor to gain in decibels.
*/
MA_API float ma_volume_linear_to_db(float factor);
/*
Helper for converting gain in decibels to a linear factor.
*/
MA_API float ma_volume_db_to_linear(float gain);
/**************************************************************************************************
Data Source
**************************************************************************************************/
typedef void ma_data_source;
#define MA_DATA_SOURCE_SELF_MANAGED_RANGE_AND_LOOP_POINT 0x00000001
typedef struct
{
ma_result (* onRead)(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
ma_result (* onSeek)(ma_data_source* pDataSource, ma_uint64 frameIndex);
ma_result (* onGetDataFormat)(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
ma_result (* onGetCursor)(ma_data_source* pDataSource, ma_uint64* pCursor);
ma_result (* onGetLength)(ma_data_source* pDataSource, ma_uint64* pLength);
ma_result (* onSetLooping)(ma_data_source* pDataSource, ma_bool32 isLooping);
ma_uint32 flags;
} ma_data_source_vtable;
typedef ma_data_source* (* ma_data_source_get_next_proc)(ma_data_source* pDataSource);
typedef struct
{
const ma_data_source_vtable* vtable;
} ma_data_source_config;
MA_API ma_data_source_config ma_data_source_config_init(void);
typedef struct
{
const ma_data_source_vtable* vtable;
ma_uint64 rangeBegInFrames;
ma_uint64 rangeEndInFrames; /* Set to -1 for unranged (default). */
ma_uint64 loopBegInFrames; /* Relative to rangeBegInFrames. */
ma_uint64 loopEndInFrames; /* Relative to rangeBegInFrames. Set to -1 for the end of the range. */
ma_data_source* pCurrent; /* When non-NULL, the data source being initialized will act as a proxy and will route all operations to pCurrent. Used in conjunction with pNext/onGetNext for seamless chaining. */
ma_data_source* pNext; /* When set to NULL, onGetNext will be used. */
ma_data_source_get_next_proc onGetNext; /* Will be used when pNext is NULL. If both are NULL, no next will be used. */
MA_ATOMIC(4, ma_bool32) isLooping;
} ma_data_source_base;
MA_API ma_result ma_data_source_init(const ma_data_source_config* pConfig, ma_data_source* pDataSource);
MA_API void ma_data_source_uninit(ma_data_source* pDataSource);
MA_API ma_result ma_data_source_read_pcm_frames(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); /* Must support pFramesOut = NULL in which case a forward seek should be performed. */
MA_API ma_result ma_data_source_seek_pcm_frames(ma_data_source* pDataSource, ma_uint64 frameCount, ma_uint64* pFramesSeeked); /* Can only seek forward. Equivalent to ma_data_source_read_pcm_frames(pDataSource, NULL, frameCount, &framesRead); */
MA_API ma_result ma_data_source_seek_to_pcm_frame(ma_data_source* pDataSource, ma_uint64 frameIndex);
MA_API ma_result ma_data_source_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
MA_API ma_result ma_data_source_get_cursor_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pCursor);
MA_API ma_result ma_data_source_get_length_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLength); /* Returns MA_NOT_IMPLEMENTED if the length is unknown or cannot be determined. Decoders can return this. */
MA_API ma_result ma_data_source_get_cursor_in_seconds(ma_data_source* pDataSource, float* pCursor);
MA_API ma_result ma_data_source_get_length_in_seconds(ma_data_source* pDataSource, float* pLength);
MA_API ma_result ma_data_source_set_looping(ma_data_source* pDataSource, ma_bool32 isLooping);
MA_API ma_bool32 ma_data_source_is_looping(const ma_data_source* pDataSource);
MA_API ma_result ma_data_source_set_range_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 rangeBegInFrames, ma_uint64 rangeEndInFrames);
MA_API void ma_data_source_get_range_in_pcm_frames(const ma_data_source* pDataSource, ma_uint64* pRangeBegInFrames, ma_uint64* pRangeEndInFrames);
MA_API ma_result ma_data_source_set_loop_point_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 loopBegInFrames, ma_uint64 loopEndInFrames);
MA_API void ma_data_source_get_loop_point_in_pcm_frames(const ma_data_source* pDataSource, ma_uint64* pLoopBegInFrames, ma_uint64* pLoopEndInFrames);
MA_API ma_result ma_data_source_set_current(ma_data_source* pDataSource, ma_data_source* pCurrentDataSource);
MA_API ma_data_source* ma_data_source_get_current(const ma_data_source* pDataSource);
MA_API ma_result ma_data_source_set_next(ma_data_source* pDataSource, ma_data_source* pNextDataSource);
MA_API ma_data_source* ma_data_source_get_next(const ma_data_source* pDataSource);
MA_API ma_result ma_data_source_set_next_callback(ma_data_source* pDataSource, ma_data_source_get_next_proc onGetNext);
MA_API ma_data_source_get_next_proc ma_data_source_get_next_callback(const ma_data_source* pDataSource);
typedef struct
{
ma_data_source_base ds;
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
ma_uint64 cursor;
ma_uint64 sizeInFrames;
const void* pData;
} ma_audio_buffer_ref;
MA_API ma_result ma_audio_buffer_ref_init(ma_format format, ma_uint32 channels, const void* pData, ma_uint64 sizeInFrames, ma_audio_buffer_ref* pAudioBufferRef);
MA_API void ma_audio_buffer_ref_uninit(ma_audio_buffer_ref* pAudioBufferRef);
MA_API ma_result ma_audio_buffer_ref_set_data(ma_audio_buffer_ref* pAudioBufferRef, const void* pData, ma_uint64 sizeInFrames);
MA_API ma_uint64 ma_audio_buffer_ref_read_pcm_frames(ma_audio_buffer_ref* pAudioBufferRef, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop);
MA_API ma_result ma_audio_buffer_ref_seek_to_pcm_frame(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameIndex);
MA_API ma_result ma_audio_buffer_ref_map(ma_audio_buffer_ref* pAudioBufferRef, void** ppFramesOut, ma_uint64* pFrameCount);
MA_API ma_result ma_audio_buffer_ref_unmap(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameCount); /* Returns MA_AT_END if the end has been reached. This should be considered successful. */
MA_API ma_bool32 ma_audio_buffer_ref_at_end(const ma_audio_buffer_ref* pAudioBufferRef);
MA_API ma_result ma_audio_buffer_ref_get_cursor_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pCursor);
MA_API ma_result ma_audio_buffer_ref_get_length_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pLength);
MA_API ma_result ma_audio_buffer_ref_get_available_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pAvailableFrames);
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
ma_uint64 sizeInFrames;
const void* pData; /* If set to NULL, will allocate a block of memory for you. */
ma_allocation_callbacks allocationCallbacks;
} ma_audio_buffer_config;
MA_API ma_audio_buffer_config ma_audio_buffer_config_init(ma_format format, ma_uint32 channels, ma_uint64 sizeInFrames, const void* pData, const ma_allocation_callbacks* pAllocationCallbacks);
typedef struct
{
ma_audio_buffer_ref ref;
ma_allocation_callbacks allocationCallbacks;
ma_bool32 ownsData; /* Used to control whether or not miniaudio owns the data buffer. If set to true, pData will be freed in ma_audio_buffer_uninit(). */
ma_uint8 _pExtraData[1]; /* For allocating a buffer with the memory located directly after the other memory of the structure. */
} ma_audio_buffer;
MA_API ma_result ma_audio_buffer_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer);
MA_API ma_result ma_audio_buffer_init_copy(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer);
MA_API ma_result ma_audio_buffer_alloc_and_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer** ppAudioBuffer); /* Always copies the data. Doesn't make sense to use this otherwise. Use ma_audio_buffer_uninit_and_free() to uninit. */
MA_API void ma_audio_buffer_uninit(ma_audio_buffer* pAudioBuffer);
MA_API void ma_audio_buffer_uninit_and_free(ma_audio_buffer* pAudioBuffer);
MA_API ma_uint64 ma_audio_buffer_read_pcm_frames(ma_audio_buffer* pAudioBuffer, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop);
MA_API ma_result ma_audio_buffer_seek_to_pcm_frame(ma_audio_buffer* pAudioBuffer, ma_uint64 frameIndex);
MA_API ma_result ma_audio_buffer_map(ma_audio_buffer* pAudioBuffer, void** ppFramesOut, ma_uint64* pFrameCount);
MA_API ma_result ma_audio_buffer_unmap(ma_audio_buffer* pAudioBuffer, ma_uint64 frameCount); /* Returns MA_AT_END if the end has been reached. This should be considered successful. */
MA_API ma_bool32 ma_audio_buffer_at_end(const ma_audio_buffer* pAudioBuffer);
MA_API ma_result ma_audio_buffer_get_cursor_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pCursor);
MA_API ma_result ma_audio_buffer_get_length_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pLength);
MA_API ma_result ma_audio_buffer_get_available_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pAvailableFrames);
/*
Paged Audio Buffer
==================
A paged audio buffer is made up of a linked list of pages. It's expandable, but not shrinkable. It
can be used for cases where audio data is streamed in asynchronously while allowing data to be read
at the same time.
This is lock-free, but not 100% thread safe. You can append a page and read from the buffer across
simultaneously across different threads, however only one thread at a time can append, and only one
thread at a time can read and seek.
*/
typedef struct ma_paged_audio_buffer_page ma_paged_audio_buffer_page;
struct ma_paged_audio_buffer_page
{
MA_ATOMIC(MA_SIZEOF_PTR, ma_paged_audio_buffer_page*) pNext;
ma_uint64 sizeInFrames;
ma_uint8 pAudioData[1];
};
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_paged_audio_buffer_page head; /* Dummy head for the lock-free algorithm. Always has a size of 0. */
MA_ATOMIC(MA_SIZEOF_PTR, ma_paged_audio_buffer_page*) pTail; /* Never null. Initially set to &head. */
} ma_paged_audio_buffer_data;
MA_API ma_result ma_paged_audio_buffer_data_init(ma_format format, ma_uint32 channels, ma_paged_audio_buffer_data* pData);
MA_API void ma_paged_audio_buffer_data_uninit(ma_paged_audio_buffer_data* pData, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_paged_audio_buffer_page* ma_paged_audio_buffer_data_get_head(ma_paged_audio_buffer_data* pData);
MA_API ma_paged_audio_buffer_page* ma_paged_audio_buffer_data_get_tail(ma_paged_audio_buffer_data* pData);
MA_API ma_result ma_paged_audio_buffer_data_get_length_in_pcm_frames(ma_paged_audio_buffer_data* pData, ma_uint64* pLength);
MA_API ma_result ma_paged_audio_buffer_data_allocate_page(ma_paged_audio_buffer_data* pData, ma_uint64 pageSizeInFrames, const void* pInitialData, const ma_allocation_callbacks* pAllocationCallbacks, ma_paged_audio_buffer_page** ppPage);
MA_API ma_result ma_paged_audio_buffer_data_free_page(ma_paged_audio_buffer_data* pData, ma_paged_audio_buffer_page* pPage, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_paged_audio_buffer_data_append_page(ma_paged_audio_buffer_data* pData, ma_paged_audio_buffer_page* pPage);
MA_API ma_result ma_paged_audio_buffer_data_allocate_and_append_page(ma_paged_audio_buffer_data* pData, ma_uint32 pageSizeInFrames, const void* pInitialData, const ma_allocation_callbacks* pAllocationCallbacks);
typedef struct
{
ma_paged_audio_buffer_data* pData; /* Must not be null. */
} ma_paged_audio_buffer_config;
MA_API ma_paged_audio_buffer_config ma_paged_audio_buffer_config_init(ma_paged_audio_buffer_data* pData);
typedef struct
{
ma_data_source_base ds;
ma_paged_audio_buffer_data* pData; /* Audio data is read from here. Cannot be null. */
ma_paged_audio_buffer_page* pCurrent;
ma_uint64 relativeCursor; /* Relative to the current page. */
ma_uint64 absoluteCursor;
} ma_paged_audio_buffer;
MA_API ma_result ma_paged_audio_buffer_init(const ma_paged_audio_buffer_config* pConfig, ma_paged_audio_buffer* pPagedAudioBuffer);
MA_API void ma_paged_audio_buffer_uninit(ma_paged_audio_buffer* pPagedAudioBuffer);
MA_API ma_result ma_paged_audio_buffer_read_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead); /* Returns MA_AT_END if no more pages available. */
MA_API ma_result ma_paged_audio_buffer_seek_to_pcm_frame(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64 frameIndex);
MA_API ma_result ma_paged_audio_buffer_get_cursor_in_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64* pCursor);
MA_API ma_result ma_paged_audio_buffer_get_length_in_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64* pLength);
/************************************************************************************************************************************************************
VFS
===
The VFS object (virtual file system) is what's used to customize file access. This is useful in cases where stdio FILE* based APIs may not be entirely
appropriate for a given situation.
************************************************************************************************************************************************************/
typedef void ma_vfs;
typedef ma_handle ma_vfs_file;
typedef enum
{
MA_OPEN_MODE_READ = 0x00000001,
MA_OPEN_MODE_WRITE = 0x00000002
} ma_open_mode_flags;
typedef enum
{
ma_seek_origin_start,
ma_seek_origin_current,
ma_seek_origin_end /* Not used by decoders. */
} ma_seek_origin;
typedef struct
{
ma_uint64 sizeInBytes;
} ma_file_info;
typedef struct
{
ma_result (* onOpen) (ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile);
ma_result (* onOpenW)(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile);
ma_result (* onClose)(ma_vfs* pVFS, ma_vfs_file file);
ma_result (* onRead) (ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead);
ma_result (* onWrite)(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten);
ma_result (* onSeek) (ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin);
ma_result (* onTell) (ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor);
ma_result (* onInfo) (ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo);
} ma_vfs_callbacks;
MA_API ma_result ma_vfs_open(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile);
MA_API ma_result ma_vfs_open_w(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile);
MA_API ma_result ma_vfs_close(ma_vfs* pVFS, ma_vfs_file file);
MA_API ma_result ma_vfs_read(ma_vfs* pVFS, ma_vfs_file file, void* pDst, size_t sizeInBytes, size_t* pBytesRead);
MA_API ma_result ma_vfs_write(ma_vfs* pVFS, ma_vfs_file file, const void* pSrc, size_t sizeInBytes, size_t* pBytesWritten);
MA_API ma_result ma_vfs_seek(ma_vfs* pVFS, ma_vfs_file file, ma_int64 offset, ma_seek_origin origin);
MA_API ma_result ma_vfs_tell(ma_vfs* pVFS, ma_vfs_file file, ma_int64* pCursor);
MA_API ma_result ma_vfs_info(ma_vfs* pVFS, ma_vfs_file file, ma_file_info* pInfo);
MA_API ma_result ma_vfs_open_and_read_file(ma_vfs* pVFS, const char* pFilePath, void** ppData, size_t* pSize, const ma_allocation_callbacks* pAllocationCallbacks);
typedef struct
{
ma_vfs_callbacks cb;
ma_allocation_callbacks allocationCallbacks; /* Only used for the wchar_t version of open() on non-Windows platforms. */
} ma_default_vfs;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
========
Decoders are independent of the main device API. Decoding APIs can be called freely inside the device's data callback, but they are not thread safe unless
you do your own synchronization.
************************************************************************************************************************************************************/
#ifndef MA_NO_DECODING
typedef struct ma_decoder ma_decoder;
typedef struct
{
ma_format preferredFormat;
ma_uint32 seekPointCount; /* Set to > 0 to generate a seektable if the decoding backend supports it. */
} ma_decoding_backend_config;
MA_API ma_decoding_backend_config ma_decoding_backend_config_init(ma_format preferredFormat, ma_uint32 seekPointCount);
typedef struct
{
ma_result (* onInit )(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source...
ma_result (* onInitFile )(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); /* Optional. */
ma_result (* onInitFileW )(void* pUserData, const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); /* Optional. */
ma_result (* onInitMemory)(void* pUserData, const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend); /* Optional. */
void (* onUninit )(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks);
} ma_decoding_backend_vtable;
typedef ma_result (* ma_decoder_read_proc)(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead); /* Returns the number of bytes read. */
typedef ma_result (* ma_decoder_seek_proc)(ma_decoder* pDecoder, ma_int64 byteOffset, ma_seek_origin origin);
typedef ma_result (* ma_decoder_tell_proc)(ma_decoder* pDecoder, ma_int64* pCursor);
typedef struct
{
ma_format format; /* Set to 0 or ma_format_unknown to use the stream's internal format. */
ma_uint32 channels; /* Set to 0 to use the stream's internal channels. */
ma_uint32 sampleRate; /* Set to 0 to use the stream's internal sample rate. */
ma_channel* pChannelMap;
ma_channel_mix_mode channelMixMode;
ma_dither_mode ditherMode;
ma_resampler_config resampling;
ma_allocation_callbacks allocationCallbacks;
ma_encoding_format encodingFormat;
ma_uint32 seekPointCount; /* When set to > 0, specifies the number of seek points to use for the generation of a seek table. Not all decoding backends support this. */
ma_decoding_backend_vtable** ppCustomBackendVTables;
ma_uint32 customBackendCount;
void* pCustomBackendUserData;
} ma_decoder_config;
struct ma_decoder
{
ma_data_source_base ds;
ma_data_source* pBackend; /* The decoding backend we'll be pulling data from. */
const ma_decoding_backend_vtable* pBackendVTable; /* The vtable for the decoding backend. This needs to be stored so we can access the onUninit() callback. */
void* pBackendUserData;
ma_decoder_read_proc onRead;
ma_decoder_seek_proc onSeek;
ma_decoder_tell_proc onTell;
void* pUserData;
ma_uint64 readPointerInPCMFrames; /* In output sample rate. Used for keeping track of how many frames are available for decoding. */
ma_format outputFormat;
ma_uint32 outputChannels;
ma_uint32 outputSampleRate;
ma_data_converter converter; /* Data conversion is achieved by running frames through this. */
void* pInputCache; /* In input format. Can be null if it's not needed. */
ma_uint64 inputCacheCap; /* The capacity of the input cache. */
ma_uint64 inputCacheConsumed; /* The number of frames that have been consumed in the cache. Used for determining the next valid frame. */
ma_uint64 inputCacheRemaining; /* The number of valid frames remaining in the cahce. */
ma_allocation_callbacks allocationCallbacks;
union
{
struct
{
ma_vfs* pVFS;
ma_vfs_file file;
} vfs;
struct
{
const ma_uint8* pData;
size_t dataSize;
size_t currentReadPos;
} memory; /* Only used for decoders that were opened against a block of memory. */
} data;
};
MA_API ma_decoder_config ma_decoder_config_init(ma_format outputFormat, ma_uint32 outputChannels, ma_uint32 outputSampleRate);
MA_API ma_decoder_config ma_decoder_config_init_default(void);
MA_API ma_result ma_decoder_init(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
MA_API ma_result ma_decoder_init_memory(const void* pData, size_t dataSize, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
MA_API ma_result ma_decoder_init_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
MA_API ma_result ma_decoder_init_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
MA_API ma_result ma_decoder_init_file(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
MA_API ma_result ma_decoder_init_file_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder);
/*
Uninitializes a decoder.
*/
MA_API ma_result ma_decoder_uninit(ma_decoder* pDecoder);
/*
Reads PCM frames from the given decoder.
This is not thread safe without your own synchronization.
*/
MA_API ma_result ma_decoder_read_pcm_frames(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
/*
Seeks to a PCM frame based on it's absolute index.
This is not thread safe without your own synchronization.
*/
MA_API ma_result ma_decoder_seek_to_pcm_frame(ma_decoder* pDecoder, ma_uint64 frameIndex);
/*
Retrieves the decoder's output data format.
*/
MA_API ma_result ma_decoder_get_data_format(ma_decoder* pDecoder, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
/*
Retrieves the current position of the read cursor in PCM frames.
*/
MA_API ma_result ma_decoder_get_cursor_in_pcm_frames(ma_decoder* pDecoder, ma_uint64* pCursor);
/*
Retrieves the length of the decoder in PCM frames.
Do not call this on streams of an undefined length, such as internet radio.
If the length is unknown or an error occurs, 0 will be returned.
This will always return 0 for Vorbis decoders. This is due to a limitation with stb_vorbis in push mode which is what miniaudio
uses internally.
For MP3's, this will decode the entire file. Do not call this in time critical scenarios.
This function is not thread safe without your own synchronization.
*/
MA_API ma_result ma_decoder_get_length_in_pcm_frames(ma_decoder* pDecoder, ma_uint64* pLength);
/*
Retrieves the number of frames that can be read before reaching the end.
This calls `ma_decoder_get_length_in_pcm_frames()` so you need to be aware of the rules for that function, in
particular ensuring you do not call it on streams of an undefined length, such as internet radio.
If the total length of the decoder cannot be retrieved, such as with Vorbis decoders, `MA_NOT_IMPLEMENTED` will be
returned.
*/
MA_API ma_result ma_decoder_get_available_frames(ma_decoder* pDecoder, ma_uint64* pAvailableFrames);
/*
Helper for opening and decoding a file into a heap allocated block of memory. Free the returned pointer with ma_free(). On input,
pConfig should be set to what you want. On output it will be set to what you got.
*/
MA_API ma_result ma_decode_from_vfs(ma_vfs* pVFS, const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut);
MA_API ma_result ma_decode_file(const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut);
MA_API ma_result ma_decode_memory(const void* pData, size_t dataSize, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut);
#endif /* MA_NO_DECODING */
/************************************************************************************************************************************************************
Encoding
========
Encoders do not perform any format conversion for you. If your target format does not support the format, and error will be returned.
************************************************************************************************************************************************************/
#ifndef MA_NO_ENCODING
typedef struct ma_encoder ma_encoder;
typedef ma_result (* ma_encoder_write_proc) (ma_encoder* pEncoder, const void* pBufferIn, size_t bytesToWrite, size_t* pBytesWritten);
typedef ma_result (* ma_encoder_seek_proc) (ma_encoder* pEncoder, ma_int64 offset, ma_seek_origin origin);
typedef ma_result (* ma_encoder_init_proc) (ma_encoder* pEncoder);
typedef void (* ma_encoder_uninit_proc) (ma_encoder* pEncoder);
typedef ma_result (* ma_encoder_write_pcm_frames_proc)(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount, ma_uint64* pFramesWritten);
typedef struct
{
ma_encoding_format encodingFormat;
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
ma_allocation_callbacks allocationCallbacks;
} ma_encoder_config;
MA_API ma_encoder_config ma_encoder_config_init(ma_encoding_format encodingFormat, ma_format format, ma_uint32 channels, ma_uint32 sampleRate);
struct ma_encoder
{
ma_encoder_config config;
ma_encoder_write_proc onWrite;
ma_encoder_seek_proc onSeek;
ma_encoder_init_proc onInit;
ma_encoder_uninit_proc onUninit;
ma_encoder_write_pcm_frames_proc onWritePCMFrames;
void* pUserData;
void* pInternalEncoder; /* <-- The drwav/drflac/stb_vorbis/etc. objects. */
union
{
struct
{
ma_vfs* pVFS;
ma_vfs_file file;
} vfs;
} data;
};
MA_API ma_result ma_encoder_init(ma_encoder_write_proc onWrite, ma_encoder_seek_proc onSeek, void* pUserData, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
MA_API ma_result ma_encoder_init_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
MA_API ma_result ma_encoder_init_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
MA_API ma_result ma_encoder_init_file(const char* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
MA_API ma_result ma_encoder_init_file_w(const wchar_t* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder);
MA_API void ma_encoder_uninit(ma_encoder* pEncoder);
MA_API ma_result ma_encoder_write_pcm_frames(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount, ma_uint64* pFramesWritten);
#endif /* MA_NO_ENCODING */
/************************************************************************************************************************************************************
Generation
************************************************************************************************************************************************************/
#ifndef MA_NO_GENERATION
typedef enum
{
ma_waveform_type_sine,
ma_waveform_type_square,
ma_waveform_type_triangle,
ma_waveform_type_sawtooth
} ma_waveform_type;
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
ma_waveform_type type;
double amplitude;
double frequency;
} ma_waveform_config;
MA_API ma_waveform_config ma_waveform_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_waveform_type type, double amplitude, double frequency);
typedef struct
{
ma_data_source_base ds;
ma_waveform_config config;
double advance;
double time;
} ma_waveform;
MA_API ma_result ma_waveform_init(const ma_waveform_config* pConfig, ma_waveform* pWaveform);
MA_API void ma_waveform_uninit(ma_waveform* pWaveform);
MA_API ma_result ma_waveform_read_pcm_frames(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
MA_API ma_result ma_waveform_seek_to_pcm_frame(ma_waveform* pWaveform, ma_uint64 frameIndex);
MA_API ma_result ma_waveform_set_amplitude(ma_waveform* pWaveform, double amplitude);
MA_API ma_result ma_waveform_set_frequency(ma_waveform* pWaveform, double frequency);
MA_API ma_result ma_waveform_set_type(ma_waveform* pWaveform, ma_waveform_type type);
MA_API ma_result ma_waveform_set_sample_rate(ma_waveform* pWaveform, ma_uint32 sampleRate);
typedef enum
{
ma_noise_type_white,
ma_noise_type_pink,
ma_noise_type_brownian
} ma_noise_type;
typedef struct
{
ma_format format;
ma_uint32 channels;
ma_noise_type type;
ma_int32 seed;
double amplitude;
ma_bool32 duplicateChannels;
} ma_noise_config;
MA_API ma_noise_config ma_noise_config_init(ma_format format, ma_uint32 channels, ma_noise_type type, ma_int32 seed, double amplitude);
typedef struct
{
ma_data_source_vtable ds;
ma_noise_config config;
ma_lcg lcg;
union
{
struct
{
double** bin;
double* accumulation;
ma_uint32* counter;
} pink;
struct
{
double* accumulation;
} brownian;
} state;
/* Memory management. */
void* _pHeap;
ma_bool32 _ownsHeap;
} ma_noise;
MA_API ma_result ma_noise_get_heap_size(const ma_noise_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_noise_init_preallocated(const ma_noise_config* pConfig, void* pHeap, ma_noise* pNoise);
MA_API ma_result ma_noise_init(const ma_noise_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_noise* pNoise);
MA_API void ma_noise_uninit(ma_noise* pNoise, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_noise_read_pcm_frames(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
MA_API ma_result ma_noise_set_amplitude(ma_noise* pNoise, double amplitude);
MA_API ma_result ma_noise_set_seed(ma_noise* pNoise, ma_int32 seed);
MA_API ma_result ma_noise_set_type(ma_noise* pNoise, ma_noise_type type);
#endif /* MA_NO_GENERATION */
/************************************************************************************************************************************************************
Resource Manager
************************************************************************************************************************************************************/
/* The resource manager cannot be enabled if there is no decoder. */
#if !defined(MA_NO_RESOURCE_MANAGER) && defined(MA_NO_DECODING)
#define MA_NO_RESOURCE_MANAGER
#endif
#ifndef MA_NO_RESOURCE_MANAGER
typedef struct ma_resource_manager ma_resource_manager;
typedef struct ma_resource_manager_data_buffer_node ma_resource_manager_data_buffer_node;
typedef struct ma_resource_manager_data_buffer ma_resource_manager_data_buffer;
typedef struct ma_resource_manager_data_stream ma_resource_manager_data_stream;
typedef struct ma_resource_manager_data_source ma_resource_manager_data_source;
typedef enum
{
MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM = 0x00000001, /* When set, does not load the entire data source in memory. Disk I/O will happen on job threads. */
MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE = 0x00000002, /* Decode data before storing in memory. When set, decoding is done at the resource manager level rather than the mixing thread. Results in faster mixing, but higher memory usage...
MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC = 0x00000004, /* When set, the resource manager will load the data source asynchronously. */
MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT = 0x00000008, /* When set, waits for initialization of the underlying data source before returning from ma_resource_manager_data_source_init(). */
MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_UNKNOWN_LENGTH = 0x00000010 /* Gives the resource manager a hint that the length of the data source is unknown and calling `ma_data_source_get_length_in_pcm_frames()` should be avoided. */
} ma_resource_manager_data_source_flags;
/*
Pipeline notifications used by the resource manager. Made up of both an async notification and a fence, both of which are optional.
*/
typedef struct
{
ma_async_notification* pNotification;
ma_fence* pFence;
} ma_resource_manager_pipeline_stage_notification;
typedef struct
{
ma_resource_manager_pipeline_stage_notification init; /* Initialization of the decoder. */
ma_resource_manager_pipeline_stage_notification done; /* Decoding fully completed. */
} ma_resource_manager_pipeline_notifications;
MA_API ma_resource_manager_pipeline_notifications ma_resource_manager_pipeline_notifications_init(void);
/* BEGIN BACKWARDS COMPATIBILITY */
/* TODO: Remove this block in version 0.12. */
#if 1
#define ma_resource_manager_job ma_job
#define ma_resource_manager_job_init ma_job_init
#define MA_JOB_TYPE_RESOURCE_MANAGER_QUEUE_FLAG_NON_BLOCKING MA_JOB_QUEUE_FLAG_NON_BLOCKING
#define ma_resource_manager_job_queue_config ma_job_queue_config
#define ma_resource_manager_job_queue_config_init ma_job_queue_config_init
#define ma_resource_manager_job_queue ma_job_queue
#define ma_resource_manager_job_queue_get_heap_size ma_job_queue_get_heap_size
#define ma_resource_manager_job_queue_init_preallocated ma_job_queue_init_preallocated
#define ma_resource_manager_job_queue_init ma_job_queue_init
#define ma_resource_manager_job_queue_uninit ma_job_queue_uninit
#define ma_resource_manager_job_queue_post ma_job_queue_post
#define ma_resource_manager_job_queue_next ma_job_queue_next
#endif
/* END BACKWARDS COMPATIBILITY */
/* Maximum job thread count will be restricted to this, but this may be removed later and replaced with a heap allocation thereby removing any limitation. */
#ifndef MA_RESOURCE_MANAGER_MAX_JOB_THREAD_COUNT
#define MA_RESOURCE_MANAGER_MAX_JOB_THREAD_COUNT 64
#endif
typedef enum
{
/* Indicates ma_resource_manager_next_job() should not block. Only valid when the job thread count is 0. */
MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING = 0x00000001,
/* Disables any kind of multithreading. Implicitly enables MA_RESOURCE_MANAGER_FLAG_NON_BLOCKING. */
MA_RESOURCE_MANAGER_FLAG_NO_THREADING = 0x00000002
} ma_resource_manager_flags;
typedef struct
{
const char* pFilePath;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
typedef enum
{
ma_resource_manager_data_supply_type_unknown = 0, /* Used for determining whether or the data supply has been initialized. */
ma_resource_manager_data_supply_type_encoded, /* Data supply is an encoded buffer. Connector is ma_decoder. */
ma_resource_manager_data_supply_type_decoded, /* Data supply is a decoded buffer. Connector is ma_audio_buffer. */
ma_resource_manager_data_supply_type_decoded_paged /* Data supply is a linked list of decoded buffers. Connector is ma_paged_audio_buffer. */
} ma_resource_manager_data_supply_type;
typedef struct
{
MA_ATOMIC(4, ma_resource_manager_data_supply_type) type; /* Read and written from different threads so needs to be accessed atomically. */
union
{
struct
{
const void* pData;
size_t sizeInBytes;
} encoded;
struct
{
const void* pData;
ma_uint64 totalFrameCount;
ma_uint64 decodedFrameCount;
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
} decoded;
struct
{
ma_paged_audio_buffer_data data;
ma_uint64 decodedFrameCount;
ma_uint32 sampleRate;
} decodedPaged;
} backend;
} ma_resource_manager_data_supply;
struct ma_resource_manager_data_buffer_node
{
ma_uint32 hashedName32; /* The hashed name. This is the key. */
ma_uint32 refCount;
MA_ATOMIC(4, ma_result) result; /* Result from asynchronous loading. When loading set to MA_BUSY. When fully loaded set to MA_SUCCESS. When deleting set to MA_UNAVAILABLE. */
MA_ATOMIC(4, ma_uint32) executionCounter; /* For allocating execution orders for jobs. */
MA_ATOMIC(4, ma_uint32) executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */
ma_bool32 isDataOwnedByResourceManager; /* Set to true when the underlying data buffer was allocated the resource manager. Set to false if it is owned by the application (via ma_resource_manager_register_*()). */
ma_resource_manager_data_supply data;
ma_resource_manager_data_buffer_node* pParent;
ma_resource_manager_data_buffer_node* pChildLo;
ma_resource_manager_data_buffer_node* pChildHi;
};
struct ma_resource_manager_data_buffer
{
ma_data_source_base ds; /* Base data source. A data buffer is a data source. */
ma_resource_manager* pResourceManager; /* A pointer to the resource manager that owns this buffer. */
ma_resource_manager_data_buffer_node* pNode; /* The data node. This is reference counted and is what supplies the data. */
ma_uint32 flags; /* The flags that were passed used to initialize the buffer. */
MA_ATOMIC(4, ma_uint32) executionCounter; /* For allocating execution orders for jobs. */
MA_ATOMIC(4, ma_uint32) executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */
ma_uint64 seekTargetInPCMFrames; /* Only updated by the public API. Never written nor read from the job thread. */
ma_bool32 seekToCursorOnNextRead; /* On the next read we need to seek to the frame cursor. */
MA_ATOMIC(4, ma_result) result; /* Keeps track of a result of decoding. Set to MA_BUSY while the buffer is still loading. Set to MA_SUCCESS when loading is finished successfully. Otherwise set to some other code. */
MA_ATOMIC(4, ma_bool32) isLooping; /* Can be read and written by different threads at the same time. Must be used atomically. */
ma_bool32 isConnectorInitialized; /* Used for asynchronous loading to ensure we don't try to initialize the connector multiple times while waiting for the node to fully load. */
union
{
ma_decoder decoder; /* Supply type is ma_resource_manager_data_supply_type_encoded */
ma_audio_buffer buffer; /* Supply type is ma_resource_manager_data_supply_type_decoded */
ma_paged_audio_buffer pagedBuffer; /* Supply type is ma_resource_manager_data_supply_type_decoded_paged */
} connector; /* Connects this object to the node's data supply. */
};
struct ma_resource_manager_data_stream
{
ma_data_source_base ds; /* Base data source. A data stream is a data source. */
ma_resource_manager* pResourceManager; /* A pointer to the resource manager that owns this data stream. */
ma_uint32 flags; /* The flags that were passed used to initialize the stream. */
ma_decoder decoder; /* Used for filling pages with data. This is only ever accessed by the job thread. The public API should never touch this. */
ma_bool32 isDecoderInitialized; /* Required for determining whether or not the decoder should be uninitialized in MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_STREAM. */
ma_uint64 totalLengthInPCMFrames; /* This is calculated when first loaded by the MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_STREAM. */
ma_uint32 relativeCursor; /* The playback cursor, relative to the current page. Only ever accessed by the public API. Never accessed by the job thread. */
MA_ATOMIC(8, ma_uint64) absoluteCursor; /* The playback cursor, in absolute position starting from the start of the file. */
ma_uint32 currentPageIndex; /* Toggles between 0 and 1. Index 0 is the first half of pPageData. Index 1 is the second half. Only ever accessed by the public API. Never accessed by the job thread. */
MA_ATOMIC(4, ma_uint32) executionCounter; /* For allocating execution orders for jobs. */
MA_ATOMIC(4, ma_uint32) executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */
/* Written by the public API, read by the job thread. */
MA_ATOMIC(4, ma_bool32) isLooping; /* Whether or not the stream is looping. It's important to set the looping flag at the data stream level for smooth loop transitions. */
/* Written by the job thread, read by the public API. */
void* pPageData; /* Buffer containing the decoded data of each page. Allocated once at initialization time. */
MA_ATOMIC(4, ma_uint32) pageFrameCount[2]; /* The number of valid PCM frames in each page. Used to determine the last valid frame. */
/* Written and read by both the public API and the job thread. These must be atomic. */
MA_ATOMIC(4, ma_result) result; /* Result from asynchronous loading. When loading set to MA_BUSY. When initialized set to MA_SUCCESS. When deleting set to MA_UNAVAILABLE. If an error occurs when loading, set to an error code. */
MA_ATOMIC(4, ma_bool32) isDecoderAtEnd; /* Whether or not the decoder has reached the end. */
MA_ATOMIC(4, ma_bool32) isPageValid[2]; /* Booleans to indicate whether or not a page is valid. Set to false by the public API, set to true by the job thread. Set to false as the pages are consumed, true when they are filled. */
MA_ATOMIC(4, ma_bool32) seekCounter; /* When 0, no seeking is being performed. When > 0, a seek is being performed and reading should be delayed with MA_BUSY. */
};
struct ma_resource_manager_data_source
{
union
{
ma_resource_manager_data_buffer buffer;
ma_resource_manager_data_stream stream;
} backend; /* Must be the first item because we need the first item to be the data source callbacks for the buffer or stream. */
ma_uint32 flags; /* The flags that were passed in to ma_resource_manager_data_source_init(). */
MA_ATOMIC(4, ma_uint32) executionCounter; /* For allocating execution orders for jobs. */
MA_ATOMIC(4, ma_uint32) executionPointer; /* For managing the order of execution for asynchronous jobs relating to this object. Incremented as jobs complete processing. */
};
typedef struct
{
ma_allocation_callbacks allocationCallbacks;
ma_log* pLog;
ma_format decodedFormat; /* The decoded format to use. Set to ma_format_unknown (default) to use the file's native format. */
ma_uint32 decodedChannels; /* The decoded channel count to use. Set to 0 (default) to use the file's native channel count. */
ma_uint32 decodedSampleRate; /* the decoded sample rate to use. Set to 0 (default) to use the file's native sample rate. */
ma_uint32 jobThreadCount; /* Set to 0 if you want to self-manage your job threads. Defaults to 1. */
ma_uint32 jobQueueCapacity; /* The maximum number of jobs that can fit in the queue at a time. Defaults to MA_JOB_TYPE_RESOURCE_MANAGER_QUEUE_CAPACITY. Cannot be zero. */
ma_uint32 flags;
ma_vfs* pVFS; /* Can be NULL in which case defaults will be used. */
ma_decoding_backend_vtable** ppCustomDecodingBackendVTables;
ma_uint32 customDecodingBackendCount;
void* pCustomDecodingBackendUserData;
} ma_resource_manager_config;
MA_API ma_resource_manager_config ma_resource_manager_config_init(void);
struct ma_resource_manager
{
ma_resource_manager_config config;
ma_resource_manager_data_buffer_node* pRootDataBufferNode; /* The root buffer in the binary tree. */
#ifndef MA_NO_THREADING
ma_mutex dataBufferBSTLock; /* For synchronizing access to the data buffer binary tree. */
ma_thread jobThreads[MA_RESOURCE_MANAGER_MAX_JOB_THREAD_COUNT]; /* The threads for executing jobs. */
#endif
ma_job_queue jobQueue; /* Multi-consumer, multi-producer job queue for managing jobs for asynchronous decoding and streaming. */
ma_default_vfs defaultVFS; /* Only used if a custom VFS is not specified. */
ma_log log; /* Only used if no log was specified in the config. */
};
/* Init. */
MA_API ma_result ma_resource_manager_init(const ma_resource_manager_config* pConfig, ma_resource_manager* pResourceManager);
MA_API void ma_resource_manager_uninit(ma_resource_manager* pResourceManager);
MA_API ma_log* ma_resource_manager_get_log(ma_resource_manager* pResourceManager);
/* Registration. */
MA_API ma_result ma_resource_manager_register_file(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags);
MA_API ma_result ma_resource_manager_register_file_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags);
MA_API ma_result ma_resource_manager_register_decoded_data(ma_resource_manager* pResourceManager, const char* pName, const void* pData, ma_uint64 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate); /* Does not copy. Increments t...
MA_API ma_result ma_resource_manager_register_decoded_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName, const void* pData, ma_uint64 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate);
MA_API ma_result ma_resource_manager_register_encoded_data(ma_resource_manager* pResourceManager, const char* pName, const void* pData, size_t sizeInBytes); /* Does not copy. Increments the reference count if already exists and returns MA_SUCCESS....
MA_API ma_result ma_resource_manager_register_encoded_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName, const void* pData, size_t sizeInBytes);
MA_API ma_result ma_resource_manager_unregister_file(ma_resource_manager* pResourceManager, const char* pFilePath);
MA_API ma_result ma_resource_manager_unregister_file_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath);
MA_API ma_result ma_resource_manager_unregister_data(ma_resource_manager* pResourceManager, const char* pName);
MA_API ma_result ma_resource_manager_unregister_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName);
/* Data Buffers. */
MA_API ma_result ma_resource_manager_data_buffer_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_buffer* pDataBuffer);
MA_API ma_result ma_resource_manager_data_buffer_init(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_buffer* pDataBuffer);
MA_API ma_result ma_resource_manager_data_buffer_init_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_buffer* pDataBuffer);
MA_API ma_result ma_resource_manager_data_buffer_init_copy(ma_resource_manager* pResourceManager, const ma_resource_manager_data_buffer* pExistingDataBuffer, ma_resource_manager_data_buffer* pDataBuffer);
MA_API ma_result ma_resource_manager_data_buffer_uninit(ma_resource_manager_data_buffer* pDataBuffer);
MA_API ma_result ma_resource_manager_data_buffer_read_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
MA_API ma_result ma_resource_manager_data_buffer_seek_to_pcm_frame(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64 frameIndex);
MA_API ma_result ma_resource_manager_data_buffer_get_data_format(ma_resource_manager_data_buffer* pDataBuffer, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
MA_API ma_result ma_resource_manager_data_buffer_get_cursor_in_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pCursor);
MA_API ma_result ma_resource_manager_data_buffer_get_length_in_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pLength);
MA_API ma_result ma_resource_manager_data_buffer_result(const ma_resource_manager_data_buffer* pDataBuffer);
MA_API ma_result ma_resource_manager_data_buffer_set_looping(ma_resource_manager_data_buffer* pDataBuffer, ma_bool32 isLooping);
MA_API ma_bool32 ma_resource_manager_data_buffer_is_looping(const ma_resource_manager_data_buffer* pDataBuffer);
MA_API ma_result ma_resource_manager_data_buffer_get_available_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pAvailableFrames);
/* Data Streams. */
MA_API ma_result ma_resource_manager_data_stream_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_stream* pDataStream);
MA_API ma_result ma_resource_manager_data_stream_init(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_stream* pDataStream);
MA_API ma_result ma_resource_manager_data_stream_init_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_stream* pDataStream);
MA_API ma_result ma_resource_manager_data_stream_uninit(ma_resource_manager_data_stream* pDataStream);
MA_API ma_result ma_resource_manager_data_stream_read_pcm_frames(ma_resource_manager_data_stream* pDataStream, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
MA_API ma_result ma_resource_manager_data_stream_seek_to_pcm_frame(ma_resource_manager_data_stream* pDataStream, ma_uint64 frameIndex);
MA_API ma_result ma_resource_manager_data_stream_get_data_format(ma_resource_manager_data_stream* pDataStream, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
MA_API ma_result ma_resource_manager_data_stream_get_cursor_in_pcm_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pCursor);
MA_API ma_result ma_resource_manager_data_stream_get_length_in_pcm_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pLength);
MA_API ma_result ma_resource_manager_data_stream_result(const ma_resource_manager_data_stream* pDataStream);
MA_API ma_result ma_resource_manager_data_stream_set_looping(ma_resource_manager_data_stream* pDataStream, ma_bool32 isLooping);
MA_API ma_bool32 ma_resource_manager_data_stream_is_looping(const ma_resource_manager_data_stream* pDataStream);
MA_API ma_result ma_resource_manager_data_stream_get_available_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pAvailableFrames);
/* Data Sources. */
MA_API ma_result ma_resource_manager_data_source_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_source* pDataSource);
MA_API ma_result ma_resource_manager_data_source_init(ma_resource_manager* pResourceManager, const char* pName, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_source* pDataSource);
MA_API ma_result ma_resource_manager_data_source_init_w(ma_resource_manager* pResourceManager, const wchar_t* pName, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_source* pDataSource);
MA_API ma_result ma_resource_manager_data_source_init_copy(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source* pExistingDataSource, ma_resource_manager_data_source* pDataSource);
MA_API ma_result ma_resource_manager_data_source_uninit(ma_resource_manager_data_source* pDataSource);
MA_API ma_result ma_resource_manager_data_source_read_pcm_frames(ma_resource_manager_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
MA_API ma_result ma_resource_manager_data_source_seek_to_pcm_frame(ma_resource_manager_data_source* pDataSource, ma_uint64 frameIndex);
MA_API ma_result ma_resource_manager_data_source_get_data_format(ma_resource_manager_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
MA_API ma_result ma_resource_manager_data_source_get_cursor_in_pcm_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pCursor);
MA_API ma_result ma_resource_manager_data_source_get_length_in_pcm_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pLength);
MA_API ma_result ma_resource_manager_data_source_result(const ma_resource_manager_data_source* pDataSource);
MA_API ma_result ma_resource_manager_data_source_set_looping(ma_resource_manager_data_source* pDataSource, ma_bool32 isLooping);
MA_API ma_bool32 ma_resource_manager_data_source_is_looping(const ma_resource_manager_data_source* pDataSource);
MA_API ma_result ma_resource_manager_data_source_get_available_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pAvailableFrames);
/* Job management. */
MA_API ma_result ma_resource_manager_post_job(ma_resource_manager* pResourceManager, const ma_job* pJob);
MA_API ma_result ma_resource_manager_post_job_quit(ma_resource_manager* pResourceManager); /* Helper for posting a quit job. */
MA_API ma_result ma_resource_manager_next_job(ma_resource_manager* pResourceManager, ma_job* pJob);
MA_API ma_result ma_resource_manager_process_job(ma_resource_manager* pResourceManager, ma_job* pJob); /* DEPRECATED. Use ma_job_process(). Will be removed in version 0.12. */
MA_API ma_result ma_resource_manager_process_next_job(ma_resource_manager* pResourceManager); /* Returns MA_CANCELLED if a MA_JOB_TYPE_QUIT job is found. In non-blocking mode, returns MA_NO_DATA_AVAILABLE if no jobs are available. */
#endif /* MA_NO_RESOURCE_MANAGER */
/************************************************************************************************************************************************************
Node Graph
************************************************************************************************************************************************************/
#ifndef MA_NO_NODE_GRAPH
/* Must never exceed 254. */
#ifndef MA_MAX_NODE_BUS_COUNT
#define MA_MAX_NODE_BUS_COUNT 254
#endif
/* Used internally by miniaudio for memory management. Must never exceed MA_MAX_NODE_BUS_COUNT. */
#ifndef MA_MAX_NODE_LOCAL_BUS_COUNT
#define MA_MAX_NODE_LOCAL_BUS_COUNT 2
#endif
/* Use this when the bus count is determined by the node instance rather than the vtable. */
#define MA_NODE_BUS_COUNT_UNKNOWN 255
typedef struct ma_node_graph ma_node_graph;
typedef void ma_node;
/* Node flags. */
typedef enum
{
MA_NODE_FLAG_PASSTHROUGH = 0x00000001,
MA_NODE_FLAG_CONTINUOUS_PROCESSING = 0x00000002,
MA_NODE_FLAG_ALLOW_NULL_INPUT = 0x00000004,
MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES = 0x00000008,
MA_NODE_FLAG_SILENT_OUTPUT = 0x00000010
} ma_node_flags;
/* The playback state of a node. Either started or stopped. */
typedef enum
{
ma_node_state_started = 0,
ma_node_state_stopped = 1
} ma_node_state;
typedef struct
{
/*
Extended processing callback. This callback is used for effects that process input and output
at different rates (i.e. they perform resampling). This is similar to the simple version, only
they take two seperate frame counts: one for input, and one for output.
On input, `pFrameCountOut` is equal to the capacity of the output buffer for each bus, whereas
`pFrameCountIn` will be equal to the number of PCM frames in each of the buffers in `ppFramesIn`.
On output, set `pFrameCountOut` to the number of PCM frames that were actually output and set
`pFrameCountIn` to the number of input frames that were consumed.
*/
void (* onProcess)(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut);
/*
A callback for retrieving the number of a input frames that are required to output the
specified number of output frames. You would only want to implement this when the node performs
resampling. This is optional, even for nodes that perform resampling, but it does offer a
small reduction in latency as it allows miniaudio to calculate the exact number of input frames
to read at a time instead of having to estimate.
*/
ma_result (* onGetRequiredInputFrameCount)(ma_node* pNode, ma_uint32 outputFrameCount, ma_uint32* pInputFrameCount);
/*
The number of input buses. This is how many sub-buffers will be contained in the `ppFramesIn`
parameters of the callbacks above.
*/
ma_uint8 inputBusCount;
/*
The number of output buses. This is how many sub-buffers will be contained in the `ppFramesOut`
parameters of the callbacks above.
*/
ma_uint8 outputBusCount;
/*
Flags describing characteristics of the node. This is currently just a placeholder for some
ideas for later on.
*/
ma_uint32 flags;
} ma_node_vtable;
typedef struct
{
const ma_node_vtable* vtable; /* Should never be null. Initialization of the node will fail if so. */
ma_node_state initialState; /* Defaults to ma_node_state_started. */
ma_uint32 inputBusCount; /* Only used if the vtable specifies an input bus count of `MA_NODE_BUS_COUNT_UNKNOWN`, otherwise must be set to `MA_NODE_BUS_COUNT_UNKNOWN` (default). */
ma_uint32 outputBusCount; /* Only used if the vtable specifies an output bus count of `MA_NODE_BUS_COUNT_UNKNOWN`, otherwise be set to `MA_NODE_BUS_COUNT_UNKNOWN` (default). */
const ma_uint32* pInputChannels; /* The number of elements are determined by the input bus count as determined by the vtable, or `inputBusCount` if the vtable specifies `MA_NODE_BUS_COUNT_UNKNOWN`. */
const ma_uint32* pOutputChannels; /* The number of elements are determined by the output bus count as determined by the vtable, or `outputBusCount` if the vtable specifies `MA_NODE_BUS_COUNT_UNKNOWN`. */
} ma_node_config;
MA_API ma_node_config ma_node_config_init(void);
/*
A node has multiple output buses. An output bus is attached to an input bus as an item in a linked
list. Think of the input bus as a linked list, with the output bus being an item in that list.
*/
typedef struct ma_node_output_bus ma_node_output_bus;
struct ma_node_output_bus
{
/* Immutable. */
ma_node* pNode; /* The node that owns this output bus. The input node. Will be null for dummy head and tail nodes. */
ma_uint8 outputBusIndex; /* The index of the output bus on pNode that this output bus represents. */
ma_uint8 channels; /* The number of channels in the audio stream for this bus. */
/* Mutable via multiple threads. Must be used atomically. The weird ordering here is for packing reasons. */
MA_ATOMIC(1, ma_uint8) inputNodeInputBusIndex; /* The index of the input bus on the input. Required for detaching. */
MA_ATOMIC(4, ma_uint32) flags; /* Some state flags for tracking the read state of the output buffer. A combination of MA_NODE_OUTPUT_BUS_FLAG_*. */
MA_ATOMIC(4, ma_uint32) refCount; /* Reference count for some thread-safety when detaching. */
MA_ATOMIC(4, ma_bool32) isAttached; /* This is used to prevent iteration of nodes that are in the middle of being detached. Used for thread safety. */
MA_ATOMIC(4, ma_spinlock) lock; /* Unfortunate lock, but significantly simplifies the implementation. Required for thread-safe attaching and detaching. */
MA_ATOMIC(4, float) volume; /* Linear. */
MA_ATOMIC(MA_SIZEOF_PTR, ma_node_output_bus*) pNext; /* If null, it's the tail node or detached. */
MA_ATOMIC(MA_SIZEOF_PTR, ma_node_output_bus*) pPrev; /* If null, it's the head node or detached. */
MA_ATOMIC(MA_SIZEOF_PTR, ma_node*) pInputNode; /* The node that this output bus is attached to. Required for detaching. */
};
/*
A node has multiple input buses. The output buses of a node are connecting to the input busses of
another. An input bus is essentially just a linked list of output buses.
*/
typedef struct ma_node_input_bus ma_node_input_bus;
struct ma_node_input_bus
{
/* Mutable via multiple threads. */
ma_node_output_bus head; /* Dummy head node for simplifying some lock-free thread-safety stuff. */
MA_ATOMIC(4, ma_uint32) nextCounter; /* This is used to determine whether or not the input bus is finding the next node in the list. Used for thread safety when detaching output buses. */
MA_ATOMIC(4, ma_spinlock) lock; /* Unfortunate lock, but significantly simplifies the implementation. Required for thread-safe attaching and detaching. */
/* Set once at startup. */
ma_uint8 channels; /* The number of channels in the audio stream for this bus. */
};
typedef struct ma_node_base ma_node_base;
struct ma_node_base
{
/* These variables are set once at startup. */
ma_node_graph* pNodeGraph; /* The graph this node belongs to. */
const ma_node_vtable* vtable;
float* pCachedData; /* Allocated on the heap. Fixed size. Needs to be stored on the heap because reading from output buses is done in separate function calls. */
ma_uint16 cachedDataCapInFramesPerBus; /* The capacity of the input data cache in frames, per bus. */
/* These variables are read and written only from the audio thread. */
ma_uint16 cachedFrameCountOut;
ma_uint16 cachedFrameCountIn;
ma_uint16 consumedFrameCountIn;
/* These variables are read and written between different threads. */
MA_ATOMIC(4, ma_node_state) state; /* When set to stopped, nothing will be read, regardless of the times in stateTimes. */
MA_ATOMIC(8, ma_uint64) stateTimes[2]; /* Indexed by ma_node_state. Specifies the time based on the global clock that a node should be considered to be in the relevant state. */
MA_ATOMIC(8, ma_uint64) localTime; /* The node's local clock. This is just a running sum of the number of output frames that have been processed. Can be modified by any thread with `ma_node_set_time()`. */
ma_uint32 inputBusCount;
ma_uint32 outputBusCount;
ma_node_input_bus* pInputBuses;
ma_node_output_bus* pOutputBuses;
/* Memory management. */
ma_node_input_bus _inputBuses[MA_MAX_NODE_LOCAL_BUS_COUNT];
ma_node_output_bus _outputBuses[MA_MAX_NODE_LOCAL_BUS_COUNT];
void* _pHeap; /* A heap allocation for internal use only. pInputBuses and/or pOutputBuses will point to this if the bus count exceeds MA_MAX_NODE_LOCAL_BUS_COUNT. */
ma_bool32 _ownsHeap; /* If set to true, the node owns the heap allocation and _pHeap will be freed in ma_node_uninit(). */
};
MA_API ma_result ma_node_get_heap_size(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_node_init_preallocated(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, void* pHeap, ma_node* pNode);
MA_API ma_result ma_node_init(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_node* pNode);
MA_API void ma_node_uninit(ma_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_node_graph* ma_node_get_node_graph(const ma_node* pNode);
MA_API ma_uint32 ma_node_get_input_bus_count(const ma_node* pNode);
MA_API ma_uint32 ma_node_get_output_bus_count(const ma_node* pNode);
MA_API ma_uint32 ma_node_get_input_channels(const ma_node* pNode, ma_uint32 inputBusIndex);
MA_API ma_uint32 ma_node_get_output_channels(const ma_node* pNode, ma_uint32 outputBusIndex);
MA_API ma_result ma_node_attach_output_bus(ma_node* pNode, ma_uint32 outputBusIndex, ma_node* pOtherNode, ma_uint32 otherNodeInputBusIndex);
MA_API ma_result ma_node_detach_output_bus(ma_node* pNode, ma_uint32 outputBusIndex);
MA_API ma_result ma_node_detach_all_output_buses(ma_node* pNode);
MA_API ma_result ma_node_set_output_bus_volume(ma_node* pNode, ma_uint32 outputBusIndex, float volume);
MA_API float ma_node_get_output_bus_volume(const ma_node* pNode, ma_uint32 outputBusIndex);
MA_API ma_result ma_node_set_state(ma_node* pNode, ma_node_state state);
MA_API ma_node_state ma_node_get_state(const ma_node* pNode);
MA_API ma_result ma_node_set_state_time(ma_node* pNode, ma_node_state state, ma_uint64 globalTime);
MA_API ma_uint64 ma_node_get_state_time(const ma_node* pNode, ma_node_state state);
MA_API ma_node_state ma_node_get_state_by_time(const ma_node* pNode, ma_uint64 globalTime);
MA_API ma_node_state ma_node_get_state_by_time_range(const ma_node* pNode, ma_uint64 globalTimeBeg, ma_uint64 globalTimeEnd);
MA_API ma_uint64 ma_node_get_time(const ma_node* pNode);
MA_API ma_result ma_node_set_time(ma_node* pNode, ma_uint64 localTime);
typedef struct
{
ma_uint32 channels;
ma_uint16 nodeCacheCapInFrames;
} ma_node_graph_config;
MA_API ma_node_graph_config ma_node_graph_config_init(ma_uint32 channels);
struct ma_node_graph
{
/* Immutable. */
ma_node_base base; /* The node graph itself is a node so it can be connected as an input to different node graph. This has zero inputs and calls ma_node_graph_read_pcm_frames() to generate it's output. */
ma_node_base endpoint; /* Special node that all nodes eventually connect to. Data is read from this node in ma_node_graph_read_pcm_frames(). */
ma_uint16 nodeCacheCapInFrames;
/* Read and written by multiple threads. */
MA_ATOMIC(4, ma_bool32) isReading;
};
MA_API ma_result ma_node_graph_init(const ma_node_graph_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_node_graph* pNodeGraph);
MA_API void ma_node_graph_uninit(ma_node_graph* pNodeGraph, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_node* ma_node_graph_get_endpoint(ma_node_graph* pNodeGraph);
MA_API ma_result ma_node_graph_read_pcm_frames(ma_node_graph* pNodeGraph, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
MA_API ma_uint32 ma_node_graph_get_channels(const ma_node_graph* pNodeGraph);
MA_API ma_uint64 ma_node_graph_get_time(const ma_node_graph* pNodeGraph);
MA_API ma_result ma_node_graph_set_time(ma_node_graph* pNodeGraph, ma_uint64 globalTime);
/* Data source node. 0 input buses, 1 output bus. Used for reading from a data source. */
typedef struct
{
ma_node_config nodeConfig;
ma_data_source* pDataSource;
} ma_data_source_node_config;
MA_API ma_data_source_node_config ma_data_source_node_config_init(ma_data_source* pDataSource);
typedef struct
{
ma_node_base base;
ma_data_source* pDataSource;
} ma_data_source_node;
MA_API ma_result ma_data_source_node_init(ma_node_graph* pNodeGraph, const ma_data_source_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source_node* pDataSourceNode);
MA_API void ma_data_source_node_uninit(ma_data_source_node* pDataSourceNode, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_data_source_node_set_looping(ma_data_source_node* pDataSourceNode, ma_bool32 isLooping);
MA_API ma_bool32 ma_data_source_node_is_looping(ma_data_source_node* pDataSourceNode);
/* Splitter Node. 1 input, 2 outputs. Used for splitting/copying a stream so it can be as input into two separate output nodes. */
typedef struct
{
ma_node_config nodeConfig;
ma_uint32 channels;
} ma_splitter_node_config;
MA_API ma_splitter_node_config ma_splitter_node_config_init(ma_uint32 channels);
typedef struct
{
ma_node_base base;
} ma_splitter_node;
MA_API ma_result ma_splitter_node_init(ma_node_graph* pNodeGraph, const ma_splitter_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_splitter_node* pSplitterNode);
MA_API void ma_splitter_node_uninit(ma_splitter_node* pSplitterNode, const ma_allocation_callbacks* pAllocationCallbacks);
/*
Biquad Node
*/
typedef struct
{
ma_node_config nodeConfig;
ma_biquad_config biquad;
} ma_biquad_node_config;
MA_API ma_biquad_node_config ma_biquad_node_config_init(ma_uint32 channels, float b0, float b1, float b2, float a0, float a1, float a2);
typedef struct
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} ma_engine_node_config;
MA_API ma_engine_node_config ma_engine_node_config_init(ma_engine* pEngine, ma_engine_node_type type, ma_uint32 flags);
/* Base node object for both ma_sound and ma_sound_group. */
typedef struct
{
ma_node_base baseNode; /* Must be the first member for compatiblity with the ma_node API. */
ma_engine* pEngine; /* A pointer to the engine. Set based on the value from the config. */
ma_uint32 sampleRate; /* The sample rate of the input data. For sounds backed by a data source, this will be the data source's sample rate. Otherwise it'll be the engine's sample rate. */
ma_fader fader;
ma_linear_resampler resampler; /* For pitch shift. */
ma_spatializer spatializer;
ma_panner panner;
MA_ATOMIC(4, float) pitch;
float oldPitch; /* For determining whether or not the resampler needs to be updated to reflect the new pitch. The resampler will be updated on the mixing thread. */
float oldDopplerPitch; /* For determining whether or not the resampler needs to be updated to take a new doppler pitch into account. */
MA_ATOMIC(4, ma_bool32) isPitchDisabled; /* When set to true, pitching will be disabled which will allow the resampler to be bypassed to save some computation. */
MA_ATOMIC(4, ma_bool32) isSpatializationDisabled; /* Set to false by default. When set to false, will not have spatialisation applied. */
MA_ATOMIC(4, ma_uint32) pinnedListenerIndex; /* The index of the listener this node should always use for spatialization. If set to MA_LISTENER_INDEX_CLOSEST the engine will use the closest listener. */
/* Memory management. */
ma_bool8 _ownsHeap;
void* _pHeap;
} ma_engine_node;
MA_API ma_result ma_engine_node_get_heap_size(const ma_engine_node_config* pConfig, size_t* pHeapSizeInBytes);
MA_API ma_result ma_engine_node_init_preallocated(const ma_engine_node_config* pConfig, void* pHeap, ma_engine_node* pEngineNode);
MA_API ma_result ma_engine_node_init(const ma_engine_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_engine_node* pEngineNode);
MA_API void ma_engine_node_uninit(ma_engine_node* pEngineNode, const ma_allocation_callbacks* pAllocationCallbacks);
#define MA_SOUND_SOURCE_CHANNEL_COUNT 0xFFFFFFFF
typedef struct
{
const char* pFilePath; /* Set this to load from the resource manager. */
const wchar_t* pFilePathW; /* Set this to load from the resource manager. */
ma_data_source* pDataSource; /* Set this to load from an existing data source. */
ma_node* pInitialAttachment; /* If set, the sound will be attached to an input of this node. This can be set to a ma_sound. If set to NULL, the sound will be attached directly to the endpoint unless MA_SOUND_FLAG_NO_DEFAULT_ATTACHM...
ma_uint32 initialAttachmentInputBusIndex; /* The index of the input bus of pInitialAttachment to attach the sound to. */
ma_uint32 channelsIn; /* Ignored if using a data source as input (the data source's channel count will be used always). Otherwise, setting to 0 will cause the engine's channel count to be used. */
ma_uint32 channelsOut; /* Set this to 0 (default) to use the engine's channel count. Set to MA_SOUND_SOURCE_CHANNEL_COUNT to use the data source's channel count (only used if using a data source as input). */
ma_uint32 flags; /* A combination of MA_SOUND_FLAG_* flags. */
ma_uint64 initialSeekPointInPCMFrames; /* Initializes the sound such that it's seeked to this location by default. */
ma_uint64 rangeBegInPCMFrames;
ma_uint64 rangeEndInPCMFrames;
ma_uint64 loopPointBegInPCMFrames;
ma_uint64 loopPointEndInPCMFrames;
ma_bool32 isLooping;
ma_fence* pDoneFence; /* Released when the resource manager has finished decoding the entire sound. Not used with streams. */
} ma_sound_config;
MA_API ma_sound_config ma_sound_config_init(void);
struct ma_sound
{
ma_engine_node engineNode; /* Must be the first member for compatibility with the ma_node API. */
ma_data_source* pDataSource;
MA_ATOMIC(8, ma_uint64) seekTarget; /* The PCM frame index to seek to in the mixing thread. Set to (~(ma_uint64)0) to not perform any seeking. */
MA_ATOMIC(4, ma_bool32) atEnd;
ma_bool8 ownsDataSource;
/*
We're declaring a resource manager data source object here to save us a malloc when loading a
sound via the resource manager, which I *think* will be the most common scenario.
*/
#ifndef MA_NO_RESOURCE_MANAGER
ma_resource_manager_data_source* pResourceManagerDataSource;
#endif
};
/* Structure specifically for sounds played with ma_engine_play_sound(). Making this a separate structure to reduce overhead. */
typedef struct ma_sound_inlined ma_sound_inlined;
struct ma_sound_inlined
{
ma_sound sound;
ma_sound_inlined* pNext;
ma_sound_inlined* pPrev;
};
/* A sound group is just a sound. */
typedef ma_sound_config ma_sound_group_config;
typedef ma_sound ma_sound_group;
MA_API ma_sound_group_config ma_sound_group_config_init(void);
typedef struct
{
#if !defined(MA_NO_RESOURCE_MANAGER)
ma_resource_manager* pResourceManager; /* Can be null in which case a resource manager will be created for you. */
#endif
#if !defined(MA_NO_DEVICE_IO)
ma_context* pContext;
ma_device* pDevice; /* If set, the caller is responsible for calling ma_engine_data_callback() in the device's data callback. */
ma_device_id* pPlaybackDeviceID; /* The ID of the playback device to use with the default listener. */
#endif
ma_log* pLog; /* When set to NULL, will use the context's log. */
ma_uint32 listenerCount; /* Must be between 1 and MA_ENGINE_MAX_LISTENERS. */
ma_uint32 channels; /* The number of channels to use when mixing and spatializing. When set to 0, will use the native channel count of the device. */
ma_uint32 sampleRate; /* The sample rate. When set to 0 will use the native channel count of the device. */
ma_uint32 periodSizeInFrames; /* If set to something other than 0, updates will always be exactly this size. The underlying device may be a different size, but from the perspective of the mixer that won't matter.*/
ma_uint32 periodSizeInMilliseconds; /* Used if periodSizeInFrames is unset. */
ma_uint32 gainSmoothTimeInFrames; /* The number of frames to interpolate the gain of spatialized sounds across. If set to 0, will use gainSmoothTimeInMilliseconds. */
ma_uint32 gainSmoothTimeInMilliseconds; /* When set to 0, gainSmoothTimeInFrames will be used. If both are set to 0, a default value will be used. */
ma_allocation_callbacks allocationCallbacks;
ma_bool32 noAutoStart; /* When set to true, requires an explicit call to ma_engine_start(). This is false by default, meaning the engine will be started automatically in ma_engine_init(). */
ma_bool32 noDevice; /* When set to true, don't create a default device. ma_engine_read_pcm_frames() can be called manually to read data. */
ma_mono_expansion_mode monoExpansionMode; /* Controls how the mono channel should be expanded to other channels when spatialization is disabled on a sound. */
ma_vfs* pResourceManagerVFS; /* A pointer to a pre-allocated VFS object to use with the resource manager. This is ignored if pResourceManager is not NULL. */
} ma_engine_config;
MA_API ma_engine_config ma_engine_config_init(void);
struct ma_engine
{
ma_node_graph nodeGraph; /* An engine is a node graph. It should be able to be plugged into any ma_node_graph API (with a cast) which means this must be the first member of this struct. */
#if !defined(MA_NO_RESOURCE_MANAGER)
ma_resource_manager* pResourceManager;
#endif
#if !defined(MA_NO_DEVICE_IO)
ma_device* pDevice; /* Optionally set via the config, otherwise allocated by the engine in ma_engine_init(). */
#endif
ma_log* pLog;
ma_uint32 sampleRate;
ma_uint32 listenerCount;
ma_spatializer_listener listeners[MA_ENGINE_MAX_LISTENERS];
ma_allocation_callbacks allocationCallbacks;
ma_bool8 ownsResourceManager;
ma_bool8 ownsDevice;
ma_spinlock inlinedSoundLock; /* For synchronizing access so the inlined sound list. */
ma_sound_inlined* pInlinedSoundHead; /* The first inlined sound. Inlined sounds are tracked in a linked list. */
MA_ATOMIC(4, ma_uint32) inlinedSoundCount; /* The total number of allocated inlined sound objects. Used for debugging. */
ma_uint32 gainSmoothTimeInFrames; /* The number of frames to interpolate the gain of spatialized sounds across. */
ma_mono_expansion_mode monoExpansionMode;
};
MA_API ma_result ma_engine_init(const ma_engine_config* pConfig, ma_engine* pEngine);
MA_API void ma_engine_uninit(ma_engine* pEngine);
MA_API ma_result ma_engine_read_pcm_frames(ma_engine* pEngine, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
MA_API ma_node_graph* ma_engine_get_node_graph(ma_engine* pEngine);
#if !defined(MA_NO_RESOURCE_MANAGER)
MA_API ma_resource_manager* ma_engine_get_resource_manager(ma_engine* pEngine);
#endif
MA_API ma_device* ma_engine_get_device(ma_engine* pEngine);
MA_API ma_log* ma_engine_get_log(ma_engine* pEngine);
MA_API ma_node* ma_engine_get_endpoint(ma_engine* pEngine);
MA_API ma_uint64 ma_engine_get_time(const ma_engine* pEngine);
MA_API ma_result ma_engine_set_time(ma_engine* pEngine, ma_uint64 globalTime);
MA_API ma_uint32 ma_engine_get_channels(const ma_engine* pEngine);
MA_API ma_uint32 ma_engine_get_sample_rate(const ma_engine* pEngine);
MA_API ma_result ma_engine_start(ma_engine* pEngine);
MA_API ma_result ma_engine_stop(ma_engine* pEngine);
MA_API ma_result ma_engine_set_volume(ma_engine* pEngine, float volume);
MA_API ma_result ma_engine_set_gain_db(ma_engine* pEngine, float gainDB);
MA_API ma_uint32 ma_engine_get_listener_count(const ma_engine* pEngine);
MA_API ma_uint32 ma_engine_find_closest_listener(const ma_engine* pEngine, float absolutePosX, float absolutePosY, float absolutePosZ);
MA_API void ma_engine_listener_set_position(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z);
MA_API ma_vec3f ma_engine_listener_get_position(const ma_engine* pEngine, ma_uint32 listenerIndex);
MA_API void ma_engine_listener_set_direction(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z);
MA_API ma_vec3f ma_engine_listener_get_direction(const ma_engine* pEngine, ma_uint32 listenerIndex);
MA_API void ma_engine_listener_set_velocity(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z);
MA_API ma_vec3f ma_engine_listener_get_velocity(const ma_engine* pEngine, ma_uint32 listenerIndex);
MA_API void ma_engine_listener_set_cone(ma_engine* pEngine, ma_uint32 listenerIndex, float innerAngleInRadians, float outerAngleInRadians, float outerGain);
MA_API void ma_engine_listener_get_cone(const ma_engine* pEngine, ma_uint32 listenerIndex, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain);
MA_API void ma_engine_listener_set_world_up(ma_engine* pEngine, ma_uint32 listenerIndex, float x, float y, float z);
MA_API ma_vec3f ma_engine_listener_get_world_up(const ma_engine* pEngine, ma_uint32 listenerIndex);
MA_API void ma_engine_listener_set_enabled(ma_engine* pEngine, ma_uint32 listenerIndex, ma_bool32 isEnabled);
MA_API ma_bool32 ma_engine_listener_is_enabled(const ma_engine* pEngine, ma_uint32 listenerIndex);
#ifndef MA_NO_RESOURCE_MANAGER
MA_API ma_result ma_engine_play_sound_ex(ma_engine* pEngine, const char* pFilePath, ma_node* pNode, ma_uint32 nodeInputBusIndex);
MA_API ma_result ma_engine_play_sound(ma_engine* pEngine, const char* pFilePath, ma_sound_group* pGroup); /* Fire and forget. */
#endif
#ifndef MA_NO_RESOURCE_MANAGER
MA_API ma_result ma_sound_init_from_file(ma_engine* pEngine, const char* pFilePath, ma_uint32 flags, ma_sound_group* pGroup, ma_fence* pDoneFence, ma_sound* pSound);
MA_API ma_result ma_sound_init_from_file_w(ma_engine* pEngine, const wchar_t* pFilePath, ma_uint32 flags, ma_sound_group* pGroup, ma_fence* pDoneFence, ma_sound* pSound);
MA_API ma_result ma_sound_init_copy(ma_engine* pEngine, const ma_sound* pExistingSound, ma_uint32 flags, ma_sound_group* pGroup, ma_sound* pSound);
#endif
MA_API ma_result ma_sound_init_from_data_source(ma_engine* pEngine, ma_data_source* pDataSource, ma_uint32 flags, ma_sound_group* pGroup, ma_sound* pSound);
MA_API ma_result ma_sound_init_ex(ma_engine* pEngine, const ma_sound_config* pConfig, ma_sound* pSound);
MA_API void ma_sound_uninit(ma_sound* pSound);
MA_API ma_engine* ma_sound_get_engine(const ma_sound* pSound);
MA_API ma_data_source* ma_sound_get_data_source(const ma_sound* pSound);
MA_API ma_result ma_sound_start(ma_sound* pSound);
MA_API ma_result ma_sound_stop(ma_sound* pSound);
MA_API void ma_sound_set_volume(ma_sound* pSound, float volume);
MA_API float ma_sound_get_volume(const ma_sound* pSound);
MA_API void ma_sound_set_pan(ma_sound* pSound, float pan);
MA_API float ma_sound_get_pan(const ma_sound* pSound);
MA_API void ma_sound_set_pan_mode(ma_sound* pSound, ma_pan_mode panMode);
MA_API ma_pan_mode ma_sound_get_pan_mode(const ma_sound* pSound);
MA_API void ma_sound_set_pitch(ma_sound* pSound, float pitch);
MA_API float ma_sound_get_pitch(const ma_sound* pSound);
MA_API void ma_sound_set_spatialization_enabled(ma_sound* pSound, ma_bool32 enabled);
MA_API ma_bool32 ma_sound_is_spatialization_enabled(const ma_sound* pSound);
MA_API void ma_sound_set_pinned_listener_index(ma_sound* pSound, ma_uint32 listenerIndex);
MA_API ma_uint32 ma_sound_get_pinned_listener_index(const ma_sound* pSound);
MA_API ma_uint32 ma_sound_get_listener_index(const ma_sound* pSound);
MA_API ma_vec3f ma_sound_get_direction_to_listener(const ma_sound* pSound);
MA_API void ma_sound_set_position(ma_sound* pSound, float x, float y, float z);
MA_API ma_vec3f ma_sound_get_position(const ma_sound* pSound);
MA_API void ma_sound_set_direction(ma_sound* pSound, float x, float y, float z);
MA_API ma_vec3f ma_sound_get_direction(const ma_sound* pSound);
MA_API void ma_sound_set_velocity(ma_sound* pSound, float x, float y, float z);
MA_API ma_vec3f ma_sound_get_velocity(const ma_sound* pSound);
MA_API void ma_sound_set_attenuation_model(ma_sound* pSound, ma_attenuation_model attenuationModel);
MA_API ma_attenuation_model ma_sound_get_attenuation_model(const ma_sound* pSound);
MA_API void ma_sound_set_positioning(ma_sound* pSound, ma_positioning positioning);
MA_API ma_positioning ma_sound_get_positioning(const ma_sound* pSound);
MA_API void ma_sound_set_rolloff(ma_sound* pSound, float rolloff);
MA_API float ma_sound_get_rolloff(const ma_sound* pSound);
MA_API void ma_sound_set_min_gain(ma_sound* pSound, float minGain);
MA_API float ma_sound_get_min_gain(const ma_sound* pSound);
MA_API void ma_sound_set_max_gain(ma_sound* pSound, float maxGain);
MA_API float ma_sound_get_max_gain(const ma_sound* pSound);
MA_API void ma_sound_set_min_distance(ma_sound* pSound, float minDistance);
MA_API float ma_sound_get_min_distance(const ma_sound* pSound);
MA_API void ma_sound_set_max_distance(ma_sound* pSound, float maxDistance);
MA_API float ma_sound_get_max_distance(const ma_sound* pSound);
MA_API void ma_sound_set_cone(ma_sound* pSound, float innerAngleInRadians, float outerAngleInRadians, float outerGain);
MA_API void ma_sound_get_cone(const ma_sound* pSound, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain);
MA_API void ma_sound_set_doppler_factor(ma_sound* pSound, float dopplerFactor);
MA_API float ma_sound_get_doppler_factor(const ma_sound* pSound);
MA_API void ma_sound_set_directional_attenuation_factor(ma_sound* pSound, float directionalAttenuationFactor);
MA_API float ma_sound_get_directional_attenuation_factor(const ma_sound* pSound);
MA_API void ma_sound_set_fade_in_pcm_frames(ma_sound* pSound, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInFrames);
MA_API void ma_sound_set_fade_in_milliseconds(ma_sound* pSound, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInMilliseconds);
MA_API float ma_sound_get_current_fade_volume(ma_sound* pSound);
MA_API void ma_sound_set_start_time_in_pcm_frames(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInFrames);
MA_API void ma_sound_set_start_time_in_milliseconds(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInMilliseconds);
MA_API void ma_sound_set_stop_time_in_pcm_frames(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInFrames);
MA_API void ma_sound_set_stop_time_in_milliseconds(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInMilliseconds);
MA_API ma_bool32 ma_sound_is_playing(const ma_sound* pSound);
MA_API ma_uint64 ma_sound_get_time_in_pcm_frames(const ma_sound* pSound);
MA_API void ma_sound_set_looping(ma_sound* pSound, ma_bool32 isLooping);
MA_API ma_bool32 ma_sound_is_looping(const ma_sound* pSound);
MA_API ma_bool32 ma_sound_at_end(const ma_sound* pSound);
MA_API ma_result ma_sound_seek_to_pcm_frame(ma_sound* pSound, ma_uint64 frameIndex); /* Just a wrapper around ma_data_source_seek_to_pcm_frame(). */
MA_API ma_result ma_sound_get_data_format(ma_sound* pSound, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
MA_API ma_result ma_sound_get_cursor_in_pcm_frames(ma_sound* pSound, ma_uint64* pCursor);
MA_API ma_result ma_sound_get_length_in_pcm_frames(ma_sound* pSound, ma_uint64* pLength);
MA_API ma_result ma_sound_get_cursor_in_seconds(ma_sound* pSound, float* pCursor);
MA_API ma_result ma_sound_get_length_in_seconds(ma_sound* pSound, float* pLength);
MA_API ma_result ma_sound_group_init(ma_engine* pEngine, ma_uint32 flags, ma_sound_group* pParentGroup, ma_sound_group* pGroup);
MA_API ma_result ma_sound_group_init_ex(ma_engine* pEngine, const ma_sound_group_config* pConfig, ma_sound_group* pGroup);
MA_API void ma_sound_group_uninit(ma_sound_group* pGroup);
MA_API ma_engine* ma_sound_group_get_engine(const ma_sound_group* pGroup);
MA_API ma_result ma_sound_group_start(ma_sound_group* pGroup);
MA_API ma_result ma_sound_group_stop(ma_sound_group* pGroup);
MA_API void ma_sound_group_set_volume(ma_sound_group* pGroup, float volume);
MA_API float ma_sound_group_get_volume(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_pan(ma_sound_group* pGroup, float pan);
MA_API float ma_sound_group_get_pan(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_pan_mode(ma_sound_group* pGroup, ma_pan_mode panMode);
MA_API ma_pan_mode ma_sound_group_get_pan_mode(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_pitch(ma_sound_group* pGroup, float pitch);
MA_API float ma_sound_group_get_pitch(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_spatialization_enabled(ma_sound_group* pGroup, ma_bool32 enabled);
MA_API ma_bool32 ma_sound_group_is_spatialization_enabled(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_pinned_listener_index(ma_sound_group* pGroup, ma_uint32 listenerIndex);
MA_API ma_uint32 ma_sound_group_get_pinned_listener_index(const ma_sound_group* pGroup);
MA_API ma_uint32 ma_sound_group_get_listener_index(const ma_sound_group* pGroup);
MA_API ma_vec3f ma_sound_group_get_direction_to_listener(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_position(ma_sound_group* pGroup, float x, float y, float z);
MA_API ma_vec3f ma_sound_group_get_position(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_direction(ma_sound_group* pGroup, float x, float y, float z);
MA_API ma_vec3f ma_sound_group_get_direction(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_velocity(ma_sound_group* pGroup, float x, float y, float z);
MA_API ma_vec3f ma_sound_group_get_velocity(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_attenuation_model(ma_sound_group* pGroup, ma_attenuation_model attenuationModel);
MA_API ma_attenuation_model ma_sound_group_get_attenuation_model(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_positioning(ma_sound_group* pGroup, ma_positioning positioning);
MA_API ma_positioning ma_sound_group_get_positioning(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_rolloff(ma_sound_group* pGroup, float rolloff);
MA_API float ma_sound_group_get_rolloff(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_min_gain(ma_sound_group* pGroup, float minGain);
MA_API float ma_sound_group_get_min_gain(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_max_gain(ma_sound_group* pGroup, float maxGain);
MA_API float ma_sound_group_get_max_gain(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_min_distance(ma_sound_group* pGroup, float minDistance);
MA_API float ma_sound_group_get_min_distance(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_max_distance(ma_sound_group* pGroup, float maxDistance);
MA_API float ma_sound_group_get_max_distance(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_cone(ma_sound_group* pGroup, float innerAngleInRadians, float outerAngleInRadians, float outerGain);
MA_API void ma_sound_group_get_cone(const ma_sound_group* pGroup, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain);
MA_API void ma_sound_group_set_doppler_factor(ma_sound_group* pGroup, float dopplerFactor);
MA_API float ma_sound_group_get_doppler_factor(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_directional_attenuation_factor(ma_sound_group* pGroup, float directionalAttenuationFactor);
MA_API float ma_sound_group_get_directional_attenuation_factor(const ma_sound_group* pGroup);
MA_API void ma_sound_group_set_fade_in_pcm_frames(ma_sound_group* pGroup, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInFrames);
MA_API void ma_sound_group_set_fade_in_milliseconds(ma_sound_group* pGroup, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInMilliseconds);
MA_API float ma_sound_group_get_current_fade_volume(ma_sound_group* pGroup);
MA_API void ma_sound_group_set_start_time_in_pcm_frames(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInFrames);
MA_API void ma_sound_group_set_start_time_in_milliseconds(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInMilliseconds);
MA_API void ma_sound_group_set_stop_time_in_pcm_frames(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInFrames);
MA_API void ma_sound_group_set_stop_time_in_milliseconds(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInMilliseconds);
MA_API ma_bool32 ma_sound_group_is_playing(const ma_sound_group* pGroup);
MA_API ma_uint64 ma_sound_group_get_time_in_pcm_frames(const ma_sound_group* pGroup);
#endif /* MA_NO_ENGINE */
#ifdef __cplusplus
}
#endif
#endif /* miniaudio_h */
/*
This is for preventing greying out of the implementation section.
*/
#if defined(Q_CREATOR_RUN) || defined(__INTELLISENSE__) || defined(__CDT_PARSER__)
#define MINIAUDIO_IMPLEMENTATION
#endif
/************************************************************************************************************************************************************
*************************************************************************************************************************************************************
IMPLEMENTATION
*************************************************************************************************************************************************************
************************************************************************************************************************************************************/
#if defined(MINIAUDIO_IMPLEMENTATION) || defined(MA_IMPLEMENTATION)
#ifndef miniaudio_c
#define miniaudio_c
#include <assert.h>
#include <limits.h> /* For INT_MAX */
#include <math.h> /* sin(), etc. */
#include <stdarg.h>
#include <stdio.h>
#if !defined(_MSC_VER) && !defined(__DMC__)
#include <strings.h> /* For strcasecmp(). */
#include <wchar.h> /* For wcslen(), wcsrtombs() */
#endif
#ifdef _MSC_VER
#include <float.h> /* For _controlfp_s constants */
#endif
#ifdef MA_WIN32
#include <windows.h>
#else
#include <stdlib.h> /* For malloc(), free(), wcstombs(). */
#include <string.h> /* For memset() */
#include <sched.h>
#include <sys/time.h> /* select() (used for ma_sleep()). */
#include <pthread.h>
#endif
#include <sys/stat.h> /* For fstat(), etc. */
#ifdef MA_EMSCRIPTEN
#include <emscripten/emscripten.h>
#endif
#if !defined(MA_64BIT) && !defined(MA_32BIT)
#ifdef _WIN32
#ifdef _WIN64
#define MA_64BIT
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#ifndef MA_REALLOC
#ifdef MA_WIN32
#define MA_REALLOC(p, sz) (((sz) > 0) ? ((p) ? HeapReAlloc(GetProcessHeap(), 0, (p), (sz)) : HeapAlloc(GetProcessHeap(), 0, (sz))) : ((VOID*)(size_t)(HeapFree(GetProcessHeap(), 0, (p)) & 0)))
#else
#define MA_REALLOC(p, sz) realloc((p), (sz))
#endif
#endif
#ifndef MA_FREE
#ifdef MA_WIN32
#define MA_FREE(p) HeapFree(GetProcessHeap(), 0, (p))
#else
#define MA_FREE(p) free((p))
#endif
#endif
#ifndef MA_ZERO_MEMORY
#ifdef MA_WIN32
#define MA_ZERO_MEMORY(p, sz) ZeroMemory((p), (sz))
#else
#define MA_ZERO_MEMORY(p, sz) memset((p), 0, (sz))
#endif
#endif
#ifndef MA_COPY_MEMORY
#ifdef MA_WIN32
#define MA_COPY_MEMORY(dst, src, sz) CopyMemory((dst), (src), (sz))
#else
#define MA_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz))
#endif
#endif
#ifndef MA_MOVE_MEMORY
#ifdef MA_WIN32
#define MA_MOVE_MEMORY(dst, src, sz) MoveMemory((dst), (src), (sz))
#else
#define MA_MOVE_MEMORY(dst, src, sz) memmove((dst), (src), (sz))
#endif
#endif
#ifndef MA_ASSERT
#ifdef MA_WIN32
#define MA_ASSERT(condition) assert(condition)
#else
#define MA_ASSERT(condition) assert(condition)
#endif
#endif
#define MA_ZERO_OBJECT(p) MA_ZERO_MEMORY((p), sizeof(*(p)))
#define ma_countof(x) (sizeof(x) / sizeof(x[0]))
#define ma_max(x, y) (((x) > (y)) ? (x) : (y))
#define ma_min(x, y) (((x) < (y)) ? (x) : (y))
#define ma_abs(x) (((x) > 0) ? (x) : -(x))
#define ma_clamp(x, lo, hi) (ma_max(lo, ma_min(x, hi)))
#define ma_offset_ptr(p, offset) (((ma_uint8*)(p)) + (offset))
#define ma_align(x, a) ((x + (a-1)) & ~(a-1))
#define ma_align_64(x) ma_align(x, 8)
#define ma_buffer_frame_capacity(buffer, channels, format) (sizeof(buffer) / ma_get_bytes_per_sample(format) / (channels))
static MA_INLINE double ma_sind(double x)
{
/* TODO: Implement custom sin(x). */
return sin(x);
}
static MA_INLINE double ma_expd(double x)
{
/* TODO: Implement custom exp(x). */
return exp(x);
}
static MA_INLINE double ma_logd(double x)
{
/* TODO: Implement custom log(x). */
return log(x);
}
static MA_INLINE double ma_powd(double x, double y)
{
/* TODO: Implement custom pow(x, y). */
return pow(x, y);
}
static MA_INLINE double ma_sqrtd(double x)
{
/* TODO: Implement custom sqrt(x). */
return sqrt(x);
}
static MA_INLINE float ma_sinf(float x)
{
return (float)ma_sind((float)x);
}
static MA_INLINE double ma_cosd(double x)
{
return ma_sind((MA_PI_D*0.5) - x);
}
static MA_INLINE float ma_cosf(float x)
{
return (float)ma_cosd((float)x);
}
static MA_INLINE double ma_log10d(double x)
{
return ma_logd(x) * 0.43429448190325182765;
}
static MA_INLINE float ma_powf(float x, float y)
{
return (float)ma_powd((double)x, (double)y);
}
static MA_INLINE float ma_log10f(float x)
{
return (float)ma_log10d((double)x);
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}
static C89ATOMIC_INLINE float c89atomic_load_explicit_f32(volatile const float* ptr, c89atomic_memory_order order)
{
c89atomic_if32 r;
r.i = c89atomic_load_explicit_32((volatile const c89atomic_uint32*)ptr, order);
return r.f;
}
static C89ATOMIC_INLINE double c89atomic_load_explicit_f64(volatile const double* ptr, c89atomic_memory_order order)
{
c89atomic_if64 r;
r.i = c89atomic_load_explicit_64((volatile const c89atomic_uint64*)ptr, order);
return r.f;
}
static C89ATOMIC_INLINE float c89atomic_exchange_explicit_f32(volatile float* dst, float src, c89atomic_memory_order order)
{
c89atomic_if32 r;
c89atomic_if32 x;
x.f = src;
r.i = c89atomic_exchange_explicit_32((volatile c89atomic_uint32*)dst, x.i, order);
return r.f;
}
static C89ATOMIC_INLINE double c89atomic_exchange_explicit_f64(volatile double* dst, double src, c89atomic_memory_order order)
{
c89atomic_if64 r;
c89atomic_if64 x;
x.f = src;
r.i = c89atomic_exchange_explicit_64((volatile c89atomic_uint64*)dst, x.i, order);
return r.f;
}
#define c89atomic_clear_f32(ptr) (float )c89atomic_clear_explicit_f32(ptr, c89atomic_memory_order_seq_cst)
#define c89atomic_clear_f64(ptr) (double)c89atomic_clear_explicit_f64(ptr, c89atomic_memory_order_seq_cst)
#define c89atomic_store_f32(dst, src) c89atomic_store_explicit_f32(dst, src, c89atomic_memory_order_seq_cst)
#define c89atomic_store_f64(dst, src) c89atomic_store_explicit_f64(dst, src, c89atomic_memory_order_seq_cst)
#define c89atomic_load_f32(ptr) (float )c89atomic_load_explicit_f32(ptr, c89atomic_memory_order_seq_cst)
#define c89atomic_load_f64(ptr) (double)c89atomic_load_explicit_f64(ptr, c89atomic_memory_order_seq_cst)
#define c89atomic_exchange_f32(dst, src) (float )c89atomic_exchange_explicit_f32(dst, src, c89atomic_memory_order_seq_cst)
#define c89atomic_exchange_f64(dst, src) (double)c89atomic_exchange_explicit_f64(dst, src, c89atomic_memory_order_seq_cst)
typedef c89atomic_flag c89atomic_spinlock;
static C89ATOMIC_INLINE void c89atomic_spinlock_lock(volatile c89atomic_spinlock* pSpinlock)
{
for (;;) {
if (c89atomic_flag_test_and_set_explicit(pSpinlock, c89atomic_memory_order_acquire) == 0) {
break;
}
while (c89atoimc_flag_load_explicit(pSpinlock, c89atomic_memory_order_relaxed) == 1) {
}
}
}
static C89ATOMIC_INLINE void c89atomic_spinlock_unlock(volatile c89atomic_spinlock* pSpinlock)
{
c89atomic_flag_clear_explicit(pSpinlock, c89atomic_memory_order_release);
}
#if defined(__cplusplus)
}
#endif
#endif
/* c89atomic.h end */
MA_API ma_uint64 ma_calculate_frame_count_after_resampling(ma_uint32 sampleRateOut, ma_uint32 sampleRateIn, ma_uint64 frameCountIn)
{
/* This is based on the calculation in ma_linear_resampler_get_expected_output_frame_count(). */
ma_uint64 outputFrameCount;
ma_uint64 preliminaryInputFrameCountFromFrac;
ma_uint64 preliminaryInputFrameCount;
if (sampleRateIn == 0 || sampleRateOut == 0 || frameCountIn == 0) {
return 0;
}
if (sampleRateOut == sampleRateIn) {
return frameCountIn;
}
outputFrameCount = (frameCountIn * sampleRateOut) / sampleRateIn;
preliminaryInputFrameCountFromFrac = (outputFrameCount * (sampleRateIn / sampleRateOut)) / sampleRateOut;
preliminaryInputFrameCount = (outputFrameCount * (sampleRateIn % sampleRateOut)) + preliminaryInputFrameCountFromFrac;
if (preliminaryInputFrameCount <= frameCountIn) {
outputFrameCount += 1;
}
return outputFrameCount;
}
#ifndef MA_DATA_CONVERTER_STACK_BUFFER_SIZE
#define MA_DATA_CONVERTER_STACK_BUFFER_SIZE 4096
#endif
#if defined(MA_WIN32)
static ma_result ma_result_from_GetLastError(DWORD error)
{
switch (error)
{
case ERROR_SUCCESS: return MA_SUCCESS;
case ERROR_PATH_NOT_FOUND: return MA_DOES_NOT_EXIST;
case ERROR_TOO_MANY_OPEN_FILES: return MA_TOO_MANY_OPEN_FILES;
case ERROR_NOT_ENOUGH_MEMORY: return MA_OUT_OF_MEMORY;
case ERROR_DISK_FULL: return MA_NO_SPACE;
case ERROR_HANDLE_EOF: return MA_AT_END;
case ERROR_NEGATIVE_SEEK: return MA_BAD_SEEK;
case ERROR_INVALID_PARAMETER: return MA_INVALID_ARGS;
case ERROR_ACCESS_DENIED: return MA_ACCESS_DENIED;
case ERROR_SEM_TIMEOUT: return MA_TIMEOUT;
case ERROR_FILE_NOT_FOUND: return MA_DOES_NOT_EXIST;
default: break;
}
return MA_ERROR;
}
#endif /* MA_WIN32 */
/*******************************************************************************
Threading
*******************************************************************************/
static MA_INLINE ma_result ma_spinlock_lock_ex(volatile ma_spinlock* pSpinlock, ma_bool32 yield)
{
if (pSpinlock == NULL) {
return MA_INVALID_ARGS;
}
for (;;) {
if (c89atomic_exchange_explicit_32(pSpinlock, 1, c89atomic_memory_order_acquire) == 0) {
break;
}
while (c89atomic_load_explicit_32(pSpinlock, c89atomic_memory_order_relaxed) == 1) {
if (yield) {
ma_yield();
}
}
}
return MA_SUCCESS;
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if (!pDevice->noDisableDenormals) {
ma_restore_denormals(prevState);
} else {
/* Do nothing. */
(void)prevState;
}
}
static ma_device_notification ma_device_notification_init(ma_device* pDevice, ma_device_notification_type type)
{
ma_device_notification notification;
MA_ZERO_OBJECT(¬ification);
notification.pDevice = pDevice;
notification.type = type;
return notification;
}
static void ma_device__on_notification(ma_device_notification notification)
{
MA_ASSERT(notification.pDevice != NULL);
if (notification.pDevice->onNotification != NULL) {
notification.pDevice->onNotification(¬ification);
}
/* TEMP FOR COMPATIBILITY: If it's a stopped notification, fire the onStop callback as well. This is only for backwards compatibility and will be removed. */
if (notification.pDevice->onStop != NULL && notification.type == ma_device_notification_type_stopped) {
notification.pDevice->onStop(notification.pDevice);
}
}
void ma_device__on_notification_started(ma_device* pDevice)
{
ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_started));
}
void ma_device__on_notification_stopped(ma_device* pDevice)
{
ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_stopped));
}
void ma_device__on_notification_rerouted(ma_device* pDevice)
{
ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_rerouted));
}
void ma_device__on_notification_interruption_began(ma_device* pDevice)
{
ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_interruption_began));
}
void ma_device__on_notification_interruption_ended(ma_device* pDevice)
{
ma_device__on_notification(ma_device_notification_init(pDevice, ma_device_notification_type_interruption_ended));
}
static void ma_device__on_data_inner(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount)
{
MA_ASSERT(pDevice != NULL);
MA_ASSERT(pDevice->onData != NULL);
if (!pDevice->noPreSilencedOutputBuffer && pFramesOut != NULL) {
ma_silence_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels);
}
pDevice->onData(pDevice, pFramesOut, pFramesIn, frameCount);
}
static void ma_device__on_data(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount)
{
MA_ASSERT(pDevice != NULL);
if (pDevice->noFixedSizedCallback) {
/* Fast path. Not using a fixed sized callback. Process directly from the specified buffers. */
ma_device__on_data_inner(pDevice, pFramesOut, pFramesIn, frameCount);
} else {
/* Slow path. Using a fixed sized callback. Need to use the intermediary buffer. */
ma_uint32 totalFramesProcessed = 0;
while (totalFramesProcessed < frameCount) {
ma_uint32 totalFramesRemaining = frameCount - totalFramesProcessed;
ma_uint32 framesToProcessThisIteration = 0;
if (pFramesIn != NULL) {
/* Capturing. Write to the intermediary buffer. If there's no room, fire the callback to empty it. */
if (pDevice->capture.intermediaryBufferLen < pDevice->capture.intermediaryBufferCap) {
/* There's some room left in the intermediary buffer. Write to it without firing the callback. */
framesToProcessThisIteration = totalFramesRemaining;
if (framesToProcessThisIteration > pDevice->capture.intermediaryBufferCap - pDevice->capture.intermediaryBufferLen) {
framesToProcessThisIteration = pDevice->capture.intermediaryBufferCap - pDevice->capture.intermediaryBufferLen;
}
ma_copy_pcm_frames(
ma_offset_pcm_frames_ptr(pDevice->capture.pIntermediaryBuffer, pDevice->capture.intermediaryBufferLen, pDevice->capture.format, pDevice->capture.channels),
ma_offset_pcm_frames_const_ptr(pFramesIn, totalFramesProcessed, pDevice->capture.format, pDevice->capture.channels),
framesToProcessThisIteration,
pDevice->capture.format, pDevice->capture.channels);
pDevice->capture.intermediaryBufferLen += framesToProcessThisIteration;
}
if (pDevice->capture.intermediaryBufferLen == pDevice->capture.intermediaryBufferCap) {
/* No room left in the intermediary buffer. Fire the data callback. */
if (pDevice->type == ma_device_type_duplex) {
/* We'll do the duplex data callback later after we've processed the playback data. */
} else {
ma_device__on_data_inner(pDevice, NULL, pDevice->capture.pIntermediaryBuffer, pDevice->capture.intermediaryBufferCap);
/* The intermediary buffer has just been drained. */
pDevice->capture.intermediaryBufferLen = 0;
}
}
}
if (pFramesOut != NULL) {
/* Playing back. Read from the intermediary buffer. If there's nothing in it, fire the callback to fill it. */
if (pDevice->playback.intermediaryBufferLen > 0) {
/* There's some content in the intermediary buffer. Read from that without firing the callback. */
if (pDevice->type == ma_device_type_duplex) {
/* The frames processed this iteration for a duplex device will always be based on the capture side. Leave it unmodified. */
} else {
framesToProcessThisIteration = totalFramesRemaining;
if (framesToProcessThisIteration > pDevice->playback.intermediaryBufferLen) {
framesToProcessThisIteration = pDevice->playback.intermediaryBufferLen;
}
}
ma_copy_pcm_frames(
ma_offset_pcm_frames_ptr(pFramesOut, totalFramesProcessed, pDevice->playback.format, pDevice->playback.channels),
ma_offset_pcm_frames_ptr(pDevice->playback.pIntermediaryBuffer, pDevice->playback.intermediaryBufferCap - pDevice->playback.intermediaryBufferLen, pDevice->playback.format, pDevice->playback.channels),
framesToProcessThisIteration,
pDevice->playback.format, pDevice->playback.channels);
pDevice->playback.intermediaryBufferLen -= framesToProcessThisIteration;
}
if (pDevice->playback.intermediaryBufferLen == 0) {
/* There's nothing in the intermediary buffer. Fire the data callback to fill it. */
if (pDevice->type == ma_device_type_duplex) {
/* In duplex mode, the data callback will be fired later. Nothing to do here. */
} else {
ma_device__on_data_inner(pDevice, pDevice->playback.pIntermediaryBuffer, NULL, pDevice->playback.intermediaryBufferCap);
/* The intermediary buffer has just been filled. */
pDevice->playback.intermediaryBufferLen = pDevice->playback.intermediaryBufferCap;
}
}
}
/* If we're in duplex mode we might need to do a refill of the data. */
if (pDevice->type == ma_device_type_duplex) {
if (pDevice->capture.intermediaryBufferLen == pDevice->capture.intermediaryBufferCap) {
ma_device__on_data_inner(pDevice, pDevice->playback.pIntermediaryBuffer, pDevice->capture.pIntermediaryBuffer, pDevice->capture.intermediaryBufferCap);
pDevice->playback.intermediaryBufferLen = pDevice->playback.intermediaryBufferCap; /* The playback buffer will have just been filled. */
pDevice->capture.intermediaryBufferLen = 0; /* The intermediary buffer has just been drained. */
}
}
/* Make sure this is only incremented once in the duplex case. */
totalFramesProcessed += framesToProcessThisIteration;
}
}
}
static void ma_device__handle_data_callback(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount)
{
float masterVolumeFactor;
ma_device_get_master_volume(pDevice, &masterVolumeFactor); /* Use ma_device_get_master_volume() to ensure the volume is loaded atomically. */
if (pDevice->onData) {
unsigned int prevDenormalState = ma_device_disable_denormals(pDevice);
{
/* Volume control of input makes things a bit awkward because the input buffer is read-only. We'll need to use a temp buffer and loop in this case. */
if (pFramesIn != NULL && masterVolumeFactor < 1) {
ma_uint8 tempFramesIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
ma_uint32 bpfCapture = ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
ma_uint32 bpfPlayback = ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
ma_uint32 totalFramesProcessed = 0;
while (totalFramesProcessed < frameCount) {
ma_uint32 framesToProcessThisIteration = frameCount - totalFramesProcessed;
if (framesToProcessThisIteration > sizeof(tempFramesIn)/bpfCapture) {
framesToProcessThisIteration = sizeof(tempFramesIn)/bpfCapture;
}
ma_copy_and_apply_volume_factor_pcm_frames(tempFramesIn, ma_offset_ptr(pFramesIn, totalFramesProcessed*bpfCapture), framesToProcessThisIteration, pDevice->capture.format, pDevice->capture.channels, masterVolumeFactor);
ma_device__on_data(pDevice, ma_offset_ptr(pFramesOut, totalFramesProcessed*bpfPlayback), tempFramesIn, framesToProcessThisIteration);
totalFramesProcessed += framesToProcessThisIteration;
}
} else {
ma_device__on_data(pDevice, pFramesOut, pFramesIn, frameCount);
}
/* Volume control and clipping for playback devices. */
if (pFramesOut != NULL) {
if (masterVolumeFactor < 1) {
if (pFramesIn == NULL) { /* <-- In full-duplex situations, the volume will have been applied to the input samples before the data callback. Applying it again post-callback will incorrectly compound it. */
ma_apply_volume_factor_pcm_frames(pFramesOut, frameCount, pDevice->playback.format, pDevice->playback.channels, masterVolumeFactor);
}
}
if (!pDevice->noClip && pDevice->playback.format == ma_format_f32) {
ma_clip_samples_f32((float*)pFramesOut, (const float*)pFramesOut, frameCount * pDevice->playback.channels); /* Intentionally specifying the same pointer for both input and output for in-place processing. */
}
}
}
ma_device_restore_denormals(pDevice, prevDenormalState);
}
}
/* A helper function for reading sample data from the client. */
static void ma_device__read_frames_from_client(ma_device* pDevice, ma_uint32 frameCount, void* pFramesOut)
{
MA_ASSERT(pDevice != NULL);
MA_ASSERT(frameCount > 0);
MA_ASSERT(pFramesOut != NULL);
if (pDevice->playback.converter.isPassthrough) {
ma_device__handle_data_callback(pDevice, pFramesOut, NULL, frameCount);
} else {
ma_result result;
ma_uint64 totalFramesReadOut;
void* pRunningFramesOut;
totalFramesReadOut = 0;
pRunningFramesOut = pFramesOut;
/*
We run slightly different logic depending on whether or not we're using a heap-allocated
buffer for caching input data. This will be the case if the data converter does not have
the ability to retrieve the required input frame count for a given output frame count.
*/
if (pDevice->playback.pInputCache != NULL) {
while (totalFramesReadOut < frameCount) {
ma_uint64 framesToReadThisIterationIn;
ma_uint64 framesToReadThisIterationOut;
/* If there's any data available in the cache, that needs to get processed first. */
if (pDevice->playback.inputCacheRemaining > 0) {
framesToReadThisIterationOut = (frameCount - totalFramesReadOut);
framesToReadThisIterationIn = framesToReadThisIterationOut;
if (framesToReadThisIterationIn > pDevice->playback.inputCacheRemaining) {
framesToReadThisIterationIn = pDevice->playback.inputCacheRemaining;
}
result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, ma_offset_pcm_frames_ptr(pDevice->playback.pInputCache, pDevice->playback.inputCacheConsumed, pDevice->playback.format, pDevice->playback.channels), &fram...
if (result != MA_SUCCESS) {
break;
}
pDevice->playback.inputCacheConsumed += framesToReadThisIterationIn;
pDevice->playback.inputCacheRemaining -= framesToReadThisIterationIn;
totalFramesReadOut += framesToReadThisIterationOut;
pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesToReadThisIterationOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
if (framesToReadThisIterationIn == 0 && framesToReadThisIterationOut == 0) {
break; /* We're done. */
}
}
/* Getting here means there's no data in the cache and we need to fill it up with data from the client. */
if (pDevice->playback.inputCacheRemaining == 0) {
ma_device__handle_data_callback(pDevice, pDevice->playback.pInputCache, NULL, (ma_uint32)pDevice->playback.inputCacheCap);
pDevice->playback.inputCacheConsumed = 0;
pDevice->playback.inputCacheRemaining = pDevice->playback.inputCacheCap;
}
}
} else {
while (totalFramesReadOut < frameCount) {
ma_uint8 pIntermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In client format. */
ma_uint64 intermediaryBufferCap = sizeof(pIntermediaryBuffer) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
ma_uint64 framesToReadThisIterationIn;
ma_uint64 framesReadThisIterationIn;
ma_uint64 framesToReadThisIterationOut;
ma_uint64 framesReadThisIterationOut;
ma_uint64 requiredInputFrameCount;
framesToReadThisIterationOut = (frameCount - totalFramesReadOut);
framesToReadThisIterationIn = framesToReadThisIterationOut;
if (framesToReadThisIterationIn > intermediaryBufferCap) {
framesToReadThisIterationIn = intermediaryBufferCap;
}
ma_data_converter_get_required_input_frame_count(&pDevice->playback.converter, framesToReadThisIterationOut, &requiredInputFrameCount);
if (framesToReadThisIterationIn > requiredInputFrameCount) {
framesToReadThisIterationIn = requiredInputFrameCount;
}
if (framesToReadThisIterationIn > 0) {
ma_device__handle_data_callback(pDevice, pIntermediaryBuffer, NULL, (ma_uint32)framesToReadThisIterationIn);
}
/*
At this point we have our decoded data in input format and now we need to convert to output format. Note that even if we didn't read any
input frames, we still want to try processing frames because there may some output frames generated from cached input data.
*/
framesReadThisIterationIn = framesToReadThisIterationIn;
framesReadThisIterationOut = framesToReadThisIterationOut;
result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, pIntermediaryBuffer, &framesReadThisIterationIn, pRunningFramesOut, &framesReadThisIterationOut);
if (result != MA_SUCCESS) {
break;
}
totalFramesReadOut += framesReadThisIterationOut;
pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesReadThisIterationOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
if (framesReadThisIterationIn == 0 && framesReadThisIterationOut == 0) {
break; /* We're done. */
}
}
}
}
}
/* A helper for sending sample data to the client. */
static void ma_device__send_frames_to_client(ma_device* pDevice, ma_uint32 frameCountInDeviceFormat, const void* pFramesInDeviceFormat)
{
MA_ASSERT(pDevice != NULL);
MA_ASSERT(frameCountInDeviceFormat > 0);
MA_ASSERT(pFramesInDeviceFormat != NULL);
if (pDevice->capture.converter.isPassthrough) {
ma_device__handle_data_callback(pDevice, NULL, pFramesInDeviceFormat, frameCountInDeviceFormat);
} else {
ma_result result;
ma_uint8 pFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
ma_uint64 framesInClientFormatCap = sizeof(pFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
ma_uint64 totalDeviceFramesProcessed = 0;
ma_uint64 totalClientFramesProcessed = 0;
const void* pRunningFramesInDeviceFormat = pFramesInDeviceFormat;
/* We just keep going until we've exhaused all of our input frames and cannot generate any more output frames. */
for (;;) {
ma_uint64 deviceFramesProcessedThisIteration;
ma_uint64 clientFramesProcessedThisIteration;
deviceFramesProcessedThisIteration = (frameCountInDeviceFormat - totalDeviceFramesProcessed);
clientFramesProcessedThisIteration = framesInClientFormatCap;
result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningFramesInDeviceFormat, &deviceFramesProcessedThisIteration, pFramesInClientFormat, &clientFramesProcessedThisIteration);
if (result != MA_SUCCESS) {
break;
}
if (clientFramesProcessedThisIteration > 0) {
ma_device__handle_data_callback(pDevice, NULL, pFramesInClientFormat, (ma_uint32)clientFramesProcessedThisIteration); /* Safe cast. */
}
pRunningFramesInDeviceFormat = ma_offset_ptr(pRunningFramesInDeviceFormat, deviceFramesProcessedThisIteration * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
totalDeviceFramesProcessed += deviceFramesProcessedThisIteration;
totalClientFramesProcessed += clientFramesProcessedThisIteration;
if (deviceFramesProcessedThisIteration == 0 && clientFramesProcessedThisIteration == 0) {
break; /* We're done. */
}
}
}
}
static ma_result ma_device__handle_duplex_callback_capture(ma_device* pDevice, ma_uint32 frameCountInDeviceFormat, const void* pFramesInDeviceFormat, ma_pcm_rb* pRB)
{
ma_result result;
ma_uint32 totalDeviceFramesProcessed = 0;
const void* pRunningFramesInDeviceFormat = pFramesInDeviceFormat;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(frameCountInDeviceFormat > 0);
MA_ASSERT(pFramesInDeviceFormat != NULL);
MA_ASSERT(pRB != NULL);
/* Write to the ring buffer. The ring buffer is in the client format which means we need to convert. */
for (;;) {
ma_uint32 framesToProcessInDeviceFormat = (frameCountInDeviceFormat - totalDeviceFramesProcessed);
ma_uint32 framesToProcessInClientFormat = MA_DATA_CONVERTER_STACK_BUFFER_SIZE / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
ma_uint64 framesProcessedInDeviceFormat;
ma_uint64 framesProcessedInClientFormat;
void* pFramesInClientFormat;
result = ma_pcm_rb_acquire_write(pRB, &framesToProcessInClientFormat, &pFramesInClientFormat);
if (result != MA_SUCCESS) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "Failed to acquire capture PCM frames from ring buffer.");
break;
}
if (framesToProcessInClientFormat == 0) {
if (ma_pcm_rb_pointer_distance(pRB) == (ma_int32)ma_pcm_rb_get_subbuffer_size(pRB)) {
break; /* Overrun. Not enough room in the ring buffer for input frame. Excess frames are dropped. */
}
}
/* Convert. */
framesProcessedInDeviceFormat = framesToProcessInDeviceFormat;
framesProcessedInClientFormat = framesToProcessInClientFormat;
result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningFramesInDeviceFormat, &framesProcessedInDeviceFormat, pFramesInClientFormat, &framesProcessedInClientFormat);
if (result != MA_SUCCESS) {
break;
}
result = ma_pcm_rb_commit_write(pRB, (ma_uint32)framesProcessedInClientFormat); /* Safe cast. */
if (result != MA_SUCCESS) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "Failed to commit capture PCM frames to ring buffer.");
break;
}
pRunningFramesInDeviceFormat = ma_offset_ptr(pRunningFramesInDeviceFormat, framesProcessedInDeviceFormat * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
totalDeviceFramesProcessed += (ma_uint32)framesProcessedInDeviceFormat; /* Safe cast. */
/* We're done when we're unable to process any client nor device frames. */
if (framesProcessedInClientFormat == 0 && framesProcessedInDeviceFormat == 0) {
break; /* Done. */
}
}
return MA_SUCCESS;
}
static ma_result ma_device__handle_duplex_callback_playback(ma_device* pDevice, ma_uint32 frameCount, void* pFramesInInternalFormat, ma_pcm_rb* pRB)
{
ma_result result;
ma_uint8 silentInputFrames[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
ma_uint32 totalFramesReadOut = 0;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(frameCount > 0);
MA_ASSERT(pFramesInInternalFormat != NULL);
MA_ASSERT(pRB != NULL);
MA_ASSERT(pDevice->playback.pInputCache != NULL);
/*
Sitting in the ring buffer should be captured data from the capture callback in external format. If there's not enough data in there for
the whole frameCount frames we just use silence instead for the input data.
*/
MA_ZERO_MEMORY(silentInputFrames, sizeof(silentInputFrames));
while (totalFramesReadOut < frameCount && ma_device_is_started(pDevice)) {
/*
We should have a buffer allocated on the heap. Any playback frames still sitting in there
need to be sent to the internal device before we process any more data from the client.
*/
if (pDevice->playback.inputCacheRemaining > 0) {
ma_uint64 framesConvertedIn = pDevice->playback.inputCacheRemaining;
ma_uint64 framesConvertedOut = (frameCount - totalFramesReadOut);
ma_data_converter_process_pcm_frames(&pDevice->playback.converter, ma_offset_pcm_frames_ptr(pDevice->playback.pInputCache, pDevice->playback.inputCacheConsumed, pDevice->playback.format, pDevice->playback.channels), &framesConvertedIn, pF...
pDevice->playback.inputCacheConsumed += framesConvertedIn;
pDevice->playback.inputCacheRemaining -= framesConvertedIn;
totalFramesReadOut += (ma_uint32)framesConvertedOut; /* Safe cast. */
pFramesInInternalFormat = ma_offset_ptr(pFramesInInternalFormat, framesConvertedOut * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
}
/* If there's no more data in the cache we'll need to fill it with some. */
if (totalFramesReadOut < frameCount && pDevice->playback.inputCacheRemaining == 0) {
ma_uint32 inputFrameCount;
void* pInputFrames;
inputFrameCount = (ma_uint32)pDevice->playback.inputCacheCap;
result = ma_pcm_rb_acquire_read(pRB, &inputFrameCount, &pInputFrames);
if (result == MA_SUCCESS) {
if (inputFrameCount > 0) {
ma_device__handle_data_callback(pDevice, pDevice->playback.pInputCache, pInputFrames, inputFrameCount);
} else {
if (ma_pcm_rb_pointer_distance(pRB) == 0) {
break; /* Underrun. */
}
}
} else {
/* No capture data available. Feed in silence. */
inputFrameCount = (ma_uint32)ma_min(pDevice->playback.inputCacheCap, sizeof(silentInputFrames) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels));
ma_device__handle_data_callback(pDevice, pDevice->playback.pInputCache, silentInputFrames, inputFrameCount);
}
pDevice->playback.inputCacheConsumed = 0;
pDevice->playback.inputCacheRemaining = inputFrameCount;
result = ma_pcm_rb_commit_read(pRB, inputFrameCount);
if (result != MA_SUCCESS) {
return result; /* Should never happen. */
}
}
}
return MA_SUCCESS;
}
/* A helper for changing the state of the device. */
static MA_INLINE void ma_device__set_state(ma_device* pDevice, ma_device_state newState)
{
c89atomic_exchange_i32((ma_int32*)&pDevice->state, (ma_int32)newState);
}
#ifdef MA_WIN32
GUID MA_GUID_KSDATAFORMAT_SUBTYPE_PCM = {0x00000001, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};
GUID MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT = {0x00000003, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};
/*GUID MA_GUID_KSDATAFORMAT_SUBTYPE_ALAW = {0x00000006, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};*/
/*GUID MA_GUID_KSDATAFORMAT_SUBTYPE_MULAW = {0x00000007, 0x0000, 0x0010, {0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71}};*/
#endif
MA_API ma_uint32 ma_get_format_priority_index(ma_format format) /* Lower = better. */
{
ma_uint32 i;
for (i = 0; i < ma_countof(g_maFormatPriorities); ++i) {
if (g_maFormatPriorities[i] == format) {
return i;
}
}
/* Getting here means the format could not be found or is equal to ma_format_unknown. */
return (ma_uint32)-1;
}
static ma_result ma_device__post_init_setup(ma_device* pDevice, ma_device_type deviceType);
static ma_bool32 ma_device_descriptor_is_valid(const ma_device_descriptor* pDeviceDescriptor)
{
if (pDeviceDescriptor == NULL) {
return MA_FALSE;
}
if (pDeviceDescriptor->format == ma_format_unknown) {
return MA_FALSE;
}
if (pDeviceDescriptor->channels == 0 || pDeviceDescriptor->channels > MA_MAX_CHANNELS) {
return MA_FALSE;
}
if (pDeviceDescriptor->sampleRate == 0) {
return MA_FALSE;
}
return MA_TRUE;
}
static ma_result ma_device_audio_thread__default_read_write(ma_device* pDevice)
{
ma_result result = MA_SUCCESS;
ma_bool32 exitLoop = MA_FALSE;
ma_uint8 capturedDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
ma_uint8 playbackDeviceData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
ma_uint32 capturedDeviceDataCapInFrames = 0;
ma_uint32 playbackDeviceDataCapInFrames = 0;
MA_ASSERT(pDevice != NULL);
/* Just some quick validation on the device type and the available callbacks. */
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
if (pDevice->pContext->callbacks.onDeviceRead == NULL) {
return MA_NOT_IMPLEMENTED;
}
capturedDeviceDataCapInFrames = sizeof(capturedDeviceData) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
if (pDevice->pContext->callbacks.onDeviceWrite == NULL) {
return MA_NOT_IMPLEMENTED;
}
playbackDeviceDataCapInFrames = sizeof(playbackDeviceData) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
}
/* NOTE: The device was started outside of this function, in the worker thread. */
while (ma_device_get_state(pDevice) == ma_device_state_started && !exitLoop) {
switch (pDevice->type) {
case ma_device_type_duplex:
{
/* The process is: onDeviceRead() -> convert -> callback -> convert -> onDeviceWrite() */
ma_uint32 totalCapturedDeviceFramesProcessed = 0;
ma_uint32 capturedDevicePeriodSizeInFrames = ma_min(pDevice->capture.internalPeriodSizeInFrames, pDevice->playback.internalPeriodSizeInFrames);
while (totalCapturedDeviceFramesProcessed < capturedDevicePeriodSizeInFrames) {
ma_uint32 capturedDeviceFramesRemaining;
ma_uint32 capturedDeviceFramesProcessed;
ma_uint32 capturedDeviceFramesToProcess;
ma_uint32 capturedDeviceFramesToTryProcessing = capturedDevicePeriodSizeInFrames - totalCapturedDeviceFramesProcessed;
if (capturedDeviceFramesToTryProcessing > capturedDeviceDataCapInFrames) {
capturedDeviceFramesToTryProcessing = capturedDeviceDataCapInFrames;
}
result = pDevice->pContext->callbacks.onDeviceRead(pDevice, capturedDeviceData, capturedDeviceFramesToTryProcessing, &capturedDeviceFramesToProcess);
if (result != MA_SUCCESS) {
exitLoop = MA_TRUE;
break;
}
capturedDeviceFramesRemaining = capturedDeviceFramesToProcess;
capturedDeviceFramesProcessed = 0;
/* At this point we have our captured data in device format and we now need to convert it to client format. */
for (;;) {
ma_uint8 capturedClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
ma_uint8 playbackClientData[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
ma_uint32 capturedClientDataCapInFrames = sizeof(capturedClientData) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
ma_uint32 playbackClientDataCapInFrames = sizeof(playbackClientData) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
ma_uint64 capturedClientFramesToProcessThisIteration = ma_min(capturedClientDataCapInFrames, playbackClientDataCapInFrames);
ma_uint64 capturedDeviceFramesToProcessThisIteration = capturedDeviceFramesRemaining;
ma_uint8* pRunningCapturedDeviceFrames = ma_offset_ptr(capturedDeviceData, capturedDeviceFramesProcessed * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
/* Convert capture data from device format to client format. */
result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningCapturedDeviceFrames, &capturedDeviceFramesToProcessThisIteration, capturedClientData, &capturedClientFramesToProcessThisIteration);
if (result != MA_SUCCESS) {
break;
}
/*
If we weren't able to generate any output frames it must mean we've exhaused all of our input. The only time this would not be the case is if capturedClientData was too small
which should never be the case when it's of the size MA_DATA_CONVERTER_STACK_BUFFER_SIZE.
*/
if (capturedClientFramesToProcessThisIteration == 0) {
break;
}
ma_device__handle_data_callback(pDevice, playbackClientData, capturedClientData, (ma_uint32)capturedClientFramesToProcessThisIteration); /* Safe cast .*/
capturedDeviceFramesProcessed += (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */
capturedDeviceFramesRemaining -= (ma_uint32)capturedDeviceFramesToProcessThisIteration; /* Safe cast. */
/* At this point the playbackClientData buffer should be holding data that needs to be written to the device. */
for (;;) {
ma_uint64 convertedClientFrameCount = capturedClientFramesToProcessThisIteration;
ma_uint64 convertedDeviceFrameCount = playbackDeviceDataCapInFrames;
result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, playbackClientData, &convertedClientFrameCount, playbackDeviceData, &convertedDeviceFrameCount);
if (result != MA_SUCCESS) {
break;
}
result = pDevice->pContext->callbacks.onDeviceWrite(pDevice, playbackDeviceData, (ma_uint32)convertedDeviceFrameCount, NULL); /* Safe cast. */
if (result != MA_SUCCESS) {
exitLoop = MA_TRUE;
break;
}
capturedClientFramesToProcessThisIteration -= (ma_uint32)convertedClientFrameCount; /* Safe cast. */
if (capturedClientFramesToProcessThisIteration == 0) {
break;
}
}
/* In case an error happened from ma_device_write__null()... */
if (result != MA_SUCCESS) {
exitLoop = MA_TRUE;
break;
}
}
/* Make sure we don't get stuck in the inner loop. */
if (capturedDeviceFramesProcessed == 0) {
break;
}
totalCapturedDeviceFramesProcessed += capturedDeviceFramesProcessed;
}
} break;
case ma_device_type_capture:
case ma_device_type_loopback:
{
ma_uint32 periodSizeInFrames = pDevice->capture.internalPeriodSizeInFrames;
ma_uint32 framesReadThisPeriod = 0;
while (framesReadThisPeriod < periodSizeInFrames) {
ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesReadThisPeriod;
ma_uint32 framesProcessed;
ma_uint32 framesToReadThisIteration = framesRemainingInPeriod;
if (framesToReadThisIteration > capturedDeviceDataCapInFrames) {
framesToReadThisIteration = capturedDeviceDataCapInFrames;
}
result = pDevice->pContext->callbacks.onDeviceRead(pDevice, capturedDeviceData, framesToReadThisIteration, &framesProcessed);
if (result != MA_SUCCESS) {
exitLoop = MA_TRUE;
break;
}
/* Make sure we don't get stuck in the inner loop. */
if (framesProcessed == 0) {
break;
}
ma_device__send_frames_to_client(pDevice, framesProcessed, capturedDeviceData);
framesReadThisPeriod += framesProcessed;
}
} break;
case ma_device_type_playback:
{
/* We write in chunks of the period size, but use a stack allocated buffer for the intermediary. */
ma_uint32 periodSizeInFrames = pDevice->playback.internalPeriodSizeInFrames;
ma_uint32 framesWrittenThisPeriod = 0;
while (framesWrittenThisPeriod < periodSizeInFrames) {
ma_uint32 framesRemainingInPeriod = periodSizeInFrames - framesWrittenThisPeriod;
ma_uint32 framesProcessed;
ma_uint32 framesToWriteThisIteration = framesRemainingInPeriod;
if (framesToWriteThisIteration > playbackDeviceDataCapInFrames) {
framesToWriteThisIteration = playbackDeviceDataCapInFrames;
}
ma_device__read_frames_from_client(pDevice, framesToWriteThisIteration, playbackDeviceData);
result = pDevice->pContext->callbacks.onDeviceWrite(pDevice, playbackDeviceData, framesToWriteThisIteration, &framesProcessed);
if (result != MA_SUCCESS) {
exitLoop = MA_TRUE;
break;
}
/* Make sure we don't get stuck in the inner loop. */
if (framesProcessed == 0) {
break;
}
framesWrittenThisPeriod += framesProcessed;
}
} break;
/* Should never get here. */
default: break;
}
}
return result;
}
/*******************************************************************************
Null Backend
*******************************************************************************/
#ifdef MA_HAS_NULL
#define MA_DEVICE_OP_NONE__NULL 0
#define MA_DEVICE_OP_START__NULL 1
#define MA_DEVICE_OP_SUSPEND__NULL 2
#define MA_DEVICE_OP_KILL__NULL 3
static ma_thread_result MA_THREADCALL ma_device_thread__null(void* pData)
{
ma_device* pDevice = (ma_device*)pData;
MA_ASSERT(pDevice != NULL);
for (;;) { /* Keep the thread alive until the device is uninitialized. */
ma_uint32 operation;
/* Wait for an operation to be requested. */
ma_event_wait(&pDevice->null_device.operationEvent);
/* At this point an event should have been triggered. */
operation = pDevice->null_device.operation;
/* Starting the device needs to put the thread into a loop. */
if (operation == MA_DEVICE_OP_START__NULL) {
/* Reset the timer just in case. */
ma_timer_init(&pDevice->null_device.timer);
/* Getting here means a suspend or kill operation has been requested. */
pDevice->null_device.operationResult = MA_SUCCESS;
ma_event_signal(&pDevice->null_device.operationCompletionEvent);
ma_semaphore_release(&pDevice->null_device.operationSemaphore);
continue;
}
/* Suspending the device means we need to stop the timer and just continue the loop. */
if (operation == MA_DEVICE_OP_SUSPEND__NULL) {
/* We need to add the current run time to the prior run time, then reset the timer. */
pDevice->null_device.priorRunTime += ma_timer_get_time_in_seconds(&pDevice->null_device.timer);
ma_timer_init(&pDevice->null_device.timer);
/* We're done. */
pDevice->null_device.operationResult = MA_SUCCESS;
ma_event_signal(&pDevice->null_device.operationCompletionEvent);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
/* Killing the device means we need to get out of this loop so that this thread can terminate. */
if (operation == MA_DEVICE_OP_KILL__NULL) {
pDevice->null_device.operationResult = MA_SUCCESS;
ma_event_signal(&pDevice->null_device.operationCompletionEvent);
ma_semaphore_release(&pDevice->null_device.operationSemaphore);
break;
}
/* Getting a signal on a "none" operation probably means an error. Return invalid operation. */
if (operation == MA_DEVICE_OP_NONE__NULL) {
MA_ASSERT(MA_FALSE); /* <-- Trigger this in debug mode to ensure developers are aware they're doing something wrong (or there's a bug in a miniaudio). */
pDevice->null_device.operationResult = MA_INVALID_OPERATION;
ma_event_signal(&pDevice->null_device.operationCompletionEvent);
ma_semaphore_release(&pDevice->null_device.operationSemaphore);
continue; /* Continue the loop. Don't terminate. */
}
}
return (ma_thread_result)0;
}
static ma_result ma_device_do_operation__null(ma_device* pDevice, ma_uint32 operation)
{
ma_result result;
/*
TODO: Need to review this and consider just using mutual exclusion. I think the original motivation
for this was to just post the event to a queue and return immediately, but that has since changed
and now this function is synchronous. I think this can be simplified to just use a mutex.
*/
/*
The first thing to do is wait for an operation slot to become available. We only have a single slot for this, but we could extend this later
to support queing of operations.
*/
result = ma_semaphore_wait(&pDevice->null_device.operationSemaphore);
if (result != MA_SUCCESS) {
return result; /* Failed to wait for the event. */
}
/*
When we get here it means the background thread is not referencing the operation code and it can be changed. After changing this we need to
signal an event to the worker thread to let it know that it can start work.
*/
pDevice->null_device.operation = operation;
/* Once the operation code has been set, the worker thread can start work. */
if (ma_event_signal(&pDevice->null_device.operationEvent) != MA_SUCCESS) {
return MA_ERROR;
}
/* We want everything to be synchronous so we're going to wait for the worker thread to complete it's operation. */
if (ma_event_wait(&pDevice->null_device.operationCompletionEvent) != MA_SUCCESS) {
return MA_ERROR;
}
return pDevice->null_device.operationResult;
}
static ma_uint64 ma_device_get_total_run_time_in_frames__null(ma_device* pDevice)
{
ma_uint32 internalSampleRate;
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
internalSampleRate = pDevice->capture.internalSampleRate;
} else {
internalSampleRate = pDevice->playback.internalSampleRate;
}
return (ma_uint64)((pDevice->null_device.priorRunTime + ma_timer_get_time_in_seconds(&pDevice->null_device.timer)) * internalSampleRate);
}
static ma_result ma_context_enumerate_devices__null(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
{
ma_bool32 cbResult = MA_TRUE;
MA_ASSERT(pContext != NULL);
MA_ASSERT(callback != NULL);
/* Playback. */
if (cbResult) {
ma_device_info deviceInfo;
MA_ZERO_OBJECT(&deviceInfo);
ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), "NULL Playback Device", (size_t)-1);
deviceInfo.isDefault = MA_TRUE; /* Only one playback and capture device for the null backend, so might as well mark as default. */
cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
}
/* Capture. */
if (cbResult) {
ma_device_info deviceInfo;
MA_ZERO_OBJECT(&deviceInfo);
ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), "NULL Capture Device", (size_t)-1);
deviceInfo.isDefault = MA_TRUE; /* Only one playback and capture device for the null backend, so might as well mark as default. */
cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
}
(void)cbResult; /* Silence a static analysis warning. */
return MA_SUCCESS;
}
static ma_result ma_context_get_device_info__null(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
{
MA_ASSERT(pContext != NULL);
if (pDeviceID != NULL && pDeviceID->nullbackend != 0) {
return MA_NO_DEVICE; /* Don't know the device. */
}
/* Name / Description */
if (deviceType == ma_device_type_playback) {
ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), "NULL Playback Device", (size_t)-1);
} else {
ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), "NULL Capture Device", (size_t)-1);
}
pDeviceInfo->isDefault = MA_TRUE; /* Only one playback and capture device for the null backend, so might as well mark as default. */
/* Support everything on the null backend. */
pDeviceInfo->nativeDataFormats[0].format = ma_format_unknown;
pDeviceInfo->nativeDataFormats[0].channels = 0;
pDeviceInfo->nativeDataFormats[0].sampleRate = 0;
pDeviceInfo->nativeDataFormats[0].flags = 0;
pDeviceInfo->nativeDataFormatCount = 1;
(void)pContext;
return MA_SUCCESS;
}
static ma_result ma_device_uninit__null(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
/* Keep it clean and wait for the device thread to finish before returning. */
ma_device_do_operation__null(pDevice, MA_DEVICE_OP_KILL__NULL);
/* Wait for the thread to finish before continuing. */
ma_thread_wait(&pDevice->null_device.deviceThread);
/* At this point the loop in the device thread is as good as terminated so we can uninitialize our events. */
ma_semaphore_uninit(&pDevice->null_device.operationSemaphore);
ma_event_uninit(&pDevice->null_device.operationCompletionEvent);
ma_event_uninit(&pDevice->null_device.operationEvent);
return MA_SUCCESS;
}
static ma_result ma_device_init__null(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
ma_result result;
MA_ASSERT(pDevice != NULL);
MA_ZERO_OBJECT(&pDevice->null_device);
if (pConfig->deviceType == ma_device_type_loopback) {
return MA_DEVICE_TYPE_NOT_SUPPORTED;
}
/* The null backend supports everything exactly as we specify it. */
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
pDescriptorCapture->format = (pDescriptorCapture->format != ma_format_unknown) ? pDescriptorCapture->format : MA_DEFAULT_FORMAT;
pDescriptorCapture->channels = (pDescriptorCapture->channels != 0) ? pDescriptorCapture->channels : MA_DEFAULT_CHANNELS;
pDescriptorCapture->sampleRate = (pDescriptorCapture->sampleRate != 0) ? pDescriptorCapture->sampleRate : MA_DEFAULT_SAMPLE_RATE;
if (pDescriptorCapture->channelMap[0] == MA_CHANNEL_NONE) {
ma_channel_map_init_standard(ma_standard_channel_map_default, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorCapture->channels);
}
pDescriptorCapture->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorCapture, pDescriptorCapture->sampleRate, pConfig->performanceProfile);
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
pDescriptorPlayback->format = (pDescriptorPlayback->format != ma_format_unknown) ? pDescriptorPlayback->format : MA_DEFAULT_FORMAT;
pDescriptorPlayback->channels = (pDescriptorPlayback->channels != 0) ? pDescriptorPlayback->channels : MA_DEFAULT_CHANNELS;
pDescriptorPlayback->sampleRate = (pDescriptorPlayback->sampleRate != 0) ? pDescriptorPlayback->sampleRate : MA_DEFAULT_SAMPLE_RATE;
if (pDescriptorPlayback->channelMap[0] == MA_CHANNEL_NONE) {
ma_channel_map_init_standard(ma_standard_channel_map_default, pDescriptorPlayback->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorPlayback->channels);
}
pDescriptorPlayback->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile);
}
/*
In order to get timing right, we need to create a thread that does nothing but keeps track of the timer. This timer is started when the
first period is "written" to it, and then stopped in ma_device_stop__null().
*/
result = ma_event_init(&pDevice->null_device.operationEvent);
if (result != MA_SUCCESS) {
return result;
}
result = ma_event_init(&pDevice->null_device.operationCompletionEvent);
if (result != MA_SUCCESS) {
return result;
}
result = ma_semaphore_init(1, &pDevice->null_device.operationSemaphore); /* <-- It's important that the initial value is set to 1. */
if (result != MA_SUCCESS) {
return result;
}
result = ma_thread_create(&pDevice->null_device.deviceThread, pDevice->pContext->threadPriority, 0, ma_device_thread__null, pDevice, &pDevice->pContext->allocationCallbacks);
if (result != MA_SUCCESS) {
return result;
}
return MA_SUCCESS;
}
static ma_result ma_device_start__null(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
ma_device_do_operation__null(pDevice, MA_DEVICE_OP_START__NULL);
c89atomic_exchange_32(&pDevice->null_device.isStarted, MA_TRUE);
return MA_SUCCESS;
}
static ma_result ma_device_stop__null(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
ma_device_do_operation__null(pDevice, MA_DEVICE_OP_SUSPEND__NULL);
c89atomic_exchange_32(&pDevice->null_device.isStarted, MA_FALSE);
return MA_SUCCESS;
}
static ma_result ma_device_write__null(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
{
ma_result result = MA_SUCCESS;
ma_uint32 totalPCMFramesProcessed;
ma_bool32 wasStartedOnEntry;
if (pFramesWritten != NULL) {
*pFramesWritten = 0;
}
wasStartedOnEntry = c89atomic_load_32(&pDevice->null_device.isStarted);
/* Keep going until everything has been read. */
totalPCMFramesProcessed = 0;
while (totalPCMFramesProcessed < frameCount) {
ma_uint64 targetFrame;
/* If there are any frames remaining in the current period, consume those first. */
if (pDevice->null_device.currentPeriodFramesRemainingPlayback > 0) {
ma_uint32 framesRemaining = (frameCount - totalPCMFramesProcessed);
ma_uint32 framesToProcess = pDevice->null_device.currentPeriodFramesRemainingPlayback;
if (framesToProcess > framesRemaining) {
framesToProcess = framesRemaining;
}
/* We don't actually do anything with pPCMFrames, so just mark it as unused to prevent a warning. */
(void)pPCMFrames;
pDevice->null_device.currentPeriodFramesRemainingPlayback -= framesToProcess;
totalPCMFramesProcessed += framesToProcess;
}
/* If we've consumed the current period we'll need to mark it as such an ensure the device is started if it's not already. */
if (pDevice->null_device.currentPeriodFramesRemainingPlayback == 0) {
pDevice->null_device.currentPeriodFramesRemainingPlayback = 0;
if (!c89atomic_load_32(&pDevice->null_device.isStarted) && !wasStartedOnEntry) {
result = ma_device_start__null(pDevice);
if (result != MA_SUCCESS) {
break;
}
}
}
/* If we've consumed the whole buffer we can return now. */
MA_ASSERT(totalPCMFramesProcessed <= frameCount);
if (totalPCMFramesProcessed == frameCount) {
break;
}
/* Getting here means we've still got more frames to consume, we but need to wait for it to become available. */
targetFrame = pDevice->null_device.lastProcessedFramePlayback;
for (;;) {
ma_uint64 currentFrame;
/* Stop waiting if the device has been stopped. */
if (!c89atomic_load_32(&pDevice->null_device.isStarted)) {
break;
}
currentFrame = ma_device_get_total_run_time_in_frames__null(pDevice);
if (currentFrame >= targetFrame) {
break;
}
/* Getting here means we haven't yet reached the target sample, so continue waiting. */
ma_sleep(10);
}
pDevice->null_device.lastProcessedFramePlayback += pDevice->playback.internalPeriodSizeInFrames;
pDevice->null_device.currentPeriodFramesRemainingPlayback = pDevice->playback.internalPeriodSizeInFrames;
}
if (pFramesWritten != NULL) {
*pFramesWritten = totalPCMFramesProcessed;
}
return result;
}
static ma_result ma_device_read__null(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
{
ma_result result = MA_SUCCESS;
ma_uint32 totalPCMFramesProcessed;
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
/* Keep going until everything has been read. */
totalPCMFramesProcessed = 0;
while (totalPCMFramesProcessed < frameCount) {
ma_uint64 targetFrame;
/* If there are any frames remaining in the current period, consume those first. */
if (pDevice->null_device.currentPeriodFramesRemainingCapture > 0) {
ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
ma_uint32 framesRemaining = (frameCount - totalPCMFramesProcessed);
ma_uint32 framesToProcess = pDevice->null_device.currentPeriodFramesRemainingCapture;
if (framesToProcess > framesRemaining) {
framesToProcess = framesRemaining;
}
/* We need to ensure the output buffer is zeroed. */
MA_ZERO_MEMORY(ma_offset_ptr(pPCMFrames, totalPCMFramesProcessed*bpf), framesToProcess*bpf);
pDevice->null_device.currentPeriodFramesRemainingCapture -= framesToProcess;
totalPCMFramesProcessed += framesToProcess;
}
/* If we've consumed the current period we'll need to mark it as such an ensure the device is started if it's not already. */
if (pDevice->null_device.currentPeriodFramesRemainingCapture == 0) {
pDevice->null_device.currentPeriodFramesRemainingCapture = 0;
}
/* If we've consumed the whole buffer we can return now. */
MA_ASSERT(totalPCMFramesProcessed <= frameCount);
if (totalPCMFramesProcessed == frameCount) {
break;
}
/* Getting here means we've still got more frames to consume, we but need to wait for it to become available. */
targetFrame = pDevice->null_device.lastProcessedFrameCapture + pDevice->capture.internalPeriodSizeInFrames;
for (;;) {
ma_uint64 currentFrame;
/* Stop waiting if the device has been stopped. */
if (!c89atomic_load_32(&pDevice->null_device.isStarted)) {
break;
}
currentFrame = ma_device_get_total_run_time_in_frames__null(pDevice);
if (currentFrame >= targetFrame) {
break;
}
/* Getting here means we haven't yet reached the target sample, so continue waiting. */
ma_sleep(10);
}
pDevice->null_device.lastProcessedFrameCapture += pDevice->capture.internalPeriodSizeInFrames;
pDevice->null_device.currentPeriodFramesRemainingCapture = pDevice->capture.internalPeriodSizeInFrames;
}
if (pFramesRead != NULL) {
*pFramesRead = totalPCMFramesProcessed;
}
return result;
}
static ma_result ma_context_uninit__null(ma_context* pContext)
{
MA_ASSERT(pContext != NULL);
MA_ASSERT(pContext->backend == ma_backend_null);
(void)pContext;
return MA_SUCCESS;
}
static ma_result ma_context_init__null(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
{
MA_ASSERT(pContext != NULL);
(void)pConfig;
(void)pContext;
pCallbacks->onContextInit = ma_context_init__null;
pCallbacks->onContextUninit = ma_context_uninit__null;
pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__null;
pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__null;
pCallbacks->onDeviceInit = ma_device_init__null;
pCallbacks->onDeviceUninit = ma_device_uninit__null;
pCallbacks->onDeviceStart = ma_device_start__null;
pCallbacks->onDeviceStop = ma_device_stop__null;
pCallbacks->onDeviceRead = ma_device_read__null;
pCallbacks->onDeviceWrite = ma_device_write__null;
pCallbacks->onDeviceDataLoop = NULL; /* Our backend is asynchronous with a blocking read-write API which means we can get miniaudio to deal with the audio thread. */
/* The null backend always works. */
return MA_SUCCESS;
}
#endif
/*******************************************************************************
WIN32 COMMON
*******************************************************************************/
#if defined(MA_WIN32)
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
/* In shared mode we are always using the format reported by the operating system. */
WAVEFORMATEXTENSIBLE* pNativeFormat = NULL;
hr = ma_IAudioClient_GetMixFormat((ma_IAudioClient*)pData->pAudioClient, (WAVEFORMATEX**)&pNativeFormat);
if (hr != S_OK) {
result = MA_FORMAT_NOT_SUPPORTED;
} else {
MA_COPY_MEMORY(&wf, pNativeFormat, sizeof(wf));
result = MA_SUCCESS;
}
ma_CoTaskMemFree(pContext, pNativeFormat);
shareMode = MA_AUDCLNT_SHAREMODE_SHARED;
}
/* Return an error if we still haven't found a format. */
if (result != MA_SUCCESS) {
errorMsg = "[WASAPI] Failed to find best device mix format.";
goto done;
}
/*
Override the native sample rate with the one requested by the caller, but only if we're not using the default sample rate. We'll use
WASAPI to perform the sample rate conversion.
*/
nativeSampleRate = wf.Format.nSamplesPerSec;
if (streamFlags & MA_AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM) {
wf.Format.nSamplesPerSec = (pData->sampleRateIn != 0) ? pData->sampleRateIn : MA_DEFAULT_SAMPLE_RATE;
wf.Format.nAvgBytesPerSec = wf.Format.nSamplesPerSec * wf.Format.nBlockAlign;
}
pData->formatOut = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)&wf);
if (pData->formatOut == ma_format_unknown) {
/*
The format isn't supported. This is almost certainly because the exclusive mode format isn't supported by miniaudio. We need to return MA_SHARE_MODE_NOT_SUPPORTED
in this case so that the caller can detect it and fall back to shared mode if desired. We should never get here if shared mode was requested, but just for
completeness we'll check for it and return MA_FORMAT_NOT_SUPPORTED.
*/
if (shareMode == MA_AUDCLNT_SHAREMODE_EXCLUSIVE) {
result = MA_SHARE_MODE_NOT_SUPPORTED;
} else {
result = MA_FORMAT_NOT_SUPPORTED;
}
errorMsg = "[WASAPI] Native format not supported.";
goto done;
}
pData->channelsOut = wf.Format.nChannels;
pData->sampleRateOut = wf.Format.nSamplesPerSec;
/* Get the internal channel map based on the channel mask. */
ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pData->channelsOut, pData->channelMapOut);
/* Period size. */
pData->periodsOut = (pData->periodsIn != 0) ? pData->periodsIn : MA_DEFAULT_PERIODS;
pData->periodSizeInFramesOut = pData->periodSizeInFramesIn;
if (pData->periodSizeInFramesOut == 0) {
if (pData->periodSizeInMillisecondsIn == 0) {
if (pData->performanceProfile == ma_performance_profile_low_latency) {
pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, wf.Format.nSamplesPerSec);
} else {
pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, wf.Format.nSamplesPerSec);
}
} else {
pData->periodSizeInFramesOut = ma_calculate_buffer_size_in_frames_from_milliseconds(pData->periodSizeInMillisecondsIn, wf.Format.nSamplesPerSec);
}
}
periodDurationInMicroseconds = ((ma_uint64)pData->periodSizeInFramesOut * 1000 * 1000) / wf.Format.nSamplesPerSec;
/* Slightly different initialization for shared and exclusive modes. We try exclusive mode first, and if it fails, fall back to shared mode. */
if (shareMode == MA_AUDCLNT_SHAREMODE_EXCLUSIVE) {
MA_REFERENCE_TIME bufferDuration = periodDurationInMicroseconds * pData->periodsOut * 10;
/*
If the periodicy is too small, Initialize() will fail with AUDCLNT_E_INVALID_DEVICE_PERIOD. In this case we should just keep increasing
it and trying it again.
*/
hr = E_FAIL;
for (;;) {
hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, bufferDuration, (WAVEFORMATEX*)&wf, NULL);
if (hr == MA_AUDCLNT_E_INVALID_DEVICE_PERIOD) {
if (bufferDuration > 500*10000) {
break;
} else {
if (bufferDuration == 0) { /* <-- Just a sanity check to prevent an infinit loop. Should never happen, but it makes me feel better. */
break;
}
bufferDuration = bufferDuration * 2;
continue;
}
} else {
break;
}
}
if (hr == MA_AUDCLNT_E_BUFFER_SIZE_NOT_ALIGNED) {
ma_uint32 bufferSizeInFrames;
hr = ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pData->pAudioClient, &bufferSizeInFrames);
if (SUCCEEDED(hr)) {
bufferDuration = (MA_REFERENCE_TIME)((10000.0 * 1000 / wf.Format.nSamplesPerSec * bufferSizeInFrames) + 0.5);
/* Unfortunately we need to release and re-acquire the audio client according to MSDN. Seems silly - why not just call IAudioClient_Initialize() again?! */
ma_IAudioClient_Release((ma_IAudioClient*)pData->pAudioClient);
#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
hr = ma_IMMDevice_Activate(pDeviceInterface, &MA_IID_IAudioClient, CLSCTX_ALL, NULL, (void**)&pData->pAudioClient);
#else
hr = ma_IUnknown_QueryInterface(pDeviceInterface, &MA_IID_IAudioClient, (void**)&pData->pAudioClient);
#endif
if (SUCCEEDED(hr)) {
hr = ma_IAudioClient_Initialize((ma_IAudioClient*)pData->pAudioClient, shareMode, streamFlags, bufferDuration, bufferDuration, (WAVEFORMATEX*)&wf, NULL);
}
}
}
if (FAILED(hr)) {
/* Failed to initialize in exclusive mode. Don't fall back to shared mode - instead tell the client about it. They can reinitialize in shared mode if they want. */
if (hr == E_ACCESSDENIED) {
errorMsg = "[WASAPI] Failed to initialize device in exclusive mode. Access denied.", result = MA_ACCESS_DENIED;
} else if (hr == MA_AUDCLNT_E_DEVICE_IN_USE) {
errorMsg = "[WASAPI] Failed to initialize device in exclusive mode. Device in use.", result = MA_BUSY;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
return result;
}
ma_IAudioClient_SetEventHandle((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, pDevice->wasapi.hEventPlayback);
pDevice->wasapi.periodSizeInFramesPlayback = data.periodSizeInFramesOut;
ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &pDevice->wasapi.actualBufferSizeInFramesPlayback);
/* We must always have a valid ID because rerouting will look at it. */
ma_wcscpy_s(pDevice->playback.id.wasapi, sizeof(pDevice->playback.id.wasapi), data.id.wasapi);
/* The descriptor needs to be updated with actual values. */
pDescriptorPlayback->format = data.formatOut;
pDescriptorPlayback->channels = data.channelsOut;
pDescriptorPlayback->sampleRate = data.sampleRateOut;
MA_COPY_MEMORY(pDescriptorPlayback->channelMap, data.channelMapOut, sizeof(data.channelMapOut));
pDescriptorPlayback->periodSizeInFrames = data.periodSizeInFramesOut;
pDescriptorPlayback->periodCount = data.periodsOut;
}
/*
We need to register a notification client to detect when the device has been disabled, unplugged or re-routed (when the default device changes). When
we are connecting to the default device we want to do automatic stream routing when the device is disabled or unplugged. Otherwise we want to just
stop the device outright and let the application handle it.
*/
#if defined(MA_WIN32_DESKTOP) || defined(MA_WIN32_GDK)
if (pConfig->wasapi.noAutoStreamRouting == MA_FALSE) {
if ((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.pDeviceID == NULL) {
pDevice->wasapi.allowCaptureAutoStreamRouting = MA_TRUE;
}
if ((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.pDeviceID == NULL) {
pDevice->wasapi.allowPlaybackAutoStreamRouting = MA_TRUE;
}
}
hr = ma_CoCreateInstance(pDevice->pContext, MA_CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, MA_IID_IMMDeviceEnumerator, (void**)&pDeviceEnumerator);
if (FAILED(hr)) {
ma_device_uninit__wasapi(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to create device enumerator.");
return ma_result_from_HRESULT(hr);
}
pDevice->wasapi.notificationClient.lpVtbl = (void*)&g_maNotificationCientVtbl;
pDevice->wasapi.notificationClient.counter = 1;
pDevice->wasapi.notificationClient.pDevice = pDevice;
hr = pDeviceEnumerator->lpVtbl->RegisterEndpointNotificationCallback(pDeviceEnumerator, &pDevice->wasapi.notificationClient);
if (SUCCEEDED(hr)) {
pDevice->wasapi.pDeviceEnumerator = (ma_ptr)pDeviceEnumerator;
} else {
/* Not the end of the world if we fail to register the notification callback. We just won't support automatic stream routing. */
ma_IMMDeviceEnumerator_Release(pDeviceEnumerator);
}
#endif
c89atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_FALSE);
c89atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_FALSE);
return MA_SUCCESS;
}
static ma_result ma_device__get_available_frames__wasapi(ma_device* pDevice, ma_IAudioClient* pAudioClient, ma_uint32* pFrameCount)
{
ma_uint32 paddingFramesCount;
HRESULT hr;
ma_share_mode shareMode;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(pFrameCount != NULL);
*pFrameCount = 0;
if ((ma_ptr)pAudioClient != pDevice->wasapi.pAudioClientPlayback && (ma_ptr)pAudioClient != pDevice->wasapi.pAudioClientCapture) {
return MA_INVALID_OPERATION;
}
/*
I've had a report that GetCurrentPadding() is returning a frame count of 0 which is preventing
higher level function calls from doing anything because it thinks nothing is available. I have
taken a look at the documentation and it looks like this is unnecessary in exclusive mode.
From Microsoft's documentation:
For an exclusive-mode rendering or capture stream that was initialized with the
AUDCLNT_STREAMFLAGS_EVENTCALLBACK flag, the client typically has no use for the padding
value reported by GetCurrentPadding. Instead, the client accesses an entire buffer during
each processing pass.
Considering this, I'm going to skip GetCurrentPadding() for exclusive mode and just report the
entire buffer. This depends on the caller making sure they wait on the event handler.
*/
shareMode = ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) ? pDevice->playback.shareMode : pDevice->capture.shareMode;
if (shareMode == ma_share_mode_shared) {
/* Shared mode. */
hr = ma_IAudioClient_GetCurrentPadding(pAudioClient, &paddingFramesCount);
if (FAILED(hr)) {
return ma_result_from_HRESULT(hr);
}
if ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) {
*pFrameCount = pDevice->wasapi.actualBufferSizeInFramesPlayback - paddingFramesCount;
} else {
*pFrameCount = paddingFramesCount;
}
} else {
/* Exclusive mode. */
if ((ma_ptr)pAudioClient == pDevice->wasapi.pAudioClientPlayback) {
*pFrameCount = pDevice->wasapi.actualBufferSizeInFramesPlayback;
} else {
*pFrameCount = pDevice->wasapi.actualBufferSizeInFramesCapture;
}
}
return MA_SUCCESS;
}
static ma_result ma_device_reroute__wasapi(ma_device* pDevice, ma_device_type deviceType)
{
ma_result result;
if (deviceType == ma_device_type_duplex) {
return MA_INVALID_ARGS;
}
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "=== CHANGING DEVICE ===\n");
result = ma_device_reinit__wasapi(pDevice, deviceType);
if (result != MA_SUCCESS) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[WASAPI] Reinitializing device after route change failed.\n");
return result;
}
ma_device__post_init_setup(pDevice, deviceType);
ma_device__on_notification_rerouted(pDevice);
return MA_SUCCESS;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
MA_ASSERT(pDevice != NULL);
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal capture device.");
return ma_result_from_HRESULT(hr);
}
c89atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_TRUE);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
hr = ma_IAudioClient_Start((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to start internal playback device.");
return ma_result_from_HRESULT(hr);
}
c89atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_TRUE);
}
return MA_SUCCESS;
}
static ma_result ma_device_stop__wasapi(ma_device* pDevice)
{
ma_result result;
HRESULT hr;
MA_ASSERT(pDevice != NULL);
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex || pDevice->type == ma_device_type_loopback) {
hr = ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to stop internal capture device.");
return ma_result_from_HRESULT(hr);
}
/* The audio client needs to be reset otherwise restarting will fail. */
hr = ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to reset internal capture device.");
return ma_result_from_HRESULT(hr);
}
/* If we have a mapped buffer we need to release it. */
if (pDevice->wasapi.pMappedBufferCapture != NULL) {
ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap);
pDevice->wasapi.pMappedBufferCapture = NULL;
pDevice->wasapi.mappedBufferCaptureCap = 0;
pDevice->wasapi.mappedBufferCaptureLen = 0;
}
c89atomic_exchange_32(&pDevice->wasapi.isStartedCapture, MA_FALSE);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
/*
The buffer needs to be drained before stopping the device. Not doing this will result in the last few frames not getting output to
the speakers. This is a problem for very short sounds because it'll result in a significant portion of it not getting played.
*/
if (c89atomic_load_32(&pDevice->wasapi.isStartedPlayback)) {
/* We need to make sure we put a timeout here or else we'll risk getting stuck in a deadlock in some cases. */
DWORD waitTime = pDevice->wasapi.actualBufferSizeInFramesPlayback / pDevice->playback.internalSampleRate;
if (pDevice->playback.shareMode == ma_share_mode_exclusive) {
WaitForSingleObject(pDevice->wasapi.hEventPlayback, waitTime);
} else {
ma_uint32 prevFramesAvaialablePlayback = (ma_uint32)-1;
ma_uint32 framesAvailablePlayback;
for (;;) {
result = ma_device__get_available_frames__wasapi(pDevice, (ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback, &framesAvailablePlayback);
if (result != MA_SUCCESS) {
break;
}
if (framesAvailablePlayback >= pDevice->wasapi.actualBufferSizeInFramesPlayback) {
break;
}
/*
Just a safety check to avoid an infinite loop. If this iteration results in a situation where the number of available frames
has not changed, get out of the loop. I don't think this should ever happen, but I think it's nice to have just in case.
*/
if (framesAvailablePlayback == prevFramesAvaialablePlayback) {
break;
}
prevFramesAvaialablePlayback = framesAvailablePlayback;
WaitForSingleObject(pDevice->wasapi.hEventPlayback, waitTime);
ResetEvent(pDevice->wasapi.hEventPlayback); /* Manual reset. */
}
}
}
hr = ma_IAudioClient_Stop((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to stop internal playback device.");
return ma_result_from_HRESULT(hr);
}
/* The audio client needs to be reset otherwise restarting will fail. */
hr = ma_IAudioClient_Reset((ma_IAudioClient*)pDevice->wasapi.pAudioClientPlayback);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to reset internal playback device.");
return ma_result_from_HRESULT(hr);
}
if (pDevice->wasapi.pMappedBufferPlayback != NULL) {
ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, pDevice->wasapi.mappedBufferPlaybackCap, 0);
pDevice->wasapi.pMappedBufferPlayback = NULL;
pDevice->wasapi.mappedBufferPlaybackCap = 0;
pDevice->wasapi.mappedBufferPlaybackLen = 0;
}
c89atomic_exchange_32(&pDevice->wasapi.isStartedPlayback, MA_FALSE);
}
return MA_SUCCESS;
}
#ifndef MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS
#define MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS 5000
#endif
static ma_result ma_device_read__wasapi(ma_device* pDevice, void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
{
ma_result result = MA_SUCCESS;
ma_uint32 totalFramesProcessed = 0;
/*
When reading, we need to get a buffer and process all of it before releasing it. Because the
frame count (frameCount) can be different to the size of the buffer, we'll need to cache the
pointer to the buffer.
*/
/* Keep running until we've processed the requested number of frames. */
while (ma_device_get_state(pDevice) == ma_device_state_started && totalFramesProcessed < frameCount) {
ma_uint32 framesRemaining = frameCount - totalFramesProcessed;
/* If we have a mapped data buffer, consume that first. */
if (pDevice->wasapi.pMappedBufferCapture != NULL) {
/* We have a cached data pointer so consume that before grabbing another one from WASAPI. */
ma_uint32 framesToProcessNow = framesRemaining;
if (framesToProcessNow > pDevice->wasapi.mappedBufferCaptureLen) {
framesToProcessNow = pDevice->wasapi.mappedBufferCaptureLen;
}
/* Now just copy the data over to the output buffer. */
ma_copy_pcm_frames(
ma_offset_pcm_frames_ptr(pFrames, totalFramesProcessed, pDevice->capture.internalFormat, pDevice->capture.internalChannels),
ma_offset_pcm_frames_const_ptr(pDevice->wasapi.pMappedBufferCapture, pDevice->wasapi.mappedBufferCaptureCap - pDevice->wasapi.mappedBufferCaptureLen, pDevice->capture.internalFormat, pDevice->capture.internalChannels),
framesToProcessNow,
pDevice->capture.internalFormat, pDevice->capture.internalChannels
);
totalFramesProcessed += framesToProcessNow;
pDevice->wasapi.mappedBufferCaptureLen -= framesToProcessNow;
/* If the data buffer has been fully consumed we need to release it. */
if (pDevice->wasapi.mappedBufferCaptureLen == 0) {
ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap);
pDevice->wasapi.pMappedBufferCapture = NULL;
pDevice->wasapi.mappedBufferCaptureCap = 0;
}
} else {
/* We don't have any cached data pointer, so grab another one. */
HRESULT hr;
DWORD flags;
/* First just ask WASAPI for a data buffer. If it's not available, we'll wait for more. */
hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pDevice->wasapi.pMappedBufferCapture, &pDevice->wasapi.mappedBufferCaptureCap, &flags, NULL, NULL);
if (hr == S_OK) {
/* We got a data buffer. Continue to the next loop iteration which will then read from the mapped pointer. */
/* Overrun detection. */
if ((flags & MA_AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) != 0) {
/* Glitched. Probably due to an overrun. */
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Data discontinuity (possible overrun). Attempting recovery. mappedBufferCaptureCap=%d\n", pDevice->wasapi.mappedBufferCaptureCap);
/*
If we got an overrun it probably means we're straddling the end of the buffer. In order to prevent
a never-ending sequence of glitches we're going to recover by completely clearing out the capture
buffer.
*/
{
ma_uint32 iterationCount = 4; /* Safety to prevent an infinite loop. */
ma_uint32 i;
for (i = 0; i < iterationCount; i += 1) {
hr = ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap);
if (FAILED(hr)) {
break;
}
hr = ma_IAudioCaptureClient_GetBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, (BYTE**)&pDevice->wasapi.pMappedBufferCapture, &pDevice->wasapi.mappedBufferCaptureCap, &flags, NULL, NULL);
if (hr == MA_AUDCLNT_S_BUFFER_EMPTY || FAILED(hr)) {
break;
}
}
}
/* We should not have a valid buffer at this point so make sure everything is empty. */
pDevice->wasapi.pMappedBufferCapture = NULL;
pDevice->wasapi.mappedBufferCaptureCap = 0;
pDevice->wasapi.mappedBufferCaptureLen = 0;
} else {
/* The data is clean. */
pDevice->wasapi.mappedBufferCaptureLen = pDevice->wasapi.mappedBufferCaptureCap;
if (flags != 0) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[WASAPI] Capture Flags: %ld\n", flags);
}
}
continue;
} else {
if (hr == MA_AUDCLNT_S_BUFFER_EMPTY || hr == MA_AUDCLNT_E_BUFFER_ERROR) {
/*
No data is available. We need to wait for more. There's two situations to consider
here. The first is normal capture mode. If this times out it probably means the
microphone isn't delivering data for whatever reason. In this case we'll just
abort the read and return whatever we were able to get. The other situations is
loopback mode, in which case a timeout probably just means the nothing is playing
through the speakers.
*/
if (WaitForSingleObject(pDevice->wasapi.hEventCapture, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) {
if (pDevice->type == ma_device_type_loopback) {
continue; /* Keep waiting in loopback mode. */
} else {
result = MA_ERROR;
break; /* Wait failed. */
}
}
/* At this point we should be able to loop back to the start of the loop and try retrieving a data buffer again. */
} else {
/* An error occured and we need to abort. */
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from capture device in preparation for reading from the device. HRESULT = %d. Stopping device.\n", (int)hr);
result = ma_result_from_HRESULT(hr);
break;
}
}
}
}
/*
If we were unable to process the entire requested frame count, but we still have a mapped buffer,
there's a good chance either an error occurred or the device was stopped mid-read. In this case
we'll need to make sure the buffer is released.
*/
if (totalFramesProcessed < frameCount && pDevice->wasapi.pMappedBufferCapture != NULL) {
ma_IAudioCaptureClient_ReleaseBuffer((ma_IAudioCaptureClient*)pDevice->wasapi.pCaptureClient, pDevice->wasapi.mappedBufferCaptureCap);
pDevice->wasapi.pMappedBufferCapture = NULL;
pDevice->wasapi.mappedBufferCaptureCap = 0;
pDevice->wasapi.mappedBufferCaptureLen = 0;
}
if (pFramesRead != NULL) {
*pFramesRead = totalFramesProcessed;
}
return result;
}
static ma_result ma_device_write__wasapi(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
{
ma_result result = MA_SUCCESS;
ma_uint32 totalFramesProcessed = 0;
/* Keep writing to the device until it's stopped or we've consumed all of our input. */
while (ma_device_get_state(pDevice) == ma_device_state_started && totalFramesProcessed < frameCount) {
ma_uint32 framesRemaining = frameCount - totalFramesProcessed;
/*
We're going to do this in a similar way to capture. We'll first check if the cached data pointer
is valid, and if so, read from that. Otherwise We will call IAudioRenderClient_GetBuffer() with
a requested buffer size equal to our actual period size. If it returns AUDCLNT_E_BUFFER_TOO_LARGE
it means we need to wait for some data to become available.
*/
if (pDevice->wasapi.pMappedBufferPlayback != NULL) {
/* We still have some space available in the mapped data buffer. Write to it. */
ma_uint32 framesToProcessNow = framesRemaining;
if (framesToProcessNow > (pDevice->wasapi.mappedBufferPlaybackCap - pDevice->wasapi.mappedBufferPlaybackLen)) {
framesToProcessNow = (pDevice->wasapi.mappedBufferPlaybackCap - pDevice->wasapi.mappedBufferPlaybackLen);
}
/* Now just copy the data over to the output buffer. */
ma_copy_pcm_frames(
ma_offset_pcm_frames_ptr(pDevice->wasapi.pMappedBufferPlayback, pDevice->wasapi.mappedBufferPlaybackLen, pDevice->playback.internalFormat, pDevice->playback.internalChannels),
ma_offset_pcm_frames_const_ptr(pFrames, totalFramesProcessed, pDevice->playback.internalFormat, pDevice->playback.internalChannels),
framesToProcessNow,
pDevice->playback.internalFormat, pDevice->playback.internalChannels
);
totalFramesProcessed += framesToProcessNow;
pDevice->wasapi.mappedBufferPlaybackLen += framesToProcessNow;
/* If the data buffer has been fully consumed we need to release it. */
if (pDevice->wasapi.mappedBufferPlaybackLen == pDevice->wasapi.mappedBufferPlaybackCap) {
ma_IAudioRenderClient_ReleaseBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, pDevice->wasapi.mappedBufferPlaybackCap, 0);
pDevice->wasapi.pMappedBufferPlayback = NULL;
pDevice->wasapi.mappedBufferPlaybackCap = 0;
pDevice->wasapi.mappedBufferPlaybackLen = 0;
/*
In exclusive mode we need to wait here. Exclusive mode is weird because GetBuffer() never
seems to return AUDCLNT_E_BUFFER_TOO_LARGE, which is what we normally use to determine
whether or not we need to wait for more data.
*/
if (pDevice->playback.shareMode == ma_share_mode_exclusive) {
if (WaitForSingleObject(pDevice->wasapi.hEventPlayback, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) {
result = MA_ERROR;
break; /* Wait failed. Probably timed out. */
}
}
}
} else {
/* We don't have a mapped data buffer so we'll need to get one. */
HRESULT hr;
ma_uint32 bufferSizeInFrames;
/* Special rules for exclusive mode. */
if (pDevice->playback.shareMode == ma_share_mode_exclusive) {
bufferSizeInFrames = pDevice->wasapi.actualBufferSizeInFramesPlayback;
} else {
bufferSizeInFrames = pDevice->wasapi.periodSizeInFramesPlayback;
}
hr = ma_IAudioRenderClient_GetBuffer((ma_IAudioRenderClient*)pDevice->wasapi.pRenderClient, bufferSizeInFrames, (BYTE**)&pDevice->wasapi.pMappedBufferPlayback);
if (hr == S_OK) {
/* We have data available. */
pDevice->wasapi.mappedBufferPlaybackCap = bufferSizeInFrames;
pDevice->wasapi.mappedBufferPlaybackLen = 0;
} else {
if (hr == MA_AUDCLNT_E_BUFFER_TOO_LARGE || hr == MA_AUDCLNT_E_BUFFER_ERROR) {
/* Not enough data available. We need to wait for more. */
if (WaitForSingleObject(pDevice->wasapi.hEventPlayback, MA_WASAPI_WAIT_TIMEOUT_MILLISECONDS) != WAIT_OBJECT_0) {
result = MA_ERROR;
break; /* Wait failed. Probably timed out. */
}
} else {
/* Some error occurred. We'll need to abort. */
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WASAPI] Failed to retrieve internal buffer from playback device in preparation for writing to the device. HRESULT = %d. Stopping device.\n", (int)hr);
result = ma_result_from_HRESULT(hr);
break;
}
}
}
}
if (pFramesWritten != NULL) {
*pFramesWritten = totalFramesProcessed;
}
return result;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
ma_IDirectSoundBuffer_Release((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer);
}
if (pDevice->dsound.pPlayback != NULL) {
ma_IDirectSound_Release((ma_IDirectSound*)pDevice->dsound.pPlayback);
}
return MA_SUCCESS;
}
static ma_result ma_config_to_WAVEFORMATEXTENSIBLE(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const ma_channel* pChannelMap, WAVEFORMATEXTENSIBLE* pWF)
{
GUID subformat;
if (format == ma_format_unknown) {
format = MA_DEFAULT_FORMAT;
}
if (channels == 0) {
channels = MA_DEFAULT_CHANNELS;
}
if (sampleRate == 0) {
sampleRate = MA_DEFAULT_SAMPLE_RATE;
}
switch (format)
{
case ma_format_u8:
case ma_format_s16:
case ma_format_s24:
/*case ma_format_s24_32:*/
case ma_format_s32:
{
subformat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM;
} break;
case ma_format_f32:
{
subformat = MA_GUID_KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
} break;
default:
return MA_FORMAT_NOT_SUPPORTED;
}
MA_ZERO_OBJECT(pWF);
pWF->Format.cbSize = sizeof(*pWF);
pWF->Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
pWF->Format.nChannels = (WORD)channels;
pWF->Format.nSamplesPerSec = (DWORD)sampleRate;
pWF->Format.wBitsPerSample = (WORD)(ma_get_bytes_per_sample(format)*8);
pWF->Format.nBlockAlign = (WORD)(pWF->Format.nChannels * pWF->Format.wBitsPerSample / 8);
pWF->Format.nAvgBytesPerSec = pWF->Format.nBlockAlign * pWF->Format.nSamplesPerSec;
pWF->Samples.wValidBitsPerSample = pWF->Format.wBitsPerSample;
pWF->dwChannelMask = ma_channel_map_to_channel_mask__win32(pChannelMap, channels);
pWF->SubFormat = subformat;
return MA_SUCCESS;
}
static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__dsound(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
{
/*
DirectSound has a minimum period size of 20ms. In practice, this doesn't seem to be enough for
reliable glitch-free processing so going to use 30ms instead.
*/
ma_uint32 minPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(30, nativeSampleRate);
ma_uint32 periodSizeInFrames;
periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, nativeSampleRate, performanceProfile);
if (periodSizeInFrames < minPeriodSizeInFrames) {
periodSizeInFrames = minPeriodSizeInFrames;
}
return periodSizeInFrames;
}
static ma_result ma_device_init__dsound(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
ma_result result;
HRESULT hr;
MA_ASSERT(pDevice != NULL);
MA_ZERO_OBJECT(&pDevice->dsound);
if (pConfig->deviceType == ma_device_type_loopback) {
return MA_DEVICE_TYPE_NOT_SUPPORTED;
}
/*
Unfortunately DirectSound uses different APIs and data structures for playback and catpure devices. We need to initialize
the capture device first because we'll want to match it's buffer size and period count on the playback side if we're using
full-duplex mode.
*/
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
WAVEFORMATEXTENSIBLE wf;
MA_DSCBUFFERDESC descDS;
ma_uint32 periodSizeInFrames;
ma_uint32 periodCount;
char rawdata[1024]; /* <-- Ugly hack to avoid a malloc() due to a crappy DirectSound API. */
WAVEFORMATEXTENSIBLE* pActualFormat;
result = ma_config_to_WAVEFORMATEXTENSIBLE(pDescriptorCapture->format, pDescriptorCapture->channels, pDescriptorCapture->sampleRate, pDescriptorCapture->channelMap, &wf);
if (result != MA_SUCCESS) {
return result;
}
result = ma_context_create_IDirectSoundCapture__dsound(pDevice->pContext, pDescriptorCapture->shareMode, pDescriptorCapture->pDeviceID, (ma_IDirectSoundCapture**)&pDevice->dsound.pCapture);
if (result != MA_SUCCESS) {
ma_device_uninit__dsound(pDevice);
return result;
}
result = ma_context_get_format_info_for_IDirectSoundCapture__dsound(pDevice->pContext, (ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &wf.Format.nChannels, &wf.Format.wBitsPerSample, &wf.Format.nSamplesPerSec);
if (result != MA_SUCCESS) {
ma_device_uninit__dsound(pDevice);
return result;
}
wf.Format.nBlockAlign = (WORD)(wf.Format.nChannels * wf.Format.wBitsPerSample / 8);
wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec;
wf.Samples.wValidBitsPerSample = wf.Format.wBitsPerSample;
wf.SubFormat = MA_GUID_KSDATAFORMAT_SUBTYPE_PCM;
/* The size of the buffer must be a clean multiple of the period count. */
periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__dsound(pDescriptorCapture, wf.Format.nSamplesPerSec, pConfig->performanceProfile);
periodCount = (pDescriptorCapture->periodCount > 0) ? pDescriptorCapture->periodCount : MA_DEFAULT_PERIODS;
MA_ZERO_OBJECT(&descDS);
descDS.dwSize = sizeof(descDS);
descDS.dwFlags = 0;
descDS.dwBufferBytes = periodSizeInFrames * periodCount * wf.Format.nBlockAlign;
descDS.lpwfxFormat = (WAVEFORMATEX*)&wf;
hr = ma_IDirectSoundCapture_CreateCaptureBuffer((ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &descDS, (ma_IDirectSoundCaptureBuffer**)&pDevice->dsound.pCaptureBuffer, NULL);
if (FAILED(hr)) {
ma_device_uninit__dsound(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCapture_CreateCaptureBuffer() failed for capture device.");
return ma_result_from_HRESULT(hr);
}
/* Get the _actual_ properties of the buffer. */
pActualFormat = (WAVEFORMATEXTENSIBLE*)rawdata;
hr = ma_IDirectSoundCaptureBuffer_GetFormat((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, (WAVEFORMATEX*)pActualFormat, sizeof(rawdata), NULL);
if (FAILED(hr)) {
ma_device_uninit__dsound(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to retrieve the actual format of the capture device's buffer.");
return ma_result_from_HRESULT(hr);
}
/* We can now start setting the output data formats. */
pDescriptorCapture->format = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)pActualFormat);
pDescriptorCapture->channels = pActualFormat->Format.nChannels;
pDescriptorCapture->sampleRate = pActualFormat->Format.nSamplesPerSec;
/* Get the native channel map based on the channel mask. */
if (pActualFormat->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
ma_channel_mask_to_channel_map__win32(pActualFormat->dwChannelMask, pDescriptorCapture->channels, pDescriptorCapture->channelMap);
} else {
ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pDescriptorCapture->channels, pDescriptorCapture->channelMap);
}
/*
After getting the actual format the size of the buffer in frames may have actually changed. However, we want this to be as close to what the
user has asked for as possible, so let's go ahead and release the old capture buffer and create a new one in this case.
*/
if (periodSizeInFrames != (descDS.dwBufferBytes / ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) / periodCount)) {
descDS.dwBufferBytes = periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) * periodCount;
ma_IDirectSoundCaptureBuffer_Release((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
hr = ma_IDirectSoundCapture_CreateCaptureBuffer((ma_IDirectSoundCapture*)pDevice->dsound.pCapture, &descDS, (ma_IDirectSoundCaptureBuffer**)&pDevice->dsound.pCaptureBuffer, NULL);
if (FAILED(hr)) {
ma_device_uninit__dsound(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Second attempt at IDirectSoundCapture_CreateCaptureBuffer() failed for capture device.");
return ma_result_from_HRESULT(hr);
}
}
/* DirectSound should give us a buffer exactly the size we asked for. */
pDescriptorCapture->periodSizeInFrames = periodSizeInFrames;
pDescriptorCapture->periodCount = periodCount;
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
WAVEFORMATEXTENSIBLE wf;
MA_DSBUFFERDESC descDSPrimary;
MA_DSCAPS caps;
char rawdata[1024]; /* <-- Ugly hack to avoid a malloc() due to a crappy DirectSound API. */
WAVEFORMATEXTENSIBLE* pActualFormat;
ma_uint32 periodSizeInFrames;
ma_uint32 periodCount;
MA_DSBUFFERDESC descDS;
result = ma_config_to_WAVEFORMATEXTENSIBLE(pDescriptorPlayback->format, pDescriptorPlayback->channels, pDescriptorPlayback->sampleRate, pDescriptorPlayback->channelMap, &wf);
if (result != MA_SUCCESS) {
return result;
}
result = ma_context_create_IDirectSound__dsound(pDevice->pContext, pDescriptorPlayback->shareMode, pDescriptorPlayback->pDeviceID, (ma_IDirectSound**)&pDevice->dsound.pPlayback);
if (result != MA_SUCCESS) {
ma_device_uninit__dsound(pDevice);
return result;
}
MA_ZERO_OBJECT(&descDSPrimary);
descDSPrimary.dwSize = sizeof(MA_DSBUFFERDESC);
descDSPrimary.dwFlags = MA_DSBCAPS_PRIMARYBUFFER | MA_DSBCAPS_CTRLVOLUME;
hr = ma_IDirectSound_CreateSoundBuffer((ma_IDirectSound*)pDevice->dsound.pPlayback, &descDSPrimary, (ma_IDirectSoundBuffer**)&pDevice->dsound.pPlaybackPrimaryBuffer, NULL);
if (FAILED(hr)) {
ma_device_uninit__dsound(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_CreateSoundBuffer() failed for playback device's primary buffer.");
return ma_result_from_HRESULT(hr);
}
/* We may want to make some adjustments to the format if we are using defaults. */
MA_ZERO_OBJECT(&caps);
caps.dwSize = sizeof(caps);
hr = ma_IDirectSound_GetCaps((ma_IDirectSound*)pDevice->dsound.pPlayback, &caps);
if (FAILED(hr)) {
ma_device_uninit__dsound(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_GetCaps() failed for playback device.");
return ma_result_from_HRESULT(hr);
}
if (pDescriptorPlayback->channels == 0) {
if ((caps.dwFlags & MA_DSCAPS_PRIMARYSTEREO) != 0) {
DWORD speakerConfig;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
wf.Format.nChannels = 2;
/* Look at the speaker configuration to get a better idea on the channel count. */
if (SUCCEEDED(ma_IDirectSound_GetSpeakerConfig((ma_IDirectSound*)pDevice->dsound.pPlayback, &speakerConfig))) {
ma_get_channels_from_speaker_config__dsound(speakerConfig, &wf.Format.nChannels, &wf.dwChannelMask);
}
} else {
/* It does not support stereo, which means we are stuck with mono. */
wf.Format.nChannels = 1;
}
}
if (pDescriptorPlayback->sampleRate == 0) {
/* We base the sample rate on the values returned by GetCaps(). */
if ((caps.dwFlags & MA_DSCAPS_CONTINUOUSRATE) != 0) {
wf.Format.nSamplesPerSec = ma_get_best_sample_rate_within_range(caps.dwMinSecondarySampleRate, caps.dwMaxSecondarySampleRate);
} else {
wf.Format.nSamplesPerSec = caps.dwMaxSecondarySampleRate;
}
}
wf.Format.nBlockAlign = (WORD)(wf.Format.nChannels * wf.Format.wBitsPerSample / 8);
wf.Format.nAvgBytesPerSec = wf.Format.nBlockAlign * wf.Format.nSamplesPerSec;
/*
From MSDN:
The method succeeds even if the hardware does not support the requested format; DirectSound sets the buffer to the closest
supported format. To determine whether this has happened, an application can call the GetFormat method for the primary buffer
and compare the result with the format that was requested with the SetFormat method.
*/
hr = ma_IDirectSoundBuffer_SetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (WAVEFORMATEX*)&wf);
if (FAILED(hr)) {
ma_device_uninit__dsound(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to set format of playback device's primary buffer.");
return ma_result_from_HRESULT(hr);
}
/* Get the _actual_ properties of the buffer. */
pActualFormat = (WAVEFORMATEXTENSIBLE*)rawdata;
hr = ma_IDirectSoundBuffer_GetFormat((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackPrimaryBuffer, (WAVEFORMATEX*)pActualFormat, sizeof(rawdata), NULL);
if (FAILED(hr)) {
ma_device_uninit__dsound(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to retrieve the actual format of the playback device's primary buffer.");
return ma_result_from_HRESULT(hr);
}
/* We now have enough information to start setting some output properties. */
pDescriptorPlayback->format = ma_format_from_WAVEFORMATEX((WAVEFORMATEX*)pActualFormat);
pDescriptorPlayback->channels = pActualFormat->Format.nChannels;
pDescriptorPlayback->sampleRate = pActualFormat->Format.nSamplesPerSec;
/* Get the internal channel map based on the channel mask. */
if (pActualFormat->Format.wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
ma_channel_mask_to_channel_map__win32(pActualFormat->dwChannelMask, pDescriptorPlayback->channels, pDescriptorPlayback->channelMap);
} else {
ma_channel_mask_to_channel_map__win32(wf.dwChannelMask, pDescriptorPlayback->channels, pDescriptorPlayback->channelMap);
}
/* The size of the buffer must be a clean multiple of the period count. */
periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__dsound(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile);
periodCount = (pDescriptorPlayback->periodCount > 0) ? pDescriptorPlayback->periodCount : MA_DEFAULT_PERIODS;
/*
Meaning of dwFlags (from MSDN):
DSBCAPS_CTRLPOSITIONNOTIFY
The buffer has position notification capability.
DSBCAPS_GLOBALFOCUS
With this flag set, an application using DirectSound can continue to play its buffers if the user switches focus to
another application, even if the new application uses DirectSound.
DSBCAPS_GETCURRENTPOSITION2
In the first version of DirectSound, the play cursor was significantly ahead of the actual playing sound on emulated
sound cards; it was directly behind the write cursor. Now, if the DSBCAPS_GETCURRENTPOSITION2 flag is specified, the
application can get a more accurate play cursor.
*/
MA_ZERO_OBJECT(&descDS);
descDS.dwSize = sizeof(descDS);
descDS.dwFlags = MA_DSBCAPS_CTRLPOSITIONNOTIFY | MA_DSBCAPS_GLOBALFOCUS | MA_DSBCAPS_GETCURRENTPOSITION2;
descDS.dwBufferBytes = periodSizeInFrames * periodCount * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels);
descDS.lpwfxFormat = (WAVEFORMATEX*)&wf;
hr = ma_IDirectSound_CreateSoundBuffer((ma_IDirectSound*)pDevice->dsound.pPlayback, &descDS, (ma_IDirectSoundBuffer**)&pDevice->dsound.pPlaybackBuffer, NULL);
if (FAILED(hr)) {
ma_device_uninit__dsound(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSound_CreateSoundBuffer() failed for playback device's secondary buffer.");
return ma_result_from_HRESULT(hr);
}
/* DirectSound should give us a buffer exactly the size we asked for. */
pDescriptorPlayback->periodSizeInFrames = periodSizeInFrames;
pDescriptorPlayback->periodCount = periodCount;
}
return MA_SUCCESS;
}
static ma_result ma_device_data_loop__dsound(ma_device* pDevice)
{
ma_result result = MA_SUCCESS;
ma_uint32 bpfDeviceCapture = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
ma_uint32 bpfDevicePlayback = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
HRESULT hr;
DWORD lockOffsetInBytesCapture;
DWORD lockSizeInBytesCapture;
DWORD mappedSizeInBytesCapture;
DWORD mappedDeviceFramesProcessedCapture;
void* pMappedDeviceBufferCapture;
DWORD lockOffsetInBytesPlayback;
DWORD lockSizeInBytesPlayback;
DWORD mappedSizeInBytesPlayback;
void* pMappedDeviceBufferPlayback;
DWORD prevReadCursorInBytesCapture = 0;
DWORD prevPlayCursorInBytesPlayback = 0;
ma_bool32 physicalPlayCursorLoopFlagPlayback = 0;
DWORD virtualWriteCursorInBytesPlayback = 0;
ma_bool32 virtualWriteCursorLoopFlagPlayback = 0;
ma_bool32 isPlaybackDeviceStarted = MA_FALSE;
ma_uint32 framesWrittenToPlaybackDevice = 0; /* For knowing whether or not the playback device needs to be started. */
ma_uint32 waitTimeInMilliseconds = 1;
MA_ASSERT(pDevice != NULL);
/* The first thing to do is start the capture device. The playback device is only started after the first period is written. */
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
hr = ma_IDirectSoundCaptureBuffer_Start((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, MA_DSCBSTART_LOOPING);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCaptureBuffer_Start() failed.");
return ma_result_from_HRESULT(hr);
}
}
while (ma_device_get_state(pDevice) == ma_device_state_started) {
switch (pDevice->type)
{
case ma_device_type_duplex:
{
DWORD physicalCaptureCursorInBytes;
DWORD physicalReadCursorInBytes;
hr = ma_IDirectSoundCaptureBuffer_GetCurrentPosition((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, &physicalCaptureCursorInBytes, &physicalReadCursorInBytes);
if (FAILED(hr)) {
return ma_result_from_HRESULT(hr);
}
/* If nothing is available we just sleep for a bit and return from this iteration. */
if (physicalReadCursorInBytes == prevReadCursorInBytesCapture) {
ma_sleep(waitTimeInMilliseconds);
continue; /* Nothing is available in the capture buffer. */
}
/*
The current position has moved. We need to map all of the captured samples and write them to the playback device, making sure
we don't return until every frame has been copied over.
*/
if (prevReadCursorInBytesCapture < physicalReadCursorInBytes) {
/* The capture position has not looped. This is the simple case. */
lockOffsetInBytesCapture = prevReadCursorInBytesCapture;
lockSizeInBytesCapture = (physicalReadCursorInBytes - prevReadCursorInBytesCapture);
} else {
/*
The capture position has looped. This is the more complex case. Map to the end of the buffer. If this does not return anything,
do it again from the start.
*/
if (prevReadCursorInBytesCapture < pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) {
/* Lock up to the end of the buffer. */
lockOffsetInBytesCapture = prevReadCursorInBytesCapture;
lockSizeInBytesCapture = (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) - prevReadCursorInBytesCapture;
} else {
/* Lock starting from the start of the buffer. */
lockOffsetInBytesCapture = 0;
lockSizeInBytesCapture = physicalReadCursorInBytes;
}
}
if (lockSizeInBytesCapture == 0) {
ma_sleep(waitTimeInMilliseconds);
continue; /* Nothing is available in the capture buffer. */
}
hr = ma_IDirectSoundCaptureBuffer_Lock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, lockOffsetInBytesCapture, lockSizeInBytesCapture, &pMappedDeviceBufferCapture, &mappedSizeInBytesCapture, NULL, NULL, 0);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from capture device in preparation for writing to the device.");
return ma_result_from_HRESULT(hr);
}
/* At this point we have some input data that we need to output. We do not return until every mapped frame of the input data is written to the playback device. */
mappedDeviceFramesProcessedCapture = 0;
for (;;) { /* Keep writing to the playback device. */
ma_uint8 inputFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
ma_uint32 inputFramesInClientFormatCap = sizeof(inputFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
ma_uint8 outputFramesInClientFormat[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
ma_uint32 outputFramesInClientFormatCap = sizeof(outputFramesInClientFormat) / ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
ma_uint32 outputFramesInClientFormatCount;
ma_uint32 outputFramesInClientFormatConsumed = 0;
ma_uint64 clientCapturedFramesToProcess = ma_min(inputFramesInClientFormatCap, outputFramesInClientFormatCap);
ma_uint64 deviceCapturedFramesToProcess = (mappedSizeInBytesCapture / bpfDeviceCapture) - mappedDeviceFramesProcessedCapture;
void* pRunningMappedDeviceBufferCapture = ma_offset_ptr(pMappedDeviceBufferCapture, mappedDeviceFramesProcessedCapture * bpfDeviceCapture);
result = ma_data_converter_process_pcm_frames(&pDevice->capture.converter, pRunningMappedDeviceBufferCapture, &deviceCapturedFramesToProcess, inputFramesInClientFormat, &clientCapturedFramesToProcess);
if (result != MA_SUCCESS) {
break;
}
outputFramesInClientFormatCount = (ma_uint32)clientCapturedFramesToProcess;
mappedDeviceFramesProcessedCapture += (ma_uint32)deviceCapturedFramesToProcess;
ma_device__handle_data_callback(pDevice, outputFramesInClientFormat, inputFramesInClientFormat, (ma_uint32)clientCapturedFramesToProcess);
/* At this point we have input and output data in client format. All we need to do now is convert it to the output device format. This may take a few passes. */
for (;;) {
ma_uint32 framesWrittenThisIteration;
DWORD physicalPlayCursorInBytes;
DWORD physicalWriteCursorInBytes;
DWORD availableBytesPlayback;
DWORD silentPaddingInBytes = 0; /* <-- Must be initialized to 0. */
/* We need the physical play and write cursors. */
if (FAILED(ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes))) {
break;
}
if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) {
physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback;
}
prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes;
/* If there's any bytes available for writing we can do that now. The space between the virtual cursor position and play cursor. */
if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
/* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */
if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) {
availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */
} else {
/* This is an error. */
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Duplex/Playback): Play cursor has moved in front of the write cursor (same loop iteration). physicalPlayCursorInBytes=%ld, virtualWriteCurs...
availableBytesPlayback = 0;
}
} else {
/* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */
if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) {
availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
} else {
/* This is an error. */
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Duplex/Playback): Write cursor has moved behind the play cursor (different loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCur...
availableBytesPlayback = 0;
}
}
/* If there's no room available for writing we need to wait for more. */
if (availableBytesPlayback == 0) {
/* If we haven't started the device yet, this will never get beyond 0. In this case we need to get the device started. */
if (!isPlaybackDeviceStarted) {
hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
if (FAILED(hr)) {
ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.");
return ma_result_from_HRESULT(hr);
}
isPlaybackDeviceStarted = MA_TRUE;
} else {
ma_sleep(waitTimeInMilliseconds);
continue;
}
}
/* Getting here means there room available somewhere. We limit this to either the end of the buffer or the physical play cursor, whichever is closest. */
lockOffsetInBytesPlayback = virtualWriteCursorInBytesPlayback;
if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
/* Same loop iteration. Go up to the end of the buffer. */
lockSizeInBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
} else {
/* Different loop iterations. Go up to the physical play cursor. */
lockSizeInBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
}
hr = ma_IDirectSoundBuffer_Lock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, lockOffsetInBytesPlayback, lockSizeInBytesPlayback, &pMappedDeviceBufferPlayback, &mappedSizeInBytesPlayback, NULL, NULL, 0);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from playback device in preparation for writing to the device.");
result = ma_result_from_HRESULT(hr);
break;
}
/*
Experiment: If the playback buffer is being starved, pad it with some silence to get it back in sync. This will cause a glitch, but it may prevent
endless glitching due to it constantly running out of data.
*/
if (isPlaybackDeviceStarted) {
DWORD bytesQueuedForPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - availableBytesPlayback;
if (bytesQueuedForPlayback < (pDevice->playback.internalPeriodSizeInFrames*bpfDevicePlayback)) {
silentPaddingInBytes = (pDevice->playback.internalPeriodSizeInFrames*2*bpfDevicePlayback) - bytesQueuedForPlayback;
if (silentPaddingInBytes > lockSizeInBytesPlayback) {
silentPaddingInBytes = lockSizeInBytesPlayback;
}
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Duplex/Playback) Playback buffer starved. availableBytesPlayback=%ld, silentPaddingInBytes=%ld\n", availableBytesPlayback, silentPaddingInB...
}
}
/* At this point we have a buffer for output. */
if (silentPaddingInBytes > 0) {
MA_ZERO_MEMORY(pMappedDeviceBufferPlayback, silentPaddingInBytes);
framesWrittenThisIteration = silentPaddingInBytes/bpfDevicePlayback;
} else {
ma_uint64 convertedFrameCountIn = (outputFramesInClientFormatCount - outputFramesInClientFormatConsumed);
ma_uint64 convertedFrameCountOut = mappedSizeInBytesPlayback/bpfDevicePlayback;
void* pConvertedFramesIn = ma_offset_ptr(outputFramesInClientFormat, outputFramesInClientFormatConsumed * bpfDevicePlayback);
void* pConvertedFramesOut = pMappedDeviceBufferPlayback;
result = ma_data_converter_process_pcm_frames(&pDevice->playback.converter, pConvertedFramesIn, &convertedFrameCountIn, pConvertedFramesOut, &convertedFrameCountOut);
if (result != MA_SUCCESS) {
break;
}
outputFramesInClientFormatConsumed += (ma_uint32)convertedFrameCountOut;
framesWrittenThisIteration = (ma_uint32)convertedFrameCountOut;
}
hr = ma_IDirectSoundBuffer_Unlock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, pMappedDeviceBufferPlayback, framesWrittenThisIteration*bpfDevicePlayback, NULL, 0);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from playback device after writing to the device.");
result = ma_result_from_HRESULT(hr);
break;
}
virtualWriteCursorInBytesPlayback += framesWrittenThisIteration*bpfDevicePlayback;
if ((virtualWriteCursorInBytesPlayback/bpfDevicePlayback) == pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods) {
virtualWriteCursorInBytesPlayback = 0;
virtualWriteCursorLoopFlagPlayback = !virtualWriteCursorLoopFlagPlayback;
}
/*
We may need to start the device. We want two full periods to be written before starting the playback device. Having an extra period adds
a bit of a buffer to prevent the playback buffer from getting starved.
*/
framesWrittenToPlaybackDevice += framesWrittenThisIteration;
if (!isPlaybackDeviceStarted && framesWrittenToPlaybackDevice >= (pDevice->playback.internalPeriodSizeInFrames*2)) {
hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
if (FAILED(hr)) {
ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.");
return ma_result_from_HRESULT(hr);
}
isPlaybackDeviceStarted = MA_TRUE;
}
if (framesWrittenThisIteration < mappedSizeInBytesPlayback/bpfDevicePlayback) {
break; /* We're finished with the output data.*/
}
}
if (clientCapturedFramesToProcess == 0) {
break; /* We just consumed every input sample. */
}
}
/* At this point we're done with the mapped portion of the capture buffer. */
hr = ma_IDirectSoundCaptureBuffer_Unlock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, pMappedDeviceBufferCapture, mappedSizeInBytesCapture, NULL, 0);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from capture device after reading from the device.");
return ma_result_from_HRESULT(hr);
}
prevReadCursorInBytesCapture = (lockOffsetInBytesCapture + mappedSizeInBytesCapture);
} break;
case ma_device_type_capture:
{
DWORD physicalCaptureCursorInBytes;
DWORD physicalReadCursorInBytes;
hr = ma_IDirectSoundCaptureBuffer_GetCurrentPosition((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, &physicalCaptureCursorInBytes, &physicalReadCursorInBytes);
if (FAILED(hr)) {
return MA_ERROR;
}
/* If the previous capture position is the same as the current position we need to wait a bit longer. */
if (prevReadCursorInBytesCapture == physicalReadCursorInBytes) {
ma_sleep(waitTimeInMilliseconds);
continue;
}
/* Getting here means we have capture data available. */
if (prevReadCursorInBytesCapture < physicalReadCursorInBytes) {
/* The capture position has not looped. This is the simple case. */
lockOffsetInBytesCapture = prevReadCursorInBytesCapture;
lockSizeInBytesCapture = (physicalReadCursorInBytes - prevReadCursorInBytesCapture);
} else {
/*
The capture position has looped. This is the more complex case. Map to the end of the buffer. If this does not return anything,
do it again from the start.
*/
if (prevReadCursorInBytesCapture < pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) {
/* Lock up to the end of the buffer. */
lockOffsetInBytesCapture = prevReadCursorInBytesCapture;
lockSizeInBytesCapture = (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture) - prevReadCursorInBytesCapture;
} else {
/* Lock starting from the start of the buffer. */
lockOffsetInBytesCapture = 0;
lockSizeInBytesCapture = physicalReadCursorInBytes;
}
}
if (lockSizeInBytesCapture < pDevice->capture.internalPeriodSizeInFrames) {
ma_sleep(waitTimeInMilliseconds);
continue; /* Nothing is available in the capture buffer. */
}
hr = ma_IDirectSoundCaptureBuffer_Lock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, lockOffsetInBytesCapture, lockSizeInBytesCapture, &pMappedDeviceBufferCapture, &mappedSizeInBytesCapture, NULL, NULL, 0);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from capture device in preparation for writing to the device.");
result = ma_result_from_HRESULT(hr);
}
if (lockSizeInBytesCapture != mappedSizeInBytesCapture) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[DirectSound] (Capture) lockSizeInBytesCapture=%ld != mappedSizeInBytesCapture=%ld\n", lockSizeInBytesCapture, mappedSizeInBytesCapture);
}
ma_device__send_frames_to_client(pDevice, mappedSizeInBytesCapture/bpfDeviceCapture, pMappedDeviceBufferCapture);
hr = ma_IDirectSoundCaptureBuffer_Unlock((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer, pMappedDeviceBufferCapture, mappedSizeInBytesCapture, NULL, 0);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from capture device after reading from the device.");
return ma_result_from_HRESULT(hr);
}
prevReadCursorInBytesCapture = lockOffsetInBytesCapture + mappedSizeInBytesCapture;
if (prevReadCursorInBytesCapture == (pDevice->capture.internalPeriodSizeInFrames*pDevice->capture.internalPeriods*bpfDeviceCapture)) {
prevReadCursorInBytesCapture = 0;
}
} break;
case ma_device_type_playback:
{
DWORD availableBytesPlayback;
DWORD physicalPlayCursorInBytes;
DWORD physicalWriteCursorInBytes;
hr = ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes);
if (FAILED(hr)) {
break;
}
if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) {
physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback;
}
prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes;
/* If there's any bytes available for writing we can do that now. The space between the virtual cursor position and play cursor. */
if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
/* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */
if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) {
availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */
} else {
/* This is an error. */
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Playback): Play cursor has moved in front of the write cursor (same loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld....
availableBytesPlayback = 0;
}
} else {
/* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */
if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) {
availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
} else {
/* This is an error. */
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[DirectSound] (Playback): Write cursor has moved behind the play cursor (different loop iterations). physicalPlayCursorInBytes=%ld, virtualWriteCursorInBytes=%ld....
availableBytesPlayback = 0;
}
}
/* If there's no room available for writing we need to wait for more. */
if (availableBytesPlayback < pDevice->playback.internalPeriodSizeInFrames) {
/* If we haven't started the device yet, this will never get beyond 0. In this case we need to get the device started. */
if (availableBytesPlayback == 0 && !isPlaybackDeviceStarted) {
hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.");
return ma_result_from_HRESULT(hr);
}
isPlaybackDeviceStarted = MA_TRUE;
} else {
ma_sleep(waitTimeInMilliseconds);
continue;
}
}
/* Getting here means there room available somewhere. We limit this to either the end of the buffer or the physical play cursor, whichever is closest. */
lockOffsetInBytesPlayback = virtualWriteCursorInBytesPlayback;
if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
/* Same loop iteration. Go up to the end of the buffer. */
lockSizeInBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
} else {
/* Different loop iterations. Go up to the physical play cursor. */
lockSizeInBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
}
hr = ma_IDirectSoundBuffer_Lock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, lockOffsetInBytesPlayback, lockSizeInBytesPlayback, &pMappedDeviceBufferPlayback, &mappedSizeInBytesPlayback, NULL, NULL, 0);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to map buffer from playback device in preparation for writing to the device.");
result = ma_result_from_HRESULT(hr);
break;
}
/* At this point we have a buffer for output. */
ma_device__read_frames_from_client(pDevice, (mappedSizeInBytesPlayback/bpfDevicePlayback), pMappedDeviceBufferPlayback);
hr = ma_IDirectSoundBuffer_Unlock((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, pMappedDeviceBufferPlayback, mappedSizeInBytesPlayback, NULL, 0);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] Failed to unlock internal buffer from playback device after writing to the device.");
result = ma_result_from_HRESULT(hr);
break;
}
virtualWriteCursorInBytesPlayback += mappedSizeInBytesPlayback;
if (virtualWriteCursorInBytesPlayback == pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) {
virtualWriteCursorInBytesPlayback = 0;
virtualWriteCursorLoopFlagPlayback = !virtualWriteCursorLoopFlagPlayback;
}
/*
We may need to start the device. We want two full periods to be written before starting the playback device. Having an extra period adds
a bit of a buffer to prevent the playback buffer from getting starved.
*/
framesWrittenToPlaybackDevice += mappedSizeInBytesPlayback/bpfDevicePlayback;
if (!isPlaybackDeviceStarted && framesWrittenToPlaybackDevice >= pDevice->playback.internalPeriodSizeInFrames) {
hr = ma_IDirectSoundBuffer_Play((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, 0, 0, MA_DSBPLAY_LOOPING);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundBuffer_Play() failed.");
return ma_result_from_HRESULT(hr);
}
isPlaybackDeviceStarted = MA_TRUE;
}
} break;
default: return MA_INVALID_ARGS; /* Invalid device type. */
}
if (result != MA_SUCCESS) {
return result;
}
}
/* Getting here means the device is being stopped. */
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
hr = ma_IDirectSoundCaptureBuffer_Stop((ma_IDirectSoundCaptureBuffer*)pDevice->dsound.pCaptureBuffer);
if (FAILED(hr)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[DirectSound] IDirectSoundCaptureBuffer_Stop() failed.");
return ma_result_from_HRESULT(hr);
}
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
/* The playback device should be drained before stopping. All we do is wait until the available bytes is equal to the size of the buffer. */
if (isPlaybackDeviceStarted) {
for (;;) {
DWORD availableBytesPlayback = 0;
DWORD physicalPlayCursorInBytes;
DWORD physicalWriteCursorInBytes;
hr = ma_IDirectSoundBuffer_GetCurrentPosition((ma_IDirectSoundBuffer*)pDevice->dsound.pPlaybackBuffer, &physicalPlayCursorInBytes, &physicalWriteCursorInBytes);
if (FAILED(hr)) {
break;
}
if (physicalPlayCursorInBytes < prevPlayCursorInBytesPlayback) {
physicalPlayCursorLoopFlagPlayback = !physicalPlayCursorLoopFlagPlayback;
}
prevPlayCursorInBytesPlayback = physicalPlayCursorInBytes;
if (physicalPlayCursorLoopFlagPlayback == virtualWriteCursorLoopFlagPlayback) {
/* Same loop iteration. The available bytes wraps all the way around from the virtual write cursor to the physical play cursor. */
if (physicalPlayCursorInBytes <= virtualWriteCursorInBytesPlayback) {
availableBytesPlayback = (pDevice->playback.internalPeriodSizeInFrames*pDevice->playback.internalPeriods*bpfDevicePlayback) - virtualWriteCursorInBytesPlayback;
availableBytesPlayback += physicalPlayCursorInBytes; /* Wrap around. */
} else {
break;
}
} else {
/* Different loop iterations. The available bytes only goes from the virtual write cursor to the physical play cursor. */
if (physicalPlayCursorInBytes >= virtualWriteCursorInBytesPlayback) {
availableBytesPlayback = physicalPlayCursorInBytes - virtualWriteCursorInBytesPlayback;
} else {
break;
}
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
GUID ManufacturerGuid;
GUID ProductGuid;
GUID NameGuid;
} MA_WAVEINCAPS2A;
typedef UINT (WINAPI * MA_PFN_waveOutGetNumDevs)(void);
typedef MMRESULT (WINAPI * MA_PFN_waveOutGetDevCapsA)(ma_uintptr uDeviceID, LPWAVEOUTCAPSA pwoc, UINT cbwoc);
typedef MMRESULT (WINAPI * MA_PFN_waveOutOpen)(LPHWAVEOUT phwo, UINT uDeviceID, LPCWAVEFORMATEX pwfx, DWORD_PTR dwCallback, DWORD_PTR dwInstance, DWORD fdwOpen);
typedef MMRESULT (WINAPI * MA_PFN_waveOutClose)(HWAVEOUT hwo);
typedef MMRESULT (WINAPI * MA_PFN_waveOutPrepareHeader)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh);
typedef MMRESULT (WINAPI * MA_PFN_waveOutUnprepareHeader)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh);
typedef MMRESULT (WINAPI * MA_PFN_waveOutWrite)(HWAVEOUT hwo, LPWAVEHDR pwh, UINT cbwh);
typedef MMRESULT (WINAPI * MA_PFN_waveOutReset)(HWAVEOUT hwo);
typedef UINT (WINAPI * MA_PFN_waveInGetNumDevs)(void);
typedef MMRESULT (WINAPI * MA_PFN_waveInGetDevCapsA)(ma_uintptr uDeviceID, LPWAVEINCAPSA pwic, UINT cbwic);
typedef MMRESULT (WINAPI * MA_PFN_waveInOpen)(LPHWAVEIN phwi, UINT uDeviceID, LPCWAVEFORMATEX pwfx, DWORD_PTR dwCallback, DWORD_PTR dwInstance, DWORD fdwOpen);
typedef MMRESULT (WINAPI * MA_PFN_waveInClose)(HWAVEIN hwi);
typedef MMRESULT (WINAPI * MA_PFN_waveInPrepareHeader)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh);
typedef MMRESULT (WINAPI * MA_PFN_waveInUnprepareHeader)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh);
typedef MMRESULT (WINAPI * MA_PFN_waveInAddBuffer)(HWAVEIN hwi, LPWAVEHDR pwh, UINT cbwh);
typedef MMRESULT (WINAPI * MA_PFN_waveInStart)(HWAVEIN hwi);
typedef MMRESULT (WINAPI * MA_PFN_waveInReset)(HWAVEIN hwi);
static ma_result ma_result_from_MMRESULT(MMRESULT resultMM)
{
switch (resultMM) {
case MMSYSERR_NOERROR: return MA_SUCCESS;
case MMSYSERR_BADDEVICEID: return MA_INVALID_ARGS;
case MMSYSERR_INVALHANDLE: return MA_INVALID_ARGS;
case MMSYSERR_NOMEM: return MA_OUT_OF_MEMORY;
case MMSYSERR_INVALFLAG: return MA_INVALID_ARGS;
case MMSYSERR_INVALPARAM: return MA_INVALID_ARGS;
case MMSYSERR_HANDLEBUSY: return MA_BUSY;
case MMSYSERR_ERROR: return MA_ERROR;
default: return MA_ERROR;
}
}
static char* ma_find_last_character(char* str, char ch)
{
char* last;
if (str == NULL) {
return NULL;
}
last = NULL;
while (*str != '\0') {
if (*str == ch) {
last = str;
}
str += 1;
}
return last;
}
static ma_uint32 ma_get_period_size_in_bytes(ma_uint32 periodSizeInFrames, ma_format format, ma_uint32 channels)
{
return periodSizeInFrames * ma_get_bytes_per_frame(format, channels);
}
/*
Our own "WAVECAPS" structure that contains generic information shared between WAVEOUTCAPS2 and WAVEINCAPS2 so
we can do things generically and typesafely. Names are being kept the same for consistency.
*/
typedef struct
{
CHAR szPname[MAXPNAMELEN];
DWORD dwFormats;
WORD wChannels;
GUID NameGuid;
} MA_WAVECAPSA;
static ma_result ma_get_best_info_from_formats_flags__winmm(DWORD dwFormats, WORD channels, WORD* pBitsPerSample, DWORD* pSampleRate)
{
WORD bitsPerSample = 0;
DWORD sampleRate = 0;
if (pBitsPerSample) {
*pBitsPerSample = 0;
}
if (pSampleRate) {
*pSampleRate = 0;
}
if (channels == 1) {
bitsPerSample = 16;
if ((dwFormats & WAVE_FORMAT_48M16) != 0) {
sampleRate = 48000;
} else if ((dwFormats & WAVE_FORMAT_44M16) != 0) {
sampleRate = 44100;
} else if ((dwFormats & WAVE_FORMAT_2M16) != 0) {
sampleRate = 22050;
} else if ((dwFormats & WAVE_FORMAT_1M16) != 0) {
sampleRate = 11025;
} else if ((dwFormats & WAVE_FORMAT_96M16) != 0) {
sampleRate = 96000;
} else {
bitsPerSample = 8;
if ((dwFormats & WAVE_FORMAT_48M08) != 0) {
sampleRate = 48000;
} else if ((dwFormats & WAVE_FORMAT_44M08) != 0) {
sampleRate = 44100;
} else if ((dwFormats & WAVE_FORMAT_2M08) != 0) {
sampleRate = 22050;
} else if ((dwFormats & WAVE_FORMAT_1M08) != 0) {
sampleRate = 11025;
} else if ((dwFormats & WAVE_FORMAT_96M08) != 0) {
sampleRate = 96000;
} else {
return MA_FORMAT_NOT_SUPPORTED;
}
}
} else {
bitsPerSample = 16;
if ((dwFormats & WAVE_FORMAT_48S16) != 0) {
sampleRate = 48000;
} else if ((dwFormats & WAVE_FORMAT_44S16) != 0) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
winMMDeviceID = 0;
if (pDeviceID != NULL) {
winMMDeviceID = (UINT)pDeviceID->winmm;
}
pDeviceInfo->id.winmm = winMMDeviceID;
/* The first ID is the default device. */
if (winMMDeviceID == 0) {
pDeviceInfo->isDefault = MA_TRUE;
}
if (deviceType == ma_device_type_playback) {
MMRESULT result;
MA_WAVEOUTCAPS2A caps;
MA_ZERO_OBJECT(&caps);
result = ((MA_PFN_waveOutGetDevCapsA)pContext->winmm.waveOutGetDevCapsA)(winMMDeviceID, (WAVEOUTCAPSA*)&caps, sizeof(caps));
if (result == MMSYSERR_NOERROR) {
return ma_context_get_device_info_from_WAVEOUTCAPS2(pContext, &caps, pDeviceInfo);
}
} else {
MMRESULT result;
MA_WAVEINCAPS2A caps;
MA_ZERO_OBJECT(&caps);
result = ((MA_PFN_waveInGetDevCapsA)pContext->winmm.waveInGetDevCapsA)(winMMDeviceID, (WAVEINCAPSA*)&caps, sizeof(caps));
if (result == MMSYSERR_NOERROR) {
return ma_context_get_device_info_from_WAVEINCAPS2(pContext, &caps, pDeviceInfo);
}
}
return MA_NO_DEVICE;
}
static ma_result ma_device_uninit__winmm(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
((MA_PFN_waveInClose)pDevice->pContext->winmm.waveInClose)((HWAVEIN)pDevice->winmm.hDeviceCapture);
CloseHandle((HANDLE)pDevice->winmm.hEventCapture);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
((MA_PFN_waveOutReset)pDevice->pContext->winmm.waveOutReset)((HWAVEOUT)pDevice->winmm.hDevicePlayback);
((MA_PFN_waveOutClose)pDevice->pContext->winmm.waveOutClose)((HWAVEOUT)pDevice->winmm.hDevicePlayback);
CloseHandle((HANDLE)pDevice->winmm.hEventPlayback);
}
ma_free(pDevice->winmm._pHeapData, &pDevice->pContext->allocationCallbacks);
MA_ZERO_OBJECT(&pDevice->winmm); /* Safety. */
return MA_SUCCESS;
}
static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__winmm(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
{
/* WinMM has a minimum period size of 40ms. */
ma_uint32 minPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(40, nativeSampleRate);
ma_uint32 periodSizeInFrames;
periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, nativeSampleRate, performanceProfile);
if (periodSizeInFrames < minPeriodSizeInFrames) {
periodSizeInFrames = minPeriodSizeInFrames;
}
return periodSizeInFrames;
}
static ma_result ma_device_init__winmm(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
const char* errorMsg = "";
ma_result errorCode = MA_ERROR;
ma_result result = MA_SUCCESS;
ma_uint32 heapSize;
UINT winMMDeviceIDPlayback = 0;
UINT winMMDeviceIDCapture = 0;
MA_ASSERT(pDevice != NULL);
MA_ZERO_OBJECT(&pDevice->winmm);
if (pConfig->deviceType == ma_device_type_loopback) {
return MA_DEVICE_TYPE_NOT_SUPPORTED;
}
/* No exlusive mode with WinMM. */
if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
return MA_SHARE_MODE_NOT_SUPPORTED;
}
if (pDescriptorPlayback->pDeviceID != NULL) {
winMMDeviceIDPlayback = (UINT)pDescriptorPlayback->pDeviceID->winmm;
}
if (pDescriptorCapture->pDeviceID != NULL) {
winMMDeviceIDCapture = (UINT)pDescriptorCapture->pDeviceID->winmm;
}
/* The capture device needs to be initialized first. */
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
WAVEINCAPSA caps;
WAVEFORMATEX wf;
MMRESULT resultMM;
/* We use an event to know when a new fragment needs to be enqueued. */
pDevice->winmm.hEventCapture = (ma_handle)CreateEventW(NULL, TRUE, TRUE, NULL);
if (pDevice->winmm.hEventCapture == NULL) {
errorMsg = "[WinMM] Failed to create event for fragment enqueing for the capture device.", errorCode = ma_result_from_GetLastError(GetLastError());
goto on_error;
}
/* The format should be based on the device's actual format. */
if (((MA_PFN_waveInGetDevCapsA)pDevice->pContext->winmm.waveInGetDevCapsA)(winMMDeviceIDCapture, &caps, sizeof(caps)) != MMSYSERR_NOERROR) {
errorMsg = "[WinMM] Failed to retrieve internal device caps.", errorCode = MA_FORMAT_NOT_SUPPORTED;
goto on_error;
}
result = ma_formats_flags_to_WAVEFORMATEX__winmm(caps.dwFormats, caps.wChannels, &wf);
if (result != MA_SUCCESS) {
errorMsg = "[WinMM] Could not find appropriate format for internal device.", errorCode = result;
goto on_error;
}
resultMM = ((MA_PFN_waveInOpen)pDevice->pContext->winmm.waveInOpen)((LPHWAVEIN)&pDevice->winmm.hDeviceCapture, winMMDeviceIDCapture, &wf, (DWORD_PTR)pDevice->winmm.hEventCapture, (DWORD_PTR)pDevice, CALLBACK_EVENT | WAVE_ALLOWSYNC);
if (resultMM != MMSYSERR_NOERROR) {
errorMsg = "[WinMM] Failed to open capture device.", errorCode = MA_FAILED_TO_OPEN_BACKEND_DEVICE;
goto on_error;
}
pDescriptorCapture->format = ma_format_from_WAVEFORMATEX(&wf);
pDescriptorCapture->channels = wf.nChannels;
pDescriptorCapture->sampleRate = wf.nSamplesPerSec;
ma_channel_map_init_standard(ma_standard_channel_map_microsoft, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorCapture->channels);
pDescriptorCapture->periodCount = pDescriptorCapture->periodCount;
pDescriptorCapture->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__winmm(pDescriptorCapture, pDescriptorCapture->sampleRate, pConfig->performanceProfile);
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
WAVEOUTCAPSA caps;
WAVEFORMATEX wf;
MMRESULT resultMM;
/* We use an event to know when a new fragment needs to be enqueued. */
pDevice->winmm.hEventPlayback = (ma_handle)CreateEventW(NULL, TRUE, TRUE, NULL);
if (pDevice->winmm.hEventPlayback == NULL) {
errorMsg = "[WinMM] Failed to create event for fragment enqueing for the playback device.", errorCode = ma_result_from_GetLastError(GetLastError());
goto on_error;
}
/* The format should be based on the device's actual format. */
if (((MA_PFN_waveOutGetDevCapsA)pDevice->pContext->winmm.waveOutGetDevCapsA)(winMMDeviceIDPlayback, &caps, sizeof(caps)) != MMSYSERR_NOERROR) {
errorMsg = "[WinMM] Failed to retrieve internal device caps.", errorCode = MA_FORMAT_NOT_SUPPORTED;
goto on_error;
}
result = ma_formats_flags_to_WAVEFORMATEX__winmm(caps.dwFormats, caps.wChannels, &wf);
if (result != MA_SUCCESS) {
errorMsg = "[WinMM] Could not find appropriate format for internal device.", errorCode = result;
goto on_error;
}
resultMM = ((MA_PFN_waveOutOpen)pDevice->pContext->winmm.waveOutOpen)((LPHWAVEOUT)&pDevice->winmm.hDevicePlayback, winMMDeviceIDPlayback, &wf, (DWORD_PTR)pDevice->winmm.hEventPlayback, (DWORD_PTR)pDevice, CALLBACK_EVENT | WAVE_ALLOWSYNC);
if (resultMM != MMSYSERR_NOERROR) {
errorMsg = "[WinMM] Failed to open playback device.", errorCode = MA_FAILED_TO_OPEN_BACKEND_DEVICE;
goto on_error;
}
pDescriptorPlayback->format = ma_format_from_WAVEFORMATEX(&wf);
pDescriptorPlayback->channels = wf.nChannels;
pDescriptorPlayback->sampleRate = wf.nSamplesPerSec;
ma_channel_map_init_standard(ma_standard_channel_map_microsoft, pDescriptorPlayback->channelMap, ma_countof(pDescriptorPlayback->channelMap), pDescriptorPlayback->channels);
pDescriptorPlayback->periodCount = pDescriptorPlayback->periodCount;
pDescriptorPlayback->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__winmm(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile);
}
/*
The heap allocated data is allocated like so:
[Capture WAVEHDRs][Playback WAVEHDRs][Capture Intermediary Buffer][Playback Intermediary Buffer]
*/
heapSize = 0;
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
heapSize += sizeof(WAVEHDR)*pDescriptorCapture->periodCount + (pDescriptorCapture->periodSizeInFrames * pDescriptorCapture->periodCount * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels));
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
heapSize += sizeof(WAVEHDR)*pDescriptorPlayback->periodCount + (pDescriptorPlayback->periodSizeInFrames * pDescriptorPlayback->periodCount * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels));
}
pDevice->winmm._pHeapData = (ma_uint8*)ma_calloc(heapSize, &pDevice->pContext->allocationCallbacks);
if (pDevice->winmm._pHeapData == NULL) {
errorMsg = "[WinMM] Failed to allocate memory for the intermediary buffer.", errorCode = MA_OUT_OF_MEMORY;
goto on_error;
}
MA_ZERO_MEMORY(pDevice->winmm._pHeapData, heapSize);
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
ma_uint32 iPeriod;
if (pConfig->deviceType == ma_device_type_capture) {
pDevice->winmm.pWAVEHDRCapture = pDevice->winmm._pHeapData;
pDevice->winmm.pIntermediaryBufferCapture = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDescriptorCapture->periodCount));
} else {
pDevice->winmm.pWAVEHDRCapture = pDevice->winmm._pHeapData;
pDevice->winmm.pIntermediaryBufferCapture = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDescriptorCapture->periodCount + pDescriptorPlayback->periodCount));
}
/* Prepare headers. */
for (iPeriod = 0; iPeriod < pDescriptorCapture->periodCount; ++iPeriod) {
ma_uint32 periodSizeInBytes = ma_get_period_size_in_bytes(pDescriptorCapture->periodSizeInFrames, pDescriptorCapture->format, pDescriptorCapture->channels);
((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].lpData = (LPSTR)(pDevice->winmm.pIntermediaryBufferCapture + (periodSizeInBytes*iPeriod));
((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwBufferLength = periodSizeInBytes;
((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwFlags = 0L;
((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwLoops = 0L;
((MA_PFN_waveInPrepareHeader)pDevice->pContext->winmm.waveInPrepareHeader)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR));
/*
The user data of the WAVEHDR structure is a single flag the controls whether or not it is ready for writing. Consider it to be named "isLocked". A value of 0 means
it's unlocked and available for writing. A value of 1 means it's locked.
*/
((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod].dwUser = 0;
}
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
ma_uint32 iPeriod;
if (pConfig->deviceType == ma_device_type_playback) {
pDevice->winmm.pWAVEHDRPlayback = pDevice->winmm._pHeapData;
pDevice->winmm.pIntermediaryBufferPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*pDescriptorPlayback->periodCount);
} else {
pDevice->winmm.pWAVEHDRPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDescriptorCapture->periodCount));
pDevice->winmm.pIntermediaryBufferPlayback = pDevice->winmm._pHeapData + (sizeof(WAVEHDR)*(pDescriptorCapture->periodCount + pDescriptorPlayback->periodCount)) + (pDescriptorCapture->periodSizeInFrames*pDescriptorCapture->periodCount*ma_g...
}
/* Prepare headers. */
for (iPeriod = 0; iPeriod < pDescriptorPlayback->periodCount; ++iPeriod) {
ma_uint32 periodSizeInBytes = ma_get_period_size_in_bytes(pDescriptorPlayback->periodSizeInFrames, pDescriptorPlayback->format, pDescriptorPlayback->channels);
((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].lpData = (LPSTR)(pDevice->winmm.pIntermediaryBufferPlayback + (periodSizeInBytes*iPeriod));
((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwBufferLength = periodSizeInBytes;
((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwFlags = 0L;
((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwLoops = 0L;
((MA_PFN_waveOutPrepareHeader)pDevice->pContext->winmm.waveOutPrepareHeader)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod], sizeof(WAVEHDR));
/*
The user data of the WAVEHDR structure is a single flag the controls whether or not it is ready for writing. Consider it to be named "isLocked". A value of 0 means
it's unlocked and available for writing. A value of 1 means it's locked.
*/
((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod].dwUser = 0;
}
}
return MA_SUCCESS;
on_error:
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
if (pDevice->winmm.pWAVEHDRCapture != NULL) {
ma_uint32 iPeriod;
for (iPeriod = 0; iPeriod < pDescriptorCapture->periodCount; ++iPeriod) {
((MA_PFN_waveInUnprepareHeader)pDevice->pContext->winmm.waveInUnprepareHeader)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((WAVEHDR*)pDevice->winmm.pWAVEHDRCapture)[iPeriod], sizeof(WAVEHDR));
}
}
((MA_PFN_waveInClose)pDevice->pContext->winmm.waveInClose)((HWAVEIN)pDevice->winmm.hDeviceCapture);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
if (pDevice->winmm.pWAVEHDRCapture != NULL) {
ma_uint32 iPeriod;
for (iPeriod = 0; iPeriod < pDescriptorPlayback->periodCount; ++iPeriod) {
((MA_PFN_waveOutUnprepareHeader)pDevice->pContext->winmm.waveOutUnprepareHeader)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &((WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback)[iPeriod], sizeof(WAVEHDR));
}
}
((MA_PFN_waveOutClose)pDevice->pContext->winmm.waveOutClose)((HWAVEOUT)pDevice->winmm.hDevicePlayback);
}
ma_free(pDevice->winmm._pHeapData, &pDevice->pContext->allocationCallbacks);
if (errorMsg != NULL && errorMsg[0] != '\0') {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "%s", errorMsg);
}
return errorCode;
}
static ma_result ma_device_start__winmm(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
MMRESULT resultMM;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
resultMM = ((MA_PFN_waveInStart)pDevice->pContext->winmm.waveInStart)((HWAVEIN)pDevice->winmm.hDeviceCapture);
if (resultMM != MMSYSERR_NOERROR) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WinMM] Failed to start backend device.");
return ma_result_from_MMRESULT(resultMM);
}
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
/* Don't need to do anything for playback. It'll be started automatically in ma_device_start__winmm(). */
}
return MA_SUCCESS;
}
static ma_result ma_device_stop__winmm(ma_device* pDevice)
{
MMRESULT resultMM;
MA_ASSERT(pDevice != NULL);
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
if (pDevice->winmm.hDeviceCapture == NULL) {
return MA_INVALID_ARGS;
}
resultMM = ((MA_PFN_waveInReset)pDevice->pContext->winmm.waveInReset)((HWAVEIN)pDevice->winmm.hDeviceCapture);
if (resultMM != MMSYSERR_NOERROR) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[WinMM] WARNING: Failed to reset capture device.");
}
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
ma_uint32 iPeriod;
WAVEHDR* pWAVEHDR;
if (pDevice->winmm.hDevicePlayback == NULL) {
return MA_INVALID_ARGS;
}
/* We need to drain the device. To do this we just loop over each header and if it's locked just wait for the event. */
pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback;
for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; iPeriod += 1) {
if (pWAVEHDR[iPeriod].dwUser == 1) { /* 1 = locked. */
if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventPlayback, INFINITE) != WAIT_OBJECT_0) {
break; /* An error occurred so just abandon ship and stop the device without draining. */
}
pWAVEHDR[iPeriod].dwUser = 0;
}
}
resultMM = ((MA_PFN_waveOutReset)pDevice->pContext->winmm.waveOutReset)((HWAVEOUT)pDevice->winmm.hDevicePlayback);
if (resultMM != MMSYSERR_NOERROR) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_WARNING, "[WinMM] WARNING: Failed to reset playback device.");
}
}
return MA_SUCCESS;
}
static ma_result ma_device_write__winmm(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
{
ma_result result = MA_SUCCESS;
MMRESULT resultMM;
ma_uint32 totalFramesWritten;
WAVEHDR* pWAVEHDR;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(pPCMFrames != NULL);
if (pFramesWritten != NULL) {
*pFramesWritten = 0;
}
pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRPlayback;
/* Keep processing as much data as possible. */
totalFramesWritten = 0;
while (totalFramesWritten < frameCount) {
/* If the current header has some space available we need to write part of it. */
if (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser == 0) { /* 0 = unlocked. */
/*
This header has room in it. We copy as much of it as we can. If we end up fully consuming the buffer we need to
write it out and move on to the next iteration.
*/
ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
ma_uint32 framesRemainingInHeader = (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwBufferLength/bpf) - pDevice->winmm.headerFramesConsumedPlayback;
ma_uint32 framesToCopy = ma_min(framesRemainingInHeader, (frameCount - totalFramesWritten));
const void* pSrc = ma_offset_ptr(pPCMFrames, totalFramesWritten*bpf);
void* pDst = ma_offset_ptr(pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].lpData, pDevice->winmm.headerFramesConsumedPlayback*bpf);
MA_COPY_MEMORY(pDst, pSrc, framesToCopy*bpf);
pDevice->winmm.headerFramesConsumedPlayback += framesToCopy;
totalFramesWritten += framesToCopy;
/* If we've consumed the buffer entirely we need to write it out to the device. */
if (pDevice->winmm.headerFramesConsumedPlayback == (pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwBufferLength/bpf)) {
pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser = 1; /* 1 = locked. */
pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwFlags &= ~WHDR_DONE; /* <-- Need to make sure the WHDR_DONE flag is unset. */
/* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */
ResetEvent((HANDLE)pDevice->winmm.hEventPlayback);
/* The device will be started here. */
resultMM = ((MA_PFN_waveOutWrite)pDevice->pContext->winmm.waveOutWrite)((HWAVEOUT)pDevice->winmm.hDevicePlayback, &pWAVEHDR[pDevice->winmm.iNextHeaderPlayback], sizeof(WAVEHDR));
if (resultMM != MMSYSERR_NOERROR) {
result = ma_result_from_MMRESULT(resultMM);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WinMM] waveOutWrite() failed.");
break;
}
/* Make sure we move to the next header. */
pDevice->winmm.iNextHeaderPlayback = (pDevice->winmm.iNextHeaderPlayback + 1) % pDevice->playback.internalPeriods;
pDevice->winmm.headerFramesConsumedPlayback = 0;
}
/* If at this point we have consumed the entire input buffer we can return. */
MA_ASSERT(totalFramesWritten <= frameCount);
if (totalFramesWritten == frameCount) {
break;
}
/* Getting here means there's more to process. */
continue;
}
/* Getting here means there isn't enough room in the buffer and we need to wait for one to become available. */
if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventPlayback, INFINITE) != WAIT_OBJECT_0) {
result = MA_ERROR;
break;
}
/* Something happened. If the next buffer has been marked as done we need to reset a bit of state. */
if ((pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwFlags & WHDR_DONE) != 0) {
pWAVEHDR[pDevice->winmm.iNextHeaderPlayback].dwUser = 0; /* 0 = unlocked (make it available for writing). */
pDevice->winmm.headerFramesConsumedPlayback = 0;
}
/* If the device has been stopped we need to break. */
if (ma_device_get_state(pDevice) != ma_device_state_started) {
break;
}
}
if (pFramesWritten != NULL) {
*pFramesWritten = totalFramesWritten;
}
return result;
}
static ma_result ma_device_read__winmm(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
{
ma_result result = MA_SUCCESS;
MMRESULT resultMM;
ma_uint32 totalFramesRead;
WAVEHDR* pWAVEHDR;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(pPCMFrames != NULL);
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
pWAVEHDR = (WAVEHDR*)pDevice->winmm.pWAVEHDRCapture;
/* Keep processing as much data as possible. */
totalFramesRead = 0;
while (totalFramesRead < frameCount) {
/* If the current header has some space available we need to write part of it. */
if (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser == 0) { /* 0 = unlocked. */
/* The buffer is available for reading. If we fully consume it we need to add it back to the buffer. */
ma_uint32 bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
ma_uint32 framesRemainingInHeader = (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwBufferLength/bpf) - pDevice->winmm.headerFramesConsumedCapture;
ma_uint32 framesToCopy = ma_min(framesRemainingInHeader, (frameCount - totalFramesRead));
const void* pSrc = ma_offset_ptr(pWAVEHDR[pDevice->winmm.iNextHeaderCapture].lpData, pDevice->winmm.headerFramesConsumedCapture*bpf);
void* pDst = ma_offset_ptr(pPCMFrames, totalFramesRead*bpf);
MA_COPY_MEMORY(pDst, pSrc, framesToCopy*bpf);
pDevice->winmm.headerFramesConsumedCapture += framesToCopy;
totalFramesRead += framesToCopy;
/* If we've consumed the buffer entirely we need to add it back to the device. */
if (pDevice->winmm.headerFramesConsumedCapture == (pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwBufferLength/bpf)) {
pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser = 1; /* 1 = locked. */
pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwFlags &= ~WHDR_DONE; /* <-- Need to make sure the WHDR_DONE flag is unset. */
/* Make sure the event is reset to a non-signaled state to ensure we don't prematurely return from WaitForSingleObject(). */
ResetEvent((HANDLE)pDevice->winmm.hEventCapture);
/* The device will be started here. */
resultMM = ((MA_PFN_waveInAddBuffer)pDevice->pContext->winmm.waveInAddBuffer)((HWAVEIN)pDevice->winmm.hDeviceCapture, &((LPWAVEHDR)pDevice->winmm.pWAVEHDRCapture)[pDevice->winmm.iNextHeaderCapture], sizeof(WAVEHDR));
if (resultMM != MMSYSERR_NOERROR) {
result = ma_result_from_MMRESULT(resultMM);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[WinMM] waveInAddBuffer() failed.");
break;
}
/* Make sure we move to the next header. */
pDevice->winmm.iNextHeaderCapture = (pDevice->winmm.iNextHeaderCapture + 1) % pDevice->capture.internalPeriods;
pDevice->winmm.headerFramesConsumedCapture = 0;
}
/* If at this point we have filled the entire input buffer we can return. */
MA_ASSERT(totalFramesRead <= frameCount);
if (totalFramesRead == frameCount) {
break;
}
/* Getting here means there's more to process. */
continue;
}
/* Getting here means there isn't enough any data left to send to the client which means we need to wait for more. */
if (WaitForSingleObject((HANDLE)pDevice->winmm.hEventCapture, INFINITE) != WAIT_OBJECT_0) {
result = MA_ERROR;
break;
}
/* Something happened. If the next buffer has been marked as done we need to reset a bit of state. */
if ((pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwFlags & WHDR_DONE) != 0) {
pWAVEHDR[pDevice->winmm.iNextHeaderCapture].dwUser = 0; /* 0 = unlocked (make it available for reading). */
pDevice->winmm.headerFramesConsumedCapture = 0;
}
/* If the device has been stopped we need to break. */
if (ma_device_get_state(pDevice) != ma_device_state_started) {
break;
}
}
if (pFramesRead != NULL) {
*pFramesRead = totalFramesRead;
}
return result;
}
static ma_result ma_context_uninit__winmm(ma_context* pContext)
{
MA_ASSERT(pContext != NULL);
MA_ASSERT(pContext->backend == ma_backend_winmm);
ma_dlclose(pContext, pContext->winmm.hWinMM);
return MA_SUCCESS;
}
static ma_result ma_context_init__winmm(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
{
MA_ASSERT(pContext != NULL);
(void)pConfig;
pContext->winmm.hWinMM = ma_dlopen(pContext, "winmm.dll");
if (pContext->winmm.hWinMM == NULL) {
return MA_NO_BACKEND;
}
pContext->winmm.waveOutGetNumDevs = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutGetNumDevs");
pContext->winmm.waveOutGetDevCapsA = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutGetDevCapsA");
pContext->winmm.waveOutOpen = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutOpen");
pContext->winmm.waveOutClose = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutClose");
pContext->winmm.waveOutPrepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutPrepareHeader");
pContext->winmm.waveOutUnprepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutUnprepareHeader");
pContext->winmm.waveOutWrite = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutWrite");
pContext->winmm.waveOutReset = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveOutReset");
pContext->winmm.waveInGetNumDevs = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInGetNumDevs");
pContext->winmm.waveInGetDevCapsA = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInGetDevCapsA");
pContext->winmm.waveInOpen = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInOpen");
pContext->winmm.waveInClose = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInClose");
pContext->winmm.waveInPrepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInPrepareHeader");
pContext->winmm.waveInUnprepareHeader = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInUnprepareHeader");
pContext->winmm.waveInAddBuffer = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInAddBuffer");
pContext->winmm.waveInStart = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInStart");
pContext->winmm.waveInReset = ma_dlsym(pContext, pContext->winmm.hWinMM, "waveInReset");
pCallbacks->onContextInit = ma_context_init__winmm;
pCallbacks->onContextUninit = ma_context_uninit__winmm;
pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__winmm;
pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__winmm;
pCallbacks->onDeviceInit = ma_device_init__winmm;
pCallbacks->onDeviceUninit = ma_device_uninit__winmm;
pCallbacks->onDeviceStart = ma_device_start__winmm;
pCallbacks->onDeviceStop = ma_device_stop__winmm;
pCallbacks->onDeviceRead = ma_device_read__winmm;
pCallbacks->onDeviceWrite = ma_device_write__winmm;
pCallbacks->onDeviceDataLoop = NULL; /* This is a blocking read-write API, so this can be NULL since miniaudio will manage the audio thread for us. */
return MA_SUCCESS;
}
#endif
/******************************************************************************
ALSA Backend
******************************************************************************/
#ifdef MA_HAS_ALSA
#include <poll.h> /* poll(), struct pollfd */
#include <sys/eventfd.h> /* eventfd() */
#ifdef MA_NO_RUNTIME_LINKING
/* asoundlib.h marks some functions with "inline" which isn't always supported. Need to emulate it. */
#if !defined(__cplusplus)
#if defined(__STRICT_ANSI__)
#if !defined(inline)
#define inline __inline__ __attribute__((always_inline))
#define MA_INLINE_DEFINED
#endif
#endif
#endif
#include <alsa/asoundlib.h>
#if defined(MA_INLINE_DEFINED)
#undef inline
#undef MA_INLINE_DEFINED
#endif
typedef snd_pcm_uframes_t ma_snd_pcm_uframes_t;
typedef snd_pcm_sframes_t ma_snd_pcm_sframes_t;
typedef snd_pcm_stream_t ma_snd_pcm_stream_t;
typedef snd_pcm_format_t ma_snd_pcm_format_t;
typedef snd_pcm_access_t ma_snd_pcm_access_t;
typedef snd_pcm_t ma_snd_pcm_t;
typedef snd_pcm_hw_params_t ma_snd_pcm_hw_params_t;
typedef snd_pcm_sw_params_t ma_snd_pcm_sw_params_t;
typedef snd_pcm_format_mask_t ma_snd_pcm_format_mask_t;
typedef snd_pcm_info_t ma_snd_pcm_info_t;
typedef snd_pcm_channel_area_t ma_snd_pcm_channel_area_t;
typedef snd_pcm_chmap_t ma_snd_pcm_chmap_t;
typedef snd_pcm_state_t ma_snd_pcm_state_t;
/* snd_pcm_stream_t */
#define MA_SND_PCM_STREAM_PLAYBACK SND_PCM_STREAM_PLAYBACK
#define MA_SND_PCM_STREAM_CAPTURE SND_PCM_STREAM_CAPTURE
/* snd_pcm_format_t */
#define MA_SND_PCM_FORMAT_UNKNOWN SND_PCM_FORMAT_UNKNOWN
#define MA_SND_PCM_FORMAT_U8 SND_PCM_FORMAT_U8
#define MA_SND_PCM_FORMAT_S16_LE SND_PCM_FORMAT_S16_LE
#define MA_SND_PCM_FORMAT_S16_BE SND_PCM_FORMAT_S16_BE
#define MA_SND_PCM_FORMAT_S24_LE SND_PCM_FORMAT_S24_LE
#define MA_SND_PCM_FORMAT_S24_BE SND_PCM_FORMAT_S24_BE
#define MA_SND_PCM_FORMAT_S32_LE SND_PCM_FORMAT_S32_LE
#define MA_SND_PCM_FORMAT_S32_BE SND_PCM_FORMAT_S32_BE
#define MA_SND_PCM_FORMAT_FLOAT_LE SND_PCM_FORMAT_FLOAT_LE
#define MA_SND_PCM_FORMAT_FLOAT_BE SND_PCM_FORMAT_FLOAT_BE
#define MA_SND_PCM_FORMAT_FLOAT64_LE SND_PCM_FORMAT_FLOAT64_LE
#define MA_SND_PCM_FORMAT_FLOAT64_BE SND_PCM_FORMAT_FLOAT64_BE
#define MA_SND_PCM_FORMAT_MU_LAW SND_PCM_FORMAT_MU_LAW
#define MA_SND_PCM_FORMAT_A_LAW SND_PCM_FORMAT_A_LAW
#define MA_SND_PCM_FORMAT_S24_3LE SND_PCM_FORMAT_S24_3LE
#define MA_SND_PCM_FORMAT_S24_3BE SND_PCM_FORMAT_S24_3BE
/* ma_snd_pcm_access_t */
#define MA_SND_PCM_ACCESS_MMAP_INTERLEAVED SND_PCM_ACCESS_MMAP_INTERLEAVED
#define MA_SND_PCM_ACCESS_MMAP_NONINTERLEAVED SND_PCM_ACCESS_MMAP_NONINTERLEAVED
#define MA_SND_PCM_ACCESS_MMAP_COMPLEX SND_PCM_ACCESS_MMAP_COMPLEX
#define MA_SND_PCM_ACCESS_RW_INTERLEAVED SND_PCM_ACCESS_RW_INTERLEAVED
#define MA_SND_PCM_ACCESS_RW_NONINTERLEAVED SND_PCM_ACCESS_RW_NONINTERLEAVED
/* Channel positions. */
#define MA_SND_CHMAP_UNKNOWN SND_CHMAP_UNKNOWN
#define MA_SND_CHMAP_NA SND_CHMAP_NA
#define MA_SND_CHMAP_MONO SND_CHMAP_MONO
#define MA_SND_CHMAP_FL SND_CHMAP_FL
#define MA_SND_CHMAP_FR SND_CHMAP_FR
#define MA_SND_CHMAP_RL SND_CHMAP_RL
#define MA_SND_CHMAP_RR SND_CHMAP_RR
#define MA_SND_CHMAP_FC SND_CHMAP_FC
#define MA_SND_CHMAP_LFE SND_CHMAP_LFE
#define MA_SND_CHMAP_SL SND_CHMAP_SL
#define MA_SND_CHMAP_SR SND_CHMAP_SR
#define MA_SND_CHMAP_RC SND_CHMAP_RC
#define MA_SND_CHMAP_FLC SND_CHMAP_FLC
#define MA_SND_CHMAP_FRC SND_CHMAP_FRC
#define MA_SND_CHMAP_RLC SND_CHMAP_RLC
#define MA_SND_CHMAP_RRC SND_CHMAP_RRC
#define MA_SND_CHMAP_FLW SND_CHMAP_FLW
#define MA_SND_CHMAP_FRW SND_CHMAP_FRW
#define MA_SND_CHMAP_FLH SND_CHMAP_FLH
#define MA_SND_CHMAP_FCH SND_CHMAP_FCH
#define MA_SND_CHMAP_FRH SND_CHMAP_FRH
#define MA_SND_CHMAP_TC SND_CHMAP_TC
#define MA_SND_CHMAP_TFL SND_CHMAP_TFL
#define MA_SND_CHMAP_TFR SND_CHMAP_TFR
#define MA_SND_CHMAP_TFC SND_CHMAP_TFC
#define MA_SND_CHMAP_TRL SND_CHMAP_TRL
#define MA_SND_CHMAP_TRR SND_CHMAP_TRR
#define MA_SND_CHMAP_TRC SND_CHMAP_TRC
#define MA_SND_CHMAP_TFLC SND_CHMAP_TFLC
#define MA_SND_CHMAP_TFRC SND_CHMAP_TFRC
#define MA_SND_CHMAP_TSL SND_CHMAP_TSL
#define MA_SND_CHMAP_TSR SND_CHMAP_TSR
#define MA_SND_CHMAP_LLFE SND_CHMAP_LLFE
#define MA_SND_CHMAP_RLFE SND_CHMAP_RLFE
#define MA_SND_CHMAP_BC SND_CHMAP_BC
#define MA_SND_CHMAP_BLC SND_CHMAP_BLC
#define MA_SND_CHMAP_BRC SND_CHMAP_BRC
/* Open mode flags. */
#define MA_SND_PCM_NO_AUTO_RESAMPLE SND_PCM_NO_AUTO_RESAMPLE
#define MA_SND_PCM_NO_AUTO_CHANNELS SND_PCM_NO_AUTO_CHANNELS
#define MA_SND_PCM_NO_AUTO_FORMAT SND_PCM_NO_AUTO_FORMAT
#else
#include <errno.h> /* For EPIPE, etc. */
typedef unsigned long ma_snd_pcm_uframes_t;
typedef long ma_snd_pcm_sframes_t;
typedef int ma_snd_pcm_stream_t;
typedef int ma_snd_pcm_format_t;
typedef int ma_snd_pcm_access_t;
typedef int ma_snd_pcm_state_t;
typedef struct ma_snd_pcm_t ma_snd_pcm_t;
typedef struct ma_snd_pcm_hw_params_t ma_snd_pcm_hw_params_t;
typedef struct ma_snd_pcm_sw_params_t ma_snd_pcm_sw_params_t;
typedef struct ma_snd_pcm_format_mask_t ma_snd_pcm_format_mask_t;
typedef struct ma_snd_pcm_info_t ma_snd_pcm_info_t;
typedef struct
{
void* addr;
unsigned int first;
unsigned int step;
} ma_snd_pcm_channel_area_t;
typedef struct
{
unsigned int channels;
unsigned int pos[1];
} ma_snd_pcm_chmap_t;
/* snd_pcm_state_t */
#define MA_SND_PCM_STATE_OPEN 0
#define MA_SND_PCM_STATE_SETUP 1
#define MA_SND_PCM_STATE_PREPARED 2
#define MA_SND_PCM_STATE_RUNNING 3
#define MA_SND_PCM_STATE_XRUN 4
#define MA_SND_PCM_STATE_DRAINING 5
#define MA_SND_PCM_STATE_PAUSED 6
#define MA_SND_PCM_STATE_SUSPENDED 7
#define MA_SND_PCM_STATE_DISCONNECTED 8
/* snd_pcm_stream_t */
#define MA_SND_PCM_STREAM_PLAYBACK 0
#define MA_SND_PCM_STREAM_CAPTURE 1
/* snd_pcm_format_t */
#define MA_SND_PCM_FORMAT_UNKNOWN -1
#define MA_SND_PCM_FORMAT_U8 1
#define MA_SND_PCM_FORMAT_S16_LE 2
#define MA_SND_PCM_FORMAT_S16_BE 3
#define MA_SND_PCM_FORMAT_S24_LE 6
#define MA_SND_PCM_FORMAT_S24_BE 7
#define MA_SND_PCM_FORMAT_S32_LE 10
#define MA_SND_PCM_FORMAT_S32_BE 11
#define MA_SND_PCM_FORMAT_FLOAT_LE 14
#define MA_SND_PCM_FORMAT_FLOAT_BE 15
#define MA_SND_PCM_FORMAT_FLOAT64_LE 16
#define MA_SND_PCM_FORMAT_FLOAT64_BE 17
#define MA_SND_PCM_FORMAT_MU_LAW 20
#define MA_SND_PCM_FORMAT_A_LAW 21
#define MA_SND_PCM_FORMAT_S24_3LE 32
#define MA_SND_PCM_FORMAT_S24_3BE 33
/* snd_pcm_access_t */
#define MA_SND_PCM_ACCESS_MMAP_INTERLEAVED 0
#define MA_SND_PCM_ACCESS_MMAP_NONINTERLEAVED 1
#define MA_SND_PCM_ACCESS_MMAP_COMPLEX 2
#define MA_SND_PCM_ACCESS_RW_INTERLEAVED 3
#define MA_SND_PCM_ACCESS_RW_NONINTERLEAVED 4
/* Channel positions. */
#define MA_SND_CHMAP_UNKNOWN 0
#define MA_SND_CHMAP_NA 1
#define MA_SND_CHMAP_MONO 2
#define MA_SND_CHMAP_FL 3
#define MA_SND_CHMAP_FR 4
#define MA_SND_CHMAP_RL 5
#define MA_SND_CHMAP_RR 6
#define MA_SND_CHMAP_FC 7
#define MA_SND_CHMAP_LFE 8
#define MA_SND_CHMAP_SL 9
#define MA_SND_CHMAP_SR 10
#define MA_SND_CHMAP_RC 11
#define MA_SND_CHMAP_FLC 12
#define MA_SND_CHMAP_FRC 13
#define MA_SND_CHMAP_RLC 14
#define MA_SND_CHMAP_RRC 15
#define MA_SND_CHMAP_FLW 16
#define MA_SND_CHMAP_FRW 17
#define MA_SND_CHMAP_FLH 18
#define MA_SND_CHMAP_FCH 19
#define MA_SND_CHMAP_FRH 20
#define MA_SND_CHMAP_TC 21
#define MA_SND_CHMAP_TFL 22
#define MA_SND_CHMAP_TFR 23
#define MA_SND_CHMAP_TFC 24
#define MA_SND_CHMAP_TRL 25
#define MA_SND_CHMAP_TRR 26
#define MA_SND_CHMAP_TRC 27
#define MA_SND_CHMAP_TFLC 28
#define MA_SND_CHMAP_TFRC 29
#define MA_SND_CHMAP_TSL 30
#define MA_SND_CHMAP_TSR 31
#define MA_SND_CHMAP_LLFE 32
#define MA_SND_CHMAP_RLFE 33
#define MA_SND_CHMAP_BC 34
#define MA_SND_CHMAP_BLC 35
#define MA_SND_CHMAP_BRC 36
/* Open mode flags. */
#define MA_SND_PCM_NO_AUTO_RESAMPLE 0x00010000
#define MA_SND_PCM_NO_AUTO_CHANNELS 0x00020000
#define MA_SND_PCM_NO_AUTO_FORMAT 0x00040000
#endif
typedef int (* ma_snd_pcm_open_proc) (ma_snd_pcm_t **pcm, const char *name, ma_snd_pcm_stream_t stream, int mode);
typedef int (* ma_snd_pcm_close_proc) (ma_snd_pcm_t *pcm);
typedef size_t (* ma_snd_pcm_hw_params_sizeof_proc) (void);
typedef int (* ma_snd_pcm_hw_params_any_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params);
typedef int (* ma_snd_pcm_hw_params_set_format_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t val);
typedef int (* ma_snd_pcm_hw_params_set_format_first_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t *format);
typedef void (* ma_snd_pcm_hw_params_get_format_mask_proc) (ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_mask_t *mask);
typedef int (* ma_snd_pcm_hw_params_set_channels_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val);
typedef int (* ma_snd_pcm_hw_params_set_channels_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val);
typedef int (* ma_snd_pcm_hw_params_set_channels_minmax_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *minimum, unsigned int *maximum);
typedef int (* ma_snd_pcm_hw_params_set_rate_resample_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val);
typedef int (* ma_snd_pcm_hw_params_set_rate_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val, int dir);
typedef int (* ma_snd_pcm_hw_params_set_rate_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
typedef int (* ma_snd_pcm_hw_params_set_buffer_size_near_proc)(ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_uframes_t *val);
typedef int (* ma_snd_pcm_hw_params_set_periods_near_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
typedef int (* ma_snd_pcm_hw_params_set_access_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_access_t _access);
typedef int (* ma_snd_pcm_hw_params_get_format_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t *format);
typedef int (* ma_snd_pcm_hw_params_get_channels_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val);
typedef int (* ma_snd_pcm_hw_params_get_channels_min_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val);
typedef int (* ma_snd_pcm_hw_params_get_channels_max_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val);
typedef int (* ma_snd_pcm_hw_params_get_rate_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir);
typedef int (* ma_snd_pcm_hw_params_get_rate_min_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir);
typedef int (* ma_snd_pcm_hw_params_get_rate_max_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *rate, int *dir);
typedef int (* ma_snd_pcm_hw_params_get_buffer_size_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_uframes_t *val);
typedef int (* ma_snd_pcm_hw_params_get_periods_proc) (const ma_snd_pcm_hw_params_t *params, unsigned int *val, int *dir);
typedef int (* ma_snd_pcm_hw_params_get_access_proc) (const ma_snd_pcm_hw_params_t *params, ma_snd_pcm_access_t *_access);
typedef int (* ma_snd_pcm_hw_params_test_format_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, ma_snd_pcm_format_t val);
typedef int (* ma_snd_pcm_hw_params_test_channels_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val);
typedef int (* ma_snd_pcm_hw_params_test_rate_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params, unsigned int val, int dir);
typedef int (* ma_snd_pcm_hw_params_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_hw_params_t *params);
typedef size_t (* ma_snd_pcm_sw_params_sizeof_proc) (void);
typedef int (* ma_snd_pcm_sw_params_current_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params);
typedef int (* ma_snd_pcm_sw_params_get_boundary_proc) (const ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t* val);
typedef int (* ma_snd_pcm_sw_params_set_avail_min_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val);
typedef int (* ma_snd_pcm_sw_params_set_start_threshold_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val);
typedef int (* ma_snd_pcm_sw_params_set_stop_threshold_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params, ma_snd_pcm_uframes_t val);
typedef int (* ma_snd_pcm_sw_params_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_sw_params_t *params);
typedef size_t (* ma_snd_pcm_format_mask_sizeof_proc) (void);
typedef int (* ma_snd_pcm_format_mask_test_proc) (const ma_snd_pcm_format_mask_t *mask, ma_snd_pcm_format_t val);
typedef ma_snd_pcm_chmap_t * (* ma_snd_pcm_get_chmap_proc) (ma_snd_pcm_t *pcm);
typedef ma_snd_pcm_state_t (* ma_snd_pcm_state_proc) (ma_snd_pcm_t *pcm);
typedef int (* ma_snd_pcm_prepare_proc) (ma_snd_pcm_t *pcm);
typedef int (* ma_snd_pcm_start_proc) (ma_snd_pcm_t *pcm);
typedef int (* ma_snd_pcm_drop_proc) (ma_snd_pcm_t *pcm);
typedef int (* ma_snd_pcm_drain_proc) (ma_snd_pcm_t *pcm);
typedef int (* ma_snd_pcm_reset_proc) (ma_snd_pcm_t *pcm);
typedef int (* ma_snd_device_name_hint_proc) (int card, const char *iface, void ***hints);
typedef char * (* ma_snd_device_name_get_hint_proc) (const void *hint, const char *id);
typedef int (* ma_snd_card_get_index_proc) (const char *name);
typedef int (* ma_snd_device_name_free_hint_proc) (void **hints);
typedef int (* ma_snd_pcm_mmap_begin_proc) (ma_snd_pcm_t *pcm, const ma_snd_pcm_channel_area_t **areas, ma_snd_pcm_uframes_t *offset, ma_snd_pcm_uframes_t *frames);
typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_mmap_commit_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_uframes_t offset, ma_snd_pcm_uframes_t frames);
typedef int (* ma_snd_pcm_recover_proc) (ma_snd_pcm_t *pcm, int err, int silent);
typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_readi_proc) (ma_snd_pcm_t *pcm, void *buffer, ma_snd_pcm_uframes_t size);
typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_writei_proc) (ma_snd_pcm_t *pcm, const void *buffer, ma_snd_pcm_uframes_t size);
typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_avail_proc) (ma_snd_pcm_t *pcm);
typedef ma_snd_pcm_sframes_t (* ma_snd_pcm_avail_update_proc) (ma_snd_pcm_t *pcm);
typedef int (* ma_snd_pcm_wait_proc) (ma_snd_pcm_t *pcm, int timeout);
typedef int (* ma_snd_pcm_nonblock_proc) (ma_snd_pcm_t *pcm, int nonblock);
typedef int (* ma_snd_pcm_info_proc) (ma_snd_pcm_t *pcm, ma_snd_pcm_info_t* info);
typedef size_t (* ma_snd_pcm_info_sizeof_proc) (void);
typedef const char* (* ma_snd_pcm_info_get_name_proc) (const ma_snd_pcm_info_t* info);
typedef int (* ma_snd_pcm_poll_descriptors_proc) (ma_snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space);
typedef int (* ma_snd_pcm_poll_descriptors_count_proc) (ma_snd_pcm_t *pcm);
typedef int (* ma_snd_pcm_poll_descriptors_revents_proc) (ma_snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents);
typedef int (* ma_snd_config_update_free_global_proc) (void);
/* This array specifies each of the common devices that can be used for both playback and capture. */
static const char* g_maCommonDeviceNamesALSA[] = {
"default",
"null",
"pulse",
"jack"
};
/* This array allows us to blacklist specific playback devices. */
static const char* g_maBlacklistedPlaybackDeviceNamesALSA[] = {
""
};
/* This array allows us to blacklist specific capture devices. */
static const char* g_maBlacklistedCaptureDeviceNamesALSA[] = {
""
};
static ma_snd_pcm_format_t ma_convert_ma_format_to_alsa_format(ma_format format)
{
ma_snd_pcm_format_t ALSAFormats[] = {
MA_SND_PCM_FORMAT_UNKNOWN, /* ma_format_unknown */
MA_SND_PCM_FORMAT_U8, /* ma_format_u8 */
MA_SND_PCM_FORMAT_S16_LE, /* ma_format_s16 */
MA_SND_PCM_FORMAT_S24_3LE, /* ma_format_s24 */
MA_SND_PCM_FORMAT_S32_LE, /* ma_format_s32 */
MA_SND_PCM_FORMAT_FLOAT_LE /* ma_format_f32 */
};
if (ma_is_big_endian()) {
ALSAFormats[0] = MA_SND_PCM_FORMAT_UNKNOWN;
ALSAFormats[1] = MA_SND_PCM_FORMAT_U8;
ALSAFormats[2] = MA_SND_PCM_FORMAT_S16_BE;
ALSAFormats[3] = MA_SND_PCM_FORMAT_S24_3BE;
ALSAFormats[4] = MA_SND_PCM_FORMAT_S32_BE;
ALSAFormats[5] = MA_SND_PCM_FORMAT_FLOAT_BE;
}
return ALSAFormats[format];
}
static ma_format ma_format_from_alsa(ma_snd_pcm_format_t formatALSA)
{
if (ma_is_little_endian()) {
switch (formatALSA) {
case MA_SND_PCM_FORMAT_S16_LE: return ma_format_s16;
case MA_SND_PCM_FORMAT_S24_3LE: return ma_format_s24;
case MA_SND_PCM_FORMAT_S32_LE: return ma_format_s32;
case MA_SND_PCM_FORMAT_FLOAT_LE: return ma_format_f32;
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if (((ma_snd_pcm_hw_params_test_channels_proc)pContext->alsa.snd_pcm_hw_params_test_channels)(pPCM, pHWParams, channels) == 0) {
/* The channel count is supported. */
/* The configuration space now needs to be restricted to the channel count before extracting the sample rate. */
((ma_snd_pcm_hw_params_set_channels_proc)pContext->alsa.snd_pcm_hw_params_set_channels)(pPCM, pHWParams, channels);
/* Only after the configuration space has been restricted to the specific channel count should we iterate over our sample rates. */
ma_context_iterate_rates_and_add_native_data_format__alsa(pContext, pPCM, pHWParams, format, channels, 0, pDeviceInfo);
} else {
/* The channel count is not supported. Skip. */
}
}
}
} else {
/* The format is not supported. Skip. */
}
}
ma_free(pHWParams, &pContext->allocationCallbacks);
((ma_snd_pcm_close_proc)pContext->alsa.snd_pcm_close)(pPCM);
return MA_SUCCESS;
}
static ma_result ma_device_uninit__alsa(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
if ((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture) {
((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture);
close(pDevice->alsa.wakeupfdCapture);
ma_free(pDevice->alsa.pPollDescriptorsCapture, &pDevice->pContext->allocationCallbacks);
}
if ((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback) {
((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback);
close(pDevice->alsa.wakeupfdPlayback);
ma_free(pDevice->alsa.pPollDescriptorsPlayback, &pDevice->pContext->allocationCallbacks);
}
return MA_SUCCESS;
}
static ma_result ma_device_init_by_type__alsa(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType)
{
ma_result result;
int resultALSA;
ma_snd_pcm_t* pPCM;
ma_bool32 isUsingMMap;
ma_snd_pcm_format_t formatALSA;
ma_format internalFormat;
ma_uint32 internalChannels;
ma_uint32 internalSampleRate;
ma_channel internalChannelMap[MA_MAX_CHANNELS];
ma_uint32 internalPeriodSizeInFrames;
ma_uint32 internalPeriods;
int openMode;
ma_snd_pcm_hw_params_t* pHWParams;
ma_snd_pcm_sw_params_t* pSWParams;
ma_snd_pcm_uframes_t bufferBoundary;
int pollDescriptorCount;
struct pollfd* pPollDescriptors;
int wakeupfd;
MA_ASSERT(pConfig != NULL);
MA_ASSERT(deviceType != ma_device_type_duplex); /* This function should only be called for playback _or_ capture, never duplex. */
MA_ASSERT(pDevice != NULL);
formatALSA = ma_convert_ma_format_to_alsa_format(pDescriptor->format);
openMode = 0;
if (pConfig->alsa.noAutoResample) {
openMode |= MA_SND_PCM_NO_AUTO_RESAMPLE;
}
if (pConfig->alsa.noAutoChannels) {
openMode |= MA_SND_PCM_NO_AUTO_CHANNELS;
}
if (pConfig->alsa.noAutoFormat) {
openMode |= MA_SND_PCM_NO_AUTO_FORMAT;
}
result = ma_context_open_pcm__alsa(pDevice->pContext, pDescriptor->shareMode, deviceType, pDescriptor->pDeviceID, openMode, &pPCM);
if (result != MA_SUCCESS) {
return result;
}
/* Hardware parameters. */
pHWParams = (ma_snd_pcm_hw_params_t*)ma_calloc(((ma_snd_pcm_hw_params_sizeof_proc)pDevice->pContext->alsa.snd_pcm_hw_params_sizeof)(), &pDevice->pContext->allocationCallbacks);
if (pHWParams == NULL) {
((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to allocate memory for hardware parameters.");
return MA_OUT_OF_MEMORY;
}
resultALSA = ((ma_snd_pcm_hw_params_any_proc)pDevice->pContext->alsa.snd_pcm_hw_params_any)(pPCM, pHWParams);
if (resultALSA < 0) {
ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize hardware parameters. snd_pcm_hw_params_any() failed.");
return ma_result_from_errno(-resultALSA);
}
/* MMAP Mode. Try using interleaved MMAP access. If this fails, fall back to standard readi/writei. */
isUsingMMap = MA_FALSE;
#if 0 /* NOTE: MMAP mode temporarily disabled. */
if (deviceType != ma_device_type_capture) { /* <-- Disabling MMAP mode for capture devices because I apparently do not have a device that supports it which means I can't test it... Contributions welcome. */
if (!pConfig->alsa.noMMap && ma_device__is_async(pDevice)) {
if (((ma_snd_pcm_hw_params_set_access_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_access)(pPCM, pHWParams, MA_SND_PCM_ACCESS_MMAP_INTERLEAVED) == 0) {
pDevice->alsa.isUsingMMap = MA_TRUE;
}
}
}
#endif
if (!isUsingMMap) {
resultALSA = ((ma_snd_pcm_hw_params_set_access_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_access)(pPCM, pHWParams, MA_SND_PCM_ACCESS_RW_INTERLEAVED);
if (resultALSA < 0) {
ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
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internalChannels = (ma_uint32)channels;
}
/* Sample Rate */
{
unsigned int sampleRate;
/*
It appears there's either a bug in ALSA, a bug in some drivers, or I'm doing something silly; but having resampling enabled causes
problems with some device configurations when used in conjunction with MMAP access mode. To fix this problem we need to disable
resampling.
To reproduce this problem, open the "plug:dmix" device, and set the sample rate to 44100. Internally, it looks like dmix uses a
sample rate of 48000. The hardware parameters will get set correctly with no errors, but it looks like the 44100 -> 48000 resampling
doesn't work properly - but only with MMAP access mode. You will notice skipping/crackling in the audio, and it'll run at a slightly
faster rate.
miniaudio has built-in support for sample rate conversion (albeit low quality at the moment), so disabling resampling should be fine
for us. The only problem is that it won't be taking advantage of any kind of hardware-accelerated resampling and it won't be very
good quality until I get a chance to improve the quality of miniaudio's software sample rate conversion.
I don't currently know if the dmix plugin is the only one with this error. Indeed, this is the only one I've been able to reproduce
this error with. In the future, we may want to restrict the disabling of resampling to only known bad plugins.
*/
((ma_snd_pcm_hw_params_set_rate_resample_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_rate_resample)(pPCM, pHWParams, 0);
sampleRate = pDescriptor->sampleRate;
if (sampleRate == 0) {
sampleRate = MA_DEFAULT_SAMPLE_RATE;
}
resultALSA = ((ma_snd_pcm_hw_params_set_rate_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_rate_near)(pPCM, pHWParams, &sampleRate, 0);
if (resultALSA < 0) {
ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Sample rate not supported. snd_pcm_hw_params_set_rate_near() failed.");
return ma_result_from_errno(-resultALSA);
}
internalSampleRate = (ma_uint32)sampleRate;
}
/* Periods. */
{
ma_uint32 periods = pDescriptor->periodCount;
resultALSA = ((ma_snd_pcm_hw_params_set_periods_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_periods_near)(pPCM, pHWParams, &periods, NULL);
if (resultALSA < 0) {
ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set period count. snd_pcm_hw_params_set_periods_near() failed.");
return ma_result_from_errno(-resultALSA);
}
internalPeriods = periods;
}
/* Buffer Size */
{
ma_snd_pcm_uframes_t actualBufferSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, internalSampleRate, pConfig->performanceProfile) * internalPeriods;
resultALSA = ((ma_snd_pcm_hw_params_set_buffer_size_near_proc)pDevice->pContext->alsa.snd_pcm_hw_params_set_buffer_size_near)(pPCM, pHWParams, &actualBufferSizeInFrames);
if (resultALSA < 0) {
ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set buffer size for device. snd_pcm_hw_params_set_buffer_size() failed.");
return ma_result_from_errno(-resultALSA);
}
internalPeriodSizeInFrames = actualBufferSizeInFrames / internalPeriods;
}
/* Apply hardware parameters. */
resultALSA = ((ma_snd_pcm_hw_params_proc)pDevice->pContext->alsa.snd_pcm_hw_params)(pPCM, pHWParams);
if (resultALSA < 0) {
ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to set hardware parameters. snd_pcm_hw_params() failed.");
return ma_result_from_errno(-resultALSA);
}
ma_free(pHWParams, &pDevice->pContext->allocationCallbacks);
pHWParams = NULL;
/* Software parameters. */
pSWParams = (ma_snd_pcm_sw_params_t*)ma_calloc(((ma_snd_pcm_sw_params_sizeof_proc)pDevice->pContext->alsa.snd_pcm_sw_params_sizeof)(), &pDevice->pContext->allocationCallbacks);
if (pSWParams == NULL) {
((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to allocate memory for software parameters.");
return MA_OUT_OF_MEMORY;
}
resultALSA = ((ma_snd_pcm_sw_params_current_proc)pDevice->pContext->alsa.snd_pcm_sw_params_current)(pPCM, pSWParams);
if (resultALSA < 0) {
ma_free(pSWParams, &pDevice->pContext->allocationCallbacks);
((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to initialize software parameters. snd_pcm_sw_params_current() failed.");
return ma_result_from_errno(-resultALSA);
}
resultALSA = ((ma_snd_pcm_sw_params_set_avail_min_proc)pDevice->pContext->alsa.snd_pcm_sw_params_set_avail_min)(pPCM, pSWParams, ma_prev_power_of_2(internalPeriodSizeInFrames));
if (resultALSA < 0) {
ma_free(pSWParams, &pDevice->pContext->allocationCallbacks);
((ma_snd_pcm_close_proc)pDevice->pContext->alsa.snd_pcm_close)(pPCM);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_sw_params_set_avail_min() failed.");
return ma_result_from_errno(-resultALSA);
}
resultALSA = ((ma_snd_pcm_sw_params_get_boundary_proc)pDevice->pContext->alsa.snd_pcm_sw_params_get_boundary)(pSWParams, &bufferBoundary);
if (resultALSA < 0) {
bufferBoundary = internalPeriodSizeInFrames * internalPeriods;
}
if (deviceType == ma_device_type_playback && !isUsingMMap) { /* Only playback devices in writei/readi mode need a start threshold. */
/*
Subtle detail here with the start threshold. When in playback-only mode (no full-duplex) we can set the start threshold to
the size of a period. But for full-duplex we need to set it such that it is at least two periods.
*/
resultALSA = ((ma_snd_pcm_sw_params_set_start_threshold_proc)pDevice->pContext->alsa.snd_pcm_sw_params_set_start_threshold)(pPCM, pSWParams, internalPeriodSizeInFrames*2);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
{
for (;;) {
unsigned short revents;
int resultALSA;
int resultPoll = poll(pPollDescriptors, pollDescriptorCount, -1);
if (resultPoll < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] poll() failed.");
return ma_result_from_errno(errno);
}
/*
Before checking the ALSA poll descriptor flag we need to check if the wakeup descriptor
has had it's POLLIN flag set. If so, we need to actually read the data and then exit
function. The wakeup descriptor will be the first item in the descriptors buffer.
*/
if ((pPollDescriptors[0].revents & POLLIN) != 0) {
ma_uint64 t;
int resultRead = read(pPollDescriptors[0].fd, &t, sizeof(t)); /* <-- Important that we read here so that the next write() does not block. */
if (resultRead < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] read() failed.");
return ma_result_from_errno(errno);
}
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] POLLIN set for wakeupfd\n");
return MA_DEVICE_NOT_STARTED;
}
/*
Getting here means that some data should be able to be read. We need to use ALSA to
translate the revents flags for us.
*/
resultALSA = ((ma_snd_pcm_poll_descriptors_revents_proc)pDevice->pContext->alsa.snd_pcm_poll_descriptors_revents)(pPCM, pPollDescriptors + 1, pollDescriptorCount - 1, &revents); /* +1, -1 to ignore the wakeup descriptor. */
if (resultALSA < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] snd_pcm_poll_descriptors_revents() failed.");
return ma_result_from_errno(-resultALSA);
}
if ((revents & POLLERR) != 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] POLLERR detected.");
return ma_result_from_errno(errno);
}
if ((revents & requiredEvent) == requiredEvent) {
break; /* We're done. Data available for reading or writing. */
}
}
return MA_SUCCESS;
}
static ma_result ma_device_wait_read__alsa(ma_device* pDevice)
{
return ma_device_wait__alsa(pDevice, (ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, (struct pollfd*)pDevice->alsa.pPollDescriptorsCapture, pDevice->alsa.pollDescriptorCountCapture + 1, POLLIN); /* +1 to account for the wakeup descriptor. */
}
static ma_result ma_device_wait_write__alsa(ma_device* pDevice)
{
return ma_device_wait__alsa(pDevice, (ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, (struct pollfd*)pDevice->alsa.pPollDescriptorsPlayback, pDevice->alsa.pollDescriptorCountPlayback + 1, POLLOUT); /* +1 to account for the wakeup descriptor. */
}
static ma_result ma_device_read__alsa(ma_device* pDevice, void* pFramesOut, ma_uint32 frameCount, ma_uint32* pFramesRead)
{
ma_snd_pcm_sframes_t resultALSA = 0;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(pFramesOut != NULL);
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
while (ma_device_get_state(pDevice) == ma_device_state_started) {
ma_result result;
/* The first thing to do is wait for data to become available for reading. This will return an error code if the device has been stopped. */
result = ma_device_wait_read__alsa(pDevice);
if (result != MA_SUCCESS) {
return result;
}
/* Getting here means we should have data available. */
resultALSA = ((ma_snd_pcm_readi_proc)pDevice->pContext->alsa.snd_pcm_readi)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, pFramesOut, frameCount);
if (resultALSA >= 0) {
break; /* Success. */
} else {
if (resultALSA == -EAGAIN) {
/*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "EGAIN (read)\n");*/
continue; /* Try again. */
} else if (resultALSA == -EPIPE) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "EPIPE (read)\n");
/* Overrun. Recover and try again. If this fails we need to return an error. */
resultALSA = ((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture, resultALSA, MA_TRUE);
if (resultALSA < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after overrun.");
return ma_result_from_errno((int)-resultALSA);
}
resultALSA = ((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMCapture);
if (resultALSA < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device after underrun.");
return ma_result_from_errno((int)-resultALSA);
}
continue; /* Try reading again. */
}
}
}
if (pFramesRead != NULL) {
*pFramesRead = resultALSA;
}
return MA_SUCCESS;
}
static ma_result ma_device_write__alsa(ma_device* pDevice, const void* pFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
{
ma_snd_pcm_sframes_t resultALSA = 0;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(pFrames != NULL);
if (pFramesWritten != NULL) {
*pFramesWritten = 0;
}
while (ma_device_get_state(pDevice) == ma_device_state_started) {
ma_result result;
/* The first thing to do is wait for space to become available for writing. This will return an error code if the device has been stopped. */
result = ma_device_wait_write__alsa(pDevice);
if (result != MA_SUCCESS) {
return result;
}
resultALSA = ((ma_snd_pcm_writei_proc)pDevice->pContext->alsa.snd_pcm_writei)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, pFrames, frameCount);
if (resultALSA >= 0) {
break; /* Success. */
} else {
if (resultALSA == -EAGAIN) {
/*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "EGAIN (write)\n");*/
continue; /* Try again. */
} else if (resultALSA == -EPIPE) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "EPIPE (write)\n");
/* Underrun. Recover and try again. If this fails we need to return an error. */
resultALSA = ((ma_snd_pcm_recover_proc)pDevice->pContext->alsa.snd_pcm_recover)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback, resultALSA, MA_TRUE); /* MA_TRUE=silent (don't print anything on error). */
if (resultALSA < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to recover device after underrun.");
return ma_result_from_errno((int)-resultALSA);
}
/*
In my testing I have had a situation where writei() does not automatically restart the device even though I've set it
up as such in the software parameters. What will happen is writei() will block indefinitely even though the number of
frames is well beyond the auto-start threshold. To work around this I've needed to add an explicit start here. Not sure
if this is me just being stupid and not recovering the device properly, but this definitely feels like something isn't
quite right here.
*/
resultALSA = ((ma_snd_pcm_start_proc)pDevice->pContext->alsa.snd_pcm_start)((ma_snd_pcm_t*)pDevice->alsa.pPCMPlayback);
if (resultALSA < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] Failed to start device after underrun.");
return ma_result_from_errno((int)-resultALSA);
}
continue; /* Try writing again. */
}
}
}
if (pFramesWritten != NULL) {
*pFramesWritten = resultALSA;
}
return MA_SUCCESS;
}
static ma_result ma_device_data_loop_wakeup__alsa(ma_device* pDevice)
{
ma_uint64 t = 1;
int resultWrite = 0;
MA_ASSERT(pDevice != NULL);
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Waking up...\n");
/* Write to an eventfd to trigger a wakeup from poll() and abort any reading or writing. */
if (pDevice->alsa.pPollDescriptorsCapture != NULL) {
resultWrite = write(pDevice->alsa.wakeupfdCapture, &t, sizeof(t));
}
if (pDevice->alsa.pPollDescriptorsPlayback != NULL) {
resultWrite = write(pDevice->alsa.wakeupfdPlayback, &t, sizeof(t));
}
if (resultWrite < 0) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[ALSA] write() failed.\n");
return ma_result_from_errno(errno);
}
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[ALSA] Waking up completed successfully.\n");
return MA_SUCCESS;
}
static ma_result ma_context_uninit__alsa(ma_context* pContext)
{
MA_ASSERT(pContext != NULL);
MA_ASSERT(pContext->backend == ma_backend_alsa);
/* Clean up memory for memory leak checkers. */
((ma_snd_config_update_free_global_proc)pContext->alsa.snd_config_update_free_global)();
#ifndef MA_NO_RUNTIME_LINKING
ma_dlclose(pContext, pContext->alsa.asoundSO);
#endif
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
pDeviceName = NULL;
}
result = ma_context_get_default_device_index__pulse(pContext, deviceType, &callbackData.defaultDeviceIndex);
if (deviceType == ma_device_type_playback) {
pOP = ((ma_pa_context_get_sink_info_by_name_proc)pContext->pulse.pa_context_get_sink_info_by_name)((ma_pa_context*)(pContext->pulse.pPulseContext), pDeviceName, ma_context_get_device_info_sink_callback__pulse, &callbackData);
} else {
pOP = ((ma_pa_context_get_source_info_by_name_proc)pContext->pulse.pa_context_get_source_info_by_name)((ma_pa_context*)(pContext->pulse.pPulseContext), pDeviceName, ma_context_get_device_info_source_callback__pulse, &callbackData);
}
if (pOP != NULL) {
ma_wait_for_operation_and_unref__pulse(pContext, pContext->pulse.pMainLoop, pOP);
} else {
result = MA_ERROR;
goto done;
}
if (!callbackData.foundDevice) {
result = MA_NO_DEVICE;
goto done;
}
done:
return result;
}
static ma_result ma_device_uninit__pulse(ma_device* pDevice)
{
ma_context* pContext;
MA_ASSERT(pDevice != NULL);
pContext = pDevice->pContext;
MA_ASSERT(pContext != NULL);
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
((ma_pa_stream_disconnect_proc)pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
((ma_pa_stream_unref_proc)pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
}
if (pDevice->type == ma_device_type_duplex) {
ma_duplex_rb_uninit(&pDevice->duplexRB);
}
((ma_pa_context_disconnect_proc)pContext->pulse.pa_context_disconnect)((ma_pa_context*)pDevice->pulse.pPulseContext);
((ma_pa_context_unref_proc)pContext->pulse.pa_context_unref)((ma_pa_context*)pDevice->pulse.pPulseContext);
((ma_pa_mainloop_free_proc)pContext->pulse.pa_mainloop_free)((ma_pa_mainloop*)pDevice->pulse.pMainLoop);
return MA_SUCCESS;
}
static ma_pa_buffer_attr ma_device__pa_buffer_attr_new(ma_uint32 periodSizeInFrames, ma_uint32 periods, const ma_pa_sample_spec* ss)
{
ma_pa_buffer_attr attr;
attr.maxlength = periodSizeInFrames * periods * ma_get_bytes_per_frame(ma_format_from_pulse(ss->format), ss->channels);
attr.tlength = attr.maxlength / periods;
attr.prebuf = (ma_uint32)-1;
attr.minreq = (ma_uint32)-1;
attr.fragsize = attr.maxlength / periods;
return attr;
}
static ma_pa_stream* ma_device__pa_stream_new__pulse(ma_device* pDevice, const char* pStreamName, const ma_pa_sample_spec* ss, const ma_pa_channel_map* cmap)
{
static int g_StreamCounter = 0;
char actualStreamName[256];
if (pStreamName != NULL) {
ma_strncpy_s(actualStreamName, sizeof(actualStreamName), pStreamName, (size_t)-1);
} else {
ma_strcpy_s(actualStreamName, sizeof(actualStreamName), "miniaudio:");
ma_itoa_s(g_StreamCounter, actualStreamName + 8, sizeof(actualStreamName)-8, 10); /* 8 = strlen("miniaudio:") */
}
g_StreamCounter += 1;
return ((ma_pa_stream_new_proc)pDevice->pContext->pulse.pa_stream_new)((ma_pa_context*)pDevice->pulse.pPulseContext, actualStreamName, ss, cmap);
}
static void ma_device_on_read__pulse(ma_pa_stream* pStream, size_t byteCount, void* pUserData)
{
ma_device* pDevice = (ma_device*)pUserData;
ma_uint32 bpf;
ma_uint32 deviceState;
ma_uint64 frameCount;
ma_uint64 framesProcessed;
MA_ASSERT(pDevice != NULL);
/*
Don't do anything if the device isn't initialized yet. Yes, this can happen because PulseAudio
can fire this callback before the stream has even started. Ridiculous.
*/
deviceState = ma_device_get_state(pDevice);
if (deviceState != ma_device_state_starting && deviceState != ma_device_state_started) {
return;
}
bpf = ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
MA_ASSERT(bpf > 0);
frameCount = byteCount / bpf;
framesProcessed = 0;
while (ma_device_get_state(pDevice) == ma_device_state_started && framesProcessed < frameCount) {
const void* pMappedPCMFrames;
size_t bytesMapped;
ma_uint64 framesMapped;
int pulseResult = ((ma_pa_stream_peek_proc)pDevice->pContext->pulse.pa_stream_peek)(pStream, &pMappedPCMFrames, &bytesMapped);
if (pulseResult < 0) {
break; /* Failed to map. Abort. */
}
framesMapped = bytesMapped / bpf;
if (framesMapped > 0) {
if (pMappedPCMFrames != NULL) {
ma_device_handle_backend_data_callback(pDevice, NULL, pMappedPCMFrames, framesMapped);
} else {
/* It's a hole. */
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[PulseAudio] ma_device_on_read__pulse: Hole.\n");
}
pulseResult = ((ma_pa_stream_drop_proc)pDevice->pContext->pulse.pa_stream_drop)(pStream);
if (pulseResult < 0) {
break; /* Failed to drop the buffer. */
}
framesProcessed += framesMapped;
} else {
/* Nothing was mapped. Just abort. */
break;
}
}
}
static ma_result ma_device_write_to_stream__pulse(ma_device* pDevice, ma_pa_stream* pStream, ma_uint64* pFramesProcessed)
{
ma_result result = MA_SUCCESS;
ma_uint64 framesProcessed = 0;
size_t bytesMapped;
ma_uint32 bpf;
ma_uint32 deviceState;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(pStream != NULL);
bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
MA_ASSERT(bpf > 0);
deviceState = ma_device_get_state(pDevice);
bytesMapped = ((ma_pa_stream_writable_size_proc)pDevice->pContext->pulse.pa_stream_writable_size)(pStream);
if (bytesMapped != (size_t)-1) {
if (bytesMapped > 0) {
ma_uint64 framesMapped;
void* pMappedPCMFrames;
int pulseResult = ((ma_pa_stream_begin_write_proc)pDevice->pContext->pulse.pa_stream_begin_write)(pStream, &pMappedPCMFrames, &bytesMapped);
if (pulseResult < 0) {
result = ma_result_from_pulse(pulseResult);
goto done;
}
framesMapped = bytesMapped / bpf;
if (deviceState == ma_device_state_started || deviceState == ma_device_state_starting) { /* Check for starting state just in case this is being used to do the initial fill. */
ma_device_handle_backend_data_callback(pDevice, pMappedPCMFrames, NULL, framesMapped);
} else {
/* Device is not started. Write silence. */
ma_silence_pcm_frames(pMappedPCMFrames, framesMapped, pDevice->playback.format, pDevice->playback.channels);
}
pulseResult = ((ma_pa_stream_write_proc)pDevice->pContext->pulse.pa_stream_write)(pStream, pMappedPCMFrames, bytesMapped, NULL, 0, MA_PA_SEEK_RELATIVE);
if (pulseResult < 0) {
result = ma_result_from_pulse(pulseResult);
goto done; /* Failed to write data to stream. */
}
framesProcessed += framesMapped;
} else {
result = MA_SUCCESS; /* No data available for writing. */
goto done;
}
} else {
result = MA_ERROR; /* Failed to retrieve the writable size. Abort. */
goto done;
}
done:
if (pFramesProcessed != NULL) {
*pFramesProcessed = framesProcessed;
}
return result;
}
static void ma_device_on_write__pulse(ma_pa_stream* pStream, size_t byteCount, void* pUserData)
{
ma_device* pDevice = (ma_device*)pUserData;
ma_uint32 bpf;
ma_uint64 frameCount;
ma_uint64 framesProcessed;
ma_uint32 deviceState;
ma_result result;
MA_ASSERT(pDevice != NULL);
/*
Don't do anything if the device isn't initialized yet. Yes, this can happen because PulseAudio
can fire this callback before the stream has even started. Ridiculous.
*/
deviceState = ma_device_get_state(pDevice);
if (deviceState != ma_device_state_starting && deviceState != ma_device_state_started) {
return;
}
bpf = ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
MA_ASSERT(bpf > 0);
frameCount = byteCount / bpf;
framesProcessed = 0;
while (framesProcessed < frameCount) {
ma_uint64 framesProcessedThisIteration;
/* Don't keep trying to process frames if the device isn't started. */
deviceState = ma_device_get_state(pDevice);
if (deviceState != ma_device_state_starting && deviceState != ma_device_state_started) {
break;
}
result = ma_device_write_to_stream__pulse(pDevice, pStream, &framesProcessedThisIteration);
if (result != MA_SUCCESS) {
break;
}
framesProcessed += framesProcessedThisIteration;
}
}
static void ma_device_on_suspended__pulse(ma_pa_stream* pStream, void* pUserData)
{
ma_device* pDevice = (ma_device*)pUserData;
int suspended;
(void)pStream;
suspended = ((ma_pa_stream_is_suspended_proc)pDevice->pContext->pulse.pa_stream_is_suspended)(pStream);
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[Pulse] Device suspended state changed. pa_stream_is_suspended() returned %d.\n", suspended);
if (suspended < 0) {
return;
}
if (suspended == 1) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[Pulse] Device suspended state changed. Suspended.\n");
ma_device__on_notification_stopped(pDevice);
} else {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "[Pulse] Device suspended state changed. Resumed.\n");
ma_device__on_notification_started(pDevice);
}
}
static void ma_device_on_rerouted__pulse(ma_pa_stream* pStream, void* pUserData)
{
ma_device* pDevice = (ma_device*)pUserData;
(void)pStream;
(void)pUserData;
ma_device__on_notification_rerouted(pDevice);
}
static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__pulse(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
{
/*
There have been reports from users where buffers of < ~20ms result glitches when running through
PipeWire. To work around this we're going to have to use a different default buffer size.
*/
const ma_uint32 defaultPeriodSizeInMilliseconds_LowLatency = 25;
const ma_uint32 defaultPeriodSizeInMilliseconds_Conservative = MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE;
MA_ASSERT(nativeSampleRate != 0);
if (pDescriptor->periodSizeInFrames == 0) {
if (pDescriptor->periodSizeInMilliseconds == 0) {
if (performanceProfile == ma_performance_profile_low_latency) {
return ma_calculate_buffer_size_in_frames_from_milliseconds(defaultPeriodSizeInMilliseconds_LowLatency, nativeSampleRate);
} else {
return ma_calculate_buffer_size_in_frames_from_milliseconds(defaultPeriodSizeInMilliseconds_Conservative, nativeSampleRate);
}
} else {
return ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptor->periodSizeInMilliseconds, nativeSampleRate);
}
} else {
return pDescriptor->periodSizeInFrames;
}
}
static ma_result ma_device_init__pulse(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
/*
Notes for PulseAudio:
- We're always using native format/channels/rate regardless of whether or not PulseAudio
supports the format directly through their own data conversion system. I'm doing this to
reduce as much variability from the PulseAudio side as possible because it's seems to be
extremely unreliable at everything it does.
- When both the period size in frames and milliseconds are 0, we default to miniaudio's
default buffer sizes rather than leaving it up to PulseAudio because I don't trust
PulseAudio to give us any kind of reasonable latency by default.
- Do not ever, *ever* forget to use MA_PA_STREAM_ADJUST_LATENCY. If you don't specify this
flag, capture mode will just not work properly until you open another PulseAudio app.
*/
ma_result result = MA_SUCCESS;
int error = 0;
const char* devPlayback = NULL;
const char* devCapture = NULL;
ma_format format = ma_format_unknown;
ma_uint32 channels = 0;
ma_uint32 sampleRate = 0;
ma_pa_sink_info sinkInfo;
ma_pa_source_info sourceInfo;
ma_pa_sample_spec ss;
ma_pa_channel_map cmap;
ma_pa_buffer_attr attr;
const ma_pa_sample_spec* pActualSS = NULL;
const ma_pa_channel_map* pActualCMap = NULL;
const ma_pa_buffer_attr* pActualAttr = NULL;
ma_uint32 iChannel;
ma_pa_stream_flags_t streamFlags;
MA_ASSERT(pDevice != NULL);
MA_ZERO_OBJECT(&pDevice->pulse);
if (pConfig->deviceType == ma_device_type_loopback) {
return MA_DEVICE_TYPE_NOT_SUPPORTED;
}
/* No exclusive mode with the PulseAudio backend. */
if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pConfig->playback.shareMode == ma_share_mode_exclusive) ||
((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pConfig->capture.shareMode == ma_share_mode_exclusive)) {
return MA_SHARE_MODE_NOT_SUPPORTED;
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
if (pDescriptorPlayback->pDeviceID != NULL) {
devPlayback = pDescriptorPlayback->pDeviceID->pulse;
}
format = pDescriptorPlayback->format;
channels = pDescriptorPlayback->channels;
sampleRate = pDescriptorPlayback->sampleRate;
}
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
if (pDescriptorCapture->pDeviceID != NULL) {
devCapture = pDescriptorCapture->pDeviceID->pulse;
}
format = pDescriptorCapture->format;
channels = pDescriptorCapture->channels;
sampleRate = pDescriptorCapture->sampleRate;
}
result = ma_init_pa_mainloop_and_pa_context__pulse(pDevice->pContext, pDevice->pContext->pulse.pApplicationName, pDevice->pContext->pulse.pServerName, MA_FALSE, &pDevice->pulse.pMainLoop, &pDevice->pulse.pPulseContext);
if (result != MA_SUCCESS) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to initialize PA mainloop and context for device.\n");
return result;
}
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
result = ma_context_get_source_info__pulse(pDevice->pContext, devCapture, &sourceInfo);
if (result != MA_SUCCESS) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve source info for capture device.");
goto on_error0;
}
ss = sourceInfo.sample_spec;
cmap = sourceInfo.channel_map;
if (ma_format_from_pulse(ss.format) == ma_format_unknown) {
if (ma_is_little_endian()) {
ss.format = MA_PA_SAMPLE_FLOAT32LE;
} else {
ss.format = MA_PA_SAMPLE_FLOAT32BE;
}
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.format not supported by miniaudio. Defaulting to PA_SAMPLE_FLOAT32.\n");
}
if (ss.rate == 0) {
ss.rate = MA_DEFAULT_SAMPLE_RATE;
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.rate = 0. Defaulting to %d.\n", ss.rate);
}
if (ss.channels == 0) {
ss.channels = MA_DEFAULT_CHANNELS;
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.channels = 0. Defaulting to %d.\n", ss.channels);
}
/* We now have enough information to calculate our actual period size in frames. */
pDescriptorCapture->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__pulse(pDescriptorCapture, ss.rate, pConfig->performanceProfile);
attr = ma_device__pa_buffer_attr_new(pDescriptorCapture->periodSizeInFrames, pDescriptorCapture->periodCount, &ss);
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Capture attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; periodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.fragsi...
pDevice->pulse.pStreamCapture = ma_device__pa_stream_new__pulse(pDevice, pConfig->pulse.pStreamNameCapture, &ss, &cmap);
if (pDevice->pulse.pStreamCapture == NULL) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio capture stream.\n");
result = MA_ERROR;
goto on_error0;
}
/* The callback needs to be set before connecting the stream. */
((ma_pa_stream_set_read_callback_proc)pDevice->pContext->pulse.pa_stream_set_read_callback)((ma_pa_stream*)pDevice->pulse.pStreamCapture, ma_device_on_read__pulse, pDevice);
/* State callback for checking when the device has been corked. */
((ma_pa_stream_set_suspended_callback_proc)pDevice->pContext->pulse.pa_stream_set_suspended_callback)((ma_pa_stream*)pDevice->pulse.pStreamCapture, ma_device_on_suspended__pulse, pDevice);
/* Rerouting notification. */
((ma_pa_stream_set_moved_callback_proc)pDevice->pContext->pulse.pa_stream_set_moved_callback)((ma_pa_stream*)pDevice->pulse.pStreamCapture, ma_device_on_rerouted__pulse, pDevice);
/* Connect after we've got all of our internal state set up. */
streamFlags = MA_PA_STREAM_START_CORKED | MA_PA_STREAM_ADJUST_LATENCY | MA_PA_STREAM_FIX_FORMAT | MA_PA_STREAM_FIX_RATE | MA_PA_STREAM_FIX_CHANNELS;
if (devCapture != NULL) {
streamFlags |= MA_PA_STREAM_DONT_MOVE;
}
error = ((ma_pa_stream_connect_record_proc)pDevice->pContext->pulse.pa_stream_connect_record)((ma_pa_stream*)pDevice->pulse.pStreamCapture, devCapture, &attr, streamFlags);
if (error != MA_PA_OK) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio capture stream.");
result = ma_result_from_pulse(error);
goto on_error1;
}
result = ma_wait_for_pa_stream_to_connect__pulse(pDevice->pContext, pDevice->pulse.pMainLoop, (ma_pa_stream*)pDevice->pulse.pStreamCapture);
if (result != MA_SUCCESS) {
goto on_error2;
}
/* Internal format. */
pActualSS = ((ma_pa_stream_get_sample_spec_proc)pDevice->pContext->pulse.pa_stream_get_sample_spec)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
if (pActualSS != NULL) {
ss = *pActualSS;
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Capture sample spec: format=%s, channels=%d, rate=%d\n", ma_get_format_name(ma_format_from_pulse(ss.format)), ss.channels, ss.rate);
} else {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Failed to retrieve capture sample spec.\n");
}
pDescriptorCapture->format = ma_format_from_pulse(ss.format);
pDescriptorCapture->channels = ss.channels;
pDescriptorCapture->sampleRate = ss.rate;
if (pDescriptorCapture->format == ma_format_unknown || pDescriptorCapture->channels == 0 || pDescriptorCapture->sampleRate == 0) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Capture sample spec is invalid. Device unusable by miniaudio. format=%s, channels=%d, sampleRate=%d.\n", ma_get_format_name(pDescriptorCapture->format), pDescripto...
result = MA_ERROR;
goto on_error4;
}
/* Internal channel map. */
/*
Bug in PipeWire. There have been reports that PipeWire is returning AUX channels when reporting
the channel map. To somewhat workaround this, I'm hacking in a hard coded channel map for mono
and stereo. In this case it should be safe to assume mono = MONO and stereo = LEFT/RIGHT. For
all other channel counts we need to just put up with whatever PipeWire reports and hope it gets
fixed sooner than later. I might remove this hack later.
*/
if (pDescriptorCapture->channels > 2) {
pActualCMap = ((ma_pa_stream_get_channel_map_proc)pDevice->pContext->pulse.pa_stream_get_channel_map)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
if (pActualCMap != NULL) {
cmap = *pActualCMap;
}
for (iChannel = 0; iChannel < pDescriptorCapture->channels; ++iChannel) {
pDescriptorCapture->channelMap[iChannel] = ma_channel_position_from_pulse(cmap.map[iChannel]);
}
} else {
/* Hack for mono and stereo. */
if (pDescriptorCapture->channels == 1) {
pDescriptorCapture->channelMap[0] = MA_CHANNEL_MONO;
} else if (pDescriptorCapture->channels == 2) {
pDescriptorCapture->channelMap[0] = MA_CHANNEL_FRONT_LEFT;
pDescriptorCapture->channelMap[1] = MA_CHANNEL_FRONT_RIGHT;
} else {
MA_ASSERT(MA_FALSE); /* Should never hit this. */
}
}
/* Buffer. */
pActualAttr = ((ma_pa_stream_get_buffer_attr_proc)pDevice->pContext->pulse.pa_stream_get_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
if (pActualAttr != NULL) {
attr = *pActualAttr;
}
if (attr.fragsize > 0) {
pDescriptorCapture->periodCount = ma_max(attr.maxlength / attr.fragsize, 1);
} else {
pDescriptorCapture->periodCount = 1;
}
pDescriptorCapture->periodSizeInFrames = attr.maxlength / ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) / pDescriptorCapture->periodCount;
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Capture actual attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; periodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr...
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
result = ma_context_get_sink_info__pulse(pDevice->pContext, devPlayback, &sinkInfo);
if (result != MA_SUCCESS) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to retrieve sink info for playback device.\n");
goto on_error2;
}
ss = sinkInfo.sample_spec;
cmap = sinkInfo.channel_map;
if (ma_format_from_pulse(ss.format) == ma_format_unknown) {
if (ma_is_little_endian()) {
ss.format = MA_PA_SAMPLE_FLOAT32LE;
} else {
ss.format = MA_PA_SAMPLE_FLOAT32BE;
}
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.format not supported by miniaudio. Defaulting to PA_SAMPLE_FLOAT32.\n");
}
if (ss.rate == 0) {
ss.rate = MA_DEFAULT_SAMPLE_RATE;
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.rate = 0. Defaulting to %d.\n", ss.rate);
}
if (ss.channels == 0) {
ss.channels = MA_DEFAULT_CHANNELS;
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] sample_spec.channels = 0. Defaulting to %d.\n", ss.channels);
}
/* We now have enough information to calculate the actual buffer size in frames. */
pDescriptorPlayback->periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__pulse(pDescriptorPlayback, ss.rate, pConfig->performanceProfile);
attr = ma_device__pa_buffer_attr_new(pDescriptorPlayback->periodSizeInFrames, pDescriptorPlayback->periodCount, &ss);
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Playback attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; periodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.minreq, attr.frags...
pDevice->pulse.pStreamPlayback = ma_device__pa_stream_new__pulse(pDevice, pConfig->pulse.pStreamNamePlayback, &ss, &cmap);
if (pDevice->pulse.pStreamPlayback == NULL) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to create PulseAudio playback stream.\n");
result = MA_ERROR;
goto on_error2;
}
/*
Note that this callback will be fired as soon as the stream is connected, even though it's started as corked. The callback needs to handle a
device state of ma_device_state_uninitialized.
*/
((ma_pa_stream_set_write_callback_proc)pDevice->pContext->pulse.pa_stream_set_write_callback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_device_on_write__pulse, pDevice);
/* State callback for checking when the device has been corked. */
((ma_pa_stream_set_suspended_callback_proc)pDevice->pContext->pulse.pa_stream_set_suspended_callback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_device_on_suspended__pulse, pDevice);
/* Rerouting notification. */
((ma_pa_stream_set_moved_callback_proc)pDevice->pContext->pulse.pa_stream_set_moved_callback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, ma_device_on_rerouted__pulse, pDevice);
/* Connect after we've got all of our internal state set up. */
streamFlags = MA_PA_STREAM_START_CORKED | MA_PA_STREAM_ADJUST_LATENCY | MA_PA_STREAM_FIX_FORMAT | MA_PA_STREAM_FIX_RATE | MA_PA_STREAM_FIX_CHANNELS;
if (devPlayback != NULL) {
streamFlags |= MA_PA_STREAM_DONT_MOVE;
}
error = ((ma_pa_stream_connect_playback_proc)pDevice->pContext->pulse.pa_stream_connect_playback)((ma_pa_stream*)pDevice->pulse.pStreamPlayback, devPlayback, &attr, streamFlags, NULL, NULL);
if (error != MA_PA_OK) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to connect PulseAudio playback stream.");
result = ma_result_from_pulse(error);
goto on_error3;
}
result = ma_wait_for_pa_stream_to_connect__pulse(pDevice->pContext, pDevice->pulse.pMainLoop, (ma_pa_stream*)pDevice->pulse.pStreamPlayback);
if (result != MA_SUCCESS) {
goto on_error3;
}
/* Internal format. */
pActualSS = ((ma_pa_stream_get_sample_spec_proc)pDevice->pContext->pulse.pa_stream_get_sample_spec)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
if (pActualSS != NULL) {
ss = *pActualSS;
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Playback sample spec: format=%s, channels=%d, rate=%d\n", ma_get_format_name(ma_format_from_pulse(ss.format)), ss.channels, ss.rate);
} else {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Failed to retrieve playback sample spec.\n");
}
pDescriptorPlayback->format = ma_format_from_pulse(ss.format);
pDescriptorPlayback->channels = ss.channels;
pDescriptorPlayback->sampleRate = ss.rate;
if (pDescriptorPlayback->format == ma_format_unknown || pDescriptorPlayback->channels == 0 || pDescriptorPlayback->sampleRate == 0) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Playback sample spec is invalid. Device unusable by miniaudio. format=%s, channels=%d, sampleRate=%d.\n", ma_get_format_name(pDescriptorPlayback->format), pDescrip...
result = MA_ERROR;
goto on_error4;
}
/* Internal channel map. */
/*
Bug in PipeWire. There have been reports that PipeWire is returning AUX channels when reporting
the channel map. To somewhat workaround this, I'm hacking in a hard coded channel map for mono
and stereo. In this case it should be safe to assume mono = MONO and stereo = LEFT/RIGHT. For
all other channel counts we need to just put up with whatever PipeWire reports and hope it gets
fixed sooner than later. I might remove this hack later.
*/
if (pDescriptorPlayback->channels > 2) {
pActualCMap = ((ma_pa_stream_get_channel_map_proc)pDevice->pContext->pulse.pa_stream_get_channel_map)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
if (pActualCMap != NULL) {
cmap = *pActualCMap;
}
for (iChannel = 0; iChannel < pDescriptorPlayback->channels; ++iChannel) {
pDescriptorPlayback->channelMap[iChannel] = ma_channel_position_from_pulse(cmap.map[iChannel]);
}
} else {
/* Hack for mono and stereo. */
if (pDescriptorPlayback->channels == 1) {
pDescriptorPlayback->channelMap[0] = MA_CHANNEL_MONO;
} else if (pDescriptorPlayback->channels == 2) {
pDescriptorPlayback->channelMap[0] = MA_CHANNEL_FRONT_LEFT;
pDescriptorPlayback->channelMap[1] = MA_CHANNEL_FRONT_RIGHT;
} else {
MA_ASSERT(MA_FALSE); /* Should never hit this. */
}
}
/* Buffer. */
pActualAttr = ((ma_pa_stream_get_buffer_attr_proc)pDevice->pContext->pulse.pa_stream_get_buffer_attr)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
if (pActualAttr != NULL) {
attr = *pActualAttr;
}
if (attr.tlength > 0) {
pDescriptorPlayback->periodCount = ma_max(attr.maxlength / attr.tlength, 1);
} else {
pDescriptorPlayback->periodCount = 1;
}
pDescriptorPlayback->periodSizeInFrames = attr.maxlength / ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels) / pDescriptorPlayback->periodCount;
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[PulseAudio] Playback actual attr: maxlength=%d, tlength=%d, prebuf=%d, minreq=%d, fragsize=%d; internalPeriodSizeInFrames=%d\n", attr.maxlength, attr.tlength, attr.prebuf, attr.min...
}
/*
We need a ring buffer for handling duplex mode. We can use the main duplex ring buffer in the main
part of the ma_device struct. We cannot, however, depend on ma_device_init() initializing this for
us later on because that will only do it if it's a fully asynchronous backend - i.e. the
onDeviceDataLoop callback is NULL, which is not the case for PulseAudio.
*/
if (pConfig->deviceType == ma_device_type_duplex) {
ma_format rbFormat = (format != ma_format_unknown) ? format : pDescriptorCapture->format;
ma_uint32 rbChannels = (channels > 0) ? channels : pDescriptorCapture->channels;
ma_uint32 rbSampleRate = (sampleRate > 0) ? sampleRate : pDescriptorCapture->sampleRate;
result = ma_duplex_rb_init(rbFormat, rbChannels, rbSampleRate, pDescriptorCapture->sampleRate, pDescriptorCapture->periodSizeInFrames, &pDevice->pContext->allocationCallbacks, &pDevice->duplexRB);
if (result != MA_SUCCESS) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[PulseAudio] Failed to initialize ring buffer. %s.\n", ma_result_description(result));
goto on_error4;
}
}
return MA_SUCCESS;
on_error4:
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
((ma_pa_stream_disconnect_proc)pDevice->pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
}
on_error3:
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
((ma_pa_stream_unref_proc)pDevice->pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamPlayback);
}
on_error2:
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
((ma_pa_stream_disconnect_proc)pDevice->pContext->pulse.pa_stream_disconnect)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
}
on_error1:
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
((ma_pa_stream_unref_proc)pDevice->pContext->pulse.pa_stream_unref)((ma_pa_stream*)pDevice->pulse.pStreamCapture);
}
on_error0:
return result;
}
static void ma_pulse_operation_complete_callback(ma_pa_stream* pStream, int success, void* pUserData)
{
ma_bool32* pIsSuccessful = (ma_bool32*)pUserData;
MA_ASSERT(pIsSuccessful != NULL);
*pIsSuccessful = (ma_bool32)success;
(void)pStream; /* Unused. */
}
static ma_result ma_device__cork_stream__pulse(ma_device* pDevice, ma_device_type deviceType, int cork)
{
ma_context* pContext = pDevice->pContext;
ma_bool32 wasSuccessful;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
pContext->pulse.pa_stream_begin_write = (ma_proc)_pa_stream_begin_write;
pContext->pulse.pa_stream_write = (ma_proc)_pa_stream_write;
pContext->pulse.pa_stream_peek = (ma_proc)_pa_stream_peek;
pContext->pulse.pa_stream_drop = (ma_proc)_pa_stream_drop;
pContext->pulse.pa_stream_writable_size = (ma_proc)_pa_stream_writable_size;
pContext->pulse.pa_stream_readable_size = (ma_proc)_pa_stream_readable_size;
#endif
/* We need to make a copy of the application and server names so we can pass them to the pa_context of each device. */
pContext->pulse.pApplicationName = ma_copy_string(pConfig->pulse.pApplicationName, &pContext->allocationCallbacks);
if (pContext->pulse.pApplicationName == NULL && pConfig->pulse.pApplicationName != NULL) {
return MA_OUT_OF_MEMORY;
}
pContext->pulse.pServerName = ma_copy_string(pConfig->pulse.pServerName, &pContext->allocationCallbacks);
if (pContext->pulse.pServerName == NULL && pConfig->pulse.pServerName != NULL) {
ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks);
return MA_OUT_OF_MEMORY;
}
result = ma_init_pa_mainloop_and_pa_context__pulse(pContext, pConfig->pulse.pApplicationName, pConfig->pulse.pServerName, pConfig->pulse.tryAutoSpawn, &pContext->pulse.pMainLoop, &pContext->pulse.pPulseContext);
if (result != MA_SUCCESS) {
ma_free(pContext->pulse.pServerName, &pContext->allocationCallbacks);
ma_free(pContext->pulse.pApplicationName, &pContext->allocationCallbacks);
#ifndef MA_NO_RUNTIME_LINKING
ma_dlclose(pContext, pContext->pulse.pulseSO);
#endif
return result;
}
/* With pa_mainloop we run a synchronous backend, but we implement our own main loop. */
pCallbacks->onContextInit = ma_context_init__pulse;
pCallbacks->onContextUninit = ma_context_uninit__pulse;
pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__pulse;
pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__pulse;
pCallbacks->onDeviceInit = ma_device_init__pulse;
pCallbacks->onDeviceUninit = ma_device_uninit__pulse;
pCallbacks->onDeviceStart = ma_device_start__pulse;
pCallbacks->onDeviceStop = ma_device_stop__pulse;
pCallbacks->onDeviceRead = NULL; /* Not used because we're implementing onDeviceDataLoop. */
pCallbacks->onDeviceWrite = NULL; /* Not used because we're implementing onDeviceDataLoop. */
pCallbacks->onDeviceDataLoop = ma_device_data_loop__pulse;
pCallbacks->onDeviceDataLoopWakeup = ma_device_data_loop_wakeup__pulse;
return MA_SUCCESS;
}
#endif
/******************************************************************************
JACK Backend
******************************************************************************/
#ifdef MA_HAS_JACK
/* It is assumed jack.h is available when compile-time linking is being used. */
#ifdef MA_NO_RUNTIME_LINKING
#include <jack/jack.h>
typedef jack_nframes_t ma_jack_nframes_t;
typedef jack_options_t ma_jack_options_t;
typedef jack_status_t ma_jack_status_t;
typedef jack_client_t ma_jack_client_t;
typedef jack_port_t ma_jack_port_t;
typedef JackProcessCallback ma_JackProcessCallback;
typedef JackBufferSizeCallback ma_JackBufferSizeCallback;
typedef JackShutdownCallback ma_JackShutdownCallback;
#define MA_JACK_DEFAULT_AUDIO_TYPE JACK_DEFAULT_AUDIO_TYPE
#define ma_JackNoStartServer JackNoStartServer
#define ma_JackPortIsInput JackPortIsInput
#define ma_JackPortIsOutput JackPortIsOutput
#define ma_JackPortIsPhysical JackPortIsPhysical
#else
typedef ma_uint32 ma_jack_nframes_t;
typedef int ma_jack_options_t;
typedef int ma_jack_status_t;
typedef struct ma_jack_client_t ma_jack_client_t;
typedef struct ma_jack_port_t ma_jack_port_t;
typedef int (* ma_JackProcessCallback) (ma_jack_nframes_t nframes, void* arg);
typedef int (* ma_JackBufferSizeCallback)(ma_jack_nframes_t nframes, void* arg);
typedef void (* ma_JackShutdownCallback) (void* arg);
#define MA_JACK_DEFAULT_AUDIO_TYPE "32 bit float mono audio"
#define ma_JackNoStartServer 1
#define ma_JackPortIsInput 1
#define ma_JackPortIsOutput 2
#define ma_JackPortIsPhysical 4
#endif
typedef ma_jack_client_t* (* ma_jack_client_open_proc) (const char* client_name, ma_jack_options_t options, ma_jack_status_t* status, ...);
typedef int (* ma_jack_client_close_proc) (ma_jack_client_t* client);
typedef int (* ma_jack_client_name_size_proc) (void);
typedef int (* ma_jack_set_process_callback_proc) (ma_jack_client_t* client, ma_JackProcessCallback process_callback, void* arg);
typedef int (* ma_jack_set_buffer_size_callback_proc)(ma_jack_client_t* client, ma_JackBufferSizeCallback bufsize_callback, void* arg);
typedef void (* ma_jack_on_shutdown_proc) (ma_jack_client_t* client, ma_JackShutdownCallback function, void* arg);
typedef ma_jack_nframes_t (* ma_jack_get_sample_rate_proc) (ma_jack_client_t* client);
typedef ma_jack_nframes_t (* ma_jack_get_buffer_size_proc) (ma_jack_client_t* client);
typedef const char** (* ma_jack_get_ports_proc) (ma_jack_client_t* client, const char* port_name_pattern, const char* type_name_pattern, unsigned long flags);
typedef int (* ma_jack_activate_proc) (ma_jack_client_t* client);
typedef int (* ma_jack_deactivate_proc) (ma_jack_client_t* client);
typedef int (* ma_jack_connect_proc) (ma_jack_client_t* client, const char* source_port, const char* destination_port);
typedef ma_jack_port_t* (* ma_jack_port_register_proc) (ma_jack_client_t* client, const char* port_name, const char* port_type, unsigned long flags, unsigned long buffer_size);
typedef const char* (* ma_jack_port_name_proc) (const ma_jack_port_t* port);
typedef void* (* ma_jack_port_get_buffer_proc) (ma_jack_port_t* port, ma_jack_nframes_t nframes);
typedef void (* ma_jack_free_proc) (void* ptr);
static ma_result ma_context_open_client__jack(ma_context* pContext, ma_jack_client_t** ppClient)
{
size_t maxClientNameSize;
char clientName[256];
ma_jack_status_t status;
ma_jack_client_t* pClient;
MA_ASSERT(pContext != NULL);
MA_ASSERT(ppClient != NULL);
if (ppClient) {
*ppClient = NULL;
}
maxClientNameSize = ((ma_jack_client_name_size_proc)pContext->jack.jack_client_name_size)(); /* Includes null terminator. */
ma_strncpy_s(clientName, ma_min(sizeof(clientName), maxClientNameSize), (pContext->jack.pClientName != NULL) ? pContext->jack.pClientName : "miniaudio", (size_t)-1);
pClient = ((ma_jack_client_open_proc)pContext->jack.jack_client_open)(clientName, (pContext->jack.tryStartServer) ? 0 : ma_JackNoStartServer, &status, NULL);
if (pClient == NULL) {
return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
}
if (ppClient) {
*ppClient = pClient;
}
return MA_SUCCESS;
}
static ma_result ma_context_enumerate_devices__jack(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
{
ma_bool32 cbResult = MA_TRUE;
MA_ASSERT(pContext != NULL);
MA_ASSERT(callback != NULL);
/* Playback. */
if (cbResult) {
ma_device_info deviceInfo;
MA_ZERO_OBJECT(&deviceInfo);
ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
deviceInfo.isDefault = MA_TRUE; /* JACK only uses default devices. */
cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
}
/* Capture. */
if (cbResult) {
ma_device_info deviceInfo;
MA_ZERO_OBJECT(&deviceInfo);
ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
deviceInfo.isDefault = MA_TRUE; /* JACK only uses default devices. */
cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
}
(void)cbResult; /* For silencing a static analysis warning. */
return MA_SUCCESS;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
pDeviceInfo->nativeDataFormats[0].sampleRate = ((ma_jack_get_sample_rate_proc)pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pClient);
pDeviceInfo->nativeDataFormats[0].channels = 0;
ppPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ((deviceType == ma_device_type_playback) ? ma_JackPortIsInput : ma_JackPortIsOutput));
if (ppPorts == NULL) {
((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pClient);
ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.");
return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
}
while (ppPorts[pDeviceInfo->nativeDataFormats[0].channels] != NULL) {
pDeviceInfo->nativeDataFormats[0].channels += 1;
}
pDeviceInfo->nativeDataFormats[0].flags = 0;
pDeviceInfo->nativeDataFormatCount = 1;
((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppPorts);
((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pClient);
(void)pContext;
return MA_SUCCESS;
}
static ma_result ma_device_uninit__jack(ma_device* pDevice)
{
ma_context* pContext;
MA_ASSERT(pDevice != NULL);
pContext = pDevice->pContext;
MA_ASSERT(pContext != NULL);
if (pDevice->jack.pClient != NULL) {
((ma_jack_client_close_proc)pContext->jack.jack_client_close)((ma_jack_client_t*)pDevice->jack.pClient);
}
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
ma_free(pDevice->jack.pIntermediaryBufferCapture, &pDevice->pContext->allocationCallbacks);
ma_free(pDevice->jack.ppPortsCapture, &pDevice->pContext->allocationCallbacks);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
ma_free(pDevice->jack.pIntermediaryBufferPlayback, &pDevice->pContext->allocationCallbacks);
ma_free(pDevice->jack.ppPortsPlayback, &pDevice->pContext->allocationCallbacks);
}
return MA_SUCCESS;
}
static void ma_device__jack_shutdown_callback(void* pUserData)
{
/* JACK died. Stop the device. */
ma_device* pDevice = (ma_device*)pUserData;
MA_ASSERT(pDevice != NULL);
ma_device_stop(pDevice);
}
static int ma_device__jack_buffer_size_callback(ma_jack_nframes_t frameCount, void* pUserData)
{
ma_device* pDevice = (ma_device*)pUserData;
MA_ASSERT(pDevice != NULL);
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
size_t newBufferSize = frameCount * (pDevice->capture.internalChannels * ma_get_bytes_per_sample(pDevice->capture.internalFormat));
float* pNewBuffer = (float*)ma_calloc(newBufferSize, &pDevice->pContext->allocationCallbacks);
if (pNewBuffer == NULL) {
return MA_OUT_OF_MEMORY;
}
ma_free(pDevice->jack.pIntermediaryBufferCapture, &pDevice->pContext->allocationCallbacks);
pDevice->jack.pIntermediaryBufferCapture = pNewBuffer;
pDevice->playback.internalPeriodSizeInFrames = frameCount;
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
size_t newBufferSize = frameCount * (pDevice->playback.internalChannels * ma_get_bytes_per_sample(pDevice->playback.internalFormat));
float* pNewBuffer = (float*)ma_calloc(newBufferSize, &pDevice->pContext->allocationCallbacks);
if (pNewBuffer == NULL) {
return MA_OUT_OF_MEMORY;
}
ma_free(pDevice->jack.pIntermediaryBufferPlayback, &pDevice->pContext->allocationCallbacks);
pDevice->jack.pIntermediaryBufferPlayback = pNewBuffer;
pDevice->playback.internalPeriodSizeInFrames = frameCount;
}
return 0;
}
static int ma_device__jack_process_callback(ma_jack_nframes_t frameCount, void* pUserData)
{
ma_device* pDevice;
ma_context* pContext;
ma_uint32 iChannel;
pDevice = (ma_device*)pUserData;
MA_ASSERT(pDevice != NULL);
pContext = pDevice->pContext;
MA_ASSERT(pContext != NULL);
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
/* Channels need to be interleaved. */
for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) {
const float* pSrc = (const float*)((ma_jack_port_get_buffer_proc)pContext->jack.jack_port_get_buffer)((ma_jack_port_t*)pDevice->jack.ppPortsCapture[iChannel], frameCount);
if (pSrc != NULL) {
float* pDst = pDevice->jack.pIntermediaryBufferCapture + iChannel;
ma_jack_nframes_t iFrame;
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
*pDst = *pSrc;
pDst += pDevice->capture.internalChannels;
pSrc += 1;
}
}
}
ma_device_handle_backend_data_callback(pDevice, NULL, pDevice->jack.pIntermediaryBufferCapture, frameCount);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
ma_device_handle_backend_data_callback(pDevice, pDevice->jack.pIntermediaryBufferPlayback, NULL, frameCount);
/* Channels need to be deinterleaved. */
for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) {
float* pDst = (float*)((ma_jack_port_get_buffer_proc)pContext->jack.jack_port_get_buffer)((ma_jack_port_t*)pDevice->jack.ppPortsPlayback[iChannel], frameCount);
if (pDst != NULL) {
const float* pSrc = pDevice->jack.pIntermediaryBufferPlayback + iChannel;
ma_jack_nframes_t iFrame;
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
*pDst = *pSrc;
pDst += 1;
pSrc += pDevice->playback.internalChannels;
}
}
}
}
return 0;
}
static ma_result ma_device_init__jack(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
ma_result result;
ma_uint32 periodSizeInFrames;
MA_ASSERT(pConfig != NULL);
MA_ASSERT(pDevice != NULL);
if (pConfig->deviceType == ma_device_type_loopback) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Loopback mode not supported.");
return MA_DEVICE_TYPE_NOT_SUPPORTED;
}
/* Only supporting default devices with JACK. */
if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->pDeviceID != NULL && pDescriptorPlayback->pDeviceID->jack != 0) ||
((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->pDeviceID != NULL && pDescriptorCapture->pDeviceID->jack != 0)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Only default devices are supported.");
return MA_NO_DEVICE;
}
/* No exclusive mode with the JACK backend. */
if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Exclusive mode not supported.");
return MA_SHARE_MODE_NOT_SUPPORTED;
}
/* Open the client. */
result = ma_context_open_client__jack(pDevice->pContext, (ma_jack_client_t**)&pDevice->jack.pClient);
if (result != MA_SUCCESS) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to open client.");
return result;
}
/* Callbacks. */
if (((ma_jack_set_process_callback_proc)pDevice->pContext->jack.jack_set_process_callback)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_process_callback, pDevice) != 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to set process callback.");
return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
}
if (((ma_jack_set_buffer_size_callback_proc)pDevice->pContext->jack.jack_set_buffer_size_callback)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_buffer_size_callback, pDevice) != 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to set buffer size callback.");
return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
}
((ma_jack_on_shutdown_proc)pDevice->pContext->jack.jack_on_shutdown)((ma_jack_client_t*)pDevice->jack.pClient, ma_device__jack_shutdown_callback, pDevice);
/* The buffer size in frames can change. */
periodSizeInFrames = ((ma_jack_get_buffer_size_proc)pDevice->pContext->jack.jack_get_buffer_size)((ma_jack_client_t*)pDevice->jack.pClient);
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
ma_uint32 iPort;
const char** ppPorts;
pDescriptorCapture->format = ma_format_f32;
pDescriptorCapture->channels = 0;
pDescriptorCapture->sampleRate = ((ma_jack_get_sample_rate_proc)pDevice->pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pDevice->jack.pClient);
ma_channel_map_init_standard(ma_standard_channel_map_alsa, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap), pDescriptorCapture->channels);
ppPorts = ((ma_jack_get_ports_proc)pDevice->pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsOutput);
if (ppPorts == NULL) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.");
return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
}
/* Need to count the number of ports first so we can allocate some memory. */
while (ppPorts[pDescriptorCapture->channels] != NULL) {
pDescriptorCapture->channels += 1;
}
pDevice->jack.ppPortsCapture = (ma_ptr*)ma_malloc(sizeof(*pDevice->jack.ppPortsCapture) * pDescriptorCapture->channels, &pDevice->pContext->allocationCallbacks);
if (pDevice->jack.ppPortsCapture == NULL) {
return MA_OUT_OF_MEMORY;
}
for (iPort = 0; iPort < pDescriptorCapture->channels; iPort += 1) {
char name[64];
ma_strcpy_s(name, sizeof(name), "capture");
ma_itoa_s((int)iPort, name+7, sizeof(name)-7, 10); /* 7 = length of "capture" */
pDevice->jack.ppPortsCapture[iPort] = ((ma_jack_port_register_proc)pDevice->pContext->jack.jack_port_register)((ma_jack_client_t*)pDevice->jack.pClient, name, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsInput, 0);
if (pDevice->jack.ppPortsCapture[iPort] == NULL) {
((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts);
ma_device_uninit__jack(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to register ports.");
return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
}
}
((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts);
pDescriptorCapture->periodSizeInFrames = periodSizeInFrames;
pDescriptorCapture->periodCount = 1; /* There's no notion of a period in JACK. Just set to 1. */
pDevice->jack.pIntermediaryBufferCapture = (float*)ma_calloc(pDescriptorCapture->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels), &pDevice->pContext->allocationCallbacks);
if (pDevice->jack.pIntermediaryBufferCapture == NULL) {
ma_device_uninit__jack(pDevice);
return MA_OUT_OF_MEMORY;
}
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
ma_uint32 iPort;
const char** ppPorts;
pDescriptorPlayback->format = ma_format_f32;
pDescriptorPlayback->channels = 0;
pDescriptorPlayback->sampleRate = ((ma_jack_get_sample_rate_proc)pDevice->pContext->jack.jack_get_sample_rate)((ma_jack_client_t*)pDevice->jack.pClient);
ma_channel_map_init_standard(ma_standard_channel_map_alsa, pDescriptorPlayback->channelMap, ma_countof(pDescriptorPlayback->channelMap), pDescriptorPlayback->channels);
ppPorts = ((ma_jack_get_ports_proc)pDevice->pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsInput);
if (ppPorts == NULL) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to query physical ports.");
return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
}
/* Need to count the number of ports first so we can allocate some memory. */
while (ppPorts[pDescriptorPlayback->channels] != NULL) {
pDescriptorPlayback->channels += 1;
}
pDevice->jack.ppPortsPlayback = (ma_ptr*)ma_malloc(sizeof(*pDevice->jack.ppPortsPlayback) * pDescriptorPlayback->channels, &pDevice->pContext->allocationCallbacks);
if (pDevice->jack.ppPortsPlayback == NULL) {
ma_free(pDevice->jack.ppPortsCapture, &pDevice->pContext->allocationCallbacks);
return MA_OUT_OF_MEMORY;
}
for (iPort = 0; iPort < pDescriptorPlayback->channels; iPort += 1) {
char name[64];
ma_strcpy_s(name, sizeof(name), "playback");
ma_itoa_s((int)iPort, name+8, sizeof(name)-8, 10); /* 8 = length of "playback" */
pDevice->jack.ppPortsPlayback[iPort] = ((ma_jack_port_register_proc)pDevice->pContext->jack.jack_port_register)((ma_jack_client_t*)pDevice->jack.pClient, name, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsOutput, 0);
if (pDevice->jack.ppPortsPlayback[iPort] == NULL) {
((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts);
ma_device_uninit__jack(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to register ports.");
return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
}
}
((ma_jack_free_proc)pDevice->pContext->jack.jack_free)((void*)ppPorts);
pDescriptorPlayback->periodSizeInFrames = periodSizeInFrames;
pDescriptorPlayback->periodCount = 1; /* There's no notion of a period in JACK. Just set to 1. */
pDevice->jack.pIntermediaryBufferPlayback = (float*)ma_calloc(pDescriptorPlayback->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels), &pDevice->pContext->allocationCallbacks);
if (pDevice->jack.pIntermediaryBufferPlayback == NULL) {
ma_device_uninit__jack(pDevice);
return MA_OUT_OF_MEMORY;
}
}
return MA_SUCCESS;
}
static ma_result ma_device_start__jack(ma_device* pDevice)
{
ma_context* pContext = pDevice->pContext;
int resultJACK;
size_t i;
resultJACK = ((ma_jack_activate_proc)pContext->jack.jack_activate)((ma_jack_client_t*)pDevice->jack.pClient);
if (resultJACK != 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to activate the JACK client.");
return MA_FAILED_TO_START_BACKEND_DEVICE;
}
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
const char** ppServerPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsOutput);
if (ppServerPorts == NULL) {
((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to retrieve physical ports.");
return MA_ERROR;
}
for (i = 0; ppServerPorts[i] != NULL; ++i) {
const char* pServerPort = ppServerPorts[i];
const char* pClientPort = ((ma_jack_port_name_proc)pContext->jack.jack_port_name)((ma_jack_port_t*)pDevice->jack.ppPortsCapture[i]);
resultJACK = ((ma_jack_connect_proc)pContext->jack.jack_connect)((ma_jack_client_t*)pDevice->jack.pClient, pServerPort, pClientPort);
if (resultJACK != 0) {
((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts);
((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to connect ports.");
return MA_ERROR;
}
}
((ma_jack_free_proc)pContext->jack.jack_free)((void*)ppServerPorts);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
const char** ppServerPorts = ((ma_jack_get_ports_proc)pContext->jack.jack_get_ports)((ma_jack_client_t*)pDevice->jack.pClient, NULL, MA_JACK_DEFAULT_AUDIO_TYPE, ma_JackPortIsPhysical | ma_JackPortIsInput);
if (ppServerPorts == NULL) {
((ma_jack_deactivate_proc)pContext->jack.jack_deactivate)((ma_jack_client_t*)pDevice->jack.pClient);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[JACK] Failed to retrieve physical ports.");
return MA_ERROR;
}
for (i = 0; ppServerPorts[i] != NULL; ++i) {
const char* pServerPort = ppServerPorts[i];
const char* pClientPort = ((ma_jack_port_name_proc)pContext->jack.jack_port_name)((ma_jack_port_t*)pDevice->jack.ppPortsPlayback[i]);
resultJACK = ((ma_jack_connect_proc)pContext->jack.jack_connect)((ma_jack_client_t*)pDevice->jack.pClient, pClientPort, pServerPort);
if (resultJACK != 0) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
for (iMALSampleRate = 0; iMALSampleRate < ma_countof(g_maStandardSampleRatePriorities); ++iMALSampleRate) {
ma_uint32 malSampleRate = g_maStandardSampleRatePriorities[iMALSampleRate];
UInt32 iCASampleRate;
for (iCASampleRate = 0; iCASampleRate < sampleRateRangeCount; ++iCASampleRate) {
AudioValueRange caSampleRate = pSampleRateRanges[iCASampleRate];
if (caSampleRate.mMinimum <= malSampleRate && caSampleRate.mMaximum >= malSampleRate) {
*pSampleRateOut = malSampleRate;
ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
return MA_SUCCESS;
}
}
}
/*
If we get here it means none of miniaudio's standard sample rates matched any of the supported sample rates from the device. In this
case we just fall back to the first one reported by Core Audio.
*/
MA_ASSERT(sampleRateRangeCount > 0);
*pSampleRateOut = pSampleRateRanges[0].mMinimum;
ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
return MA_SUCCESS;
} else {
/* Find the closest match to this sample rate. */
UInt32 currentAbsoluteDifference = INT32_MAX;
UInt32 iCurrentClosestRange = (UInt32)-1;
UInt32 iRange;
for (iRange = 0; iRange < sampleRateRangeCount; ++iRange) {
if (pSampleRateRanges[iRange].mMinimum <= sampleRateIn && pSampleRateRanges[iRange].mMaximum >= sampleRateIn) {
*pSampleRateOut = sampleRateIn;
ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
return MA_SUCCESS;
} else {
UInt32 absoluteDifference;
if (pSampleRateRanges[iRange].mMinimum > sampleRateIn) {
absoluteDifference = pSampleRateRanges[iRange].mMinimum - sampleRateIn;
} else {
absoluteDifference = sampleRateIn - pSampleRateRanges[iRange].mMaximum;
}
if (currentAbsoluteDifference > absoluteDifference) {
currentAbsoluteDifference = absoluteDifference;
iCurrentClosestRange = iRange;
}
}
}
MA_ASSERT(iCurrentClosestRange != (UInt32)-1);
*pSampleRateOut = pSampleRateRanges[iCurrentClosestRange].mMinimum;
ma_free(pSampleRateRanges, &pContext->allocationCallbacks);
return MA_SUCCESS;
}
/* Should never get here, but it would mean we weren't able to find any suitable sample rates. */
/*ma_free(pSampleRateRanges, &pContext->allocationCallbacks);*/
/*return MA_ERROR;*/
}
#endif
static ma_result ma_get_AudioObject_closest_buffer_size_in_frames(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32 bufferSizeInFramesIn, ma_uint32* pBufferSizeInFramesOut)
{
AudioObjectPropertyAddress propAddress;
AudioValueRange bufferSizeRange;
UInt32 dataSize;
OSStatus status;
MA_ASSERT(pContext != NULL);
MA_ASSERT(pBufferSizeInFramesOut != NULL);
*pBufferSizeInFramesOut = 0; /* Safety. */
propAddress.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
propAddress.mElement = kAudioObjectPropertyElementMaster;
dataSize = sizeof(bufferSizeRange);
status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &bufferSizeRange);
if (status != noErr) {
return ma_result_from_OSStatus(status);
}
/* This is just a clamp. */
if (bufferSizeInFramesIn < bufferSizeRange.mMinimum) {
*pBufferSizeInFramesOut = (ma_uint32)bufferSizeRange.mMinimum;
} else if (bufferSizeInFramesIn > bufferSizeRange.mMaximum) {
*pBufferSizeInFramesOut = (ma_uint32)bufferSizeRange.mMaximum;
} else {
*pBufferSizeInFramesOut = bufferSizeInFramesIn;
}
return MA_SUCCESS;
}
static ma_result ma_set_AudioObject_buffer_size_in_frames(ma_context* pContext, AudioObjectID deviceObjectID, ma_device_type deviceType, ma_uint32* pPeriodSizeInOut)
{
ma_result result;
ma_uint32 chosenBufferSizeInFrames;
AudioObjectPropertyAddress propAddress;
UInt32 dataSize;
OSStatus status;
MA_ASSERT(pContext != NULL);
result = ma_get_AudioObject_closest_buffer_size_in_frames(pContext, deviceObjectID, deviceType, *pPeriodSizeInOut, &chosenBufferSizeInFrames);
if (result != MA_SUCCESS) {
return result;
}
/* Try setting the size of the buffer... If this fails we just use whatever is currently set. */
propAddress.mSelector = kAudioDevicePropertyBufferFrameSize;
propAddress.mScope = (deviceType == ma_device_type_playback) ? kAudioObjectPropertyScopeOutput : kAudioObjectPropertyScopeInput;
propAddress.mElement = kAudioObjectPropertyElementMaster;
((ma_AudioObjectSetPropertyData_proc)pContext->coreaudio.AudioObjectSetPropertyData)(deviceObjectID, &propAddress, 0, NULL, sizeof(chosenBufferSizeInFrames), &chosenBufferSizeInFrames);
/* Get the actual size of the buffer. */
dataSize = sizeof(*pPeriodSizeInOut);
status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(deviceObjectID, &propAddress, 0, NULL, &dataSize, &chosenBufferSizeInFrames);
if (status != noErr) {
return ma_result_from_OSStatus(status);
}
*pPeriodSizeInOut = chosenBufferSizeInFrames;
return MA_SUCCESS;
}
static ma_result ma_find_default_AudioObjectID(ma_context* pContext, ma_device_type deviceType, AudioObjectID* pDeviceObjectID)
{
AudioObjectPropertyAddress propAddressDefaultDevice;
UInt32 defaultDeviceObjectIDSize = sizeof(AudioObjectID);
AudioObjectID defaultDeviceObjectID;
OSStatus status;
MA_ASSERT(pContext != NULL);
MA_ASSERT(pDeviceObjectID != NULL);
/* Safety. */
*pDeviceObjectID = 0;
propAddressDefaultDevice.mScope = kAudioObjectPropertyScopeGlobal;
propAddressDefaultDevice.mElement = kAudioObjectPropertyElementMaster;
if (deviceType == ma_device_type_playback) {
propAddressDefaultDevice.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
} else {
propAddressDefaultDevice.mSelector = kAudioHardwarePropertyDefaultInputDevice;
}
defaultDeviceObjectIDSize = sizeof(AudioObjectID);
status = ((ma_AudioObjectGetPropertyData_proc)pContext->coreaudio.AudioObjectGetPropertyData)(kAudioObjectSystemObject, &propAddressDefaultDevice, 0, NULL, &defaultDeviceObjectIDSize, &defaultDeviceObjectID);
if (status == noErr) {
*pDeviceObjectID = defaultDeviceObjectID;
return MA_SUCCESS;
}
/* If we get here it means we couldn't find the device. */
return MA_NO_DEVICE;
}
static ma_result ma_find_AudioObjectID(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, AudioObjectID* pDeviceObjectID)
{
MA_ASSERT(pContext != NULL);
MA_ASSERT(pDeviceObjectID != NULL);
/* Safety. */
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
formatScope = (deviceType == ma_device_type_playback) ? kAudioUnitScope_Input : kAudioUnitScope_Output;
formatElement = (deviceType == ma_device_type_playback) ? MA_COREAUDIO_OUTPUT_BUS : MA_COREAUDIO_INPUT_BUS;
propSize = sizeof(bestFormat);
status = ((ma_AudioUnitGetProperty_proc)pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, &propSize);
if (status != noErr) {
((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(audioUnit);
return ma_result_from_OSStatus(status);
}
((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(audioUnit);
audioUnit = NULL;
/* Only a single format is being reported for iOS. */
pDeviceInfo->nativeDataFormatCount = 1;
result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pDeviceInfo->nativeDataFormats[0].format);
if (result != MA_SUCCESS) {
return result;
}
pDeviceInfo->nativeDataFormats[0].channels = bestFormat.mChannelsPerFrame;
/*
It looks like Apple are wanting to push the whole AVAudioSession thing. Thus, we need to use that to determine device settings. To do
this we just get the shared instance and inspect.
*/
@autoreleasepool {
AVAudioSession* pAudioSession = [AVAudioSession sharedInstance];
MA_ASSERT(pAudioSession != NULL);
pDeviceInfo->nativeDataFormats[0].sampleRate = (ma_uint32)pAudioSession.sampleRate;
}
}
#endif
(void)pDeviceInfo; /* Unused. */
return MA_SUCCESS;
}
static AudioBufferList* ma_allocate_AudioBufferList__coreaudio(ma_uint32 sizeInFrames, ma_format format, ma_uint32 channels, ma_stream_layout layout, const ma_allocation_callbacks* pAllocationCallbacks)
{
AudioBufferList* pBufferList;
UInt32 audioBufferSizeInBytes;
size_t allocationSize;
MA_ASSERT(sizeInFrames > 0);
MA_ASSERT(format != ma_format_unknown);
MA_ASSERT(channels > 0);
allocationSize = sizeof(AudioBufferList) - sizeof(AudioBuffer); /* Subtract sizeof(AudioBuffer) because that part is dynamically sized. */
if (layout == ma_stream_layout_interleaved) {
/* Interleaved case. This is the simple case because we just have one buffer. */
allocationSize += sizeof(AudioBuffer) * 1;
} else {
/* Non-interleaved case. This is the more complex case because there's more than one buffer. */
allocationSize += sizeof(AudioBuffer) * channels;
}
allocationSize += sizeInFrames * ma_get_bytes_per_frame(format, channels);
pBufferList = (AudioBufferList*)ma_malloc(allocationSize, pAllocationCallbacks);
if (pBufferList == NULL) {
return NULL;
}
audioBufferSizeInBytes = (UInt32)(sizeInFrames * ma_get_bytes_per_sample(format));
if (layout == ma_stream_layout_interleaved) {
pBufferList->mNumberBuffers = 1;
pBufferList->mBuffers[0].mNumberChannels = channels;
pBufferList->mBuffers[0].mDataByteSize = audioBufferSizeInBytes * channels;
pBufferList->mBuffers[0].mData = (ma_uint8*)pBufferList + sizeof(AudioBufferList);
} else {
ma_uint32 iBuffer;
pBufferList->mNumberBuffers = channels;
for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) {
pBufferList->mBuffers[iBuffer].mNumberChannels = 1;
pBufferList->mBuffers[iBuffer].mDataByteSize = audioBufferSizeInBytes;
pBufferList->mBuffers[iBuffer].mData = (ma_uint8*)pBufferList + ((sizeof(AudioBufferList) - sizeof(AudioBuffer)) + (sizeof(AudioBuffer) * channels)) + (audioBufferSizeInBytes * iBuffer);
}
}
return pBufferList;
}
static ma_result ma_device_realloc_AudioBufferList__coreaudio(ma_device* pDevice, ma_uint32 sizeInFrames, ma_format format, ma_uint32 channels, ma_stream_layout layout)
{
MA_ASSERT(pDevice != NULL);
MA_ASSERT(format != ma_format_unknown);
MA_ASSERT(channels > 0);
/* Only resize the buffer if necessary. */
if (pDevice->coreaudio.audioBufferCapInFrames < sizeInFrames) {
AudioBufferList* pNewAudioBufferList;
pNewAudioBufferList = ma_allocate_AudioBufferList__coreaudio(sizeInFrames, format, channels, layout, &pDevice->pContext->allocationCallbacks);
if (pNewAudioBufferList == NULL) {
return MA_OUT_OF_MEMORY;
}
/* At this point we'll have a new AudioBufferList and we can free the old one. */
ma_free(pDevice->coreaudio.pAudioBufferList, &pDevice->pContext->allocationCallbacks);
pDevice->coreaudio.pAudioBufferList = pNewAudioBufferList;
pDevice->coreaudio.audioBufferCapInFrames = sizeInFrames;
}
/* Getting here means the capacity of the audio is fine. */
return MA_SUCCESS;
}
static OSStatus ma_on_output__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pActionFlags, const AudioTimeStamp* pTimeStamp, UInt32 busNumber, UInt32 frameCount, AudioBufferList* pBufferList)
{
ma_device* pDevice = (ma_device*)pUserData;
ma_stream_layout layout;
MA_ASSERT(pDevice != NULL);
/*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "INFO: Output Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", (int)busNumber, (int)frameCount, (int)pBufferList->mNumberBuffers);*/
/* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */
layout = ma_stream_layout_interleaved;
if (pBufferList->mBuffers[0].mNumberChannels != pDevice->playback.internalChannels) {
layout = ma_stream_layout_deinterleaved;
}
if (layout == ma_stream_layout_interleaved) {
/* For now we can assume everything is interleaved. */
UInt32 iBuffer;
for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; ++iBuffer) {
if (pBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->playback.internalChannels) {
ma_uint32 frameCountForThisBuffer = pBufferList->mBuffers[iBuffer].mDataByteSize / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
if (frameCountForThisBuffer > 0) {
ma_device_handle_backend_data_callback(pDevice, pBufferList->mBuffers[iBuffer].mData, NULL, frameCountForThisBuffer);
}
/*a_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", (int)frameCount, (int)pBufferList->mBuffers[iBuffer].mNumberChannels, (int)pBufferList->mBuffers[iBuffer].mDataBy...
} else {
/*
This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's
not interleaved, in which case we can't handle right now since miniaudio does not yet support non-interleaved streams. We just
output silence here.
*/
MA_ZERO_MEMORY(pBufferList->mBuffers[iBuffer].mData, pBufferList->mBuffers[iBuffer].mDataByteSize);
/*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", (int)frameCount, (int)pBufferList->mBuffers[iBuffer].mNumberChannels, (int)pBufferL...
}
}
} else {
/* This is the deinterleaved case. We need to update each buffer in groups of internalChannels. This assumes each buffer is the same size. */
MA_ASSERT(pDevice->playback.internalChannels <= MA_MAX_CHANNELS); /* This should heve been validated at initialization time. */
/*
For safety we'll check that the internal channels is a multiple of the buffer count. If it's not it means something
very strange has happened and we're not going to support it.
*/
if ((pBufferList->mNumberBuffers % pDevice->playback.internalChannels) == 0) {
ma_uint8 tempBuffer[4096];
UInt32 iBuffer;
for (iBuffer = 0; iBuffer < pBufferList->mNumberBuffers; iBuffer += pDevice->playback.internalChannels) {
ma_uint32 frameCountPerBuffer = pBufferList->mBuffers[iBuffer].mDataByteSize / ma_get_bytes_per_sample(pDevice->playback.internalFormat);
ma_uint32 framesRemaining = frameCountPerBuffer;
while (framesRemaining > 0) {
void* ppDeinterleavedBuffers[MA_MAX_CHANNELS];
ma_uint32 iChannel;
ma_uint32 framesToRead = sizeof(tempBuffer) / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
if (framesToRead > framesRemaining) {
framesToRead = framesRemaining;
}
ma_device_handle_backend_data_callback(pDevice, tempBuffer, NULL, framesToRead);
for (iChannel = 0; iChannel < pDevice->playback.internalChannels; ++iChannel) {
ppDeinterleavedBuffers[iChannel] = (void*)ma_offset_ptr(pBufferList->mBuffers[iBuffer+iChannel].mData, (frameCountPerBuffer - framesRemaining) * ma_get_bytes_per_sample(pDevice->playback.internalFormat));
}
ma_deinterleave_pcm_frames(pDevice->playback.internalFormat, pDevice->playback.internalChannels, framesToRead, tempBuffer, ppDeinterleavedBuffers);
framesRemaining -= framesToRead;
}
}
}
}
(void)pActionFlags;
(void)pTimeStamp;
(void)busNumber;
(void)frameCount;
return noErr;
}
static OSStatus ma_on_input__coreaudio(void* pUserData, AudioUnitRenderActionFlags* pActionFlags, const AudioTimeStamp* pTimeStamp, UInt32 busNumber, UInt32 frameCount, AudioBufferList* pUnusedBufferList)
{
ma_device* pDevice = (ma_device*)pUserData;
AudioBufferList* pRenderedBufferList;
ma_result result;
ma_stream_layout layout;
ma_uint32 iBuffer;
OSStatus status;
MA_ASSERT(pDevice != NULL);
pRenderedBufferList = (AudioBufferList*)pDevice->coreaudio.pAudioBufferList;
MA_ASSERT(pRenderedBufferList);
/* We need to check whether or not we are outputting interleaved or non-interleaved samples. The way we do this is slightly different for each type. */
layout = ma_stream_layout_interleaved;
if (pRenderedBufferList->mBuffers[0].mNumberChannels != pDevice->capture.internalChannels) {
layout = ma_stream_layout_deinterleaved;
}
/*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "INFO: Input Callback: busNumber=%d, frameCount=%d, mNumberBuffers=%d\n", (int)busNumber, (int)frameCount, (int)pRenderedBufferList->mNumberBuffers);*/
/*
There has been a situation reported where frame count passed into this function is greater than the capacity of
our capture buffer. There doesn't seem to be a reliable way to determine what the maximum frame count will be,
so we need to instead resort to dynamically reallocating our buffer to ensure it's large enough to capture the
number of frames requested by this callback.
*/
result = ma_device_realloc_AudioBufferList__coreaudio(pDevice, frameCount, pDevice->capture.internalFormat, pDevice->capture.internalChannels, layout);
if (result != MA_SUCCESS) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "Failed to allocate AudioBufferList for capture.\n");
return noErr;
}
pRenderedBufferList = (AudioBufferList*)pDevice->coreaudio.pAudioBufferList;
MA_ASSERT(pRenderedBufferList);
/*
When you call AudioUnitRender(), Core Audio tries to be helpful by setting the mDataByteSize to the number of bytes
that were actually rendered. The problem with this is that the next call can fail with -50 due to the size no longer
being set to the capacity of the buffer, but instead the size in bytes of the previous render. This will cause a
problem when a future call to this callback specifies a larger number of frames.
To work around this we need to explicitly set the size of each buffer to their respective size in bytes.
*/
for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) {
pRenderedBufferList->mBuffers[iBuffer].mDataByteSize = pDevice->coreaudio.audioBufferCapInFrames * ma_get_bytes_per_sample(pDevice->capture.internalFormat) * pRenderedBufferList->mBuffers[iBuffer].mNumberChannels;
}
status = ((ma_AudioUnitRender_proc)pDevice->pContext->coreaudio.AudioUnitRender)((AudioUnit)pDevice->coreaudio.audioUnitCapture, pActionFlags, pTimeStamp, busNumber, frameCount, pRenderedBufferList);
if (status != noErr) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " ERROR: AudioUnitRender() failed with %d.\n", (int)status);
return status;
}
if (layout == ma_stream_layout_interleaved) {
for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; ++iBuffer) {
if (pRenderedBufferList->mBuffers[iBuffer].mNumberChannels == pDevice->capture.internalChannels) {
ma_device_handle_backend_data_callback(pDevice, NULL, pRenderedBufferList->mBuffers[iBuffer].mData, frameCount);
/*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " mDataByteSize=%d.\n", (int)pRenderedBufferList->mBuffers[iBuffer].mDataByteSize);*/
} else {
/*
This case is where the number of channels in the output buffer do not match our internal channels. It could mean that it's
not interleaved, in which case we can't handle right now since miniaudio does not yet support non-interleaved streams.
*/
ma_uint8 silentBuffer[4096];
ma_uint32 framesRemaining;
MA_ZERO_MEMORY(silentBuffer, sizeof(silentBuffer));
framesRemaining = frameCount;
while (framesRemaining > 0) {
ma_uint32 framesToSend = sizeof(silentBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
if (framesToSend > framesRemaining) {
framesToSend = framesRemaining;
}
ma_device_handle_backend_data_callback(pDevice, NULL, silentBuffer, framesToSend);
framesRemaining -= framesToSend;
}
/*ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, " WARNING: Outputting silence. frameCount=%d, mNumberChannels=%d, mDataByteSize=%d\n", (int)frameCount, (int)pRenderedBufferList->mBuffers[iBuffer].mNumberChannels, (int)...
}
}
} else {
/* This is the deinterleaved case. We need to interleave the audio data before sending it to the client. This assumes each buffer is the same size. */
MA_ASSERT(pDevice->capture.internalChannels <= MA_MAX_CHANNELS); /* This should have been validated at initialization time. */
/*
For safety we'll check that the internal channels is a multiple of the buffer count. If it's not it means something
very strange has happened and we're not going to support it.
*/
if ((pRenderedBufferList->mNumberBuffers % pDevice->capture.internalChannels) == 0) {
ma_uint8 tempBuffer[4096];
for (iBuffer = 0; iBuffer < pRenderedBufferList->mNumberBuffers; iBuffer += pDevice->capture.internalChannels) {
ma_uint32 framesRemaining = frameCount;
while (framesRemaining > 0) {
void* ppDeinterleavedBuffers[MA_MAX_CHANNELS];
ma_uint32 iChannel;
ma_uint32 framesToSend = sizeof(tempBuffer) / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
if (framesToSend > framesRemaining) {
framesToSend = framesRemaining;
}
for (iChannel = 0; iChannel < pDevice->capture.internalChannels; ++iChannel) {
ppDeinterleavedBuffers[iChannel] = (void*)ma_offset_ptr(pRenderedBufferList->mBuffers[iBuffer+iChannel].mData, (frameCount - framesRemaining) * ma_get_bytes_per_sample(pDevice->capture.internalFormat));
}
ma_interleave_pcm_frames(pDevice->capture.internalFormat, pDevice->capture.internalChannels, framesToSend, (const void**)ppDeinterleavedBuffers, tempBuffer);
ma_device_handle_backend_data_callback(pDevice, NULL, tempBuffer, framesToSend);
framesRemaining -= framesToSend;
}
}
}
}
(void)pActionFlags;
(void)pTimeStamp;
(void)busNumber;
(void)frameCount;
(void)pUnusedBufferList;
return noErr;
}
static void on_start_stop__coreaudio(void* pUserData, AudioUnit audioUnit, AudioUnitPropertyID propertyID, AudioUnitScope scope, AudioUnitElement element)
{
ma_device* pDevice = (ma_device*)pUserData;
MA_ASSERT(pDevice != NULL);
/* Don't do anything if it looks like we're just reinitializing due to a device switch. */
if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isSwitchingPlaybackDevice) ||
((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isSwitchingCaptureDevice)) {
return;
}
/*
There's been a report of a deadlock here when triggered by ma_device_uninit(). It looks like
AudioUnitGetProprty (called below) and AudioComponentInstanceDispose (called in ma_device_uninit)
can try waiting on the same lock. I'm going to try working around this by not calling any Core
Audio APIs in the callback when the device has been stopped or uninitialized.
*/
if (ma_device_get_state(pDevice) == ma_device_state_uninitialized || ma_device_get_state(pDevice) == ma_device_state_stopping || ma_device_get_state(pDevice) == ma_device_state_stopped) {
ma_device__on_notification_stopped(pDevice);
} else {
UInt32 isRunning;
UInt32 isRunningSize = sizeof(isRunning);
OSStatus status = ((ma_AudioUnitGetProperty_proc)pDevice->pContext->coreaudio.AudioUnitGetProperty)(audioUnit, kAudioOutputUnitProperty_IsRunning, scope, element, &isRunning, &isRunningSize);
if (status != noErr) {
goto done; /* Don't really know what to do in this case... just ignore it, I suppose... */
}
if (!isRunning) {
/*
The stop event is a bit annoying in Core Audio because it will be called when we automatically switch the default device. Some scenarios to consider:
1) When the device is unplugged, this will be called _before_ the default device change notification.
2) When the device is changed via the default device change notification, this will be called _after_ the switch.
For case #1, we just check if there's a new default device available. If so, we just ignore the stop event. For case #2 we check a flag.
*/
if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isDefaultPlaybackDevice) ||
((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isDefaultCaptureDevice)) {
/*
It looks like the device is switching through an external event, such as the user unplugging the device or changing the default device
via the operating system's sound settings. If we're re-initializing the device, we just terminate because we want the stopping of the
device to be seamless to the client (we don't want them receiving the stopped event and thinking that the device has stopped when it
hasn't!).
*/
if (((audioUnit == pDevice->coreaudio.audioUnitPlayback) && pDevice->coreaudio.isSwitchingPlaybackDevice) ||
((audioUnit == pDevice->coreaudio.audioUnitCapture) && pDevice->coreaudio.isSwitchingCaptureDevice)) {
goto done;
}
/*
Getting here means the device is not reinitializing which means it may have been unplugged. From what I can see, it looks like Core Audio
will try switching to the new default device seamlessly. We need to somehow find a way to determine whether or not Core Audio will most
likely be successful in switching to the new device.
TODO: Try to predict if Core Audio will switch devices. If not, the stopped callback needs to be posted.
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_StreamFormat, formatScope, formatElement, &bestFormat, sizeof(bestFormat));
if (status != noErr) {
((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
return ma_result_from_OSStatus(status);
}
#endif
result = ma_format_from_AudioStreamBasicDescription(&bestFormat, &pData->formatOut);
if (result != MA_SUCCESS) {
((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
return result;
}
if (pData->formatOut == ma_format_unknown) {
((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
return MA_FORMAT_NOT_SUPPORTED;
}
pData->channelsOut = bestFormat.mChannelsPerFrame;
pData->sampleRateOut = bestFormat.mSampleRate;
}
/* Clamp the channel count for safety. */
if (pData->channelsOut > MA_MAX_CHANNELS) {
pData->channelsOut = MA_MAX_CHANNELS;
}
/*
Internal channel map. This is weird in my testing. If I use the AudioObject to get the
channel map, the channel descriptions are set to "Unknown" for some reason. To work around
this it looks like retrieving it from the AudioUnit will work. However, and this is where
it gets weird, it doesn't seem to work with capture devices, nor at all on iOS... Therefore
I'm going to fall back to a default assumption in these cases.
*/
#if defined(MA_APPLE_DESKTOP)
result = ma_get_AudioUnit_channel_map(pContext, pData->audioUnit, deviceType, pData->channelMapOut, pData->channelsOut);
if (result != MA_SUCCESS) {
#if 0
/* Try falling back to the channel map from the AudioObject. */
result = ma_get_AudioObject_channel_map(pContext, deviceObjectID, deviceType, pData->channelMapOut, pData->channelsOut);
if (result != MA_SUCCESS) {
return result;
}
#else
/* Fall back to default assumptions. */
ma_channel_map_init_standard(ma_standard_channel_map_default, pData->channelMapOut, ma_countof(pData->channelMapOut), pData->channelsOut);
#endif
}
#else
/* TODO: Figure out how to get the channel map using AVAudioSession. */
ma_channel_map_init_standard(ma_standard_channel_map_default, pData->channelMapOut, ma_countof(pData->channelMapOut), pData->channelsOut);
#endif
/* Buffer size. Not allowing this to be configurable on iOS. */
if (pData->periodSizeInFramesIn == 0) {
if (pData->periodSizeInMillisecondsIn == 0) {
if (pData->performanceProfile == ma_performance_profile_low_latency) {
actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, pData->sampleRateOut);
} else {
actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, pData->sampleRateOut);
}
} else {
actualPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pData->periodSizeInMillisecondsIn, pData->sampleRateOut);
}
} else {
actualPeriodSizeInFrames = pData->periodSizeInFramesIn;
}
#if defined(MA_APPLE_DESKTOP)
result = ma_set_AudioObject_buffer_size_in_frames(pContext, deviceObjectID, deviceType, &actualPeriodSizeInFrames);
if (result != MA_SUCCESS) {
return result;
}
#else
/*
On iOS, the size of the IO buffer needs to be specified in seconds and is a floating point
number. I don't trust any potential truncation errors due to converting from float to integer
so I'm going to explicitly set the actual period size to the next power of 2.
*/
@autoreleasepool {
AVAudioSession* pAudioSession = [AVAudioSession sharedInstance];
MA_ASSERT(pAudioSession != NULL);
[pAudioSession setPreferredIOBufferDuration:((float)actualPeriodSizeInFrames / pAudioSession.sampleRate) error:nil];
actualPeriodSizeInFrames = ma_next_power_of_2((ma_uint32)(pAudioSession.IOBufferDuration * pAudioSession.sampleRate));
}
#endif
/*
During testing I discovered that the buffer size can be too big. You'll get an error like this:
kAudioUnitErr_TooManyFramesToProcess : inFramesToProcess=4096, mMaxFramesPerSlice=512
Note how inFramesToProcess is smaller than mMaxFramesPerSlice. To fix, we need to set kAudioUnitProperty_MaximumFramesPerSlice to that
of the size of our buffer, or do it the other way around and set our buffer size to the kAudioUnitProperty_MaximumFramesPerSlice.
*/
status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &actualPeriodSizeInFrames, sizeof(actualPeriodSizeInFrames));
if (status != noErr) {
((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
return ma_result_from_OSStatus(status);
}
pData->periodSizeInFramesOut = (ma_uint32)actualPeriodSizeInFrames;
/* We need a buffer list if this is an input device. We render into this in the input callback. */
if (deviceType == ma_device_type_capture) {
ma_bool32 isInterleaved = (bestFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) == 0;
AudioBufferList* pBufferList;
pBufferList = ma_allocate_AudioBufferList__coreaudio(pData->periodSizeInFramesOut, pData->formatOut, pData->channelsOut, (isInterleaved) ? ma_stream_layout_interleaved : ma_stream_layout_deinterleaved, &pContext->allocationCallbacks);
if (pBufferList == NULL) {
((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
return MA_OUT_OF_MEMORY;
}
pData->pAudioBufferList = pBufferList;
}
/* Callbacks. */
callbackInfo.inputProcRefCon = pDevice_DoNotReference;
if (deviceType == ma_device_type_playback) {
callbackInfo.inputProc = ma_on_output__coreaudio;
status = ((ma_AudioUnitSetProperty_proc)pContext->coreaudio.AudioUnitSetProperty)(pData->audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, 0, &callbackInfo, sizeof(callbackInfo));
if (status != noErr) {
((ma_AudioComponentInstanceDispose_proc)pContext->coreaudio.AudioComponentInstanceDispose)(pData->audioUnit);
return ma_result_from_OSStatus(status);
}
} else {
callbackInfo.inputProc = ma_on_input__coreaudio;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
#endif
static ma_result ma_context_init__coreaudio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
{
#if !defined(MA_APPLE_MOBILE)
ma_result result;
#endif
MA_ASSERT(pConfig != NULL);
MA_ASSERT(pContext != NULL);
#if defined(MA_APPLE_MOBILE)
@autoreleasepool {
AVAudioSession* pAudioSession = [AVAudioSession sharedInstance];
AVAudioSessionCategoryOptions options = pConfig->coreaudio.sessionCategoryOptions;
MA_ASSERT(pAudioSession != NULL);
if (pConfig->coreaudio.sessionCategory == ma_ios_session_category_default) {
/*
I'm going to use trial and error to determine our default session category. First we'll try PlayAndRecord. If that fails
we'll try Playback and if that fails we'll try record. If all of these fail we'll just not set the category.
*/
#if !defined(MA_APPLE_TV) && !defined(MA_APPLE_WATCH)
options |= AVAudioSessionCategoryOptionDefaultToSpeaker;
#endif
if ([pAudioSession setCategory: AVAudioSessionCategoryPlayAndRecord withOptions:options error:nil]) {
/* Using PlayAndRecord */
} else if ([pAudioSession setCategory: AVAudioSessionCategoryPlayback withOptions:options error:nil]) {
/* Using Playback */
} else if ([pAudioSession setCategory: AVAudioSessionCategoryRecord withOptions:options error:nil]) {
/* Using Record */
} else {
/* Leave as default? */
}
} else {
if (pConfig->coreaudio.sessionCategory != ma_ios_session_category_none) {
#if defined(__IPHONE_12_0)
if (![pAudioSession setCategory: ma_to_AVAudioSessionCategory(pConfig->coreaudio.sessionCategory) withOptions:options error:nil]) {
return MA_INVALID_OPERATION; /* Failed to set session category. */
}
#else
/* Ignore the session category on version 11 and older, but post a warning. */
ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_WARNING, "Session category only supported in iOS 12 and newer.");
#endif
}
}
if (!pConfig->coreaudio.noAudioSessionActivate) {
if (![pAudioSession setActive:true error:nil]) {
ma_log_postf(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "Failed to activate audio session.");
return MA_FAILED_TO_INIT_BACKEND;
}
}
}
#endif
#if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE)
pContext->coreaudio.hCoreFoundation = ma_dlopen(pContext, "CoreFoundation.framework/CoreFoundation");
if (pContext->coreaudio.hCoreFoundation == NULL) {
return MA_API_NOT_FOUND;
}
pContext->coreaudio.CFStringGetCString = ma_dlsym(pContext, pContext->coreaudio.hCoreFoundation, "CFStringGetCString");
pContext->coreaudio.CFRelease = ma_dlsym(pContext, pContext->coreaudio.hCoreFoundation, "CFRelease");
pContext->coreaudio.hCoreAudio = ma_dlopen(pContext, "CoreAudio.framework/CoreAudio");
if (pContext->coreaudio.hCoreAudio == NULL) {
ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
return MA_API_NOT_FOUND;
}
pContext->coreaudio.AudioObjectGetPropertyData = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectGetPropertyData");
pContext->coreaudio.AudioObjectGetPropertyDataSize = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectGetPropertyDataSize");
pContext->coreaudio.AudioObjectSetPropertyData = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectSetPropertyData");
pContext->coreaudio.AudioObjectAddPropertyListener = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectAddPropertyListener");
pContext->coreaudio.AudioObjectRemovePropertyListener = ma_dlsym(pContext, pContext->coreaudio.hCoreAudio, "AudioObjectRemovePropertyListener");
/*
It looks like Apple has moved some APIs from AudioUnit into AudioToolbox on more recent versions of macOS. They are still
defined in AudioUnit, but just in case they decide to remove them from there entirely I'm going to implement a fallback.
The way it'll work is that it'll first try AudioUnit, and if the required symbols are not present there we'll fall back to
AudioToolbox.
*/
pContext->coreaudio.hAudioUnit = ma_dlopen(pContext, "AudioUnit.framework/AudioUnit");
if (pContext->coreaudio.hAudioUnit == NULL) {
ma_dlclose(pContext, pContext->coreaudio.hCoreAudio);
ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
return MA_API_NOT_FOUND;
}
if (ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentFindNext") == NULL) {
/* Couldn't find the required symbols in AudioUnit, so fall back to AudioToolbox. */
ma_dlclose(pContext, pContext->coreaudio.hAudioUnit);
pContext->coreaudio.hAudioUnit = ma_dlopen(pContext, "AudioToolbox.framework/AudioToolbox");
if (pContext->coreaudio.hAudioUnit == NULL) {
ma_dlclose(pContext, pContext->coreaudio.hCoreAudio);
ma_dlclose(pContext, pContext->coreaudio.hCoreFoundation);
return MA_API_NOT_FOUND;
}
}
pContext->coreaudio.AudioComponentFindNext = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentFindNext");
pContext->coreaudio.AudioComponentInstanceDispose = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentInstanceDispose");
pContext->coreaudio.AudioComponentInstanceNew = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioComponentInstanceNew");
pContext->coreaudio.AudioOutputUnitStart = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioOutputUnitStart");
pContext->coreaudio.AudioOutputUnitStop = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioOutputUnitStop");
pContext->coreaudio.AudioUnitAddPropertyListener = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitAddPropertyListener");
pContext->coreaudio.AudioUnitGetPropertyInfo = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitGetPropertyInfo");
pContext->coreaudio.AudioUnitGetProperty = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitGetProperty");
pContext->coreaudio.AudioUnitSetProperty = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitSetProperty");
pContext->coreaudio.AudioUnitInitialize = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitInitialize");
pContext->coreaudio.AudioUnitRender = ma_dlsym(pContext, pContext->coreaudio.hAudioUnit, "AudioUnitRender");
#else
pContext->coreaudio.CFStringGetCString = (ma_proc)CFStringGetCString;
pContext->coreaudio.CFRelease = (ma_proc)CFRelease;
#if defined(MA_APPLE_DESKTOP)
pContext->coreaudio.AudioObjectGetPropertyData = (ma_proc)AudioObjectGetPropertyData;
pContext->coreaudio.AudioObjectGetPropertyDataSize = (ma_proc)AudioObjectGetPropertyDataSize;
pContext->coreaudio.AudioObjectSetPropertyData = (ma_proc)AudioObjectSetPropertyData;
pContext->coreaudio.AudioObjectAddPropertyListener = (ma_proc)AudioObjectAddPropertyListener;
pContext->coreaudio.AudioObjectRemovePropertyListener = (ma_proc)AudioObjectRemovePropertyListener;
#endif
pContext->coreaudio.AudioComponentFindNext = (ma_proc)AudioComponentFindNext;
pContext->coreaudio.AudioComponentInstanceDispose = (ma_proc)AudioComponentInstanceDispose;
pContext->coreaudio.AudioComponentInstanceNew = (ma_proc)AudioComponentInstanceNew;
pContext->coreaudio.AudioOutputUnitStart = (ma_proc)AudioOutputUnitStart;
pContext->coreaudio.AudioOutputUnitStop = (ma_proc)AudioOutputUnitStop;
pContext->coreaudio.AudioUnitAddPropertyListener = (ma_proc)AudioUnitAddPropertyListener;
pContext->coreaudio.AudioUnitGetPropertyInfo = (ma_proc)AudioUnitGetPropertyInfo;
pContext->coreaudio.AudioUnitGetProperty = (ma_proc)AudioUnitGetProperty;
pContext->coreaudio.AudioUnitSetProperty = (ma_proc)AudioUnitSetProperty;
pContext->coreaudio.AudioUnitInitialize = (ma_proc)AudioUnitInitialize;
pContext->coreaudio.AudioUnitRender = (ma_proc)AudioUnitRender;
#endif
/* Audio component. */
{
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
#if defined(MA_APPLE_DESKTOP)
desc.componentSubType = kAudioUnitSubType_HALOutput;
#else
desc.componentSubType = kAudioUnitSubType_RemoteIO;
#endif
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
pContext->coreaudio.component = ((ma_AudioComponentFindNext_proc)pContext->coreaudio.AudioComponentFindNext)(NULL, &desc);
if (pContext->coreaudio.component == NULL) {
#if !defined(MA_NO_RUNTIME_LINKING) && !defined(MA_APPLE_MOBILE)
ma_dlclose(pContext, pContext->coreaudio.hAudioUnit);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
if (channels == 0) {
if (strlen(pDeviceName) > strlen("rsnd/") && strncmp(pDeviceName, "rsnd/", strlen("rsnd/")) == 0) {
channels = ma_find_best_channels_from_sio_cap__sndio(&caps, deviceType, format);
} else {
channels = MA_DEFAULT_CHANNELS;
}
}
}
if (sampleRate == 0) {
sampleRate = ma_find_best_sample_rate_from_sio_cap__sndio(&caps, pConfig->deviceType, format, channels);
}
((ma_sio_initpar_proc)pDevice->pContext->sndio.sio_initpar)(&par);
par.msb = 0;
par.le = ma_is_little_endian();
switch (format) {
case ma_format_u8:
{
par.bits = 8;
par.bps = 1;
par.sig = 0;
} break;
case ma_format_s24:
{
par.bits = 24;
par.bps = 3;
par.sig = 1;
} break;
case ma_format_s32:
{
par.bits = 32;
par.bps = 4;
par.sig = 1;
} break;
case ma_format_s16:
case ma_format_f32:
case ma_format_unknown:
default:
{
par.bits = 16;
par.bps = 2;
par.sig = 1;
} break;
}
if (deviceType == ma_device_type_capture) {
par.rchan = channels;
} else {
par.pchan = channels;
}
par.rate = sampleRate;
internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, par.rate, pConfig->performanceProfile);
par.round = internalPeriodSizeInFrames;
par.appbufsz = par.round * pDescriptor->periodCount;
if (((ma_sio_setpar_proc)pDevice->pContext->sndio.sio_setpar)((struct ma_sio_hdl*)handle, &par) == 0) {
((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)handle);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to set buffer size.");
return MA_ERROR;
}
if (((ma_sio_getpar_proc)pDevice->pContext->sndio.sio_getpar)((struct ma_sio_hdl*)handle, &par) == 0) {
((ma_sio_close_proc)pDevice->pContext->sndio.sio_close)((struct ma_sio_hdl*)handle);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to retrieve buffer size.");
return MA_ERROR;
}
internalFormat = ma_format_from_sio_enc__sndio(par.bits, par.bps, par.sig, par.le, par.msb);
internalChannels = (deviceType == ma_device_type_capture) ? par.rchan : par.pchan;
internalSampleRate = par.rate;
internalPeriods = par.appbufsz / par.round;
internalPeriodSizeInFrames = par.round;
if (deviceType == ma_device_type_capture) {
pDevice->sndio.handleCapture = handle;
} else {
pDevice->sndio.handlePlayback = handle;
}
pDescriptor->format = internalFormat;
pDescriptor->channels = internalChannels;
pDescriptor->sampleRate = internalSampleRate;
ma_channel_map_init_standard(ma_standard_channel_map_sndio, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), internalChannels);
pDescriptor->periodSizeInFrames = internalPeriodSizeInFrames;
pDescriptor->periodCount = internalPeriods;
return MA_SUCCESS;
}
static ma_result ma_device_init__sndio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
MA_ASSERT(pDevice != NULL);
MA_ZERO_OBJECT(&pDevice->sndio);
if (pConfig->deviceType == ma_device_type_loopback) {
return MA_DEVICE_TYPE_NOT_SUPPORTED;
}
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
ma_result result = ma_device_init_handle__sndio(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
if (result != MA_SUCCESS) {
return result;
}
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
ma_result result = ma_device_init_handle__sndio(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
if (result != MA_SUCCESS) {
return result;
}
}
return MA_SUCCESS;
}
static ma_result ma_device_start__sndio(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
((ma_sio_start_proc)pDevice->pContext->sndio.sio_start)((struct ma_sio_hdl*)pDevice->sndio.handleCapture);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
((ma_sio_start_proc)pDevice->pContext->sndio.sio_start)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback); /* <-- Doesn't actually playback until data is written. */
}
return MA_SUCCESS;
}
static ma_result ma_device_stop__sndio(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
/*
From the documentation:
The sio_stop() function puts the audio subsystem in the same state as before sio_start() is called. It stops recording, drains the play buffer and then
stops playback. If samples to play are queued but playback hasn't started yet then playback is forced immediately; playback will actually stop once the
buffer is drained. In no case are samples in the play buffer discarded.
Therefore, sio_stop() performs all of the necessary draining for us.
*/
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
((ma_sio_stop_proc)pDevice->pContext->sndio.sio_stop)((struct ma_sio_hdl*)pDevice->sndio.handleCapture);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
((ma_sio_stop_proc)pDevice->pContext->sndio.sio_stop)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback);
}
return MA_SUCCESS;
}
static ma_result ma_device_write__sndio(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
{
int result;
if (pFramesWritten != NULL) {
*pFramesWritten = 0;
}
result = ((ma_sio_write_proc)pDevice->pContext->sndio.sio_write)((struct ma_sio_hdl*)pDevice->sndio.handlePlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
if (result == 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to send data from the client to the device.");
return MA_IO_ERROR;
}
if (pFramesWritten != NULL) {
*pFramesWritten = frameCount;
}
return MA_SUCCESS;
}
static ma_result ma_device_read__sndio(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
{
int result;
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
result = ((ma_sio_read_proc)pDevice->pContext->sndio.sio_read)((struct ma_sio_hdl*)pDevice->sndio.handleCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
if (result == 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[sndio] Failed to read data from the device to be sent to the device.");
return MA_IO_ERROR;
}
if (pFramesRead != NULL) {
*pFramesRead = frameCount;
}
return MA_SUCCESS;
}
static ma_result ma_context_uninit__sndio(ma_context* pContext)
{
MA_ASSERT(pContext != NULL);
MA_ASSERT(pContext->backend == ma_backend_sndio);
(void)pContext;
return MA_SUCCESS;
}
static ma_result ma_context_init__sndio(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
{
#ifndef MA_NO_RUNTIME_LINKING
const char* libsndioNames[] = {
"libsndio.so"
};
size_t i;
for (i = 0; i < ma_countof(libsndioNames); ++i) {
pContext->sndio.sndioSO = ma_dlopen(pContext, libsndioNames[i]);
if (pContext->sndio.sndioSO != NULL) {
break;
}
}
if (pContext->sndio.sndioSO == NULL) {
return MA_NO_BACKEND;
}
pContext->sndio.sio_open = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_open");
pContext->sndio.sio_close = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_close");
pContext->sndio.sio_setpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_setpar");
pContext->sndio.sio_getpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_getpar");
pContext->sndio.sio_getcap = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_getcap");
pContext->sndio.sio_write = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_write");
pContext->sndio.sio_read = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_read");
pContext->sndio.sio_start = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_start");
pContext->sndio.sio_stop = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_stop");
pContext->sndio.sio_initpar = (ma_proc)ma_dlsym(pContext, pContext->sndio.sndioSO, "sio_initpar");
#else
pContext->sndio.sio_open = sio_open;
pContext->sndio.sio_close = sio_close;
pContext->sndio.sio_setpar = sio_setpar;
pContext->sndio.sio_getpar = sio_getpar;
pContext->sndio.sio_getcap = sio_getcap;
pContext->sndio.sio_write = sio_write;
pContext->sndio.sio_read = sio_read;
pContext->sndio.sio_start = sio_start;
pContext->sndio.sio_stop = sio_stop;
pContext->sndio.sio_initpar = sio_initpar;
#endif
pCallbacks->onContextInit = ma_context_init__sndio;
pCallbacks->onContextUninit = ma_context_uninit__sndio;
pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__sndio;
pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__sndio;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
*/
AUDIO_INITINFO(&fdInfo);
/* We get the driver to do as much of the data conversion as possible. */
if (deviceType == ma_device_type_capture) {
fdInfo.mode = AUMODE_RECORD;
ma_encoding_from_format__audio4(ma_best_format_from_fd__audio4(fd, pDescriptor->format), &fdInfo.record.encoding, &fdInfo.record.precision);
if (pDescriptor->channels != 0) {
fdInfo.record.channels = ma_clamp(pDescriptor->channels, 1, 12); /* From the documentation: `channels` ranges from 1 to 12. */
}
if (pDescriptor->sampleRate != 0) {
fdInfo.record.sample_rate = ma_clamp(pDescriptor->sampleRate, 1000, 192000); /* From the documentation: `frequency` ranges from 1000Hz to 192000Hz. (They mean `sample_rate` instead of `frequency`.) */
}
} else {
fdInfo.mode = AUMODE_PLAY;
ma_encoding_from_format__audio4(ma_best_format_from_fd__audio4(fd, pDescriptor->format), &fdInfo.play.encoding, &fdInfo.play.precision);
if (pDescriptor->channels != 0) {
fdInfo.play.channels = ma_clamp(pDescriptor->channels, 1, 12); /* From the documentation: `channels` ranges from 1 to 12. */
}
if (pDescriptor->sampleRate != 0) {
fdInfo.play.sample_rate = ma_clamp(pDescriptor->sampleRate, 1000, 192000); /* From the documentation: `frequency` ranges from 1000Hz to 192000Hz. (They mean `sample_rate` instead of `frequency`.) */
}
}
if (ioctl(fd, AUDIO_SETINFO, &fdInfo) < 0) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to set device format. AUDIO_SETINFO failed.");
return ma_result_from_errno(errno);
}
if (ioctl(fd, AUDIO_GETINFO, &fdInfo) < 0) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] AUDIO_GETINFO failed.");
return ma_result_from_errno(errno);
}
if (deviceType == ma_device_type_capture) {
internalFormat = ma_format_from_prinfo__audio4(&fdInfo.record);
internalChannels = fdInfo.record.channels;
internalSampleRate = fdInfo.record.sample_rate;
} else {
internalFormat = ma_format_from_prinfo__audio4(&fdInfo.play);
internalChannels = fdInfo.play.channels;
internalSampleRate = fdInfo.play.sample_rate;
}
if (internalFormat == ma_format_unknown) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.");
return MA_FORMAT_NOT_SUPPORTED;
}
/* Buffer. */
{
ma_uint32 internalPeriodSizeInBytes;
internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, internalSampleRate, pConfig->performanceProfile);
internalPeriodSizeInBytes = internalPeriodSizeInFrames * ma_get_bytes_per_frame(internalFormat, internalChannels);
if (internalPeriodSizeInBytes < 16) {
internalPeriodSizeInBytes = 16;
}
internalPeriods = pDescriptor->periodCount;
if (internalPeriods < 2) {
internalPeriods = 2;
}
/* What miniaudio calls a period, audio4 calls a block. */
AUDIO_INITINFO(&fdInfo);
fdInfo.hiwat = internalPeriods;
fdInfo.lowat = internalPeriods-1;
fdInfo.blocksize = internalPeriodSizeInBytes;
if (ioctl(fd, AUDIO_SETINFO, &fdInfo) < 0) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to set internal buffer size. AUDIO_SETINFO failed.");
return ma_result_from_errno(errno);
}
internalPeriods = fdInfo.hiwat;
internalPeriodSizeInFrames = fdInfo.blocksize / ma_get_bytes_per_frame(internalFormat, internalChannels);
}
}
#else
{
struct audio_swpar fdPar;
/* We need to retrieve the format of the device so we can know the channel count and sample rate. Then we can calculate the buffer size. */
if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to retrieve initial device parameters.");
return ma_result_from_errno(errno);
}
internalFormat = ma_format_from_swpar__audio4(&fdPar);
internalChannels = (deviceType == ma_device_type_capture) ? fdPar.rchan : fdPar.pchan;
internalSampleRate = fdPar.rate;
if (internalFormat == ma_format_unknown) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.");
return MA_FORMAT_NOT_SUPPORTED;
}
/* Buffer. */
{
ma_uint32 internalPeriodSizeInBytes;
internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, internalSampleRate, pConfig->performanceProfile);
/* What miniaudio calls a period, audio4 calls a block. */
internalPeriodSizeInBytes = internalPeriodSizeInFrames * ma_get_bytes_per_frame(internalFormat, internalChannels);
if (internalPeriodSizeInBytes < 16) {
internalPeriodSizeInBytes = 16;
}
fdPar.nblks = pDescriptor->periodCount;
fdPar.round = internalPeriodSizeInBytes;
if (ioctl(fd, AUDIO_SETPAR, &fdPar) < 0) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to set device parameters.");
return ma_result_from_errno(errno);
}
if (ioctl(fd, AUDIO_GETPAR, &fdPar) < 0) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to retrieve actual device parameters.");
return ma_result_from_errno(errno);
}
}
internalFormat = ma_format_from_swpar__audio4(&fdPar);
internalChannels = (deviceType == ma_device_type_capture) ? fdPar.rchan : fdPar.pchan;
internalSampleRate = fdPar.rate;
internalPeriods = fdPar.nblks;
internalPeriodSizeInFrames = fdPar.round / ma_get_bytes_per_frame(internalFormat, internalChannels);
}
#endif
if (internalFormat == ma_format_unknown) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] The device's internal device format is not supported by miniaudio. The device is unusable.");
return MA_FORMAT_NOT_SUPPORTED;
}
if (deviceType == ma_device_type_capture) {
pDevice->audio4.fdCapture = fd;
} else {
pDevice->audio4.fdPlayback = fd;
}
pDescriptor->format = internalFormat;
pDescriptor->channels = internalChannels;
pDescriptor->sampleRate = internalSampleRate;
ma_channel_map_init_standard(ma_standard_channel_map_sound4, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), internalChannels);
pDescriptor->periodSizeInFrames = internalPeriodSizeInFrames;
pDescriptor->periodCount = internalPeriods;
return MA_SUCCESS;
}
static ma_result ma_device_init__audio4(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
MA_ASSERT(pDevice != NULL);
MA_ZERO_OBJECT(&pDevice->audio4);
if (pConfig->deviceType == ma_device_type_loopback) {
return MA_DEVICE_TYPE_NOT_SUPPORTED;
}
pDevice->audio4.fdCapture = -1;
pDevice->audio4.fdPlayback = -1;
/*
The version of the operating system dictates whether or not the device is exclusive or shared. NetBSD
introduced in-kernel mixing which means it's shared. All other BSD flavours are exclusive as far as
I'm aware.
*/
#if defined(__NetBSD_Version__) && __NetBSD_Version__ >= 800000000
/* NetBSD 8.0+ */
if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
return MA_SHARE_MODE_NOT_SUPPORTED;
}
#else
/* All other flavors. */
#endif
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
ma_result result = ma_device_init_fd__audio4(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
if (result != MA_SUCCESS) {
return result;
}
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
if (pDevice->audio4.fdPlayback == -1) {
return MA_INVALID_ARGS;
}
}
return MA_SUCCESS;
}
static ma_result ma_device_stop_fd__audio4(ma_device* pDevice, int fd)
{
if (fd == -1) {
return MA_INVALID_ARGS;
}
#if !defined(MA_AUDIO4_USE_NEW_API)
if (ioctl(fd, AUDIO_FLUSH, 0) < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to stop device. AUDIO_FLUSH failed.");
return ma_result_from_errno(errno);
}
#else
if (ioctl(fd, AUDIO_STOP, 0) < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to stop device. AUDIO_STOP failed.");
return ma_result_from_errno(errno);
}
#endif
return MA_SUCCESS;
}
static ma_result ma_device_stop__audio4(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
ma_result result;
result = ma_device_stop_fd__audio4(pDevice, pDevice->audio4.fdCapture);
if (result != MA_SUCCESS) {
return result;
}
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
ma_result result;
/* Drain the device first. If this fails we'll just need to flush without draining. Unfortunately draining isn't available on newer version of OpenBSD. */
#if !defined(MA_AUDIO4_USE_NEW_API)
ioctl(pDevice->audio4.fdPlayback, AUDIO_DRAIN, 0);
#endif
/* Here is where the device is stopped immediately. */
result = ma_device_stop_fd__audio4(pDevice, pDevice->audio4.fdPlayback);
if (result != MA_SUCCESS) {
return result;
}
}
return MA_SUCCESS;
}
static ma_result ma_device_write__audio4(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
{
int result;
if (pFramesWritten != NULL) {
*pFramesWritten = 0;
}
result = write(pDevice->audio4.fdPlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
if (result < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to write data to the device.");
return ma_result_from_errno(errno);
}
if (pFramesWritten != NULL) {
*pFramesWritten = (ma_uint32)result / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
}
return MA_SUCCESS;
}
static ma_result ma_device_read__audio4(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
{
int result;
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
result = read(pDevice->audio4.fdCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
if (result < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[audio4] Failed to read data from the device.");
return ma_result_from_errno(errno);
}
if (pFramesRead != NULL) {
*pFramesRead = (ma_uint32)result / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
}
return MA_SUCCESS;
}
static ma_result ma_context_uninit__audio4(ma_context* pContext)
{
MA_ASSERT(pContext != NULL);
MA_ASSERT(pContext->backend == ma_backend_audio4);
(void)pContext;
return MA_SUCCESS;
}
static ma_result ma_context_init__audio4(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
{
MA_ASSERT(pContext != NULL);
(void)pConfig;
pCallbacks->onContextInit = ma_context_init__audio4;
pCallbacks->onContextUninit = ma_context_uninit__audio4;
pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__audio4;
pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__audio4;
pCallbacks->onDeviceInit = ma_device_init__audio4;
pCallbacks->onDeviceUninit = ma_device_uninit__audio4;
pCallbacks->onDeviceStart = ma_device_start__audio4;
pCallbacks->onDeviceStop = ma_device_stop__audio4;
pCallbacks->onDeviceRead = ma_device_read__audio4;
pCallbacks->onDeviceWrite = ma_device_write__audio4;
pCallbacks->onDeviceDataLoop = NULL;
return MA_SUCCESS;
}
#endif /* audio4 */
/******************************************************************************
OSS Backend
******************************************************************************/
#ifdef MA_HAS_OSS
#include <sys/ioctl.h>
#include <unistd.h>
#include <fcntl.h>
#include <sys/soundcard.h>
#ifndef SNDCTL_DSP_HALT
#define SNDCTL_DSP_HALT SNDCTL_DSP_RESET
#endif
#define MA_OSS_DEFAULT_DEVICE_NAME "/dev/dsp"
static int ma_open_temp_device__oss()
{
/* The OSS sample code uses "/dev/mixer" as the device for getting system properties so I'm going to do the same. */
int fd = open("/dev/mixer", O_RDONLY, 0);
if (fd >= 0) {
return fd;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
MA_ASSERT(pConfig != NULL);
MA_ASSERT(deviceType != ma_device_type_duplex);
pDeviceID = pDescriptor->pDeviceID;
shareMode = pDescriptor->shareMode;
ossFormat = ma_format_to_oss((pDescriptor->format != ma_format_unknown) ? pDescriptor->format : ma_format_s16); /* Use s16 by default because OSS doesn't like floating point. */
ossChannels = (int)(pDescriptor->channels > 0) ? pDescriptor->channels : MA_DEFAULT_CHANNELS;
ossSampleRate = (int)(pDescriptor->sampleRate > 0) ? pDescriptor->sampleRate : MA_DEFAULT_SAMPLE_RATE;
result = ma_context_open_device__oss(pDevice->pContext, deviceType, pDeviceID, shareMode, &fd);
if (result != MA_SUCCESS) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.");
return result;
}
/*
The OSS documantation is very clear about the order we should be initializing the device's properties:
1) Format
2) Channels
3) Sample rate.
*/
/* Format. */
ossResult = ioctl(fd, SNDCTL_DSP_SETFMT, &ossFormat);
if (ossResult == -1) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to set format.");
return ma_result_from_errno(errno);
}
/* Channels. */
ossResult = ioctl(fd, SNDCTL_DSP_CHANNELS, &ossChannels);
if (ossResult == -1) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to set channel count.");
return ma_result_from_errno(errno);
}
/* Sample Rate. */
ossResult = ioctl(fd, SNDCTL_DSP_SPEED, &ossSampleRate);
if (ossResult == -1) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to set sample rate.");
return ma_result_from_errno(errno);
}
/*
Buffer.
The documentation says that the fragment settings should be set as soon as possible, but I'm not sure if
it should be done before or after format/channels/rate.
OSS wants the fragment size in bytes and a power of 2. When setting, we specify the power, not the actual
value.
*/
{
ma_uint32 periodSizeInFrames;
ma_uint32 periodSizeInBytes;
ma_uint32 ossFragmentSizePower;
periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, (ma_uint32)ossSampleRate, pConfig->performanceProfile);
periodSizeInBytes = ma_round_to_power_of_2(periodSizeInFrames * ma_get_bytes_per_frame(ma_format_from_oss(ossFormat), ossChannels));
if (periodSizeInBytes < 16) {
periodSizeInBytes = 16;
}
ossFragmentSizePower = 4;
periodSizeInBytes >>= 4;
while (periodSizeInBytes >>= 1) {
ossFragmentSizePower += 1;
}
ossFragment = (int)((pConfig->periods << 16) | ossFragmentSizePower);
ossResult = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &ossFragment);
if (ossResult == -1) {
close(fd);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to set fragment size and period count.");
return ma_result_from_errno(errno);
}
}
/* Internal settings. */
if (deviceType == ma_device_type_capture) {
pDevice->oss.fdCapture = fd;
} else {
pDevice->oss.fdPlayback = fd;
}
pDescriptor->format = ma_format_from_oss(ossFormat);
pDescriptor->channels = ossChannels;
pDescriptor->sampleRate = ossSampleRate;
ma_channel_map_init_standard(ma_standard_channel_map_sound4, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), pDescriptor->channels);
pDescriptor->periodCount = (ma_uint32)(ossFragment >> 16);
pDescriptor->periodSizeInFrames = (ma_uint32)(1 << (ossFragment & 0xFFFF)) / ma_get_bytes_per_frame(pDescriptor->format, pDescriptor->channels);
if (pDescriptor->format == ma_format_unknown) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] The device's internal format is not supported by miniaudio.");
return MA_FORMAT_NOT_SUPPORTED;
}
return MA_SUCCESS;
}
static ma_result ma_device_init__oss(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
MA_ASSERT(pDevice != NULL);
MA_ASSERT(pConfig != NULL);
MA_ZERO_OBJECT(&pDevice->oss);
if (pConfig->deviceType == ma_device_type_loopback) {
return MA_DEVICE_TYPE_NOT_SUPPORTED;
}
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
ma_result result = ma_device_init_fd__oss(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
if (result != MA_SUCCESS) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.");
return result;
}
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
ma_result result = ma_device_init_fd__oss(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
if (result != MA_SUCCESS) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open device.");
return result;
}
}
return MA_SUCCESS;
}
/*
Note on Starting and Stopping
=============================
In the past I was using SNDCTL_DSP_HALT to stop the device, however this results in issues when
trying to resume the device again. If we use SNDCTL_DSP_HALT, the next write() or read() will
fail. Instead what we need to do is just not write or read to and from the device when the
device is not running.
As a result, both the start and stop functions for OSS are just empty stubs. The starting and
stopping logic is handled by ma_device_write__oss() and ma_device_read__oss(). These will check
the device state, and if the device is stopped they will simply not do any kind of processing.
The downside to this technique is that I've noticed a fairly lengthy delay in stopping the
device, up to a second. This is on a virtual machine, and as such might just be due to the
virtual drivers, but I'm not fully sure. I am not sure how to work around this problem so for
the moment that's just how it's going to have to be.
When starting the device, OSS will automatically start it when write() or read() is called.
*/
static ma_result ma_device_start__oss(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
/* The device is automatically started with reading and writing. */
(void)pDevice;
return MA_SUCCESS;
}
static ma_result ma_device_stop__oss(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
/* See note above on why this is empty. */
(void)pDevice;
return MA_SUCCESS;
}
static ma_result ma_device_write__oss(ma_device* pDevice, const void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesWritten)
{
int resultOSS;
ma_uint32 deviceState;
if (pFramesWritten != NULL) {
*pFramesWritten = 0;
}
/* Don't do any processing if the device is stopped. */
deviceState = ma_device_get_state(pDevice);
if (deviceState != ma_device_state_started && deviceState != ma_device_state_starting) {
return MA_SUCCESS;
}
resultOSS = write(pDevice->oss.fdPlayback, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
if (resultOSS < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to send data from the client to the device.");
return ma_result_from_errno(errno);
}
if (pFramesWritten != NULL) {
*pFramesWritten = (ma_uint32)resultOSS / ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
}
return MA_SUCCESS;
}
static ma_result ma_device_read__oss(ma_device* pDevice, void* pPCMFrames, ma_uint32 frameCount, ma_uint32* pFramesRead)
{
int resultOSS;
ma_uint32 deviceState;
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
/* Don't do any processing if the device is stopped. */
deviceState = ma_device_get_state(pDevice);
if (deviceState != ma_device_state_started && deviceState != ma_device_state_starting) {
return MA_SUCCESS;
}
resultOSS = read(pDevice->oss.fdCapture, pPCMFrames, frameCount * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels));
if (resultOSS < 0) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OSS] Failed to read data from the device to be sent to the client.");
return ma_result_from_errno(errno);
}
if (pFramesRead != NULL) {
*pFramesRead = (ma_uint32)resultOSS / ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
}
return MA_SUCCESS;
}
static ma_result ma_context_uninit__oss(ma_context* pContext)
{
MA_ASSERT(pContext != NULL);
MA_ASSERT(pContext->backend == ma_backend_oss);
(void)pContext;
return MA_SUCCESS;
}
static ma_result ma_context_init__oss(ma_context* pContext, const ma_context_config* pConfig, ma_backend_callbacks* pCallbacks)
{
int fd;
int ossVersion;
int result;
MA_ASSERT(pContext != NULL);
(void)pConfig;
/* Try opening a temporary device first so we can get version information. This is closed at the end. */
fd = ma_open_temp_device__oss();
if (fd == -1) {
ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to open temporary device for retrieving system properties."); /* Looks liks OSS isn't installed, or there are no available devices. */
return MA_NO_BACKEND;
}
/* Grab the OSS version. */
ossVersion = 0;
result = ioctl(fd, OSS_GETVERSION, &ossVersion);
if (result == -1) {
close(fd);
ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_ERROR, "[OSS] Failed to retrieve OSS version.");
return MA_NO_BACKEND;
}
/* The file handle to temp device is no longer needed. Close ASAP. */
close(fd);
pContext->oss.versionMajor = ((ossVersion & 0xFF0000) >> 16);
pContext->oss.versionMinor = ((ossVersion & 0x00FF00) >> 8);
pCallbacks->onContextInit = ma_context_init__oss;
pCallbacks->onContextUninit = ma_context_uninit__oss;
pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__oss;
pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__oss;
pCallbacks->onDeviceInit = ma_device_init__oss;
pCallbacks->onDeviceUninit = ma_device_uninit__oss;
pCallbacks->onDeviceStart = ma_device_start__oss;
pCallbacks->onDeviceStop = ma_device_stop__oss;
pCallbacks->onDeviceRead = ma_device_read__oss;
pCallbacks->onDeviceWrite = ma_device_write__oss;
pCallbacks->onDeviceDataLoop = NULL;
return MA_SUCCESS;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
{
switch (contentType) {
case ma_aaudio_content_type_movie: return MA_AAUDIO_CONTENT_TYPE_MOVIE;
case ma_aaudio_content_type_music: return MA_AAUDIO_CONTENT_TYPE_MUSIC;
case ma_aaudio_content_type_sonification: return MA_AAUDIO_CONTENT_TYPE_SONIFICATION;
case ma_aaudio_content_type_speech: return MA_AAUDIO_CONTENT_TYPE_SPEECH;
default: break;
}
return MA_AAUDIO_CONTENT_TYPE_SPEECH;
}
static ma_aaudio_input_preset_t ma_to_input_preset__aaudio(ma_aaudio_input_preset inputPreset)
{
switch (inputPreset) {
case ma_aaudio_input_preset_generic: return MA_AAUDIO_INPUT_PRESET_GENERIC;
case ma_aaudio_input_preset_camcorder: return MA_AAUDIO_INPUT_PRESET_CAMCORDER;
case ma_aaudio_input_preset_unprocessed: return MA_AAUDIO_INPUT_PRESET_UNPROCESSED;
case ma_aaudio_input_preset_voice_recognition: return MA_AAUDIO_INPUT_PRESET_VOICE_RECOGNITION;
case ma_aaudio_input_preset_voice_communication: return MA_AAUDIO_INPUT_PRESET_VOICE_COMMUNICATION;
case ma_aaudio_input_preset_voice_performance: return MA_AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE;
default: break;
}
return MA_AAUDIO_INPUT_PRESET_GENERIC;
}
static void ma_stream_error_callback__aaudio(ma_AAudioStream* pStream, void* pUserData, ma_aaudio_result_t error)
{
ma_device* pDevice = (ma_device*)pUserData;
MA_ASSERT(pDevice != NULL);
(void)error;
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[AAudio] ERROR CALLBACK: error=%d, AAudioStream_getState()=%d\n", error, ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream));
/*
From the documentation for AAudio, when a device is disconnected all we can do is stop it. However, we cannot stop it from the callback - we need
to do it from another thread. Therefore we are going to use an event thread for the AAudio backend to do this cleanly and safely.
*/
if (((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream) == MA_AAUDIO_STREAM_STATE_DISCONNECTED) {
/* We need to post a job to the job thread for processing. This will reroute the device by reinitializing the stream. */
ma_result result;
ma_job job = ma_job_init(MA_JOB_TYPE_DEVICE_AAUDIO_REROUTE);
job.data.device.aaudio.reroute.pDevice = pDevice;
if (pStream == pDevice->aaudio.pStreamCapture) {
job.data.device.aaudio.reroute.deviceType = ma_device_type_capture;
} else {
job.data.device.aaudio.reroute.deviceType = ma_device_type_playback;
}
result = ma_device_job_thread_post(&pDevice->pContext->aaudio.jobThread, &job);
if (result != MA_SUCCESS) {
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[AAudio] Device Disconnected. Failed to post job for rerouting.\n");
return;
}
}
}
static ma_aaudio_data_callback_result_t ma_stream_data_callback_capture__aaudio(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t frameCount)
{
ma_device* pDevice = (ma_device*)pUserData;
MA_ASSERT(pDevice != NULL);
ma_device_handle_backend_data_callback(pDevice, NULL, pAudioData, frameCount);
(void)pStream;
return MA_AAUDIO_CALLBACK_RESULT_CONTINUE;
}
static ma_aaudio_data_callback_result_t ma_stream_data_callback_playback__aaudio(ma_AAudioStream* pStream, void* pUserData, void* pAudioData, int32_t frameCount)
{
ma_device* pDevice = (ma_device*)pUserData;
MA_ASSERT(pDevice != NULL);
ma_device_handle_backend_data_callback(pDevice, pAudioData, NULL, frameCount);
(void)pStream;
return MA_AAUDIO_CALLBACK_RESULT_CONTINUE;
}
static ma_result ma_create_and_configure_AAudioStreamBuilder__aaudio(ma_context* pContext, const ma_device_id* pDeviceID, ma_device_type deviceType, ma_share_mode shareMode, const ma_device_descriptor* pDescriptor, const ma_device_config* pConfig, ma...
{
ma_AAudioStreamBuilder* pBuilder;
ma_aaudio_result_t resultAA;
ma_uint32 bufferCapacityInFrames;
/* Safety. */
*ppBuilder = NULL;
resultAA = ((MA_PFN_AAudio_createStreamBuilder)pContext->aaudio.AAudio_createStreamBuilder)(&pBuilder);
if (resultAA != MA_AAUDIO_OK) {
return ma_result_from_aaudio(resultAA);
}
if (pDeviceID != NULL) {
((MA_PFN_AAudioStreamBuilder_setDeviceId)pContext->aaudio.AAudioStreamBuilder_setDeviceId)(pBuilder, pDeviceID->aaudio);
}
((MA_PFN_AAudioStreamBuilder_setDirection)pContext->aaudio.AAudioStreamBuilder_setDirection)(pBuilder, (deviceType == ma_device_type_playback) ? MA_AAUDIO_DIRECTION_OUTPUT : MA_AAUDIO_DIRECTION_INPUT);
((MA_PFN_AAudioStreamBuilder_setSharingMode)pContext->aaudio.AAudioStreamBuilder_setSharingMode)(pBuilder, (shareMode == ma_share_mode_shared) ? MA_AAUDIO_SHARING_MODE_SHARED : MA_AAUDIO_SHARING_MODE_EXCLUSIVE);
/* If we have a device descriptor make sure we configure the stream builder to take our requested parameters. */
if (pDescriptor != NULL) {
MA_ASSERT(pConfig != NULL); /* We must have a device config if we also have a descriptor. The config is required for AAudio specific configuration options. */
if (pDescriptor->sampleRate != 0) {
((MA_PFN_AAudioStreamBuilder_setSampleRate)pContext->aaudio.AAudioStreamBuilder_setSampleRate)(pBuilder, pDescriptor->sampleRate);
}
if (deviceType == ma_device_type_capture) {
if (pDescriptor->channels != 0) {
((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pDescriptor->channels);
}
if (pDescriptor->format != ma_format_unknown) {
((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pDescriptor->format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT);
}
} else {
if (pDescriptor->channels != 0) {
((MA_PFN_AAudioStreamBuilder_setChannelCount)pContext->aaudio.AAudioStreamBuilder_setChannelCount)(pBuilder, pDescriptor->channels);
}
if (pDescriptor->format != ma_format_unknown) {
((MA_PFN_AAudioStreamBuilder_setFormat)pContext->aaudio.AAudioStreamBuilder_setFormat)(pBuilder, (pDescriptor->format == ma_format_s16) ? MA_AAUDIO_FORMAT_PCM_I16 : MA_AAUDIO_FORMAT_PCM_FLOAT);
}
}
/*
AAudio is annoying when it comes to it's buffer calculation stuff because it doesn't let you
retrieve the actual sample rate until after you've opened the stream. But you need to configure
the buffer capacity before you open the stream... :/
To solve, we're just going to assume MA_DEFAULT_SAMPLE_RATE (48000) and move on.
*/
bufferCapacityInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptor, pDescriptor->sampleRate, pConfig->performanceProfile) * pDescriptor->periodCount;
((MA_PFN_AAudioStreamBuilder_setBufferCapacityInFrames)pContext->aaudio.AAudioStreamBuilder_setBufferCapacityInFrames)(pBuilder, bufferCapacityInFrames);
((MA_PFN_AAudioStreamBuilder_setFramesPerDataCallback)pContext->aaudio.AAudioStreamBuilder_setFramesPerDataCallback)(pBuilder, bufferCapacityInFrames / pDescriptor->periodCount);
if (deviceType == ma_device_type_capture) {
if (pConfig->aaudio.inputPreset != ma_aaudio_input_preset_default && pContext->aaudio.AAudioStreamBuilder_setInputPreset != NULL) {
((MA_PFN_AAudioStreamBuilder_setInputPreset)pContext->aaudio.AAudioStreamBuilder_setInputPreset)(pBuilder, ma_to_input_preset__aaudio(pConfig->aaudio.inputPreset));
}
((MA_PFN_AAudioStreamBuilder_setDataCallback)pContext->aaudio.AAudioStreamBuilder_setDataCallback)(pBuilder, ma_stream_data_callback_capture__aaudio, (void*)pDevice);
} else {
if (pConfig->aaudio.usage != ma_aaudio_usage_default && pContext->aaudio.AAudioStreamBuilder_setUsage != NULL) {
((MA_PFN_AAudioStreamBuilder_setUsage)pContext->aaudio.AAudioStreamBuilder_setUsage)(pBuilder, ma_to_usage__aaudio(pConfig->aaudio.usage));
}
if (pConfig->aaudio.contentType != ma_aaudio_content_type_default && pContext->aaudio.AAudioStreamBuilder_setContentType != NULL) {
((MA_PFN_AAudioStreamBuilder_setContentType)pContext->aaudio.AAudioStreamBuilder_setContentType)(pBuilder, ma_to_content_type__aaudio(pConfig->aaudio.contentType));
}
((MA_PFN_AAudioStreamBuilder_setDataCallback)pContext->aaudio.AAudioStreamBuilder_setDataCallback)(pBuilder, ma_stream_data_callback_playback__aaudio, (void*)pDevice);
}
/* Not sure how this affects things, but since there's a mapping between miniaudio's performance profiles and AAudio's performance modes, let go ahead and set it. */
((MA_PFN_AAudioStreamBuilder_setPerformanceMode)pContext->aaudio.AAudioStreamBuilder_setPerformanceMode)(pBuilder, (pConfig->performanceProfile == ma_performance_profile_low_latency) ? MA_AAUDIO_PERFORMANCE_MODE_LOW_LATENCY : MA_AAUDIO_PERFOR...
/* We need to set an error callback to detect device changes. */
if (pDevice != NULL) { /* <-- pDevice should never be null if pDescriptor is not null, which is always the case if we hit this branch. Check anyway for safety. */
((MA_PFN_AAudioStreamBuilder_setErrorCallback)pContext->aaudio.AAudioStreamBuilder_setErrorCallback)(pBuilder, ma_stream_error_callback__aaudio, (void*)pDevice);
}
}
*ppBuilder = pBuilder;
return MA_SUCCESS;
}
static ma_result ma_open_stream_and_close_builder__aaudio(ma_context* pContext, ma_AAudioStreamBuilder* pBuilder, ma_AAudioStream** ppStream)
{
ma_result result;
result = ma_result_from_aaudio(((MA_PFN_AAudioStreamBuilder_openStream)pContext->aaudio.AAudioStreamBuilder_openStream)(pBuilder, ppStream));
((MA_PFN_AAudioStreamBuilder_delete)pContext->aaudio.AAudioStreamBuilder_delete)(pBuilder);
return result;
}
static ma_result ma_open_stream_basic__aaudio(ma_context* pContext, const ma_device_id* pDeviceID, ma_device_type deviceType, ma_share_mode shareMode, ma_AAudioStream** ppStream)
{
ma_result result;
ma_AAudioStreamBuilder* pBuilder;
*ppStream = NULL;
result = ma_create_and_configure_AAudioStreamBuilder__aaudio(pContext, pDeviceID, deviceType, shareMode, NULL, NULL, NULL, &pBuilder);
if (result != MA_SUCCESS) {
return result;
}
return ma_open_stream_and_close_builder__aaudio(pContext, pBuilder, ppStream);
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
static ma_result ma_context_get_device_info__aaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
{
ma_AAudioStream* pStream;
ma_result result;
MA_ASSERT(pContext != NULL);
/* ID */
if (pDeviceID != NULL) {
pDeviceInfo->id.aaudio = pDeviceID->aaudio;
} else {
pDeviceInfo->id.aaudio = MA_AAUDIO_UNSPECIFIED;
}
/* Name */
if (deviceType == ma_device_type_playback) {
ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
} else {
ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
}
pDeviceInfo->nativeDataFormatCount = 0;
/* We'll need to open the device to get accurate sample rate and channel count information. */
result = ma_open_stream_basic__aaudio(pContext, pDeviceID, deviceType, ma_share_mode_shared, &pStream);
if (result != MA_SUCCESS) {
return result;
}
ma_context_add_native_data_format_from_AAudioStream__aaudio(pContext, pStream, 0, pDeviceInfo);
ma_close_stream__aaudio(pContext, pStream);
pStream = NULL;
return MA_SUCCESS;
}
static ma_result ma_device_uninit__aaudio(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamCapture);
pDevice->aaudio.pStreamCapture = NULL;
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
ma_close_stream__aaudio(pDevice->pContext, (ma_AAudioStream*)pDevice->aaudio.pStreamPlayback);
pDevice->aaudio.pStreamPlayback = NULL;
}
return MA_SUCCESS;
}
static ma_result ma_device_init_by_type__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_type deviceType, ma_device_descriptor* pDescriptor, ma_AAudioStream** ppStream)
{
ma_result result;
int32_t bufferCapacityInFrames;
int32_t framesPerDataCallback;
ma_AAudioStream* pStream;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(pConfig != NULL);
MA_ASSERT(pDescriptor != NULL);
*ppStream = NULL; /* Safety. */
/* First step is to open the stream. From there we'll be able to extract the internal configuration. */
result = ma_open_stream__aaudio(pDevice, pConfig, deviceType, pDescriptor, &pStream);
if (result != MA_SUCCESS) {
return result; /* Failed to open the AAudio stream. */
}
/* Now extract the internal configuration. */
pDescriptor->format = (((MA_PFN_AAudioStream_getFormat)pDevice->pContext->aaudio.AAudioStream_getFormat)(pStream) == MA_AAUDIO_FORMAT_PCM_I16) ? ma_format_s16 : ma_format_f32;
pDescriptor->channels = ((MA_PFN_AAudioStream_getChannelCount)pDevice->pContext->aaudio.AAudioStream_getChannelCount)(pStream);
pDescriptor->sampleRate = ((MA_PFN_AAudioStream_getSampleRate)pDevice->pContext->aaudio.AAudioStream_getSampleRate)(pStream);
/* For the channel map we need to be sure we don't overflow any buffers. */
if (pDescriptor->channels <= MA_MAX_CHANNELS) {
ma_channel_map_init_standard(ma_standard_channel_map_default, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), pDescriptor->channels); /* <-- Cannot find info on channel order, so assuming a default. */
} else {
ma_channel_map_init_blank(pDescriptor->channelMap, MA_MAX_CHANNELS); /* Too many channels. Use a blank channel map. */
}
bufferCapacityInFrames = ((MA_PFN_AAudioStream_getBufferCapacityInFrames)pDevice->pContext->aaudio.AAudioStream_getBufferCapacityInFrames)(pStream);
framesPerDataCallback = ((MA_PFN_AAudioStream_getFramesPerDataCallback)pDevice->pContext->aaudio.AAudioStream_getFramesPerDataCallback)(pStream);
if (framesPerDataCallback > 0) {
pDescriptor->periodSizeInFrames = framesPerDataCallback;
pDescriptor->periodCount = bufferCapacityInFrames / framesPerDataCallback;
} else {
pDescriptor->periodSizeInFrames = bufferCapacityInFrames;
pDescriptor->periodCount = 1;
}
*ppStream = pStream;
return MA_SUCCESS;
}
static ma_result ma_device_init__aaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
ma_result result;
MA_ASSERT(pDevice != NULL);
if (pConfig->deviceType == ma_device_type_loopback) {
return MA_DEVICE_TYPE_NOT_SUPPORTED;
}
pDevice->aaudio.usage = pConfig->aaudio.usage;
pDevice->aaudio.contentType = pConfig->aaudio.contentType;
pDevice->aaudio.inputPreset = pConfig->aaudio.inputPreset;
pDevice->aaudio.noAutoStartAfterReroute = pConfig->aaudio.noAutoStartAfterReroute;
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
result = ma_device_init_by_type__aaudio(pDevice, pConfig, ma_device_type_capture, pDescriptorCapture, (ma_AAudioStream**)&pDevice->aaudio.pStreamCapture);
if (result != MA_SUCCESS) {
return result;
}
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
result = ma_device_init_by_type__aaudio(pDevice, pConfig, ma_device_type_playback, pDescriptorPlayback, (ma_AAudioStream**)&pDevice->aaudio.pStreamPlayback);
if (result != MA_SUCCESS) {
return result;
}
}
return MA_SUCCESS;
}
static ma_result ma_device_start_stream__aaudio(ma_device* pDevice, ma_AAudioStream* pStream)
{
ma_aaudio_result_t resultAA;
ma_aaudio_stream_state_t currentState;
MA_ASSERT(pDevice != NULL);
resultAA = ((MA_PFN_AAudioStream_requestStart)pDevice->pContext->aaudio.AAudioStream_requestStart)(pStream);
if (resultAA != MA_AAUDIO_OK) {
return ma_result_from_aaudio(resultAA);
}
/* Do we actually need to wait for the device to transition into it's started state? */
/* The device should be in either a starting or started state. If it's not set to started we need to wait for it to transition. It should go from starting to started. */
currentState = ((MA_PFN_AAudioStream_getState)pDevice->pContext->aaudio.AAudioStream_getState)(pStream);
if (currentState != MA_AAUDIO_STREAM_STATE_STARTED) {
ma_result result;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
if (deviceType == ma_device_type_playback) {
pDeviceInfo->id.opensl = SL_DEFAULTDEVICEID_AUDIOOUTPUT;
ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
} else {
pDeviceInfo->id.opensl = SL_DEFAULTDEVICEID_AUDIOINPUT;
ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
}
pDeviceInfo->isDefault = MA_TRUE;
goto return_detailed_info;
return_detailed_info:
/*
For now we're just outputting a set of values that are supported by the API but not necessarily supported
by the device natively. Later on we should work on this so that it more closely reflects the device's
actual native format.
*/
pDeviceInfo->nativeDataFormatCount = 0;
#if defined(MA_ANDROID) && __ANDROID_API__ >= 21
ma_context_add_data_format__opensl(pContext, ma_format_f32, pDeviceInfo);
#endif
ma_context_add_data_format__opensl(pContext, ma_format_s16, pDeviceInfo);
ma_context_add_data_format__opensl(pContext, ma_format_u8, pDeviceInfo);
return MA_SUCCESS;
}
#ifdef MA_ANDROID
/*void ma_buffer_queue_callback_capture__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, SLuint32 eventFlags, const void* pBuffer, SLuint32 bufferSize, SLuint32 dataUsed, void* pContext)*/
static void ma_buffer_queue_callback_capture__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, void* pUserData)
{
ma_device* pDevice = (ma_device*)pUserData;
size_t periodSizeInBytes;
ma_uint8* pBuffer;
SLresult resultSL;
MA_ASSERT(pDevice != NULL);
(void)pBufferQueue;
/*
For now, don't do anything unless the buffer was fully processed. From what I can tell, it looks like
OpenSL|ES 1.1 improves on buffer queues to the point that we could much more intelligently handle this,
but unfortunately it looks like Android is only supporting OpenSL|ES 1.0.1 for now :(
*/
/* Don't do anything if the device is not started. */
if (ma_device_get_state(pDevice) != ma_device_state_started) {
return;
}
/* Don't do anything if the device is being drained. */
if (pDevice->opensl.isDrainingCapture) {
return;
}
periodSizeInBytes = pDevice->capture.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
pBuffer = pDevice->opensl.pBufferCapture + (pDevice->opensl.currentBufferIndexCapture * periodSizeInBytes);
ma_device_handle_backend_data_callback(pDevice, NULL, pBuffer, pDevice->capture.internalPeriodSizeInFrames);
resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, pBuffer, periodSizeInBytes);
if (resultSL != SL_RESULT_SUCCESS) {
return;
}
pDevice->opensl.currentBufferIndexCapture = (pDevice->opensl.currentBufferIndexCapture + 1) % pDevice->capture.internalPeriods;
}
static void ma_buffer_queue_callback_playback__opensl_android(SLAndroidSimpleBufferQueueItf pBufferQueue, void* pUserData)
{
ma_device* pDevice = (ma_device*)pUserData;
size_t periodSizeInBytes;
ma_uint8* pBuffer;
SLresult resultSL;
MA_ASSERT(pDevice != NULL);
(void)pBufferQueue;
/* Don't do anything if the device is not started. */
if (ma_device_get_state(pDevice) != ma_device_state_started) {
return;
}
/* Don't do anything if the device is being drained. */
if (pDevice->opensl.isDrainingPlayback) {
return;
}
periodSizeInBytes = pDevice->playback.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
pBuffer = pDevice->opensl.pBufferPlayback + (pDevice->opensl.currentBufferIndexPlayback * periodSizeInBytes);
ma_device_handle_backend_data_callback(pDevice, pBuffer, NULL, pDevice->playback.internalPeriodSizeInFrames);
resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, pBuffer, periodSizeInBytes);
if (resultSL != SL_RESULT_SUCCESS) {
return;
}
pDevice->opensl.currentBufferIndexPlayback = (pDevice->opensl.currentBufferIndexPlayback + 1) % pDevice->playback.internalPeriods;
}
#endif
static ma_result ma_device_uninit__opensl(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it before uninitializing the device. */
if (g_maOpenSLInitCounter == 0) {
return MA_INVALID_OPERATION;
}
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
if (pDevice->opensl.pAudioRecorderObj) {
MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->Destroy((SLObjectItf)pDevice->opensl.pAudioRecorderObj);
}
ma_free(pDevice->opensl.pBufferCapture, &pDevice->pContext->allocationCallbacks);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
if (pDevice->opensl.pAudioPlayerObj) {
MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->Destroy((SLObjectItf)pDevice->opensl.pAudioPlayerObj);
}
if (pDevice->opensl.pOutputMixObj) {
MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->Destroy((SLObjectItf)pDevice->opensl.pOutputMixObj);
}
ma_free(pDevice->opensl.pBufferPlayback, &pDevice->pContext->allocationCallbacks);
}
return MA_SUCCESS;
}
#if defined(MA_ANDROID) && __ANDROID_API__ >= 21
typedef SLAndroidDataFormat_PCM_EX ma_SLDataFormat_PCM;
#else
typedef SLDataFormat_PCM ma_SLDataFormat_PCM;
#endif
static ma_result ma_SLDataFormat_PCM_init__opensl(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, const ma_channel* channelMap, ma_SLDataFormat_PCM* pDataFormat)
{
/* We need to convert our format/channels/rate so that they aren't set to default. */
if (format == ma_format_unknown) {
format = MA_DEFAULT_FORMAT;
}
if (channels == 0) {
channels = MA_DEFAULT_CHANNELS;
}
if (sampleRate == 0) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
pcm.formatType = SL_DATAFORMAT_PCM;
pcm.numChannels = 1;
((SLDataFormat_PCM*)&pcm)->samplesPerSec = SL_SAMPLINGRATE_16; /* The name of the sample rate variable is different between SLAndroidDataFormat_PCM_EX and SLDataFormat_PCM. */
pcm.bitsPerSample = 16;
pcm.containerSize = pcm.bitsPerSample; /* Always tightly packed for now. */
pcm.channelMask = 0;
resultSL = (*g_maEngineSL)->CreateAudioRecorder(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioRecorderObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired);
}
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create audio recorder.");
return ma_result_from_OpenSL(resultSL);
}
/* Set the recording preset before realizing the player. */
if (pConfig->opensl.recordingPreset != ma_opensl_recording_preset_default) {
resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDCONFIGURATION, &pRecorderConfig);
if (resultSL == SL_RESULT_SUCCESS) {
SLint32 recordingPreset = ma_to_recording_preset__opensl(pConfig->opensl.recordingPreset);
resultSL = (*pRecorderConfig)->SetConfiguration(pRecorderConfig, SL_ANDROID_KEY_RECORDING_PRESET, &recordingPreset, sizeof(SLint32));
if (resultSL != SL_RESULT_SUCCESS) {
/* Failed to set the configuration. Just keep going. */
}
}
}
resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->Realize((SLObjectItf)pDevice->opensl.pAudioRecorderObj, SL_BOOLEAN_FALSE);
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize audio recorder.");
return ma_result_from_OpenSL(resultSL);
}
resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_RECORD, &pDevice->opensl.pAudioRecorder);
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_RECORD interface.");
return ma_result_from_OpenSL(resultSL);
}
resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioRecorderObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioRecorderObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &pDevice->opensl.pBufferQueueCapture);
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_ANDROIDSIMPLEBUFFERQUEUE interface.");
return ma_result_from_OpenSL(resultSL);
}
resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->RegisterCallback((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, ma_buffer_queue_callback_capture__opensl_android, pDevice);
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to register buffer queue callback.");
return ma_result_from_OpenSL(resultSL);
}
/* The internal format is determined by the "pcm" object. */
ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDescriptorCapture->format, &pDescriptorCapture->channels, &pDescriptorCapture->sampleRate, pDescriptorCapture->channelMap, ma_countof(pDescriptorCapture->channelMap));
/* Buffer. */
pDescriptorCapture->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorCapture, pDescriptorCapture->sampleRate, pConfig->performanceProfile);
pDevice->opensl.currentBufferIndexCapture = 0;
bufferSizeInBytes = pDescriptorCapture->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorCapture->format, pDescriptorCapture->channels) * pDescriptorCapture->periodCount;
pDevice->opensl.pBufferCapture = (ma_uint8*)ma_calloc(bufferSizeInBytes, &pDevice->pContext->allocationCallbacks);
if (pDevice->opensl.pBufferCapture == NULL) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to allocate memory for data buffer.");
return MA_OUT_OF_MEMORY;
}
MA_ZERO_MEMORY(pDevice->opensl.pBufferCapture, bufferSizeInBytes);
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
ma_SLDataFormat_PCM pcm;
SLDataSource source;
SLDataLocator_OutputMix outmixLocator;
SLDataSink sink;
SLAndroidConfigurationItf pPlayerConfig;
ma_SLDataFormat_PCM_init__opensl(pDescriptorPlayback->format, pDescriptorPlayback->channels, pDescriptorPlayback->sampleRate, pDescriptorPlayback->channelMap, &pcm);
resultSL = (*g_maEngineSL)->CreateOutputMix(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pOutputMixObj, 0, NULL, NULL);
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create output mix.");
return ma_result_from_OpenSL(resultSL);
}
resultSL = MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->Realize((SLObjectItf)pDevice->opensl.pOutputMixObj, SL_BOOLEAN_FALSE);
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize output mix object.");
return ma_result_from_OpenSL(resultSL);
}
resultSL = MA_OPENSL_OBJ(pDevice->opensl.pOutputMixObj)->GetInterface((SLObjectItf)pDevice->opensl.pOutputMixObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_OUTPUTMIX, &pDevice->opensl.pOutputMix);
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_OUTPUTMIX interface.");
return ma_result_from_OpenSL(resultSL);
}
/* Set the output device. */
if (pDescriptorPlayback->pDeviceID != NULL) {
SLuint32 deviceID_OpenSL = pDescriptorPlayback->pDeviceID->opensl;
MA_OPENSL_OUTPUTMIX(pDevice->opensl.pOutputMix)->ReRoute((SLOutputMixItf)pDevice->opensl.pOutputMix, 1, &deviceID_OpenSL);
}
queue.numBuffers = pDescriptorPlayback->periodCount;
source.pLocator = &queue;
source.pFormat = (SLDataFormat_PCM*)&pcm;
outmixLocator.locatorType = SL_DATALOCATOR_OUTPUTMIX;
outmixLocator.outputMix = (SLObjectItf)pDevice->opensl.pOutputMixObj;
sink.pLocator = &outmixLocator;
sink.pFormat = NULL;
resultSL = (*g_maEngineSL)->CreateAudioPlayer(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioPlayerObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired);
if (resultSL == SL_RESULT_CONTENT_UNSUPPORTED || resultSL == SL_RESULT_PARAMETER_INVALID) {
/* Unsupported format. Fall back to something safer and try again. If this fails, just abort. */
pcm.formatType = SL_DATAFORMAT_PCM;
pcm.numChannels = 2;
((SLDataFormat_PCM*)&pcm)->samplesPerSec = SL_SAMPLINGRATE_16;
pcm.bitsPerSample = 16;
pcm.containerSize = pcm.bitsPerSample; /* Always tightly packed for now. */
pcm.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
resultSL = (*g_maEngineSL)->CreateAudioPlayer(g_maEngineSL, (SLObjectItf*)&pDevice->opensl.pAudioPlayerObj, &source, &sink, ma_countof(itfIDs), itfIDs, itfIDsRequired);
}
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to create audio player.");
return ma_result_from_OpenSL(resultSL);
}
/* Set the stream type before realizing the player. */
if (pConfig->opensl.streamType != ma_opensl_stream_type_default) {
resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDCONFIGURATION, &pPlayerConfig);
if (resultSL == SL_RESULT_SUCCESS) {
SLint32 streamType = ma_to_stream_type__opensl(pConfig->opensl.streamType);
resultSL = (*pPlayerConfig)->SetConfiguration(pPlayerConfig, SL_ANDROID_KEY_STREAM_TYPE, &streamType, sizeof(SLint32));
if (resultSL != SL_RESULT_SUCCESS) {
/* Failed to set the configuration. Just keep going. */
}
}
}
resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->Realize((SLObjectItf)pDevice->opensl.pAudioPlayerObj, SL_BOOLEAN_FALSE);
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to realize audio player.");
return ma_result_from_OpenSL(resultSL);
}
resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_PLAY, &pDevice->opensl.pAudioPlayer);
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_PLAY interface.");
return ma_result_from_OpenSL(resultSL);
}
resultSL = MA_OPENSL_OBJ(pDevice->opensl.pAudioPlayerObj)->GetInterface((SLObjectItf)pDevice->opensl.pAudioPlayerObj, (SLInterfaceID)pDevice->pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &pDevice->opensl.pBufferQueuePlayback);
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to retrieve SL_IID_ANDROIDSIMPLEBUFFERQUEUE interface.");
return ma_result_from_OpenSL(resultSL);
}
resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->RegisterCallback((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, ma_buffer_queue_callback_playback__opensl_android, pDevice);
if (resultSL != SL_RESULT_SUCCESS) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to register buffer queue callback.");
return ma_result_from_OpenSL(resultSL);
}
/* The internal format is determined by the "pcm" object. */
ma_deconstruct_SLDataFormat_PCM__opensl(&pcm, &pDescriptorPlayback->format, &pDescriptorPlayback->channels, &pDescriptorPlayback->sampleRate, pDescriptorPlayback->channelMap, ma_countof(pDescriptorPlayback->channelMap));
/* Buffer. */
pDescriptorPlayback->periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_descriptor(pDescriptorPlayback, pDescriptorPlayback->sampleRate, pConfig->performanceProfile);
pDevice->opensl.currentBufferIndexPlayback = 0;
bufferSizeInBytes = pDescriptorPlayback->periodSizeInFrames * ma_get_bytes_per_frame(pDescriptorPlayback->format, pDescriptorPlayback->channels) * pDescriptorPlayback->periodCount;
pDevice->opensl.pBufferPlayback = (ma_uint8*)ma_calloc(bufferSizeInBytes, &pDevice->pContext->allocationCallbacks);
if (pDevice->opensl.pBufferPlayback == NULL) {
ma_device_uninit__opensl(pDevice);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to allocate memory for data buffer.");
return MA_OUT_OF_MEMORY;
}
MA_ZERO_MEMORY(pDevice->opensl.pBufferPlayback, bufferSizeInBytes);
}
return MA_SUCCESS;
#else
return MA_NO_BACKEND; /* Non-Android implementations are not supported. */
#endif
}
static ma_result ma_device_start__opensl(ma_device* pDevice)
{
SLresult resultSL;
size_t periodSizeInBytes;
ma_uint32 iPeriod;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it and then attempted to start the device. */
if (g_maOpenSLInitCounter == 0) {
return MA_INVALID_OPERATION;
}
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
resultSL = MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_RECORDING);
if (resultSL != SL_RESULT_SUCCESS) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to start internal capture device.");
return ma_result_from_OpenSL(resultSL);
}
periodSizeInBytes = pDevice->capture.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->capture.internalFormat, pDevice->capture.internalChannels);
for (iPeriod = 0; iPeriod < pDevice->capture.internalPeriods; ++iPeriod) {
resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueueCapture)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture, pDevice->opensl.pBufferCapture + (periodSizeInBytes * iPeriod), periodSizeInBytes);
if (resultSL != SL_RESULT_SUCCESS) {
MA_OPENSL_RECORD(pDevice->opensl.pAudioRecorder)->SetRecordState((SLRecordItf)pDevice->opensl.pAudioRecorder, SL_RECORDSTATE_STOPPED);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to enqueue buffer for capture device.");
return ma_result_from_OpenSL(resultSL);
}
}
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
resultSL = MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_PLAYING);
if (resultSL != SL_RESULT_SUCCESS) {
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to start internal playback device.");
return ma_result_from_OpenSL(resultSL);
}
/* In playback mode (no duplex) we need to load some initial buffers. In duplex mode we need to enqueu silent buffers. */
if (pDevice->type == ma_device_type_duplex) {
MA_ZERO_MEMORY(pDevice->opensl.pBufferPlayback, pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels));
} else {
ma_device__read_frames_from_client(pDevice, pDevice->playback.internalPeriodSizeInFrames * pDevice->playback.internalPeriods, pDevice->opensl.pBufferPlayback);
}
periodSizeInBytes = pDevice->playback.internalPeriodSizeInFrames * ma_get_bytes_per_frame(pDevice->playback.internalFormat, pDevice->playback.internalChannels);
for (iPeriod = 0; iPeriod < pDevice->playback.internalPeriods; ++iPeriod) {
resultSL = MA_OPENSL_BUFFERQUEUE(pDevice->opensl.pBufferQueuePlayback)->Enqueue((SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback, pDevice->opensl.pBufferPlayback + (periodSizeInBytes * iPeriod), periodSizeInBytes);
if (resultSL != SL_RESULT_SUCCESS) {
MA_OPENSL_PLAY(pDevice->opensl.pAudioPlayer)->SetPlayState((SLPlayItf)pDevice->opensl.pAudioPlayer, SL_PLAYSTATE_STOPPED);
ma_log_post(ma_device_get_log(pDevice), MA_LOG_LEVEL_ERROR, "[OpenSL] Failed to enqueue buffer for playback device.");
return ma_result_from_OpenSL(resultSL);
}
}
}
return MA_SUCCESS;
}
static ma_result ma_device_drain__opensl(ma_device* pDevice, ma_device_type deviceType)
{
SLAndroidSimpleBufferQueueItf pBufferQueue;
MA_ASSERT(deviceType == ma_device_type_capture || deviceType == ma_device_type_playback);
if (pDevice->type == ma_device_type_capture) {
pBufferQueue = (SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueueCapture;
pDevice->opensl.isDrainingCapture = MA_TRUE;
} else {
pBufferQueue = (SLAndroidSimpleBufferQueueItf)pDevice->opensl.pBufferQueuePlayback;
pDevice->opensl.isDrainingPlayback = MA_TRUE;
}
for (;;) {
SLAndroidSimpleBufferQueueState state;
MA_OPENSL_BUFFERQUEUE(pBufferQueue)->GetState(pBufferQueue, &state);
if (state.count == 0) {
break;
}
ma_sleep(10);
}
if (pDevice->type == ma_device_type_capture) {
pDevice->opensl.isDrainingCapture = MA_FALSE;
} else {
pDevice->opensl.isDrainingPlayback = MA_FALSE;
}
return MA_SUCCESS;
}
static ma_result ma_device_stop__opensl(ma_device* pDevice)
{
SLresult resultSL;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(g_maOpenSLInitCounter > 0); /* <-- If you trigger this it means you've either not initialized the context, or you've uninitialized it before stopping/uninitializing the device. */
if (g_maOpenSLInitCounter == 0) {
return MA_INVALID_OPERATION;
}
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
ma_device_drain__opensl(pDevice, ma_device_type_capture);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
#else
pContext->opensl.SL_IID_ENGINE = (ma_handle)SL_IID_ENGINE;
pContext->opensl.SL_IID_AUDIOIODEVICECAPABILITIES = (ma_handle)SL_IID_AUDIOIODEVICECAPABILITIES;
pContext->opensl.SL_IID_ANDROIDSIMPLEBUFFERQUEUE = (ma_handle)SL_IID_ANDROIDSIMPLEBUFFERQUEUE;
pContext->opensl.SL_IID_RECORD = (ma_handle)SL_IID_RECORD;
pContext->opensl.SL_IID_PLAY = (ma_handle)SL_IID_PLAY;
pContext->opensl.SL_IID_OUTPUTMIX = (ma_handle)SL_IID_OUTPUTMIX;
pContext->opensl.SL_IID_ANDROIDCONFIGURATION = (ma_handle)SL_IID_ANDROIDCONFIGURATION;
pContext->opensl.slCreateEngine = (ma_proc)slCreateEngine;
#endif
/* Initialize global data first if applicable. */
ma_spinlock_lock(&g_maOpenSLSpinlock);
{
result = ma_context_init_engine_nolock__opensl(pContext);
}
ma_spinlock_unlock(&g_maOpenSLSpinlock);
if (result != MA_SUCCESS) {
ma_dlclose(pContext, pContext->opensl.libOpenSLES);
ma_log_post(ma_context_get_log(pContext), MA_LOG_LEVEL_INFO, "[OpenSL] Failed to initialize OpenSL engine.");
return result;
}
pCallbacks->onContextInit = ma_context_init__opensl;
pCallbacks->onContextUninit = ma_context_uninit__opensl;
pCallbacks->onContextEnumerateDevices = ma_context_enumerate_devices__opensl;
pCallbacks->onContextGetDeviceInfo = ma_context_get_device_info__opensl;
pCallbacks->onDeviceInit = ma_device_init__opensl;
pCallbacks->onDeviceUninit = ma_device_uninit__opensl;
pCallbacks->onDeviceStart = ma_device_start__opensl;
pCallbacks->onDeviceStop = ma_device_stop__opensl;
pCallbacks->onDeviceRead = NULL; /* Not needed because OpenSL|ES is asynchronous. */
pCallbacks->onDeviceWrite = NULL; /* Not needed because OpenSL|ES is asynchronous. */
pCallbacks->onDeviceDataLoop = NULL; /* Not needed because OpenSL|ES is asynchronous. */
return MA_SUCCESS;
}
#endif /* OpenSL|ES */
/******************************************************************************
Web Audio Backend
******************************************************************************/
#ifdef MA_HAS_WEBAUDIO
#include <emscripten/emscripten.h>
static ma_bool32 ma_is_capture_supported__webaudio()
{
return EM_ASM_INT({
return (navigator.mediaDevices !== undefined && navigator.mediaDevices.getUserMedia !== undefined);
}, 0) != 0; /* Must pass in a dummy argument for C99 compatibility. */
}
#ifdef __cplusplus
extern "C" {
#endif
void EMSCRIPTEN_KEEPALIVE ma_device_process_pcm_frames_capture__webaudio(ma_device* pDevice, int frameCount, float* pFrames)
{
ma_device_handle_backend_data_callback(pDevice, NULL, pFrames, (ma_uint32)frameCount);
}
void EMSCRIPTEN_KEEPALIVE ma_device_process_pcm_frames_playback__webaudio(ma_device* pDevice, int frameCount, float* pFrames)
{
ma_device_handle_backend_data_callback(pDevice, pFrames, NULL, (ma_uint32)frameCount);
}
#ifdef __cplusplus
}
#endif
static ma_result ma_context_enumerate_devices__webaudio(ma_context* pContext, ma_enum_devices_callback_proc callback, void* pUserData)
{
ma_bool32 cbResult = MA_TRUE;
MA_ASSERT(pContext != NULL);
MA_ASSERT(callback != NULL);
/* Only supporting default devices for now. */
/* Playback. */
if (cbResult) {
ma_device_info deviceInfo;
MA_ZERO_OBJECT(&deviceInfo);
ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
deviceInfo.isDefault = MA_TRUE; /* Only supporting default devices. */
cbResult = callback(pContext, ma_device_type_playback, &deviceInfo, pUserData);
}
/* Capture. */
if (cbResult) {
if (ma_is_capture_supported__webaudio()) {
ma_device_info deviceInfo;
MA_ZERO_OBJECT(&deviceInfo);
ma_strncpy_s(deviceInfo.name, sizeof(deviceInfo.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
deviceInfo.isDefault = MA_TRUE; /* Only supporting default devices. */
cbResult = callback(pContext, ma_device_type_capture, &deviceInfo, pUserData);
}
}
return MA_SUCCESS;
}
static ma_result ma_context_get_device_info__webaudio(ma_context* pContext, ma_device_type deviceType, const ma_device_id* pDeviceID, ma_device_info* pDeviceInfo)
{
MA_ASSERT(pContext != NULL);
if (deviceType == ma_device_type_capture && !ma_is_capture_supported__webaudio()) {
return MA_NO_DEVICE;
}
MA_ZERO_MEMORY(pDeviceInfo->id.webaudio, sizeof(pDeviceInfo->id.webaudio));
/* Only supporting default devices for now. */
(void)pDeviceID;
if (deviceType == ma_device_type_playback) {
ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
} else {
ma_strncpy_s(pDeviceInfo->name, sizeof(pDeviceInfo->name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
}
/* Only supporting default devices. */
pDeviceInfo->isDefault = MA_TRUE;
/* Web Audio can support any number of channels and sample rates. It only supports f32 formats, however. */
pDeviceInfo->nativeDataFormats[0].flags = 0;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
pDeviceInfo->nativeDataFormatCount = 1;
return MA_SUCCESS;
}
static void ma_device_uninit_by_index__webaudio(ma_device* pDevice, ma_device_type deviceType, int deviceIndex)
{
MA_ASSERT(pDevice != NULL);
EM_ASM({
var device = miniaudio.get_device_by_index($0);
/* Make sure all nodes are disconnected and marked for collection. */
if (device.scriptNode !== undefined) {
device.scriptNode.onaudioprocess = function(e) {}; /* We want to reset the callback to ensure it doesn't get called after AudioContext.close() has returned. Shouldn't happen since we're disconnecting, but just to be safe... */
device.scriptNode.disconnect();
device.scriptNode = undefined;
}
if (device.streamNode !== undefined) {
device.streamNode.disconnect();
device.streamNode = undefined;
}
/*
Stop the device. I think there is a chance the callback could get fired after calling this, hence why we want
to clear the callback before closing.
*/
device.webaudio.close();
device.webaudio = undefined;
/* Can't forget to free the intermediary buffer. This is the buffer that's shared between JavaScript and C. */
if (device.intermediaryBuffer !== undefined) {
Module._free(device.intermediaryBuffer);
device.intermediaryBuffer = undefined;
device.intermediaryBufferView = undefined;
device.intermediaryBufferSizeInBytes = undefined;
}
/* Make sure the device is untracked so the slot can be reused later. */
miniaudio.untrack_device_by_index($0);
}, deviceIndex, deviceType);
}
static ma_result ma_device_uninit__webaudio(ma_device* pDevice)
{
MA_ASSERT(pDevice != NULL);
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_capture, pDevice->webaudio.indexCapture);
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_playback, pDevice->webaudio.indexPlayback);
}
return MA_SUCCESS;
}
static ma_uint32 ma_calculate_period_size_in_frames_from_descriptor__webaudio(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
{
/*
There have been reports of the default buffer size being too small on some browsers. There have been reports of the default buffer
size being too small on some browsers. If we're using default buffer size, we'll make sure the period size is a big biffer than our
standard defaults.
*/
ma_uint32 periodSizeInFrames;
if (pDescriptor->periodSizeInFrames == 0) {
if (pDescriptor->periodSizeInMilliseconds == 0) {
if (performanceProfile == ma_performance_profile_low_latency) {
periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(33, nativeSampleRate); /* 1 frame @ 30 FPS */
} else {
periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(333, nativeSampleRate);
}
} else {
periodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptor->periodSizeInMilliseconds, nativeSampleRate);
}
} else {
periodSizeInFrames = pDescriptor->periodSizeInFrames;
}
/* The size of the buffer must be a power of 2 and between 256 and 16384. */
if (periodSizeInFrames < 256) {
periodSizeInFrames = 256;
} else if (periodSizeInFrames > 16384) {
periodSizeInFrames = 16384;
} else {
periodSizeInFrames = ma_next_power_of_2(periodSizeInFrames);
}
return periodSizeInFrames;
}
static ma_result ma_device_init_by_type__webaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptor, ma_device_type deviceType)
{
int deviceIndex;
ma_uint32 channels;
ma_uint32 sampleRate;
ma_uint32 periodSizeInFrames;
MA_ASSERT(pDevice != NULL);
MA_ASSERT(pConfig != NULL);
MA_ASSERT(deviceType != ma_device_type_duplex);
if (deviceType == ma_device_type_capture && !ma_is_capture_supported__webaudio()) {
return MA_NO_DEVICE;
}
/* We're going to calculate some stuff in C just to simplify the JS code. */
channels = (pDescriptor->channels > 0) ? pDescriptor->channels : MA_DEFAULT_CHANNELS;
sampleRate = (pDescriptor->sampleRate > 0) ? pDescriptor->sampleRate : MA_DEFAULT_SAMPLE_RATE;
periodSizeInFrames = ma_calculate_period_size_in_frames_from_descriptor__webaudio(pDescriptor, sampleRate, pConfig->performanceProfile);
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_DEBUG, "periodSizeInFrames = %d\n", (int)periodSizeInFrames);
/* We create the device on the JavaScript side and reference it using an index. We use this to make it possible to reference the device between JavaScript and C. */
deviceIndex = EM_ASM_INT({
var channels = $0;
var sampleRate = $1;
var bufferSize = $2; /* In PCM frames. */
var isCapture = $3;
var pDevice = $4;
if (typeof(window.miniaudio) === 'undefined') {
return -1; /* Context not initialized. */
}
var device = {};
/* The AudioContext must be created in a suspended state. */
device.webaudio = new (window.AudioContext || window.webkitAudioContext)({sampleRate:sampleRate});
device.webaudio.suspend();
device.state = 1; /* ma_device_state_stopped */
/*
We need an intermediary buffer which we use for JavaScript and C interop. This buffer stores interleaved f32 PCM data. Because it's passed between
JavaScript and C it needs to be allocated and freed using Module._malloc() and Module._free().
*/
device.intermediaryBufferSizeInBytes = channels * bufferSize * 4;
device.intermediaryBuffer = Module._malloc(device.intermediaryBufferSizeInBytes);
device.intermediaryBufferView = new Float32Array(Module.HEAPF32.buffer, device.intermediaryBuffer, device.intermediaryBufferSizeInBytes);
/*
Both playback and capture devices use a ScriptProcessorNode for performing per-sample operations.
ScriptProcessorNode is actually deprecated so this is likely to be temporary. The way this works for playback is very simple. You just set a callback
that's periodically fired, just like a normal audio callback function. But apparently this design is "flawed" and is now deprecated in favour of
something called AudioWorklets which _forces_ you to load a _separate_ .js file at run time... nice... Hopefully ScriptProcessorNode will continue to
work for years to come, but this may need to change to use AudioSourceBufferNode instead, which I think is what Emscripten uses for it's built-in SDL
implementation. I'll be avoiding that insane AudioWorklet API like the plague...
For capture it is a bit unintuitive. We use the ScriptProccessorNode _only_ to get the raw PCM data. It is connected to an AudioContext just like the
playback case, however we just output silence to the AudioContext instead of passing any real data. It would make more sense to me to use the
MediaRecorder API, but unfortunately you need to specify a MIME time (Opus, Vorbis, etc.) for the binary blob that's returned to the client, but I've
been unable to figure out how to get this as raw PCM. The closest I can think is to use the MIME type for WAV files and just parse it, but I don't know
how well this would work. Although ScriptProccessorNode is deprecated, in practice it seems to have pretty good browser support so I'm leaving it like
this for now. If anyone knows how I could get raw PCM data using the MediaRecorder API please let me know!
*/
device.scriptNode = device.webaudio.createScriptProcessor(bufferSize, (isCapture) ? channels : 0, (isCapture) ? 0 : channels);
if (isCapture) {
device.scriptNode.onaudioprocess = function(e) {
if (device.intermediaryBuffer === undefined) {
return; /* This means the device has been uninitialized. */
}
if (device.intermediaryBufferView.length == 0) {
/* Recreate intermediaryBufferView when losing reference to the underlying buffer, probably due to emscripten resizing heap. */
device.intermediaryBufferView = new Float32Array(Module.HEAPF32.buffer, device.intermediaryBuffer, device.intermediaryBufferSizeInBytes);
}
/* Make sure silence it output to the AudioContext destination. Not doing this will cause sound to come out of the speakers! */
for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) {
e.outputBuffer.getChannelData(iChannel).fill(0.0);
}
/* There are some situations where we may want to send silence to the client. */
var sendSilence = false;
if (device.streamNode === undefined) {
sendSilence = true;
}
/* Sanity check. This will never happen, right? */
if (e.inputBuffer.numberOfChannels != channels) {
console.log("Capture: Channel count mismatch. " + e.inputBufer.numberOfChannels + " != " + channels + ". Sending silence.");
sendSilence = true;
}
/* This looped design guards against the situation where e.inputBuffer is a different size to the original buffer size. Should never happen in practice. */
var totalFramesProcessed = 0;
while (totalFramesProcessed < e.inputBuffer.length) {
var framesRemaining = e.inputBuffer.length - totalFramesProcessed;
var framesToProcess = framesRemaining;
if (framesToProcess > (device.intermediaryBufferSizeInBytes/channels/4)) {
framesToProcess = (device.intermediaryBufferSizeInBytes/channels/4);
}
/* We need to do the reverse of the playback case. We need to interleave the input data and copy it into the intermediary buffer. Then we send it to the client. */
if (sendSilence) {
device.intermediaryBufferView.fill(0.0);
} else {
for (var iFrame = 0; iFrame < framesToProcess; ++iFrame) {
for (var iChannel = 0; iChannel < e.inputBuffer.numberOfChannels; ++iChannel) {
device.intermediaryBufferView[iFrame*channels + iChannel] = e.inputBuffer.getChannelData(iChannel)[totalFramesProcessed + iFrame];
}
}
}
/* Send data to the client from our intermediary buffer. */
ccall("ma_device_process_pcm_frames_capture__webaudio", "undefined", ["number", "number", "number"], [pDevice, framesToProcess, device.intermediaryBuffer]);
totalFramesProcessed += framesToProcess;
}
};
navigator.mediaDevices.getUserMedia({audio:true, video:false})
.then(function(stream) {
device.streamNode = device.webaudio.createMediaStreamSource(stream);
device.streamNode.connect(device.scriptNode);
device.scriptNode.connect(device.webaudio.destination);
})
.catch(function(error) {
/* I think this should output silence... */
device.scriptNode.connect(device.webaudio.destination);
});
} else {
device.scriptNode.onaudioprocess = function(e) {
if (device.intermediaryBuffer === undefined) {
return; /* This means the device has been uninitialized. */
}
if(device.intermediaryBufferView.length == 0) {
/* Recreate intermediaryBufferView when losing reference to the underlying buffer, probably due to emscripten resizing heap. */
device.intermediaryBufferView = new Float32Array(Module.HEAPF32.buffer, device.intermediaryBuffer, device.intermediaryBufferSizeInBytes);
}
var outputSilence = false;
/* Sanity check. This will never happen, right? */
if (e.outputBuffer.numberOfChannels != channels) {
console.log("Playback: Channel count mismatch. " + e.outputBufer.numberOfChannels + " != " + channels + ". Outputting silence.");
outputSilence = true;
return;
}
/* This looped design guards against the situation where e.outputBuffer is a different size to the original buffer size. Should never happen in practice. */
var totalFramesProcessed = 0;
while (totalFramesProcessed < e.outputBuffer.length) {
var framesRemaining = e.outputBuffer.length - totalFramesProcessed;
var framesToProcess = framesRemaining;
if (framesToProcess > (device.intermediaryBufferSizeInBytes/channels/4)) {
framesToProcess = (device.intermediaryBufferSizeInBytes/channels/4);
}
/* Read data from the client into our intermediary buffer. */
ccall("ma_device_process_pcm_frames_playback__webaudio", "undefined", ["number", "number", "number"], [pDevice, framesToProcess, device.intermediaryBuffer]);
/* At this point we'll have data in our intermediary buffer which we now need to deinterleave and copy over to the output buffers. */
if (outputSilence) {
for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) {
e.outputBuffer.getChannelData(iChannel).fill(0.0);
}
} else {
for (var iChannel = 0; iChannel < e.outputBuffer.numberOfChannels; ++iChannel) {
var outputBuffer = e.outputBuffer.getChannelData(iChannel);
var intermediaryBuffer = device.intermediaryBufferView;
for (var iFrame = 0; iFrame < framesToProcess; ++iFrame) {
outputBuffer[totalFramesProcessed + iFrame] = intermediaryBuffer[iFrame*channels + iChannel];
}
}
}
totalFramesProcessed += framesToProcess;
}
};
device.scriptNode.connect(device.webaudio.destination);
}
return miniaudio.track_device(device);
}, channels, sampleRate, periodSizeInFrames, deviceType == ma_device_type_capture, pDevice);
if (deviceIndex < 0) {
return MA_FAILED_TO_OPEN_BACKEND_DEVICE;
}
if (deviceType == ma_device_type_capture) {
pDevice->webaudio.indexCapture = deviceIndex;
} else {
pDevice->webaudio.indexPlayback = deviceIndex;
}
pDescriptor->format = ma_format_f32;
pDescriptor->channels = channels;
ma_channel_map_init_standard(ma_standard_channel_map_webaudio, pDescriptor->channelMap, ma_countof(pDescriptor->channelMap), pDescriptor->channels);
pDescriptor->sampleRate = EM_ASM_INT({ return miniaudio.get_device_by_index($0).webaudio.sampleRate; }, deviceIndex);
pDescriptor->periodSizeInFrames = periodSizeInFrames;
pDescriptor->periodCount = 1;
return MA_SUCCESS;
}
static ma_result ma_device_init__webaudio(ma_device* pDevice, const ma_device_config* pConfig, ma_device_descriptor* pDescriptorPlayback, ma_device_descriptor* pDescriptorCapture)
{
ma_result result;
if (pConfig->deviceType == ma_device_type_loopback) {
return MA_DEVICE_TYPE_NOT_SUPPORTED;
}
/* No exclusive mode with Web Audio. */
if (((pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) && pDescriptorPlayback->shareMode == ma_share_mode_exclusive) ||
((pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) && pDescriptorCapture->shareMode == ma_share_mode_exclusive)) {
return MA_SHARE_MODE_NOT_SUPPORTED;
}
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex) {
result = ma_device_init_by_type__webaudio(pDevice, pConfig, pDescriptorCapture, ma_device_type_capture);
if (result != MA_SUCCESS) {
return result;
}
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
result = ma_device_init_by_type__webaudio(pDevice, pConfig, pDescriptorPlayback, ma_device_type_playback);
if (result != MA_SUCCESS) {
if (pConfig->deviceType == ma_device_type_duplex) {
ma_device_uninit_by_index__webaudio(pDevice, ma_device_type_capture, pDevice->webaudio.indexCapture);
}
return result;
}
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
/* Converting from internal device format to client format. */
ma_data_converter_config converterConfig = ma_data_converter_config_init_default();
converterConfig.formatIn = pDevice->capture.internalFormat;
converterConfig.channelsIn = pDevice->capture.internalChannels;
converterConfig.sampleRateIn = pDevice->capture.internalSampleRate;
converterConfig.pChannelMapIn = pDevice->capture.internalChannelMap;
converterConfig.formatOut = pDevice->capture.format;
converterConfig.channelsOut = pDevice->capture.channels;
converterConfig.sampleRateOut = pDevice->sampleRate;
converterConfig.pChannelMapOut = pDevice->capture.channelMap;
converterConfig.channelMixMode = pDevice->capture.channelMixMode;
converterConfig.allowDynamicSampleRate = MA_FALSE;
converterConfig.resampling.algorithm = pDevice->resampling.algorithm;
converterConfig.resampling.linear.lpfOrder = pDevice->resampling.linear.lpfOrder;
converterConfig.resampling.pBackendVTable = pDevice->resampling.pBackendVTable;
converterConfig.resampling.pBackendUserData = pDevice->resampling.pBackendUserData;
/* Make sure the old converter is uninitialized first. */
if (ma_device_get_state(pDevice) != ma_device_state_uninitialized) {
ma_data_converter_uninit(&pDevice->capture.converter, &pDevice->pContext->allocationCallbacks);
}
result = ma_data_converter_init(&converterConfig, &pDevice->pContext->allocationCallbacks, &pDevice->capture.converter);
if (result != MA_SUCCESS) {
return result;
}
}
if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
/* Converting from client format to device format. */
ma_data_converter_config converterConfig = ma_data_converter_config_init_default();
converterConfig.formatIn = pDevice->playback.format;
converterConfig.channelsIn = pDevice->playback.channels;
converterConfig.sampleRateIn = pDevice->sampleRate;
converterConfig.pChannelMapIn = pDevice->playback.channelMap;
converterConfig.formatOut = pDevice->playback.internalFormat;
converterConfig.channelsOut = pDevice->playback.internalChannels;
converterConfig.sampleRateOut = pDevice->playback.internalSampleRate;
converterConfig.pChannelMapOut = pDevice->playback.internalChannelMap;
converterConfig.channelMixMode = pDevice->playback.channelMixMode;
converterConfig.allowDynamicSampleRate = MA_FALSE;
converterConfig.resampling.algorithm = pDevice->resampling.algorithm;
converterConfig.resampling.linear.lpfOrder = pDevice->resampling.linear.lpfOrder;
converterConfig.resampling.pBackendVTable = pDevice->resampling.pBackendVTable;
converterConfig.resampling.pBackendUserData = pDevice->resampling.pBackendUserData;
/* Make sure the old converter is uninitialized first. */
if (ma_device_get_state(pDevice) != ma_device_state_uninitialized) {
ma_data_converter_uninit(&pDevice->playback.converter, &pDevice->pContext->allocationCallbacks);
}
result = ma_data_converter_init(&converterConfig, &pDevice->pContext->allocationCallbacks, &pDevice->playback.converter);
if (result != MA_SUCCESS) {
return result;
}
}
/*
In playback mode, if the data converter does not support retrieval of the required number of
input frames given a number of output frames, we need to fall back to a heap-allocated cache.
*/
if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
ma_uint64 unused;
pDevice->playback.inputCacheConsumed = 0;
pDevice->playback.inputCacheRemaining = 0;
if (deviceType == ma_device_type_duplex || ma_data_converter_get_required_input_frame_count(&pDevice->playback.converter, 1, &unused) != MA_SUCCESS) {
/* We need a heap allocated cache. We want to size this based on the period size. */
void* pNewInputCache;
ma_uint64 newInputCacheCap;
ma_uint64 newInputCacheSizeInBytes;
newInputCacheCap = ma_calculate_frame_count_after_resampling(pDevice->playback.internalSampleRate, pDevice->sampleRate, pDevice->playback.internalPeriodSizeInFrames);
newInputCacheSizeInBytes = newInputCacheCap * ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
if (newInputCacheSizeInBytes > MA_SIZE_MAX) {
ma_free(pDevice->playback.pInputCache, &pDevice->pContext->allocationCallbacks);
pDevice->playback.pInputCache = NULL;
pDevice->playback.inputCacheCap = 0;
return MA_OUT_OF_MEMORY; /* Allocation too big. Should never hit this, but makes the cast below safer for 32-bit builds. */
}
pNewInputCache = ma_realloc(pDevice->playback.pInputCache, (size_t)newInputCacheSizeInBytes, &pDevice->pContext->allocationCallbacks);
if (pNewInputCache == NULL) {
ma_free(pDevice->playback.pInputCache, &pDevice->pContext->allocationCallbacks);
pDevice->playback.pInputCache = NULL;
pDevice->playback.inputCacheCap = 0;
return MA_OUT_OF_MEMORY;
}
pDevice->playback.pInputCache = pNewInputCache;
pDevice->playback.inputCacheCap = newInputCacheCap;
} else {
/* Heap allocation not required. Make sure we clear out the old cache just in case this function was called in response to a route change. */
ma_free(pDevice->playback.pInputCache, &pDevice->pContext->allocationCallbacks);
pDevice->playback.pInputCache = NULL;
pDevice->playback.inputCacheCap = 0;
}
}
return MA_SUCCESS;
}
MA_API ma_result ma_device_post_init(ma_device* pDevice, ma_device_type deviceType, const ma_device_descriptor* pDescriptorPlayback, const ma_device_descriptor* pDescriptorCapture)
{
ma_result result;
if (pDevice == NULL) {
return MA_INVALID_ARGS;
}
/* Capture. */
if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) {
if (ma_device_descriptor_is_valid(pDescriptorCapture) == MA_FALSE) {
return MA_INVALID_ARGS;
}
pDevice->capture.internalFormat = pDescriptorCapture->format;
pDevice->capture.internalChannels = pDescriptorCapture->channels;
pDevice->capture.internalSampleRate = pDescriptorCapture->sampleRate;
MA_COPY_MEMORY(pDevice->capture.internalChannelMap, pDescriptorCapture->channelMap, sizeof(pDescriptorCapture->channelMap));
pDevice->capture.internalPeriodSizeInFrames = pDescriptorCapture->periodSizeInFrames;
pDevice->capture.internalPeriods = pDescriptorCapture->periodCount;
if (pDevice->capture.internalPeriodSizeInFrames == 0) {
pDevice->capture.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptorCapture->periodSizeInMilliseconds, pDescriptorCapture->sampleRate);
}
}
/* Playback. */
if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
if (ma_device_descriptor_is_valid(pDescriptorPlayback) == MA_FALSE) {
return MA_INVALID_ARGS;
}
pDevice->playback.internalFormat = pDescriptorPlayback->format;
pDevice->playback.internalChannels = pDescriptorPlayback->channels;
pDevice->playback.internalSampleRate = pDescriptorPlayback->sampleRate;
MA_COPY_MEMORY(pDevice->playback.internalChannelMap, pDescriptorPlayback->channelMap, sizeof(pDescriptorPlayback->channelMap));
pDevice->playback.internalPeriodSizeInFrames = pDescriptorPlayback->periodSizeInFrames;
pDevice->playback.internalPeriods = pDescriptorPlayback->periodCount;
if (pDevice->playback.internalPeriodSizeInFrames == 0) {
pDevice->playback.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptorPlayback->periodSizeInMilliseconds, pDescriptorPlayback->sampleRate);
}
}
/*
The name of the device can be retrieved from device info. This may be temporary and replaced with a `ma_device_get_info(pDevice, deviceType)` instead.
For loopback devices, we need to retrieve the name of the playback device.
*/
{
ma_device_info deviceInfo;
if (deviceType == ma_device_type_capture || deviceType == ma_device_type_duplex || deviceType == ma_device_type_loopback) {
result = ma_device_get_info(pDevice, (deviceType == ma_device_type_loopback) ? ma_device_type_playback : ma_device_type_capture, &deviceInfo);
if (result == MA_SUCCESS) {
ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), deviceInfo.name, (size_t)-1);
} else {
/* We failed to retrieve the device info. Fall back to a default name. */
if (pDescriptorCapture->pDeviceID == NULL) {
ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
} else {
ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), "Capture Device", (size_t)-1);
}
}
}
if (deviceType == ma_device_type_playback || deviceType == ma_device_type_duplex) {
result = ma_device_get_info(pDevice, ma_device_type_playback, &deviceInfo);
if (result == MA_SUCCESS) {
ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), deviceInfo.name, (size_t)-1);
} else {
/* We failed to retrieve the device info. Fall back to a default name. */
if (pDescriptorPlayback->pDeviceID == NULL) {
ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
} else {
ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), "Playback Device", (size_t)-1);
}
}
}
}
/* Update data conversion. */
return ma_device__post_init_setup(pDevice, deviceType); /* TODO: Should probably rename ma_device__post_init_setup() to something better. */
}
static ma_thread_result MA_THREADCALL ma_worker_thread(void* pData)
{
ma_device* pDevice = (ma_device*)pData;
MA_ASSERT(pDevice != NULL);
#ifdef MA_WIN32
ma_CoInitializeEx(pDevice->pContext, NULL, MA_COINIT_VALUE);
#endif
/*
When the device is being initialized it's initial state is set to ma_device_state_uninitialized. Before returning from
ma_device_init(), the state needs to be set to something valid. In miniaudio the device's default state immediately
after initialization is stopped, so therefore we need to mark the device as such. miniaudio will wait on the worker
thread to signal an event to know when the worker thread is ready for action.
*/
ma_device__set_state(pDevice, ma_device_state_stopped);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
MA_ZERO_OBJECT(&descriptorPlayback);
descriptorPlayback.pDeviceID = pConfig->playback.pDeviceID;
descriptorPlayback.shareMode = pConfig->playback.shareMode;
descriptorPlayback.format = pConfig->playback.format;
descriptorPlayback.channels = pConfig->playback.channels;
descriptorPlayback.sampleRate = pConfig->sampleRate;
ma_channel_map_copy_or_default(descriptorPlayback.channelMap, ma_countof(descriptorPlayback.channelMap), pConfig->playback.pChannelMap, pConfig->playback.channels);
descriptorPlayback.periodSizeInFrames = pConfig->periodSizeInFrames;
descriptorPlayback.periodSizeInMilliseconds = pConfig->periodSizeInMilliseconds;
descriptorPlayback.periodCount = pConfig->periods;
if (descriptorPlayback.periodCount == 0) {
descriptorPlayback.periodCount = MA_DEFAULT_PERIODS;
}
MA_ZERO_OBJECT(&descriptorCapture);
descriptorCapture.pDeviceID = pConfig->capture.pDeviceID;
descriptorCapture.shareMode = pConfig->capture.shareMode;
descriptorCapture.format = pConfig->capture.format;
descriptorCapture.channels = pConfig->capture.channels;
descriptorCapture.sampleRate = pConfig->sampleRate;
ma_channel_map_copy_or_default(descriptorCapture.channelMap, ma_countof(descriptorCapture.channelMap), pConfig->capture.pChannelMap, pConfig->capture.channels);
descriptorCapture.periodSizeInFrames = pConfig->periodSizeInFrames;
descriptorCapture.periodSizeInMilliseconds = pConfig->periodSizeInMilliseconds;
descriptorCapture.periodCount = pConfig->periods;
if (descriptorCapture.periodCount == 0) {
descriptorCapture.periodCount = MA_DEFAULT_PERIODS;
}
result = pContext->callbacks.onDeviceInit(pDevice, pConfig, &descriptorPlayback, &descriptorCapture);
if (result != MA_SUCCESS) {
ma_event_uninit(&pDevice->startEvent);
ma_event_uninit(&pDevice->wakeupEvent);
ma_mutex_uninit(&pDevice->startStopLock);
return result;
}
#if 0
/*
On output the descriptors will contain the *actual* data format of the device. We need this to know how to convert the data between
the requested format and the internal format.
*/
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) {
if (!ma_device_descriptor_is_valid(&descriptorCapture)) {
ma_device_uninit(pDevice);
return MA_INVALID_ARGS;
}
pDevice->capture.internalFormat = descriptorCapture.format;
pDevice->capture.internalChannels = descriptorCapture.channels;
pDevice->capture.internalSampleRate = descriptorCapture.sampleRate;
ma_channel_map_copy(pDevice->capture.internalChannelMap, descriptorCapture.channelMap, descriptorCapture.channels);
pDevice->capture.internalPeriodSizeInFrames = descriptorCapture.periodSizeInFrames;
pDevice->capture.internalPeriods = descriptorCapture.periodCount;
if (pDevice->capture.internalPeriodSizeInFrames == 0) {
pDevice->capture.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(descriptorCapture.periodSizeInMilliseconds, descriptorCapture.sampleRate);
}
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
if (!ma_device_descriptor_is_valid(&descriptorPlayback)) {
ma_device_uninit(pDevice);
return MA_INVALID_ARGS;
}
pDevice->playback.internalFormat = descriptorPlayback.format;
pDevice->playback.internalChannels = descriptorPlayback.channels;
pDevice->playback.internalSampleRate = descriptorPlayback.sampleRate;
ma_channel_map_copy(pDevice->playback.internalChannelMap, descriptorPlayback.channelMap, descriptorPlayback.channels);
pDevice->playback.internalPeriodSizeInFrames = descriptorPlayback.periodSizeInFrames;
pDevice->playback.internalPeriods = descriptorPlayback.periodCount;
if (pDevice->playback.internalPeriodSizeInFrames == 0) {
pDevice->playback.internalPeriodSizeInFrames = ma_calculate_buffer_size_in_frames_from_milliseconds(descriptorPlayback.periodSizeInMilliseconds, descriptorPlayback.sampleRate);
}
}
/*
The name of the device can be retrieved from device info. This may be temporary and replaced with a `ma_device_get_info(pDevice, deviceType)` instead.
For loopback devices, we need to retrieve the name of the playback device.
*/
{
ma_device_info deviceInfo;
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) {
result = ma_device_get_info(pDevice, (pConfig->deviceType == ma_device_type_loopback) ? ma_device_type_playback : ma_device_type_capture, &deviceInfo);
if (result == MA_SUCCESS) {
ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), deviceInfo.name, (size_t)-1);
} else {
/* We failed to retrieve the device info. Fall back to a default name. */
if (descriptorCapture.pDeviceID == NULL) {
ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), MA_DEFAULT_CAPTURE_DEVICE_NAME, (size_t)-1);
} else {
ma_strncpy_s(pDevice->capture.name, sizeof(pDevice->capture.name), "Capture Device", (size_t)-1);
}
}
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
result = ma_device_get_info(pDevice, ma_device_type_playback, &deviceInfo);
if (result == MA_SUCCESS) {
ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), deviceInfo.name, (size_t)-1);
} else {
/* We failed to retrieve the device info. Fall back to a default name. */
if (descriptorPlayback.pDeviceID == NULL) {
ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), MA_DEFAULT_PLAYBACK_DEVICE_NAME, (size_t)-1);
} else {
ma_strncpy_s(pDevice->playback.name, sizeof(pDevice->playback.name), "Playback Device", (size_t)-1);
}
}
}
}
ma_device__post_init_setup(pDevice, pConfig->deviceType);
#endif
result = ma_device_post_init(pDevice, pConfig->deviceType, &descriptorPlayback, &descriptorCapture);
if (result != MA_SUCCESS) {
ma_device_uninit(pDevice);
return result;
}
/*
If we're using fixed sized callbacks we'll need to make use of an intermediary buffer. Needs to
be done after post_init_setup() because we'll need access to the sample rate.
*/
if (pConfig->noFixedSizedCallback == MA_FALSE) {
/* We're using a fixed sized data callback so we'll need an intermediary buffer. */
ma_uint32 intermediaryBufferCap = pConfig->periodSizeInFrames;
if (intermediaryBufferCap == 0) {
intermediaryBufferCap = ma_calculate_buffer_size_in_frames_from_milliseconds(pConfig->periodSizeInMilliseconds, pDevice->sampleRate);
}
if (pConfig->deviceType == ma_device_type_capture || pConfig->deviceType == ma_device_type_duplex || pConfig->deviceType == ma_device_type_loopback) {
ma_uint32 intermediaryBufferSizeInBytes;
pDevice->capture.intermediaryBufferLen = 0;
pDevice->capture.intermediaryBufferCap = intermediaryBufferCap;
if (pDevice->capture.intermediaryBufferCap == 0) {
pDevice->capture.intermediaryBufferCap = pDevice->capture.internalPeriodSizeInFrames;
}
intermediaryBufferSizeInBytes = pDevice->capture.intermediaryBufferCap * ma_get_bytes_per_frame(pDevice->capture.format, pDevice->capture.channels);
pDevice->capture.pIntermediaryBuffer = ma_malloc((size_t)intermediaryBufferSizeInBytes, &pContext->allocationCallbacks);
if (pDevice->capture.pIntermediaryBuffer == NULL) {
ma_device_uninit(pDevice);
return MA_OUT_OF_MEMORY;
}
/* Silence the buffer for safety. */
ma_silence_pcm_frames(pDevice->capture.pIntermediaryBuffer, pDevice->capture.intermediaryBufferCap, pDevice->capture.format, pDevice->capture.channels);
pDevice->capture.intermediaryBufferLen = pDevice->capture.intermediaryBufferCap;
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
ma_uint64 intermediaryBufferSizeInBytes;
pDevice->playback.intermediaryBufferLen = 0;
if (pConfig->deviceType == ma_device_type_duplex) {
pDevice->playback.intermediaryBufferCap = pDevice->capture.intermediaryBufferCap; /* In duplex mode, make sure the intermediary buffer is always the same size as the capture side. */
} else {
pDevice->playback.intermediaryBufferCap = intermediaryBufferCap;
if (pDevice->playback.intermediaryBufferCap == 0) {
pDevice->playback.intermediaryBufferCap = pDevice->playback.internalPeriodSizeInFrames;
}
}
intermediaryBufferSizeInBytes = pDevice->playback.intermediaryBufferCap * ma_get_bytes_per_frame(pDevice->playback.format, pDevice->playback.channels);
pDevice->playback.pIntermediaryBuffer = ma_malloc((size_t)intermediaryBufferSizeInBytes, &pContext->allocationCallbacks);
if (pDevice->playback.pIntermediaryBuffer == NULL) {
ma_device_uninit(pDevice);
return MA_OUT_OF_MEMORY;
}
/* Silence the buffer for safety. */
ma_silence_pcm_frames(pDevice->playback.pIntermediaryBuffer, pDevice->playback.intermediaryBufferCap, pDevice->playback.format, pDevice->playback.channels);
pDevice->playback.intermediaryBufferLen = 0;
}
} else {
/* Not using a fixed sized data callback so no need for an intermediary buffer. */
}
/* Some backends don't require the worker thread. */
if (!ma_context_is_backend_asynchronous(pContext)) {
/* The worker thread. */
result = ma_thread_create(&pDevice->thread, pContext->threadPriority, pContext->threadStackSize, ma_worker_thread, pDevice, &pContext->allocationCallbacks);
if (result != MA_SUCCESS) {
ma_device_uninit(pDevice);
return result;
}
/* Wait for the worker thread to put the device into it's stopped state for real. */
ma_event_wait(&pDevice->stopEvent);
MA_ASSERT(ma_device_get_state(pDevice) == ma_device_state_stopped);
} else {
/*
If the backend is asynchronous and the device is duplex, we'll need an intermediary ring buffer. Note that this needs to be done
after ma_device__post_init_setup().
*/
if (ma_context_is_backend_asynchronous(pContext)) {
if (pConfig->deviceType == ma_device_type_duplex) {
result = ma_duplex_rb_init(pDevice->capture.format, pDevice->capture.channels, pDevice->sampleRate, pDevice->capture.internalSampleRate, pDevice->capture.internalPeriodSizeInFrames, &pDevice->pContext->allocationCallbacks, &pDevice->d...
if (result != MA_SUCCESS) {
ma_device_uninit(pDevice);
return result;
}
}
}
ma_device__set_state(pDevice, ma_device_state_stopped);
}
/* Log device information. */
{
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, "[%s]\n", ma_get_backend_name(pDevice->pContext->backend));
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_duplex) {
char name[MA_MAX_DEVICE_NAME_LENGTH + 1];
ma_device_get_name(pDevice, ma_device_type_capture, name, sizeof(name), NULL);
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " %s (%s)\n", name, "Capture");
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Format: %s -> %s\n", ma_get_format_name(pDevice->capture.internalFormat), ma_get_format_name(pDevice->capture.format));
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Channels: %d -> %d\n", pDevice->capture.internalChannels, pDevice->capture.channels);
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Sample Rate: %d -> %d\n", pDevice->capture.internalSampleRate, pDevice->sampleRate);
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Buffer Size: %d*%d (%d)\n", pDevice->capture.internalPeriodSizeInFrames, pDevice->capture.internalPeriods, (pDevice->capture.internalPeriodSizeInFrames * pDevice->capture.in...
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Conversion:\n");
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Pre Format Conversion: %s\n", pDevice->capture.converter.hasPreFormatConversion ? "YES" : "NO");
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Post Format Conversion: %s\n", pDevice->capture.converter.hasPostFormatConversion ? "YES" : "NO");
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Channel Routing: %s\n", pDevice->capture.converter.hasChannelConverter ? "YES" : "NO");
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Resampling: %s\n", pDevice->capture.converter.hasResampler ? "YES" : "NO");
ma_log_postf(ma_device_get_log(pDevice), MA_LOG_LEVEL_INFO, " Passthrough: %s\n", pDevice->capture.converter.isPassthrough ? "YES" : "NO");
}
if (pDevice->type == ma_device_type_playback || pDevice->type == ma_device_type_duplex) {
char name[MA_MAX_DEVICE_NAME_LENGTH + 1];
ma_device_get_name(pDevice, ma_device_type_playback, name, sizeof(name), NULL);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
{
if (pDevice == NULL) {
return MA_INVALID_ARGS;
}
if (volume < 0.0f) {
return MA_INVALID_ARGS;
}
c89atomic_exchange_f32(&pDevice->masterVolumeFactor, volume);
return MA_SUCCESS;
}
MA_API ma_result ma_device_get_master_volume(ma_device* pDevice, float* pVolume)
{
if (pVolume == NULL) {
return MA_INVALID_ARGS;
}
if (pDevice == NULL) {
*pVolume = 0;
return MA_INVALID_ARGS;
}
*pVolume = c89atomic_load_f32(&pDevice->masterVolumeFactor);
return MA_SUCCESS;
}
MA_API ma_result ma_device_set_master_volume_db(ma_device* pDevice, float gainDB)
{
if (gainDB > 0) {
return MA_INVALID_ARGS;
}
return ma_device_set_master_volume(pDevice, ma_volume_db_to_linear(gainDB));
}
MA_API ma_result ma_device_get_master_volume_db(ma_device* pDevice, float* pGainDB)
{
float factor;
ma_result result;
if (pGainDB == NULL) {
return MA_INVALID_ARGS;
}
result = ma_device_get_master_volume(pDevice, &factor);
if (result != MA_SUCCESS) {
*pGainDB = 0;
return result;
}
*pGainDB = ma_volume_linear_to_db(factor);
return MA_SUCCESS;
}
MA_API ma_result ma_device_handle_backend_data_callback(ma_device* pDevice, void* pOutput, const void* pInput, ma_uint32 frameCount)
{
if (pDevice == NULL) {
return MA_INVALID_ARGS;
}
if (pOutput == NULL && pInput == NULL) {
return MA_INVALID_ARGS;
}
if (pDevice->type == ma_device_type_duplex) {
if (pInput != NULL) {
ma_device__handle_duplex_callback_capture(pDevice, frameCount, pInput, &pDevice->duplexRB.rb);
}
if (pOutput != NULL) {
ma_device__handle_duplex_callback_playback(pDevice, frameCount, pOutput, &pDevice->duplexRB.rb);
}
} else {
if (pDevice->type == ma_device_type_capture || pDevice->type == ma_device_type_loopback) {
if (pInput == NULL) {
return MA_INVALID_ARGS;
}
ma_device__send_frames_to_client(pDevice, frameCount, pInput);
}
if (pDevice->type == ma_device_type_playback) {
if (pOutput == NULL) {
return MA_INVALID_ARGS;
}
ma_device__read_frames_from_client(pDevice, frameCount, pOutput);
}
}
return MA_SUCCESS;
}
MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_descriptor(const ma_device_descriptor* pDescriptor, ma_uint32 nativeSampleRate, ma_performance_profile performanceProfile)
{
if (pDescriptor == NULL) {
return 0;
}
/*
We must have a non-0 native sample rate, but some backends don't allow retrieval of this at the
time when the size of the buffer needs to be determined. In this case we need to just take a best
guess and move on. We'll try using the sample rate in pDescriptor first. If that's not set we'll
just fall back to MA_DEFAULT_SAMPLE_RATE.
*/
if (nativeSampleRate == 0) {
nativeSampleRate = pDescriptor->sampleRate;
}
if (nativeSampleRate == 0) {
nativeSampleRate = MA_DEFAULT_SAMPLE_RATE;
}
MA_ASSERT(nativeSampleRate != 0);
if (pDescriptor->periodSizeInFrames == 0) {
if (pDescriptor->periodSizeInMilliseconds == 0) {
if (performanceProfile == ma_performance_profile_low_latency) {
return ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_LOW_LATENCY, nativeSampleRate);
} else {
return ma_calculate_buffer_size_in_frames_from_milliseconds(MA_DEFAULT_PERIOD_SIZE_IN_MILLISECONDS_CONSERVATIVE, nativeSampleRate);
}
} else {
return ma_calculate_buffer_size_in_frames_from_milliseconds(pDescriptor->periodSizeInMilliseconds, nativeSampleRate);
}
} else {
return pDescriptor->periodSizeInFrames;
}
}
#endif /* MA_NO_DEVICE_IO */
MA_API ma_uint32 ma_calculate_buffer_size_in_milliseconds_from_frames(ma_uint32 bufferSizeInFrames, ma_uint32 sampleRate)
{
/* Prevent a division by zero. */
if (sampleRate == 0) {
return 0;
}
return bufferSizeInFrames*1000 / sampleRate;
}
MA_API ma_uint32 ma_calculate_buffer_size_in_frames_from_milliseconds(ma_uint32 bufferSizeInMilliseconds, ma_uint32 sampleRate)
{
/* Prevent a division by zero. */
if (sampleRate == 0) {
return 0;
}
return bufferSizeInMilliseconds*sampleRate / 1000;
}
MA_API void ma_copy_pcm_frames(void* dst, const void* src, ma_uint64 frameCount, ma_format format, ma_uint32 channels)
{
if (dst == src) {
return; /* No-op. */
}
ma_copy_memory_64(dst, src, frameCount * ma_get_bytes_per_frame(format, channels));
}
MA_API void ma_silence_pcm_frames(void* p, ma_uint64 frameCount, ma_format format, ma_uint32 channels)
{
if (format == ma_format_u8) {
ma_uint64 sampleCount = frameCount * channels;
ma_uint64 iSample;
for (iSample = 0; iSample < sampleCount; iSample += 1) {
((ma_uint8*)p)[iSample] = 128;
}
} else {
ma_zero_memory_64(p, frameCount * ma_get_bytes_per_frame(format, channels));
}
}
MA_API void* ma_offset_pcm_frames_ptr(void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels)
{
return ma_offset_ptr(p, offsetInFrames * ma_get_bytes_per_frame(format, channels));
}
MA_API const void* ma_offset_pcm_frames_const_ptr(const void* p, ma_uint64 offsetInFrames, ma_format format, ma_uint32 channels)
{
return ma_offset_ptr(p, offsetInFrames * ma_get_bytes_per_frame(format, channels));
}
MA_API void ma_clip_samples_u8(ma_uint8* pDst, const ma_int16* pSrc, ma_uint64 count)
{
ma_uint64 iSample;
MA_ASSERT(pDst != NULL);
MA_ASSERT(pSrc != NULL);
for (iSample = 0; iSample < count; iSample += 1) {
pDst[iSample] = ma_clip_u8(pSrc[iSample]);
}
}
MA_API void ma_clip_samples_s16(ma_int16* pDst, const ma_int32* pSrc, ma_uint64 count)
{
ma_uint64 iSample;
MA_ASSERT(pDst != NULL);
MA_ASSERT(pSrc != NULL);
for (iSample = 0; iSample < count; iSample += 1) {
pDst[iSample] = ma_clip_s16(pSrc[iSample]);
}
}
MA_API void ma_clip_samples_s24(ma_uint8* pDst, const ma_int64* pSrc, ma_uint64 count)
{
ma_uint64 iSample;
MA_ASSERT(pDst != NULL);
MA_ASSERT(pSrc != NULL);
for (iSample = 0; iSample < count; iSample += 1) {
ma_int64 s = ma_clip_s24(pSrc[iSample]);
pDst[iSample*3 + 0] = (ma_uint8)((s & 0x000000FF) >> 0);
pDst[iSample*3 + 1] = (ma_uint8)((s & 0x0000FF00) >> 8);
pDst[iSample*3 + 2] = (ma_uint8)((s & 0x00FF0000) >> 16);
}
}
MA_API void ma_clip_samples_s32(ma_int32* pDst, const ma_int64* pSrc, ma_uint64 count)
{
ma_uint64 iSample;
MA_ASSERT(pDst != NULL);
MA_ASSERT(pSrc != NULL);
for (iSample = 0; iSample < count; iSample += 1) {
pDst[iSample] = ma_clip_s32(pSrc[iSample]);
}
}
MA_API void ma_clip_samples_f32(float* pDst, const float* pSrc, ma_uint64 count)
{
ma_uint64 iSample;
MA_ASSERT(pDst != NULL);
MA_ASSERT(pSrc != NULL);
for (iSample = 0; iSample < count; iSample += 1) {
pDst[iSample] = ma_clip_f32(pSrc[iSample]);
}
}
MA_API void ma_clip_pcm_frames(void* pDst, const void* pSrc, ma_uint64 frameCount, ma_format format, ma_uint32 channels)
{
ma_uint64 sampleCount;
MA_ASSERT(pDst != NULL);
MA_ASSERT(pSrc != NULL);
sampleCount = frameCount * channels;
switch (format) {
case ma_format_u8: ma_clip_samples_u8( (ma_uint8*)pDst, (const ma_int16*)pSrc, sampleCount); break;
case ma_format_s16: ma_clip_samples_s16((ma_int16*)pDst, (const ma_int32*)pSrc, sampleCount); break;
case ma_format_s24: ma_clip_samples_s24((ma_uint8*)pDst, (const ma_int64*)pSrc, sampleCount); break;
case ma_format_s32: ma_clip_samples_s32((ma_int32*)pDst, (const ma_int64*)pSrc, sampleCount); break;
case ma_format_f32: ma_clip_samples_f32(( float*)pDst, (const float*)pSrc, sampleCount); break;
/* Do nothing if we don't know the format. We're including these here to silence a compiler warning about enums not being handled by the switch. */
case ma_format_unknown:
case ma_format_count:
break;
}
}
MA_API void ma_copy_and_apply_volume_factor_u8(ma_uint8* pSamplesOut, const ma_uint8* pSamplesIn, ma_uint64 sampleCount, float factor)
{
ma_uint64 iSample;
if (pSamplesOut == NULL || pSamplesIn == NULL) {
return;
}
for (iSample = 0; iSample < sampleCount; iSample += 1) {
pSamplesOut[iSample] = (ma_uint8)(pSamplesIn[iSample] * factor);
}
}
MA_API void ma_copy_and_apply_volume_factor_s16(ma_int16* pSamplesOut, const ma_int16* pSamplesIn, ma_uint64 sampleCount, float factor)
{
ma_uint64 iSample;
if (pSamplesOut == NULL || pSamplesIn == NULL) {
return;
}
for (iSample = 0; iSample < sampleCount; iSample += 1) {
pSamplesOut[iSample] = (ma_int16)(pSamplesIn[iSample] * factor);
}
}
MA_API void ma_copy_and_apply_volume_factor_s24(void* pSamplesOut, const void* pSamplesIn, ma_uint64 sampleCount, float factor)
{
ma_uint64 iSample;
ma_uint8* pSamplesOut8;
ma_uint8* pSamplesIn8;
if (pSamplesOut == NULL || pSamplesIn == NULL) {
return;
}
pSamplesOut8 = (ma_uint8*)pSamplesOut;
pSamplesIn8 = (ma_uint8*)pSamplesIn;
for (iSample = 0; iSample < sampleCount; iSample += 1) {
ma_int32 sampleS32;
sampleS32 = (ma_int32)(((ma_uint32)(pSamplesIn8[iSample*3+0]) << 8) | ((ma_uint32)(pSamplesIn8[iSample*3+1]) << 16) | ((ma_uint32)(pSamplesIn8[iSample*3+2])) << 24);
sampleS32 = (ma_int32)(sampleS32 * factor);
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ma_uint64 iSample;
if (pSamplesOut == NULL || pSamplesIn == NULL) {
return;
}
for (iSample = 0; iSample < sampleCount; iSample += 1) {
pSamplesOut[iSample] = (ma_int32)(pSamplesIn[iSample] * factor);
}
}
MA_API void ma_copy_and_apply_volume_factor_f32(float* pSamplesOut, const float* pSamplesIn, ma_uint64 sampleCount, float factor)
{
ma_uint64 iSample;
if (pSamplesOut == NULL || pSamplesIn == NULL) {
return;
}
if (factor == 1) {
if (pSamplesOut == pSamplesIn) {
/* In place. No-op. */
} else {
/* Just a copy. */
for (iSample = 0; iSample < sampleCount; iSample += 1) {
pSamplesOut[iSample] = pSamplesIn[iSample];
}
}
} else {
for (iSample = 0; iSample < sampleCount; iSample += 1) {
pSamplesOut[iSample] = pSamplesIn[iSample] * factor;
}
}
}
MA_API void ma_apply_volume_factor_u8(ma_uint8* pSamples, ma_uint64 sampleCount, float factor)
{
ma_copy_and_apply_volume_factor_u8(pSamples, pSamples, sampleCount, factor);
}
MA_API void ma_apply_volume_factor_s16(ma_int16* pSamples, ma_uint64 sampleCount, float factor)
{
ma_copy_and_apply_volume_factor_s16(pSamples, pSamples, sampleCount, factor);
}
MA_API void ma_apply_volume_factor_s24(void* pSamples, ma_uint64 sampleCount, float factor)
{
ma_copy_and_apply_volume_factor_s24(pSamples, pSamples, sampleCount, factor);
}
MA_API void ma_apply_volume_factor_s32(ma_int32* pSamples, ma_uint64 sampleCount, float factor)
{
ma_copy_and_apply_volume_factor_s32(pSamples, pSamples, sampleCount, factor);
}
MA_API void ma_apply_volume_factor_f32(float* pSamples, ma_uint64 sampleCount, float factor)
{
ma_copy_and_apply_volume_factor_f32(pSamples, pSamples, sampleCount, factor);
}
MA_API void ma_copy_and_apply_volume_factor_pcm_frames_u8(ma_uint8* pFramesOut, const ma_uint8* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
{
ma_copy_and_apply_volume_factor_u8(pFramesOut, pFramesIn, frameCount*channels, factor);
}
MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s16(ma_int16* pFramesOut, const ma_int16* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
{
ma_copy_and_apply_volume_factor_s16(pFramesOut, pFramesIn, frameCount*channels, factor);
}
MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s24(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
{
ma_copy_and_apply_volume_factor_s24(pFramesOut, pFramesIn, frameCount*channels, factor);
}
MA_API void ma_copy_and_apply_volume_factor_pcm_frames_s32(ma_int32* pFramesOut, const ma_int32* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
{
ma_copy_and_apply_volume_factor_s32(pFramesOut, pFramesIn, frameCount*channels, factor);
}
MA_API void ma_copy_and_apply_volume_factor_pcm_frames_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float factor)
{
ma_copy_and_apply_volume_factor_f32(pFramesOut, pFramesIn, frameCount*channels, factor);
}
MA_API void ma_copy_and_apply_volume_factor_pcm_frames(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor)
{
switch (format)
{
case ma_format_u8: ma_copy_and_apply_volume_factor_pcm_frames_u8 ((ma_uint8*)pFramesOut, (const ma_uint8*)pFramesIn, frameCount, channels, factor); return;
case ma_format_s16: ma_copy_and_apply_volume_factor_pcm_frames_s16((ma_int16*)pFramesOut, (const ma_int16*)pFramesIn, frameCount, channels, factor); return;
case ma_format_s24: ma_copy_and_apply_volume_factor_pcm_frames_s24( pFramesOut, pFramesIn, frameCount, channels, factor); return;
case ma_format_s32: ma_copy_and_apply_volume_factor_pcm_frames_s32((ma_int32*)pFramesOut, (const ma_int32*)pFramesIn, frameCount, channels, factor); return;
case ma_format_f32: ma_copy_and_apply_volume_factor_pcm_frames_f32( (float*)pFramesOut, (const float*)pFramesIn, frameCount, channels, factor); return;
default: return; /* Do nothing. */
}
}
MA_API void ma_apply_volume_factor_pcm_frames_u8(ma_uint8* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
{
ma_copy_and_apply_volume_factor_pcm_frames_u8(pFrames, pFrames, frameCount, channels, factor);
}
MA_API void ma_apply_volume_factor_pcm_frames_s16(ma_int16* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
{
ma_copy_and_apply_volume_factor_pcm_frames_s16(pFrames, pFrames, frameCount, channels, factor);
}
MA_API void ma_apply_volume_factor_pcm_frames_s24(void* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
{
ma_copy_and_apply_volume_factor_pcm_frames_s24(pFrames, pFrames, frameCount, channels, factor);
}
MA_API void ma_apply_volume_factor_pcm_frames_s32(ma_int32* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
{
ma_copy_and_apply_volume_factor_pcm_frames_s32(pFrames, pFrames, frameCount, channels, factor);
}
MA_API void ma_apply_volume_factor_pcm_frames_f32(float* pFrames, ma_uint64 frameCount, ma_uint32 channels, float factor)
{
ma_copy_and_apply_volume_factor_pcm_frames_f32(pFrames, pFrames, frameCount, channels, factor);
}
MA_API void ma_apply_volume_factor_pcm_frames(void* pFramesOut, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float factor)
{
ma_copy_and_apply_volume_factor_pcm_frames(pFramesOut, pFramesOut, frameCount, format, channels, factor);
}
MA_API void ma_copy_and_apply_volume_factor_per_channel_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, ma_uint32 channels, float* pChannelGains)
{
ma_uint64 iFrame;
if (channels == 2) {
/* TODO: Do an optimized implementation for stereo and mono. Can do a SIMD optimized implementation as well. */
}
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOut[iFrame * channels + iChannel] = pFramesIn[iFrame * channels + iChannel] * pChannelGains[iChannel];
}
}
}
static MA_INLINE ma_int16 ma_apply_volume_unclipped_u8(ma_int16 x, ma_int16 volume)
{
return (ma_int16)(((ma_int32)x * (ma_int32)volume) >> 8);
}
static MA_INLINE ma_int32 ma_apply_volume_unclipped_s16(ma_int32 x, ma_int16 volume)
{
return (ma_int32)((x * volume) >> 8);
}
static MA_INLINE ma_int64 ma_apply_volume_unclipped_s24(ma_int64 x, ma_int16 volume)
{
return (ma_int64)((x * volume) >> 8);
}
static MA_INLINE ma_int64 ma_apply_volume_unclipped_s32(ma_int64 x, ma_int16 volume)
{
return (ma_int64)((x * volume) >> 8);
}
static MA_INLINE float ma_apply_volume_unclipped_f32(float x, float volume)
{
return x * volume;
}
MA_API void ma_copy_and_apply_volume_and_clip_samples_u8(ma_uint8* pDst, const ma_int16* pSrc, ma_uint64 count, float volume)
{
ma_uint64 iSample;
ma_int16 volumeFixed;
MA_ASSERT(pDst != NULL);
MA_ASSERT(pSrc != NULL);
volumeFixed = ma_float_to_fixed_16(volume);
for (iSample = 0; iSample < count; iSample += 1) {
pDst[iSample] = ma_clip_u8(ma_apply_volume_unclipped_u8(pSrc[iSample], volumeFixed));
}
}
MA_API void ma_copy_and_apply_volume_and_clip_samples_s16(ma_int16* pDst, const ma_int32* pSrc, ma_uint64 count, float volume)
{
ma_uint64 iSample;
ma_int16 volumeFixed;
MA_ASSERT(pDst != NULL);
MA_ASSERT(pSrc != NULL);
volumeFixed = ma_float_to_fixed_16(volume);
for (iSample = 0; iSample < count; iSample += 1) {
pDst[iSample] = ma_clip_s16(ma_apply_volume_unclipped_s16(pSrc[iSample], volumeFixed));
}
}
MA_API void ma_copy_and_apply_volume_and_clip_samples_s24(ma_uint8* pDst, const ma_int64* pSrc, ma_uint64 count, float volume)
{
ma_uint64 iSample;
ma_int16 volumeFixed;
MA_ASSERT(pDst != NULL);
MA_ASSERT(pSrc != NULL);
volumeFixed = ma_float_to_fixed_16(volume);
for (iSample = 0; iSample < count; iSample += 1) {
ma_int64 s = ma_clip_s24(ma_apply_volume_unclipped_s24(pSrc[iSample], volumeFixed));
pDst[iSample*3 + 0] = (ma_uint8)((s & 0x000000FF) >> 0);
pDst[iSample*3 + 1] = (ma_uint8)((s & 0x0000FF00) >> 8);
pDst[iSample*3 + 2] = (ma_uint8)((s & 0x00FF0000) >> 16);
}
}
MA_API void ma_copy_and_apply_volume_and_clip_samples_s32(ma_int32* pDst, const ma_int64* pSrc, ma_uint64 count, float volume)
{
ma_uint64 iSample;
ma_int16 volumeFixed;
MA_ASSERT(pDst != NULL);
MA_ASSERT(pSrc != NULL);
volumeFixed = ma_float_to_fixed_16(volume);
for (iSample = 0; iSample < count; iSample += 1) {
pDst[iSample] = ma_clip_s32(ma_apply_volume_unclipped_s32(pSrc[iSample], volumeFixed));
}
}
MA_API void ma_copy_and_apply_volume_and_clip_samples_f32(float* pDst, const float* pSrc, ma_uint64 count, float volume)
{
ma_uint64 iSample;
MA_ASSERT(pDst != NULL);
MA_ASSERT(pSrc != NULL);
/* For the f32 case we need to make sure this supports in-place processing where the input and output buffers are the same. */
for (iSample = 0; iSample < count; iSample += 1) {
pDst[iSample] = ma_clip_f32(ma_apply_volume_unclipped_f32(pSrc[iSample], volume));
}
}
MA_API void ma_copy_and_apply_volume_and_clip_pcm_frames(void* pDst, const void* pSrc, ma_uint64 frameCount, ma_format format, ma_uint32 channels, float volume)
{
MA_ASSERT(pDst != NULL);
MA_ASSERT(pSrc != NULL);
if (volume == 1) {
ma_clip_pcm_frames(pDst, pSrc, frameCount, format, channels); /* Optimized case for volume = 1. */
} else if (volume == 0) {
ma_silence_pcm_frames(pDst, frameCount, format, channels); /* Optimized case for volume = 0. */
} else {
ma_uint64 sampleCount = frameCount * channels;
switch (format) {
case ma_format_u8: ma_copy_and_apply_volume_and_clip_samples_u8( (ma_uint8*)pDst, (const ma_int16*)pSrc, sampleCount, volume); break;
case ma_format_s16: ma_copy_and_apply_volume_and_clip_samples_s16((ma_int16*)pDst, (const ma_int32*)pSrc, sampleCount, volume); break;
case ma_format_s24: ma_copy_and_apply_volume_and_clip_samples_s24((ma_uint8*)pDst, (const ma_int64*)pSrc, sampleCount, volume); break;
case ma_format_s32: ma_copy_and_apply_volume_and_clip_samples_s32((ma_int32*)pDst, (const ma_int64*)pSrc, sampleCount, volume); break;
case ma_format_f32: ma_copy_and_apply_volume_and_clip_samples_f32(( float*)pDst, (const float*)pSrc, sampleCount, volume); break;
/* Do nothing if we don't know the format. We're including these here to silence a compiler warning about enums not being handled by the switch. */
case ma_format_unknown:
case ma_format_count:
break;
}
}
}
MA_API float ma_volume_linear_to_db(float factor)
{
return 20*ma_log10f(factor);
}
MA_API float ma_volume_db_to_linear(float gain)
{
return ma_powf(10, gain/20.0f);
}
/**************************************************************************************************************************************************************
Format Conversion
**************************************************************************************************************************************************************/
static MA_INLINE ma_int16 ma_pcm_sample_f32_to_s16(float x)
{
return (ma_int16)(x * 32767.0f);
}
static MA_INLINE ma_int16 ma_pcm_sample_u8_to_s16_no_scale(ma_uint8 x)
{
return (ma_int16)((ma_int16)x - 128);
}
static MA_INLINE ma_int64 ma_pcm_sample_s24_to_s32_no_scale(const ma_uint8* x)
{
return (ma_int64)(((ma_uint64)x[0] << 40) | ((ma_uint64)x[1] << 48) | ((ma_uint64)x[2] << 56)) >> 40; /* Make sure the sign bits are maintained. */
}
static MA_INLINE void ma_pcm_sample_s32_to_s24_no_scale(ma_int64 x, ma_uint8* s24)
{
s24[0] = (ma_uint8)((x & 0x000000FF) >> 0);
s24[1] = (ma_uint8)((x & 0x0000FF00) >> 8);
s24[2] = (ma_uint8)((x & 0x00FF0000) >> 16);
}
/* u8 */
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
float x = (float)src_u8[i];
x = x * 0.00784313725490196078f; /* 0..255 to 0..2 */
x = x - 1; /* 0..2 to -1..1 */
dst_f32[i] = x;
}
(void)ditherMode;
}
static MA_INLINE void ma_pcm_u8_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_u8_to_f32__reference(dst, src, count, ditherMode);
}
#if defined(MA_SUPPORT_SSE2)
static MA_INLINE void ma_pcm_u8_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_AVX2)
static MA_INLINE void ma_pcm_u8_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_NEON)
static MA_INLINE void ma_pcm_u8_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
MA_API void ma_pcm_u8_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_u8_to_f32__reference(dst, src, count, ditherMode);
#else
# if MA_PREFERRED_SIMD == MA_SIMD_AVX2
if (ma_has_avx2()) {
ma_pcm_u8_to_f32__avx2(dst, src, count, ditherMode);
} else
#elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
if (ma_has_sse2()) {
ma_pcm_u8_to_f32__sse2(dst, src, count, ditherMode);
} else
#elif MA_PREFERRED_SIMD == MA_SIMD_NEON
if (ma_has_neon()) {
ma_pcm_u8_to_f32__neon(dst, src, count, ditherMode);
} else
#endif
{
ma_pcm_u8_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
}
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
static MA_INLINE void ma_pcm_interleave_u8__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_uint8* dst_u8 = (ma_uint8*)dst;
const ma_uint8** src_u8 = (const ma_uint8**)src;
ma_uint64 iFrame;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; iChannel += 1) {
dst_u8[iFrame*channels + iChannel] = src_u8[iChannel][iFrame];
}
}
}
#else
static MA_INLINE void ma_pcm_interleave_u8__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_uint8* dst_u8 = (ma_uint8*)dst;
const ma_uint8** src_u8 = (const ma_uint8**)src;
if (channels == 1) {
ma_copy_memory_64(dst, src[0], frameCount * sizeof(ma_uint8));
} else if (channels == 2) {
ma_uint64 iFrame;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
dst_u8[iFrame*2 + 0] = src_u8[0][iFrame];
dst_u8[iFrame*2 + 1] = src_u8[1][iFrame];
}
} else {
ma_uint64 iFrame;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; iChannel += 1) {
dst_u8[iFrame*channels + iChannel] = src_u8[iChannel][iFrame];
}
}
}
}
#endif
MA_API void ma_pcm_interleave_u8(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_interleave_u8__reference(dst, src, frameCount, channels);
#else
ma_pcm_interleave_u8__optimized(dst, src, frameCount, channels);
#endif
}
static MA_INLINE void ma_pcm_deinterleave_u8__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_uint8** dst_u8 = (ma_uint8**)dst;
const ma_uint8* src_u8 = (const ma_uint8*)src;
ma_uint64 iFrame;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; iChannel += 1) {
dst_u8[iChannel][iFrame] = src_u8[iFrame*channels + iChannel];
}
}
}
static MA_INLINE void ma_pcm_deinterleave_u8__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_pcm_deinterleave_u8__reference(dst, src, frameCount, channels);
}
MA_API void ma_pcm_deinterleave_u8(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_deinterleave_u8__reference(dst, src, frameCount, channels);
#else
ma_pcm_deinterleave_u8__optimized(dst, src, frameCount, channels);
#endif
}
/* s16 */
static MA_INLINE void ma_pcm_s16_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_uint8* dst_u8 = (ma_uint8*)dst;
const ma_int16* src_s16 = (const ma_int16*)src;
if (ditherMode == ma_dither_mode_none) {
ma_uint64 i;
for (i = 0; i < count; i += 1) {
ma_int16 x = src_s16[i];
x = (ma_int16)(x >> 8);
x = (ma_int16)(x + 128);
dst_u8[i] = (ma_uint8)x;
}
} else {
ma_uint64 i;
for (i = 0; i < count; i += 1) {
ma_int16 x = src_s16[i];
/* Dither. Don't overflow. */
ma_int32 dither = ma_dither_s32(ditherMode, -0x80, 0x7F);
if ((x + dither) <= 0x7FFF) {
x = (ma_int16)(x + dither);
} else {
x = 0x7FFF;
}
x = (ma_int16)(x >> 8);
x = (ma_int16)(x + 128);
dst_u8[i] = (ma_uint8)x;
}
}
}
static MA_INLINE void ma_pcm_s16_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s16_to_u8__reference(dst, src, count, ditherMode);
}
#if defined(MA_SUPPORT_SSE2)
static MA_INLINE void ma_pcm_s16_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_AVX2)
static MA_INLINE void ma_pcm_s16_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_NEON)
static MA_INLINE void ma_pcm_s16_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s16_to_u8__optimized(dst, src, count, ditherMode);
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
#else
/* The fast way. */
x = x * 0.000030517578125f; /* -32768..32767 to -1..0.999969482421875 */
#endif
dst_f32[i] = x;
}
(void)ditherMode;
}
static MA_INLINE void ma_pcm_s16_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s16_to_f32__reference(dst, src, count, ditherMode);
}
#if defined(MA_SUPPORT_SSE2)
static MA_INLINE void ma_pcm_s16_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_AVX2)
static MA_INLINE void ma_pcm_s16_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_NEON)
static MA_INLINE void ma_pcm_s16_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
MA_API void ma_pcm_s16_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_s16_to_f32__reference(dst, src, count, ditherMode);
#else
# if MA_PREFERRED_SIMD == MA_SIMD_AVX2
if (ma_has_avx2()) {
ma_pcm_s16_to_f32__avx2(dst, src, count, ditherMode);
} else
#elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
if (ma_has_sse2()) {
ma_pcm_s16_to_f32__sse2(dst, src, count, ditherMode);
} else
#elif MA_PREFERRED_SIMD == MA_SIMD_NEON
if (ma_has_neon()) {
ma_pcm_s16_to_f32__neon(dst, src, count, ditherMode);
} else
#endif
{
ma_pcm_s16_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
}
static MA_INLINE void ma_pcm_interleave_s16__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_int16* dst_s16 = (ma_int16*)dst;
const ma_int16** src_s16 = (const ma_int16**)src;
ma_uint64 iFrame;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; iChannel += 1) {
dst_s16[iFrame*channels + iChannel] = src_s16[iChannel][iFrame];
}
}
}
static MA_INLINE void ma_pcm_interleave_s16__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_pcm_interleave_s16__reference(dst, src, frameCount, channels);
}
MA_API void ma_pcm_interleave_s16(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_interleave_s16__reference(dst, src, frameCount, channels);
#else
ma_pcm_interleave_s16__optimized(dst, src, frameCount, channels);
#endif
}
static MA_INLINE void ma_pcm_deinterleave_s16__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_int16** dst_s16 = (ma_int16**)dst;
const ma_int16* src_s16 = (const ma_int16*)src;
ma_uint64 iFrame;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; iChannel += 1) {
dst_s16[iChannel][iFrame] = src_s16[iFrame*channels + iChannel];
}
}
}
static MA_INLINE void ma_pcm_deinterleave_s16__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_pcm_deinterleave_s16__reference(dst, src, frameCount, channels);
}
MA_API void ma_pcm_deinterleave_s16(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_deinterleave_s16__reference(dst, src, frameCount, channels);
#else
ma_pcm_deinterleave_s16__optimized(dst, src, frameCount, channels);
#endif
}
/* s24 */
static MA_INLINE void ma_pcm_s24_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_uint8* dst_u8 = (ma_uint8*)dst;
const ma_uint8* src_s24 = (const ma_uint8*)src;
if (ditherMode == ma_dither_mode_none) {
ma_uint64 i;
for (i = 0; i < count; i += 1) {
dst_u8[i] = (ma_uint8)((ma_int8)src_s24[i*3 + 2] + 128);
}
} else {
ma_uint64 i;
for (i = 0; i < count; i += 1) {
ma_int32 x = (ma_int32)(((ma_uint32)(src_s24[i*3+0]) << 8) | ((ma_uint32)(src_s24[i*3+1]) << 16) | ((ma_uint32)(src_s24[i*3+2])) << 24);
/* Dither. Don't overflow. */
ma_int32 dither = ma_dither_s32(ditherMode, -0x800000, 0x7FFFFF);
if ((ma_int64)x + dither <= 0x7FFFFFFF) {
x = x + dither;
} else {
x = 0x7FFFFFFF;
}
x = x >> 24;
x = x + 128;
dst_u8[i] = (ma_uint8)x;
}
}
}
static MA_INLINE void ma_pcm_s24_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s24_to_u8__reference(dst, src, count, ditherMode);
}
#if defined(MA_SUPPORT_SSE2)
static MA_INLINE void ma_pcm_s24_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_AVX2)
static MA_INLINE void ma_pcm_s24_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_NEON)
static MA_INLINE void ma_pcm_s24_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s24_to_u8__optimized(dst, src, count, ditherMode);
}
#endif
MA_API void ma_pcm_s24_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
#else
/* The fast way. */
x = x * 0.00000011920928955078125f; /* -8388608..8388607 to -1..0.999969482421875 */
#endif
dst_f32[i] = x;
}
(void)ditherMode;
}
static MA_INLINE void ma_pcm_s24_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s24_to_f32__reference(dst, src, count, ditherMode);
}
#if defined(MA_SUPPORT_SSE2)
static MA_INLINE void ma_pcm_s24_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_AVX2)
static MA_INLINE void ma_pcm_s24_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_NEON)
static MA_INLINE void ma_pcm_s24_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
MA_API void ma_pcm_s24_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_s24_to_f32__reference(dst, src, count, ditherMode);
#else
# if MA_PREFERRED_SIMD == MA_SIMD_AVX2
if (ma_has_avx2()) {
ma_pcm_s24_to_f32__avx2(dst, src, count, ditherMode);
} else
#elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
if (ma_has_sse2()) {
ma_pcm_s24_to_f32__sse2(dst, src, count, ditherMode);
} else
#elif MA_PREFERRED_SIMD == MA_SIMD_NEON
if (ma_has_neon()) {
ma_pcm_s24_to_f32__neon(dst, src, count, ditherMode);
} else
#endif
{
ma_pcm_s24_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
}
static MA_INLINE void ma_pcm_interleave_s24__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_uint8* dst8 = (ma_uint8*)dst;
const ma_uint8** src8 = (const ma_uint8**)src;
ma_uint64 iFrame;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; iChannel += 1) {
dst8[iFrame*3*channels + iChannel*3 + 0] = src8[iChannel][iFrame*3 + 0];
dst8[iFrame*3*channels + iChannel*3 + 1] = src8[iChannel][iFrame*3 + 1];
dst8[iFrame*3*channels + iChannel*3 + 2] = src8[iChannel][iFrame*3 + 2];
}
}
}
static MA_INLINE void ma_pcm_interleave_s24__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_pcm_interleave_s24__reference(dst, src, frameCount, channels);
}
MA_API void ma_pcm_interleave_s24(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_interleave_s24__reference(dst, src, frameCount, channels);
#else
ma_pcm_interleave_s24__optimized(dst, src, frameCount, channels);
#endif
}
static MA_INLINE void ma_pcm_deinterleave_s24__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_uint8** dst8 = (ma_uint8**)dst;
const ma_uint8* src8 = (const ma_uint8*)src;
ma_uint32 iFrame;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; iChannel += 1) {
dst8[iChannel][iFrame*3 + 0] = src8[iFrame*3*channels + iChannel*3 + 0];
dst8[iChannel][iFrame*3 + 1] = src8[iFrame*3*channels + iChannel*3 + 1];
dst8[iChannel][iFrame*3 + 2] = src8[iFrame*3*channels + iChannel*3 + 2];
}
}
}
static MA_INLINE void ma_pcm_deinterleave_s24__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_pcm_deinterleave_s24__reference(dst, src, frameCount, channels);
}
MA_API void ma_pcm_deinterleave_s24(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_deinterleave_s24__reference(dst, src, frameCount, channels);
#else
ma_pcm_deinterleave_s24__optimized(dst, src, frameCount, channels);
#endif
}
/* s32 */
static MA_INLINE void ma_pcm_s32_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_uint8* dst_u8 = (ma_uint8*)dst;
const ma_int32* src_s32 = (const ma_int32*)src;
if (ditherMode == ma_dither_mode_none) {
ma_uint64 i;
for (i = 0; i < count; i += 1) {
ma_int32 x = src_s32[i];
x = x >> 24;
x = x + 128;
dst_u8[i] = (ma_uint8)x;
}
} else {
ma_uint64 i;
for (i = 0; i < count; i += 1) {
ma_int32 x = src_s32[i];
/* Dither. Don't overflow. */
ma_int32 dither = ma_dither_s32(ditherMode, -0x800000, 0x7FFFFF);
if ((ma_int64)x + dither <= 0x7FFFFFFF) {
x = x + dither;
} else {
x = 0x7FFFFFFF;
}
x = x >> 24;
x = x + 128;
dst_u8[i] = (ma_uint8)x;
}
}
}
static MA_INLINE void ma_pcm_s32_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s32_to_u8__reference(dst, src, count, ditherMode);
}
#if defined(MA_SUPPORT_SSE2)
static MA_INLINE void ma_pcm_s32_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_AVX2)
static MA_INLINE void ma_pcm_s32_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_NEON)
static MA_INLINE void ma_pcm_s32_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s32_to_u8__optimized(dst, src, count, ditherMode);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
x = x - 1;
#else
x = x / 2147483648.0;
#endif
dst_f32[i] = (float)x;
}
(void)ditherMode; /* No dithering for s32 -> f32. */
}
static MA_INLINE void ma_pcm_s32_to_f32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s32_to_f32__reference(dst, src, count, ditherMode);
}
#if defined(MA_SUPPORT_SSE2)
static MA_INLINE void ma_pcm_s32_to_f32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_AVX2)
static MA_INLINE void ma_pcm_s32_to_f32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_NEON)
static MA_INLINE void ma_pcm_s32_to_f32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
MA_API void ma_pcm_s32_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_s32_to_f32__reference(dst, src, count, ditherMode);
#else
# if MA_PREFERRED_SIMD == MA_SIMD_AVX2
if (ma_has_avx2()) {
ma_pcm_s32_to_f32__avx2(dst, src, count, ditherMode);
} else
#elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
if (ma_has_sse2()) {
ma_pcm_s32_to_f32__sse2(dst, src, count, ditherMode);
} else
#elif MA_PREFERRED_SIMD == MA_SIMD_NEON
if (ma_has_neon()) {
ma_pcm_s32_to_f32__neon(dst, src, count, ditherMode);
} else
#endif
{
ma_pcm_s32_to_f32__optimized(dst, src, count, ditherMode);
}
#endif
}
static MA_INLINE void ma_pcm_interleave_s32__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_int32* dst_s32 = (ma_int32*)dst;
const ma_int32** src_s32 = (const ma_int32**)src;
ma_uint64 iFrame;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; iChannel += 1) {
dst_s32[iFrame*channels + iChannel] = src_s32[iChannel][iFrame];
}
}
}
static MA_INLINE void ma_pcm_interleave_s32__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_pcm_interleave_s32__reference(dst, src, frameCount, channels);
}
MA_API void ma_pcm_interleave_s32(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_interleave_s32__reference(dst, src, frameCount, channels);
#else
ma_pcm_interleave_s32__optimized(dst, src, frameCount, channels);
#endif
}
static MA_INLINE void ma_pcm_deinterleave_s32__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_int32** dst_s32 = (ma_int32**)dst;
const ma_int32* src_s32 = (const ma_int32*)src;
ma_uint64 iFrame;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; iChannel += 1) {
dst_s32[iChannel][iFrame] = src_s32[iFrame*channels + iChannel];
}
}
}
static MA_INLINE void ma_pcm_deinterleave_s32__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_pcm_deinterleave_s32__reference(dst, src, frameCount, channels);
}
MA_API void ma_pcm_deinterleave_s32(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_deinterleave_s32__reference(dst, src, frameCount, channels);
#else
ma_pcm_deinterleave_s32__optimized(dst, src, frameCount, channels);
#endif
}
/* f32 */
static MA_INLINE void ma_pcm_f32_to_u8__reference(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_uint64 i;
ma_uint8* dst_u8 = (ma_uint8*)dst;
const float* src_f32 = (const float*)src;
float ditherMin = 0;
float ditherMax = 0;
if (ditherMode != ma_dither_mode_none) {
ditherMin = 1.0f / -128;
ditherMax = 1.0f / 127;
}
for (i = 0; i < count; i += 1) {
float x = src_f32[i];
x = x + ma_dither_f32(ditherMode, ditherMin, ditherMax);
x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */
x = x + 1; /* -1..1 to 0..2 */
x = x * 127.5f; /* 0..2 to 0..255 */
dst_u8[i] = (ma_uint8)x;
}
}
static MA_INLINE void ma_pcm_f32_to_u8__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_f32_to_u8__reference(dst, src, count, ditherMode);
}
#if defined(MA_SUPPORT_SSE2)
static MA_INLINE void ma_pcm_f32_to_u8__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_AVX2)
static MA_INLINE void ma_pcm_f32_to_u8__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_NEON)
static MA_INLINE void ma_pcm_f32_to_u8__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_f32_to_u8__optimized(dst, src, count, ditherMode);
}
#endif
MA_API void ma_pcm_f32_to_u8(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_f32_to_u8__reference(dst, src, count, ditherMode);
#else
# if MA_PREFERRED_SIMD == MA_SIMD_AVX2
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(void)ditherMode; /* No dithering for f32 -> s32. */
}
static MA_INLINE void ma_pcm_f32_to_s32__optimized(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_f32_to_s32__reference(dst, src, count, ditherMode);
}
#if defined(MA_SUPPORT_SSE2)
static MA_INLINE void ma_pcm_f32_to_s32__sse2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_AVX2)
static MA_INLINE void ma_pcm_f32_to_s32__avx2(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode);
}
#endif
#if defined(MA_SUPPORT_NEON)
static MA_INLINE void ma_pcm_f32_to_s32__neon(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode);
}
#endif
MA_API void ma_pcm_f32_to_s32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_f32_to_s32__reference(dst, src, count, ditherMode);
#else
# if MA_PREFERRED_SIMD == MA_SIMD_AVX2
if (ma_has_avx2()) {
ma_pcm_f32_to_s32__avx2(dst, src, count, ditherMode);
} else
#elif MA_PREFERRED_SIMD == MA_SIMD_SSE2
if (ma_has_sse2()) {
ma_pcm_f32_to_s32__sse2(dst, src, count, ditherMode);
} else
#elif MA_PREFERRED_SIMD == MA_SIMD_NEON
if (ma_has_neon()) {
ma_pcm_f32_to_s32__neon(dst, src, count, ditherMode);
} else
#endif
{
ma_pcm_f32_to_s32__optimized(dst, src, count, ditherMode);
}
#endif
}
MA_API void ma_pcm_f32_to_f32(void* dst, const void* src, ma_uint64 count, ma_dither_mode ditherMode)
{
(void)ditherMode;
ma_copy_memory_64(dst, src, count * sizeof(float));
}
static void ma_pcm_interleave_f32__reference(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
float* dst_f32 = (float*)dst;
const float** src_f32 = (const float**)src;
ma_uint64 iFrame;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; iChannel += 1) {
dst_f32[iFrame*channels + iChannel] = src_f32[iChannel][iFrame];
}
}
}
static void ma_pcm_interleave_f32__optimized(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_pcm_interleave_f32__reference(dst, src, frameCount, channels);
}
MA_API void ma_pcm_interleave_f32(void* dst, const void** src, ma_uint64 frameCount, ma_uint32 channels)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_interleave_f32__reference(dst, src, frameCount, channels);
#else
ma_pcm_interleave_f32__optimized(dst, src, frameCount, channels);
#endif
}
static void ma_pcm_deinterleave_f32__reference(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
float** dst_f32 = (float**)dst;
const float* src_f32 = (const float*)src;
ma_uint64 iFrame;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; iChannel += 1) {
dst_f32[iChannel][iFrame] = src_f32[iFrame*channels + iChannel];
}
}
}
static void ma_pcm_deinterleave_f32__optimized(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
ma_pcm_deinterleave_f32__reference(dst, src, frameCount, channels);
}
MA_API void ma_pcm_deinterleave_f32(void** dst, const void* src, ma_uint64 frameCount, ma_uint32 channels)
{
#ifdef MA_USE_REFERENCE_CONVERSION_APIS
ma_pcm_deinterleave_f32__reference(dst, src, frameCount, channels);
#else
ma_pcm_deinterleave_f32__optimized(dst, src, frameCount, channels);
#endif
}
MA_API void ma_pcm_convert(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 sampleCount, ma_dither_mode ditherMode)
{
if (formatOut == formatIn) {
ma_copy_memory_64(pOut, pIn, sampleCount * ma_get_bytes_per_sample(formatOut));
return;
}
switch (formatIn)
{
case ma_format_u8:
{
switch (formatOut)
{
case ma_format_s16: ma_pcm_u8_to_s16(pOut, pIn, sampleCount, ditherMode); return;
case ma_format_s24: ma_pcm_u8_to_s24(pOut, pIn, sampleCount, ditherMode); return;
case ma_format_s32: ma_pcm_u8_to_s32(pOut, pIn, sampleCount, ditherMode); return;
case ma_format_f32: ma_pcm_u8_to_f32(pOut, pIn, sampleCount, ditherMode); return;
default: break;
}
} break;
case ma_format_s16:
{
switch (formatOut)
{
case ma_format_u8: ma_pcm_s16_to_u8( pOut, pIn, sampleCount, ditherMode); return;
case ma_format_s24: ma_pcm_s16_to_s24(pOut, pIn, sampleCount, ditherMode); return;
case ma_format_s32: ma_pcm_s16_to_s32(pOut, pIn, sampleCount, ditherMode); return;
case ma_format_f32: ma_pcm_s16_to_f32(pOut, pIn, sampleCount, ditherMode); return;
default: break;
}
} break;
case ma_format_s24:
{
switch (formatOut)
{
case ma_format_u8: ma_pcm_s24_to_u8( pOut, pIn, sampleCount, ditherMode); return;
case ma_format_s16: ma_pcm_s24_to_s16(pOut, pIn, sampleCount, ditherMode); return;
case ma_format_s32: ma_pcm_s24_to_s32(pOut, pIn, sampleCount, ditherMode); return;
case ma_format_f32: ma_pcm_s24_to_f32(pOut, pIn, sampleCount, ditherMode); return;
default: break;
}
} break;
case ma_format_s32:
{
switch (formatOut)
{
case ma_format_u8: ma_pcm_s32_to_u8( pOut, pIn, sampleCount, ditherMode); return;
case ma_format_s16: ma_pcm_s32_to_s16(pOut, pIn, sampleCount, ditherMode); return;
case ma_format_s24: ma_pcm_s32_to_s24(pOut, pIn, sampleCount, ditherMode); return;
case ma_format_f32: ma_pcm_s32_to_f32(pOut, pIn, sampleCount, ditherMode); return;
default: break;
}
} break;
case ma_format_f32:
{
switch (formatOut)
{
case ma_format_u8: ma_pcm_f32_to_u8( pOut, pIn, sampleCount, ditherMode); return;
case ma_format_s16: ma_pcm_f32_to_s16(pOut, pIn, sampleCount, ditherMode); return;
case ma_format_s24: ma_pcm_f32_to_s24(pOut, pIn, sampleCount, ditherMode); return;
case ma_format_s32: ma_pcm_f32_to_s32(pOut, pIn, sampleCount, ditherMode); return;
default: break;
}
} break;
default: break;
}
}
MA_API void ma_convert_pcm_frames_format(void* pOut, ma_format formatOut, const void* pIn, ma_format formatIn, ma_uint64 frameCount, ma_uint32 channels, ma_dither_mode ditherMode)
{
ma_pcm_convert(pOut, formatOut, pIn, formatIn, frameCount * channels, ditherMode);
}
MA_API void ma_deinterleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void* pInterleavedPCMFrames, void** ppDeinterleavedPCMFrames)
{
if (pInterleavedPCMFrames == NULL || ppDeinterleavedPCMFrames == NULL) {
return; /* Invalid args. */
}
/* For efficiency we do this per format. */
switch (format) {
case ma_format_s16:
{
const ma_int16* pSrcS16 = (const ma_int16*)pInterleavedPCMFrames;
ma_uint64 iPCMFrame;
for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; ++iChannel) {
ma_int16* pDstS16 = (ma_int16*)ppDeinterleavedPCMFrames[iChannel];
pDstS16[iPCMFrame] = pSrcS16[iPCMFrame*channels+iChannel];
}
}
} break;
case ma_format_f32:
{
const float* pSrcF32 = (const float*)pInterleavedPCMFrames;
ma_uint64 iPCMFrame;
for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; ++iChannel) {
float* pDstF32 = (float*)ppDeinterleavedPCMFrames[iChannel];
pDstF32[iPCMFrame] = pSrcF32[iPCMFrame*channels+iChannel];
}
}
} break;
default:
{
ma_uint32 sampleSizeInBytes = ma_get_bytes_per_sample(format);
ma_uint64 iPCMFrame;
for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; ++iChannel) {
void* pDst = ma_offset_ptr(ppDeinterleavedPCMFrames[iChannel], iPCMFrame*sampleSizeInBytes);
const void* pSrc = ma_offset_ptr(pInterleavedPCMFrames, (iPCMFrame*channels+iChannel)*sampleSizeInBytes);
memcpy(pDst, pSrc, sampleSizeInBytes);
}
}
} break;
}
}
MA_API void ma_interleave_pcm_frames(ma_format format, ma_uint32 channels, ma_uint64 frameCount, const void** ppDeinterleavedPCMFrames, void* pInterleavedPCMFrames)
{
switch (format)
{
case ma_format_s16:
{
ma_int16* pDstS16 = (ma_int16*)pInterleavedPCMFrames;
ma_uint64 iPCMFrame;
for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; ++iChannel) {
const ma_int16* pSrcS16 = (const ma_int16*)ppDeinterleavedPCMFrames[iChannel];
pDstS16[iPCMFrame*channels+iChannel] = pSrcS16[iPCMFrame];
}
}
} break;
case ma_format_f32:
{
float* pDstF32 = (float*)pInterleavedPCMFrames;
ma_uint64 iPCMFrame;
for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; ++iChannel) {
const float* pSrcF32 = (const float*)ppDeinterleavedPCMFrames[iChannel];
pDstF32[iPCMFrame*channels+iChannel] = pSrcF32[iPCMFrame];
}
}
} break;
default:
{
ma_uint32 sampleSizeInBytes = ma_get_bytes_per_sample(format);
ma_uint64 iPCMFrame;
for (iPCMFrame = 0; iPCMFrame < frameCount; ++iPCMFrame) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; ++iChannel) {
void* pDst = ma_offset_ptr(pInterleavedPCMFrames, (iPCMFrame*channels+iChannel)*sampleSizeInBytes);
const void* pSrc = ma_offset_ptr(ppDeinterleavedPCMFrames[iChannel], iPCMFrame*sampleSizeInBytes);
memcpy(pDst, pSrc, sampleSizeInBytes);
}
}
} break;
}
}
/**************************************************************************************************************************************************************
Biquad Filter
**************************************************************************************************************************************************************/
#ifndef MA_BIQUAD_FIXED_POINT_SHIFT
#define MA_BIQUAD_FIXED_POINT_SHIFT 14
#endif
static ma_int32 ma_biquad_float_to_fp(double x)
{
return (ma_int32)(x * (1 << MA_BIQUAD_FIXED_POINT_SHIFT));
}
MA_API ma_biquad_config ma_biquad_config_init(ma_format format, ma_uint32 channels, double b0, double b1, double b2, double a0, double a1, double a2)
{
ma_biquad_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.b0 = b0;
config.b1 = b1;
config.b2 = b2;
config.a0 = a0;
config.a1 = a1;
config.a2 = a2;
return config;
}
typedef struct
{
size_t sizeInBytes;
size_t r1Offset;
size_t r2Offset;
} ma_biquad_heap_layout;
static ma_result ma_biquad_get_heap_layout(const ma_biquad_config* pConfig, ma_biquad_heap_layout* pHeapLayout)
{
MA_ASSERT(pHeapLayout != NULL);
MA_ZERO_OBJECT(pHeapLayout);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
if (pConfig->a0 == 0) {
return MA_INVALID_ARGS; /* Division by zero. */
}
/* Only supporting f32 and s16. */
if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
return MA_INVALID_ARGS;
}
/* The format cannot be changed after initialization. */
if (pBQ->format != ma_format_unknown && pBQ->format != pConfig->format) {
return MA_INVALID_OPERATION;
}
/* The channel count cannot be changed after initialization. */
if (pBQ->channels != 0 && pBQ->channels != pConfig->channels) {
return MA_INVALID_OPERATION;
}
pBQ->format = pConfig->format;
pBQ->channels = pConfig->channels;
/* Normalize. */
if (pConfig->format == ma_format_f32) {
pBQ->b0.f32 = (float)(pConfig->b0 / pConfig->a0);
pBQ->b1.f32 = (float)(pConfig->b1 / pConfig->a0);
pBQ->b2.f32 = (float)(pConfig->b2 / pConfig->a0);
pBQ->a1.f32 = (float)(pConfig->a1 / pConfig->a0);
pBQ->a2.f32 = (float)(pConfig->a2 / pConfig->a0);
} else {
pBQ->b0.s32 = ma_biquad_float_to_fp(pConfig->b0 / pConfig->a0);
pBQ->b1.s32 = ma_biquad_float_to_fp(pConfig->b1 / pConfig->a0);
pBQ->b2.s32 = ma_biquad_float_to_fp(pConfig->b2 / pConfig->a0);
pBQ->a1.s32 = ma_biquad_float_to_fp(pConfig->a1 / pConfig->a0);
pBQ->a2.s32 = ma_biquad_float_to_fp(pConfig->a2 / pConfig->a0);
}
return MA_SUCCESS;
}
MA_API ma_result ma_biquad_clear_cache(ma_biquad* pBQ)
{
if (pBQ == NULL) {
return MA_INVALID_ARGS;
}
if (pBQ->format == ma_format_f32) {
pBQ->pR1->f32 = 0;
pBQ->pR2->f32 = 0;
} else {
pBQ->pR1->s32 = 0;
pBQ->pR2->s32 = 0;
}
return MA_SUCCESS;
}
static MA_INLINE void ma_biquad_process_pcm_frame_f32__direct_form_2_transposed(ma_biquad* pBQ, float* pY, const float* pX)
{
ma_uint32 c;
const ma_uint32 channels = pBQ->channels;
const float b0 = pBQ->b0.f32;
const float b1 = pBQ->b1.f32;
const float b2 = pBQ->b2.f32;
const float a1 = pBQ->a1.f32;
const float a2 = pBQ->a2.f32;
MA_ASSUME(channels > 0);
for (c = 0; c < channels; c += 1) {
float r1 = pBQ->pR1[c].f32;
float r2 = pBQ->pR2[c].f32;
float x = pX[c];
float y;
y = b0*x + r1;
r1 = b1*x - a1*y + r2;
r2 = b2*x - a2*y;
pY[c] = y;
pBQ->pR1[c].f32 = r1;
pBQ->pR2[c].f32 = r2;
}
}
static MA_INLINE void ma_biquad_process_pcm_frame_f32(ma_biquad* pBQ, float* pY, const float* pX)
{
ma_biquad_process_pcm_frame_f32__direct_form_2_transposed(pBQ, pY, pX);
}
static MA_INLINE void ma_biquad_process_pcm_frame_s16__direct_form_2_transposed(ma_biquad* pBQ, ma_int16* pY, const ma_int16* pX)
{
ma_uint32 c;
const ma_uint32 channels = pBQ->channels;
const ma_int32 b0 = pBQ->b0.s32;
const ma_int32 b1 = pBQ->b1.s32;
const ma_int32 b2 = pBQ->b2.s32;
const ma_int32 a1 = pBQ->a1.s32;
const ma_int32 a2 = pBQ->a2.s32;
MA_ASSUME(channels > 0);
for (c = 0; c < channels; c += 1) {
ma_int32 r1 = pBQ->pR1[c].s32;
ma_int32 r2 = pBQ->pR2[c].s32;
ma_int32 x = pX[c];
ma_int32 y;
y = (b0*x + r1) >> MA_BIQUAD_FIXED_POINT_SHIFT;
r1 = (b1*x - a1*y + r2);
r2 = (b2*x - a2*y);
pY[c] = (ma_int16)ma_clamp(y, -32768, 32767);
pBQ->pR1[c].s32 = r1;
pBQ->pR2[c].s32 = r2;
}
}
static MA_INLINE void ma_biquad_process_pcm_frame_s16(ma_biquad* pBQ, ma_int16* pY, const ma_int16* pX)
{
ma_biquad_process_pcm_frame_s16__direct_form_2_transposed(pBQ, pY, pX);
}
MA_API ma_result ma_biquad_process_pcm_frames(ma_biquad* pBQ, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
ma_uint32 n;
if (pBQ == NULL || pFramesOut == NULL || pFramesIn == NULL) {
return MA_INVALID_ARGS;
}
/* Note that the logic below needs to support in-place filtering. That is, it must support the case where pFramesOut and pFramesIn are the same. */
if (pBQ->format == ma_format_f32) {
/* */ float* pY = ( float*)pFramesOut;
const float* pX = (const float*)pFramesIn;
for (n = 0; n < frameCount; n += 1) {
ma_biquad_process_pcm_frame_f32__direct_form_2_transposed(pBQ, pY, pX);
pY += pBQ->channels;
pX += pBQ->channels;
}
} else if (pBQ->format == ma_format_s16) {
/* */ ma_int16* pY = ( ma_int16*)pFramesOut;
const ma_int16* pX = (const ma_int16*)pFramesIn;
for (n = 0; n < frameCount; n += 1) {
ma_biquad_process_pcm_frame_s16__direct_form_2_transposed(pBQ, pY, pX);
pY += pBQ->channels;
pX += pBQ->channels;
}
} else {
MA_ASSERT(MA_FALSE);
return MA_INVALID_ARGS; /* Format not supported. Should never hit this because it's checked in ma_biquad_init() and ma_biquad_reinit(). */
}
return MA_SUCCESS;
}
MA_API ma_uint32 ma_biquad_get_latency(const ma_biquad* pBQ)
{
if (pBQ == NULL) {
return 0;
}
return 2;
}
/**************************************************************************************************************************************************************
Low-Pass Filter
**************************************************************************************************************************************************************/
MA_API ma_lpf1_config ma_lpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency)
{
ma_lpf1_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.cutoffFrequency = cutoffFrequency;
config.q = 0.5;
return config;
}
MA_API ma_lpf2_config ma_lpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q)
{
ma_lpf2_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.cutoffFrequency = cutoffFrequency;
config.q = q;
/* Q cannot be 0 or else it'll result in a division by 0. In this case just default to 0.707107. */
if (config.q == 0) {
config.q = 0.707107;
}
return config;
}
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if (pLPF == NULL) {
return;
}
if (pLPF->_ownsHeap) {
ma_free(pLPF->_pHeap, pAllocationCallbacks);
}
}
MA_API ma_result ma_lpf1_reinit(const ma_lpf1_config* pConfig, ma_lpf1* pLPF)
{
double a;
if (pLPF == NULL || pConfig == NULL) {
return MA_INVALID_ARGS;
}
/* Only supporting f32 and s16. */
if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
return MA_INVALID_ARGS;
}
/* The format cannot be changed after initialization. */
if (pLPF->format != ma_format_unknown && pLPF->format != pConfig->format) {
return MA_INVALID_OPERATION;
}
/* The channel count cannot be changed after initialization. */
if (pLPF->channels != 0 && pLPF->channels != pConfig->channels) {
return MA_INVALID_OPERATION;
}
pLPF->format = pConfig->format;
pLPF->channels = pConfig->channels;
a = ma_expd(-2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate);
if (pConfig->format == ma_format_f32) {
pLPF->a.f32 = (float)a;
} else {
pLPF->a.s32 = ma_biquad_float_to_fp(a);
}
return MA_SUCCESS;
}
MA_API ma_result ma_lpf1_clear_cache(ma_lpf1* pLPF)
{
if (pLPF == NULL) {
return MA_INVALID_ARGS;
}
if (pLPF->format == ma_format_f32) {
pLPF->a.f32 = 0;
} else {
pLPF->a.s32 = 0;
}
return MA_SUCCESS;
}
static MA_INLINE void ma_lpf1_process_pcm_frame_f32(ma_lpf1* pLPF, float* pY, const float* pX)
{
ma_uint32 c;
const ma_uint32 channels = pLPF->channels;
const float a = pLPF->a.f32;
const float b = 1 - a;
MA_ASSUME(channels > 0);
for (c = 0; c < channels; c += 1) {
float r1 = pLPF->pR1[c].f32;
float x = pX[c];
float y;
y = b*x + a*r1;
pY[c] = y;
pLPF->pR1[c].f32 = y;
}
}
static MA_INLINE void ma_lpf1_process_pcm_frame_s16(ma_lpf1* pLPF, ma_int16* pY, const ma_int16* pX)
{
ma_uint32 c;
const ma_uint32 channels = pLPF->channels;
const ma_int32 a = pLPF->a.s32;
const ma_int32 b = ((1 << MA_BIQUAD_FIXED_POINT_SHIFT) - a);
MA_ASSUME(channels > 0);
for (c = 0; c < channels; c += 1) {
ma_int32 r1 = pLPF->pR1[c].s32;
ma_int32 x = pX[c];
ma_int32 y;
y = (b*x + a*r1) >> MA_BIQUAD_FIXED_POINT_SHIFT;
pY[c] = (ma_int16)y;
pLPF->pR1[c].s32 = (ma_int32)y;
}
}
MA_API ma_result ma_lpf1_process_pcm_frames(ma_lpf1* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
ma_uint32 n;
if (pLPF == NULL || pFramesOut == NULL || pFramesIn == NULL) {
return MA_INVALID_ARGS;
}
/* Note that the logic below needs to support in-place filtering. That is, it must support the case where pFramesOut and pFramesIn are the same. */
if (pLPF->format == ma_format_f32) {
/* */ float* pY = ( float*)pFramesOut;
const float* pX = (const float*)pFramesIn;
for (n = 0; n < frameCount; n += 1) {
ma_lpf1_process_pcm_frame_f32(pLPF, pY, pX);
pY += pLPF->channels;
pX += pLPF->channels;
}
} else if (pLPF->format == ma_format_s16) {
/* */ ma_int16* pY = ( ma_int16*)pFramesOut;
const ma_int16* pX = (const ma_int16*)pFramesIn;
for (n = 0; n < frameCount; n += 1) {
ma_lpf1_process_pcm_frame_s16(pLPF, pY, pX);
pY += pLPF->channels;
pX += pLPF->channels;
}
} else {
MA_ASSERT(MA_FALSE);
return MA_INVALID_ARGS; /* Format not supported. Should never hit this because it's checked in ma_biquad_init() and ma_biquad_reinit(). */
}
return MA_SUCCESS;
}
MA_API ma_uint32 ma_lpf1_get_latency(const ma_lpf1* pLPF)
{
if (pLPF == NULL) {
return 0;
}
return 1;
}
static MA_INLINE ma_biquad_config ma_lpf2__get_biquad_config(const ma_lpf2_config* pConfig)
{
ma_biquad_config bqConfig;
double q;
double w;
double s;
double c;
double a;
MA_ASSERT(pConfig != NULL);
q = pConfig->q;
w = 2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate;
s = ma_sind(w);
c = ma_cosd(w);
a = s / (2*q);
bqConfig.b0 = (1 - c) / 2;
bqConfig.b1 = 1 - c;
bqConfig.b2 = (1 - c) / 2;
bqConfig.a0 = 1 + a;
bqConfig.a1 = -2 * c;
bqConfig.a2 = 1 - a;
bqConfig.format = pConfig->format;
bqConfig.channels = pConfig->channels;
return bqConfig;
}
MA_API ma_result ma_lpf2_get_heap_size(const ma_lpf2_config* pConfig, size_t* pHeapSizeInBytes)
{
ma_biquad_config bqConfig;
bqConfig = ma_lpf2__get_biquad_config(pConfig);
return ma_biquad_get_heap_size(&bqConfig, pHeapSizeInBytes);
}
MA_API ma_result ma_lpf2_init_preallocated(const ma_lpf2_config* pConfig, void* pHeap, ma_lpf2* pLPF)
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return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_lpf2_init_preallocated(pConfig, pHeap, pLPF);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pLPF->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
return MA_SUCCESS;
}
MA_API void ma_lpf2_uninit(ma_lpf2* pLPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pLPF == NULL) {
return;
}
ma_biquad_uninit(&pLPF->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
}
MA_API ma_result ma_lpf2_reinit(const ma_lpf2_config* pConfig, ma_lpf2* pLPF)
{
ma_result result;
ma_biquad_config bqConfig;
if (pLPF == NULL || pConfig == NULL) {
return MA_INVALID_ARGS;
}
bqConfig = ma_lpf2__get_biquad_config(pConfig);
result = ma_biquad_reinit(&bqConfig, &pLPF->bq);
if (result != MA_SUCCESS) {
return result;
}
return MA_SUCCESS;
}
MA_API ma_result ma_lpf2_clear_cache(ma_lpf2* pLPF)
{
if (pLPF == NULL) {
return MA_INVALID_ARGS;
}
ma_biquad_clear_cache(&pLPF->bq);
return MA_SUCCESS;
}
static MA_INLINE void ma_lpf2_process_pcm_frame_s16(ma_lpf2* pLPF, ma_int16* pFrameOut, const ma_int16* pFrameIn)
{
ma_biquad_process_pcm_frame_s16(&pLPF->bq, pFrameOut, pFrameIn);
}
static MA_INLINE void ma_lpf2_process_pcm_frame_f32(ma_lpf2* pLPF, float* pFrameOut, const float* pFrameIn)
{
ma_biquad_process_pcm_frame_f32(&pLPF->bq, pFrameOut, pFrameIn);
}
MA_API ma_result ma_lpf2_process_pcm_frames(ma_lpf2* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
if (pLPF == NULL) {
return MA_INVALID_ARGS;
}
return ma_biquad_process_pcm_frames(&pLPF->bq, pFramesOut, pFramesIn, frameCount);
}
MA_API ma_uint32 ma_lpf2_get_latency(const ma_lpf2* pLPF)
{
if (pLPF == NULL) {
return 0;
}
return ma_biquad_get_latency(&pLPF->bq);
}
MA_API ma_lpf_config ma_lpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
{
ma_lpf_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.cutoffFrequency = cutoffFrequency;
config.order = ma_min(order, MA_MAX_FILTER_ORDER);
return config;
}
typedef struct
{
size_t sizeInBytes;
size_t lpf1Offset;
size_t lpf2Offset; /* Offset of the first second order filter. Subsequent filters will come straight after, and will each have the same heap size. */
} ma_lpf_heap_layout;
static void ma_lpf_calculate_sub_lpf_counts(ma_uint32 order, ma_uint32* pLPF1Count, ma_uint32* pLPF2Count)
{
MA_ASSERT(pLPF1Count != NULL);
MA_ASSERT(pLPF2Count != NULL);
*pLPF1Count = order % 2;
*pLPF2Count = order / 2;
}
static ma_result ma_lpf_get_heap_layout(const ma_lpf_config* pConfig, ma_lpf_heap_layout* pHeapLayout)
{
ma_result result;
ma_uint32 lpf1Count;
ma_uint32 lpf2Count;
ma_uint32 ilpf1;
ma_uint32 ilpf2;
MA_ASSERT(pHeapLayout != NULL);
MA_ZERO_OBJECT(pHeapLayout);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->channels == 0) {
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pHeap = NULL;
}
result = ma_lpf_init_preallocated(pConfig, pHeap, pLPF);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pLPF->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
MA_API void ma_lpf_uninit(ma_lpf* pLPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_uint32 ilpf1;
ma_uint32 ilpf2;
if (pLPF == NULL) {
return;
}
for (ilpf1 = 0; ilpf1 < pLPF->lpf1Count; ilpf1 += 1) {
ma_lpf1_uninit(&pLPF->pLPF1[ilpf1], pAllocationCallbacks);
}
for (ilpf2 = 0; ilpf2 < pLPF->lpf2Count; ilpf2 += 1) {
ma_lpf2_uninit(&pLPF->pLPF2[ilpf2], pAllocationCallbacks);
}
if (pLPF->_ownsHeap) {
ma_free(pLPF->_pHeap, pAllocationCallbacks);
}
}
MA_API ma_result ma_lpf_reinit(const ma_lpf_config* pConfig, ma_lpf* pLPF)
{
return ma_lpf_reinit__internal(pConfig, NULL, pLPF, /*isNew*/MA_FALSE);
}
MA_API ma_result ma_lpf_clear_cache(ma_lpf* pLPF)
{
ma_uint32 ilpf1;
ma_uint32 ilpf2;
if (pLPF == NULL) {
return MA_INVALID_ARGS;
}
for (ilpf1 = 0; ilpf1 < pLPF->lpf1Count; ilpf1 += 1) {
ma_lpf1_clear_cache(&pLPF->pLPF1[ilpf1]);
}
for (ilpf2 = 0; ilpf2 < pLPF->lpf2Count; ilpf2 += 1) {
ma_lpf2_clear_cache(&pLPF->pLPF2[ilpf2]);
}
return MA_SUCCESS;
}
static MA_INLINE void ma_lpf_process_pcm_frame_f32(ma_lpf* pLPF, float* pY, const void* pX)
{
ma_uint32 ilpf1;
ma_uint32 ilpf2;
MA_ASSERT(pLPF->format == ma_format_f32);
MA_COPY_MEMORY(pY, pX, ma_get_bytes_per_frame(pLPF->format, pLPF->channels));
for (ilpf1 = 0; ilpf1 < pLPF->lpf1Count; ilpf1 += 1) {
ma_lpf1_process_pcm_frame_f32(&pLPF->pLPF1[ilpf1], pY, pY);
}
for (ilpf2 = 0; ilpf2 < pLPF->lpf2Count; ilpf2 += 1) {
ma_lpf2_process_pcm_frame_f32(&pLPF->pLPF2[ilpf2], pY, pY);
}
}
static MA_INLINE void ma_lpf_process_pcm_frame_s16(ma_lpf* pLPF, ma_int16* pY, const ma_int16* pX)
{
ma_uint32 ilpf1;
ma_uint32 ilpf2;
MA_ASSERT(pLPF->format == ma_format_s16);
MA_COPY_MEMORY(pY, pX, ma_get_bytes_per_frame(pLPF->format, pLPF->channels));
for (ilpf1 = 0; ilpf1 < pLPF->lpf1Count; ilpf1 += 1) {
ma_lpf1_process_pcm_frame_s16(&pLPF->pLPF1[ilpf1], pY, pY);
}
for (ilpf2 = 0; ilpf2 < pLPF->lpf2Count; ilpf2 += 1) {
ma_lpf2_process_pcm_frame_s16(&pLPF->pLPF2[ilpf2], pY, pY);
}
}
MA_API ma_result ma_lpf_process_pcm_frames(ma_lpf* pLPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
ma_result result;
ma_uint32 ilpf1;
ma_uint32 ilpf2;
if (pLPF == NULL) {
return MA_INVALID_ARGS;
}
/* Faster path for in-place. */
if (pFramesOut == pFramesIn) {
for (ilpf1 = 0; ilpf1 < pLPF->lpf1Count; ilpf1 += 1) {
result = ma_lpf1_process_pcm_frames(&pLPF->pLPF1[ilpf1], pFramesOut, pFramesOut, frameCount);
if (result != MA_SUCCESS) {
return result;
}
}
for (ilpf2 = 0; ilpf2 < pLPF->lpf2Count; ilpf2 += 1) {
result = ma_lpf2_process_pcm_frames(&pLPF->pLPF2[ilpf2], pFramesOut, pFramesOut, frameCount);
if (result != MA_SUCCESS) {
return result;
}
}
}
/* Slightly slower path for copying. */
if (pFramesOut != pFramesIn) {
ma_uint32 iFrame;
/* */ if (pLPF->format == ma_format_f32) {
/* */ float* pFramesOutF32 = ( float*)pFramesOut;
const float* pFramesInF32 = (const float*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_lpf_process_pcm_frame_f32(pLPF, pFramesOutF32, pFramesInF32);
pFramesOutF32 += pLPF->channels;
pFramesInF32 += pLPF->channels;
}
} else if (pLPF->format == ma_format_s16) {
/* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_lpf_process_pcm_frame_s16(pLPF, pFramesOutS16, pFramesInS16);
pFramesOutS16 += pLPF->channels;
pFramesInS16 += pLPF->channels;
}
} else {
MA_ASSERT(MA_FALSE);
return MA_INVALID_OPERATION; /* Should never hit this. */
}
}
return MA_SUCCESS;
}
MA_API ma_uint32 ma_lpf_get_latency(const ma_lpf* pLPF)
{
if (pLPF == NULL) {
return 0;
}
return pLPF->lpf2Count*2 + pLPF->lpf1Count;
}
/**************************************************************************************************************************************************************
High-Pass Filtering
**************************************************************************************************************************************************************/
MA_API ma_hpf1_config ma_hpf1_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency)
{
ma_hpf1_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.cutoffFrequency = cutoffFrequency;
return config;
}
MA_API ma_hpf2_config ma_hpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q)
{
ma_hpf2_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.cutoffFrequency = cutoffFrequency;
config.q = q;
/* Q cannot be 0 or else it'll result in a division by 0. In this case just default to 0.707107. */
if (config.q == 0) {
config.q = 0.707107;
}
return config;
}
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pHeap = NULL;
}
result = ma_hpf1_init_preallocated(pConfig, pHeap, pLPF);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pLPF->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
MA_API void ma_hpf1_uninit(ma_hpf1* pHPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pHPF == NULL) {
return;
}
if (pHPF->_ownsHeap) {
ma_free(pHPF->_pHeap, pAllocationCallbacks);
}
}
MA_API ma_result ma_hpf1_reinit(const ma_hpf1_config* pConfig, ma_hpf1* pHPF)
{
double a;
if (pHPF == NULL || pConfig == NULL) {
return MA_INVALID_ARGS;
}
/* Only supporting f32 and s16. */
if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
return MA_INVALID_ARGS;
}
/* The format cannot be changed after initialization. */
if (pHPF->format != ma_format_unknown && pHPF->format != pConfig->format) {
return MA_INVALID_OPERATION;
}
/* The channel count cannot be changed after initialization. */
if (pHPF->channels != 0 && pHPF->channels != pConfig->channels) {
return MA_INVALID_OPERATION;
}
pHPF->format = pConfig->format;
pHPF->channels = pConfig->channels;
a = ma_expd(-2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate);
if (pConfig->format == ma_format_f32) {
pHPF->a.f32 = (float)a;
} else {
pHPF->a.s32 = ma_biquad_float_to_fp(a);
}
return MA_SUCCESS;
}
static MA_INLINE void ma_hpf1_process_pcm_frame_f32(ma_hpf1* pHPF, float* pY, const float* pX)
{
ma_uint32 c;
const ma_uint32 channels = pHPF->channels;
const float a = 1 - pHPF->a.f32;
const float b = 1 - a;
MA_ASSUME(channels > 0);
for (c = 0; c < channels; c += 1) {
float r1 = pHPF->pR1[c].f32;
float x = pX[c];
float y;
y = b*x - a*r1;
pY[c] = y;
pHPF->pR1[c].f32 = y;
}
}
static MA_INLINE void ma_hpf1_process_pcm_frame_s16(ma_hpf1* pHPF, ma_int16* pY, const ma_int16* pX)
{
ma_uint32 c;
const ma_uint32 channels = pHPF->channels;
const ma_int32 a = ((1 << MA_BIQUAD_FIXED_POINT_SHIFT) - pHPF->a.s32);
const ma_int32 b = ((1 << MA_BIQUAD_FIXED_POINT_SHIFT) - a);
MA_ASSUME(channels > 0);
for (c = 0; c < channels; c += 1) {
ma_int32 r1 = pHPF->pR1[c].s32;
ma_int32 x = pX[c];
ma_int32 y;
y = (b*x - a*r1) >> MA_BIQUAD_FIXED_POINT_SHIFT;
pY[c] = (ma_int16)y;
pHPF->pR1[c].s32 = (ma_int32)y;
}
}
MA_API ma_result ma_hpf1_process_pcm_frames(ma_hpf1* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
ma_uint32 n;
if (pHPF == NULL || pFramesOut == NULL || pFramesIn == NULL) {
return MA_INVALID_ARGS;
}
/* Note that the logic below needs to support in-place filtering. That is, it must support the case where pFramesOut and pFramesIn are the same. */
if (pHPF->format == ma_format_f32) {
/* */ float* pY = ( float*)pFramesOut;
const float* pX = (const float*)pFramesIn;
for (n = 0; n < frameCount; n += 1) {
ma_hpf1_process_pcm_frame_f32(pHPF, pY, pX);
pY += pHPF->channels;
pX += pHPF->channels;
}
} else if (pHPF->format == ma_format_s16) {
/* */ ma_int16* pY = ( ma_int16*)pFramesOut;
const ma_int16* pX = (const ma_int16*)pFramesIn;
for (n = 0; n < frameCount; n += 1) {
ma_hpf1_process_pcm_frame_s16(pHPF, pY, pX);
pY += pHPF->channels;
pX += pHPF->channels;
}
} else {
MA_ASSERT(MA_FALSE);
return MA_INVALID_ARGS; /* Format not supported. Should never hit this because it's checked in ma_biquad_init() and ma_biquad_reinit(). */
}
return MA_SUCCESS;
}
MA_API ma_uint32 ma_hpf1_get_latency(const ma_hpf1* pHPF)
{
if (pHPF == NULL) {
return 0;
}
return 1;
}
static MA_INLINE ma_biquad_config ma_hpf2__get_biquad_config(const ma_hpf2_config* pConfig)
{
ma_biquad_config bqConfig;
double q;
double w;
double s;
double c;
double a;
MA_ASSERT(pConfig != NULL);
q = pConfig->q;
w = 2 * MA_PI_D * pConfig->cutoffFrequency / pConfig->sampleRate;
s = ma_sind(w);
c = ma_cosd(w);
a = s / (2*q);
bqConfig.b0 = (1 + c) / 2;
bqConfig.b1 = -(1 + c);
bqConfig.b2 = (1 + c) / 2;
bqConfig.a0 = 1 + a;
bqConfig.a1 = -2 * c;
bqConfig.a2 = 1 - a;
bqConfig.format = pConfig->format;
bqConfig.channels = pConfig->channels;
return bqConfig;
}
MA_API ma_result ma_hpf2_get_heap_size(const ma_hpf2_config* pConfig, size_t* pHeapSizeInBytes)
{
ma_biquad_config bqConfig;
bqConfig = ma_hpf2__get_biquad_config(pConfig);
return ma_biquad_get_heap_size(&bqConfig, pHeapSizeInBytes);
}
MA_API ma_result ma_hpf2_init_preallocated(const ma_hpf2_config* pConfig, void* pHeap, ma_hpf2* pHPF)
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return MA_SUCCESS;
}
MA_API ma_result ma_hpf2_init(const ma_hpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf2* pHPF)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_hpf2_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_hpf2_init_preallocated(pConfig, pHeap, pHPF);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pHPF->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
return MA_SUCCESS;
}
MA_API void ma_hpf2_uninit(ma_hpf2* pHPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pHPF == NULL) {
return;
}
ma_biquad_uninit(&pHPF->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
}
MA_API ma_result ma_hpf2_reinit(const ma_hpf2_config* pConfig, ma_hpf2* pHPF)
{
ma_result result;
ma_biquad_config bqConfig;
if (pHPF == NULL || pConfig == NULL) {
return MA_INVALID_ARGS;
}
bqConfig = ma_hpf2__get_biquad_config(pConfig);
result = ma_biquad_reinit(&bqConfig, &pHPF->bq);
if (result != MA_SUCCESS) {
return result;
}
return MA_SUCCESS;
}
static MA_INLINE void ma_hpf2_process_pcm_frame_s16(ma_hpf2* pHPF, ma_int16* pFrameOut, const ma_int16* pFrameIn)
{
ma_biquad_process_pcm_frame_s16(&pHPF->bq, pFrameOut, pFrameIn);
}
static MA_INLINE void ma_hpf2_process_pcm_frame_f32(ma_hpf2* pHPF, float* pFrameOut, const float* pFrameIn)
{
ma_biquad_process_pcm_frame_f32(&pHPF->bq, pFrameOut, pFrameIn);
}
MA_API ma_result ma_hpf2_process_pcm_frames(ma_hpf2* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
if (pHPF == NULL) {
return MA_INVALID_ARGS;
}
return ma_biquad_process_pcm_frames(&pHPF->bq, pFramesOut, pFramesIn, frameCount);
}
MA_API ma_uint32 ma_hpf2_get_latency(const ma_hpf2* pHPF)
{
if (pHPF == NULL) {
return 0;
}
return ma_biquad_get_latency(&pHPF->bq);
}
MA_API ma_hpf_config ma_hpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
{
ma_hpf_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.cutoffFrequency = cutoffFrequency;
config.order = ma_min(order, MA_MAX_FILTER_ORDER);
return config;
}
typedef struct
{
size_t sizeInBytes;
size_t hpf1Offset;
size_t hpf2Offset; /* Offset of the first second order filter. Subsequent filters will come straight after, and will each have the same heap size. */
} ma_hpf_heap_layout;
static void ma_hpf_calculate_sub_hpf_counts(ma_uint32 order, ma_uint32* pHPF1Count, ma_uint32* pHPF2Count)
{
MA_ASSERT(pHPF1Count != NULL);
MA_ASSERT(pHPF2Count != NULL);
*pHPF1Count = order % 2;
*pHPF2Count = order / 2;
}
static ma_result ma_hpf_get_heap_layout(const ma_hpf_config* pConfig, ma_hpf_heap_layout* pHeapLayout)
{
ma_result result;
ma_uint32 hpf1Count;
ma_uint32 hpf2Count;
ma_uint32 ihpf1;
ma_uint32 ihpf2;
MA_ASSERT(pHeapLayout != NULL);
MA_ZERO_OBJECT(pHeapLayout);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->channels == 0) {
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return ma_hpf_reinit__internal(pConfig, pHeap, pLPF, /*isNew*/MA_TRUE);
}
MA_API ma_result ma_hpf_init(const ma_hpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf* pHPF)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_hpf_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_hpf_init_preallocated(pConfig, pHeap, pHPF);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pHPF->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
MA_API void ma_hpf_uninit(ma_hpf* pHPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_uint32 ihpf1;
ma_uint32 ihpf2;
if (pHPF == NULL) {
return;
}
for (ihpf1 = 0; ihpf1 < pHPF->hpf1Count; ihpf1 += 1) {
ma_hpf1_uninit(&pHPF->pHPF1[ihpf1], pAllocationCallbacks);
}
for (ihpf2 = 0; ihpf2 < pHPF->hpf2Count; ihpf2 += 1) {
ma_hpf2_uninit(&pHPF->pHPF2[ihpf2], pAllocationCallbacks);
}
if (pHPF->_ownsHeap) {
ma_free(pHPF->_pHeap, pAllocationCallbacks);
}
}
MA_API ma_result ma_hpf_reinit(const ma_hpf_config* pConfig, ma_hpf* pHPF)
{
return ma_hpf_reinit__internal(pConfig, NULL, pHPF, /*isNew*/MA_FALSE);
}
MA_API ma_result ma_hpf_process_pcm_frames(ma_hpf* pHPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
ma_result result;
ma_uint32 ihpf1;
ma_uint32 ihpf2;
if (pHPF == NULL) {
return MA_INVALID_ARGS;
}
/* Faster path for in-place. */
if (pFramesOut == pFramesIn) {
for (ihpf1 = 0; ihpf1 < pHPF->hpf1Count; ihpf1 += 1) {
result = ma_hpf1_process_pcm_frames(&pHPF->pHPF1[ihpf1], pFramesOut, pFramesOut, frameCount);
if (result != MA_SUCCESS) {
return result;
}
}
for (ihpf2 = 0; ihpf2 < pHPF->hpf2Count; ihpf2 += 1) {
result = ma_hpf2_process_pcm_frames(&pHPF->pHPF2[ihpf2], pFramesOut, pFramesOut, frameCount);
if (result != MA_SUCCESS) {
return result;
}
}
}
/* Slightly slower path for copying. */
if (pFramesOut != pFramesIn) {
ma_uint32 iFrame;
/* */ if (pHPF->format == ma_format_f32) {
/* */ float* pFramesOutF32 = ( float*)pFramesOut;
const float* pFramesInF32 = (const float*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
MA_COPY_MEMORY(pFramesOutF32, pFramesInF32, ma_get_bytes_per_frame(pHPF->format, pHPF->channels));
for (ihpf1 = 0; ihpf1 < pHPF->hpf1Count; ihpf1 += 1) {
ma_hpf1_process_pcm_frame_f32(&pHPF->pHPF1[ihpf1], pFramesOutF32, pFramesOutF32);
}
for (ihpf2 = 0; ihpf2 < pHPF->hpf2Count; ihpf2 += 1) {
ma_hpf2_process_pcm_frame_f32(&pHPF->pHPF2[ihpf2], pFramesOutF32, pFramesOutF32);
}
pFramesOutF32 += pHPF->channels;
pFramesInF32 += pHPF->channels;
}
} else if (pHPF->format == ma_format_s16) {
/* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
MA_COPY_MEMORY(pFramesOutS16, pFramesInS16, ma_get_bytes_per_frame(pHPF->format, pHPF->channels));
for (ihpf1 = 0; ihpf1 < pHPF->hpf1Count; ihpf1 += 1) {
ma_hpf1_process_pcm_frame_s16(&pHPF->pHPF1[ihpf1], pFramesOutS16, pFramesOutS16);
}
for (ihpf2 = 0; ihpf2 < pHPF->hpf2Count; ihpf2 += 1) {
ma_hpf2_process_pcm_frame_s16(&pHPF->pHPF2[ihpf2], pFramesOutS16, pFramesOutS16);
}
pFramesOutS16 += pHPF->channels;
pFramesInS16 += pHPF->channels;
}
} else {
MA_ASSERT(MA_FALSE);
return MA_INVALID_OPERATION; /* Should never hit this. */
}
}
return MA_SUCCESS;
}
MA_API ma_uint32 ma_hpf_get_latency(const ma_hpf* pHPF)
{
if (pHPF == NULL) {
return 0;
}
return pHPF->hpf2Count*2 + pHPF->hpf1Count;
}
/**************************************************************************************************************************************************************
Band-Pass Filtering
**************************************************************************************************************************************************************/
MA_API ma_bpf2_config ma_bpf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, double q)
{
ma_bpf2_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.cutoffFrequency = cutoffFrequency;
config.q = q;
/* Q cannot be 0 or else it'll result in a division by 0. In this case just default to 0.707107. */
if (config.q == 0) {
config.q = 0.707107;
}
return config;
}
static MA_INLINE ma_biquad_config ma_bpf2__get_biquad_config(const ma_bpf2_config* pConfig)
{
ma_biquad_config bqConfig;
double q;
double w;
double s;
double c;
double a;
MA_ASSERT(pConfig != NULL);
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return MA_SUCCESS;
}
MA_API ma_result ma_bpf2_init(const ma_bpf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf2* pBPF)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_bpf2_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_bpf2_init_preallocated(pConfig, pHeap, pBPF);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pBPF->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
return MA_SUCCESS;
}
MA_API void ma_bpf2_uninit(ma_bpf2* pBPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pBPF == NULL) {
return;
}
ma_biquad_uninit(&pBPF->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
}
MA_API ma_result ma_bpf2_reinit(const ma_bpf2_config* pConfig, ma_bpf2* pBPF)
{
ma_result result;
ma_biquad_config bqConfig;
if (pBPF == NULL || pConfig == NULL) {
return MA_INVALID_ARGS;
}
bqConfig = ma_bpf2__get_biquad_config(pConfig);
result = ma_biquad_reinit(&bqConfig, &pBPF->bq);
if (result != MA_SUCCESS) {
return result;
}
return MA_SUCCESS;
}
static MA_INLINE void ma_bpf2_process_pcm_frame_s16(ma_bpf2* pBPF, ma_int16* pFrameOut, const ma_int16* pFrameIn)
{
ma_biquad_process_pcm_frame_s16(&pBPF->bq, pFrameOut, pFrameIn);
}
static MA_INLINE void ma_bpf2_process_pcm_frame_f32(ma_bpf2* pBPF, float* pFrameOut, const float* pFrameIn)
{
ma_biquad_process_pcm_frame_f32(&pBPF->bq, pFrameOut, pFrameIn);
}
MA_API ma_result ma_bpf2_process_pcm_frames(ma_bpf2* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
if (pBPF == NULL) {
return MA_INVALID_ARGS;
}
return ma_biquad_process_pcm_frames(&pBPF->bq, pFramesOut, pFramesIn, frameCount);
}
MA_API ma_uint32 ma_bpf2_get_latency(const ma_bpf2* pBPF)
{
if (pBPF == NULL) {
return 0;
}
return ma_biquad_get_latency(&pBPF->bq);
}
MA_API ma_bpf_config ma_bpf_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
{
ma_bpf_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.cutoffFrequency = cutoffFrequency;
config.order = ma_min(order, MA_MAX_FILTER_ORDER);
return config;
}
typedef struct
{
size_t sizeInBytes;
size_t bpf2Offset;
} ma_bpf_heap_layout;
static ma_result ma_bpf_get_heap_layout(const ma_bpf_config* pConfig, ma_bpf_heap_layout* pHeapLayout)
{
ma_result result;
ma_uint32 bpf2Count;
ma_uint32 ibpf2;
MA_ASSERT(pHeapLayout != NULL);
MA_ZERO_OBJECT(pHeapLayout);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->order > MA_MAX_FILTER_ORDER) {
return MA_INVALID_ARGS;
}
/* We must have an even number of order. */
if ((pConfig->order & 0x1) != 0) {
return MA_INVALID_ARGS;
}
bpf2Count = pConfig->channels / 2;
pHeapLayout->sizeInBytes = 0;
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return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pBPF);
return ma_bpf_reinit__internal(pConfig, pHeap, pBPF, /*isNew*/MA_TRUE);
}
MA_API ma_result ma_bpf_init(const ma_bpf_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf* pBPF)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_bpf_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_bpf_init_preallocated(pConfig, pHeap, pBPF);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pBPF->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
MA_API void ma_bpf_uninit(ma_bpf* pBPF, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_uint32 ibpf2;
if (pBPF == NULL) {
return;
}
for (ibpf2 = 0; ibpf2 < pBPF->bpf2Count; ibpf2 += 1) {
ma_bpf2_uninit(&pBPF->pBPF2[ibpf2], pAllocationCallbacks);
}
if (pBPF->_ownsHeap) {
ma_free(pBPF->_pHeap, pAllocationCallbacks);
}
}
MA_API ma_result ma_bpf_reinit(const ma_bpf_config* pConfig, ma_bpf* pBPF)
{
return ma_bpf_reinit__internal(pConfig, NULL, pBPF, /*isNew*/MA_FALSE);
}
MA_API ma_result ma_bpf_process_pcm_frames(ma_bpf* pBPF, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
ma_result result;
ma_uint32 ibpf2;
if (pBPF == NULL) {
return MA_INVALID_ARGS;
}
/* Faster path for in-place. */
if (pFramesOut == pFramesIn) {
for (ibpf2 = 0; ibpf2 < pBPF->bpf2Count; ibpf2 += 1) {
result = ma_bpf2_process_pcm_frames(&pBPF->pBPF2[ibpf2], pFramesOut, pFramesOut, frameCount);
if (result != MA_SUCCESS) {
return result;
}
}
}
/* Slightly slower path for copying. */
if (pFramesOut != pFramesIn) {
ma_uint32 iFrame;
/* */ if (pBPF->format == ma_format_f32) {
/* */ float* pFramesOutF32 = ( float*)pFramesOut;
const float* pFramesInF32 = (const float*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
MA_COPY_MEMORY(pFramesOutF32, pFramesInF32, ma_get_bytes_per_frame(pBPF->format, pBPF->channels));
for (ibpf2 = 0; ibpf2 < pBPF->bpf2Count; ibpf2 += 1) {
ma_bpf2_process_pcm_frame_f32(&pBPF->pBPF2[ibpf2], pFramesOutF32, pFramesOutF32);
}
pFramesOutF32 += pBPF->channels;
pFramesInF32 += pBPF->channels;
}
} else if (pBPF->format == ma_format_s16) {
/* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
MA_COPY_MEMORY(pFramesOutS16, pFramesInS16, ma_get_bytes_per_frame(pBPF->format, pBPF->channels));
for (ibpf2 = 0; ibpf2 < pBPF->bpf2Count; ibpf2 += 1) {
ma_bpf2_process_pcm_frame_s16(&pBPF->pBPF2[ibpf2], pFramesOutS16, pFramesOutS16);
}
pFramesOutS16 += pBPF->channels;
pFramesInS16 += pBPF->channels;
}
} else {
MA_ASSERT(MA_FALSE);
return MA_INVALID_OPERATION; /* Should never hit this. */
}
}
return MA_SUCCESS;
}
MA_API ma_uint32 ma_bpf_get_latency(const ma_bpf* pBPF)
{
if (pBPF == NULL) {
return 0;
}
return pBPF->bpf2Count*2;
}
/**************************************************************************************************************************************************************
Notching Filter
**************************************************************************************************************************************************************/
MA_API ma_notch2_config ma_notch2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double q, double frequency)
{
ma_notch2_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.q = q;
config.frequency = frequency;
if (config.q == 0) {
config.q = 0.707107;
}
return config;
}
static MA_INLINE ma_biquad_config ma_notch2__get_biquad_config(const ma_notch2_config* pConfig)
{
ma_biquad_config bqConfig;
double q;
double w;
double s;
double c;
double a;
MA_ASSERT(pConfig != NULL);
q = pConfig->q;
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return MA_SUCCESS;
}
MA_API ma_result ma_notch2_init(const ma_notch2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_notch2* pFilter)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_notch2_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_notch2_init_preallocated(pConfig, pHeap, pFilter);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pFilter->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
return MA_SUCCESS;
}
MA_API void ma_notch2_uninit(ma_notch2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pFilter == NULL) {
return;
}
ma_biquad_uninit(&pFilter->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
}
MA_API ma_result ma_notch2_reinit(const ma_notch2_config* pConfig, ma_notch2* pFilter)
{
ma_result result;
ma_biquad_config bqConfig;
if (pFilter == NULL || pConfig == NULL) {
return MA_INVALID_ARGS;
}
bqConfig = ma_notch2__get_biquad_config(pConfig);
result = ma_biquad_reinit(&bqConfig, &pFilter->bq);
if (result != MA_SUCCESS) {
return result;
}
return MA_SUCCESS;
}
static MA_INLINE void ma_notch2_process_pcm_frame_s16(ma_notch2* pFilter, ma_int16* pFrameOut, const ma_int16* pFrameIn)
{
ma_biquad_process_pcm_frame_s16(&pFilter->bq, pFrameOut, pFrameIn);
}
static MA_INLINE void ma_notch2_process_pcm_frame_f32(ma_notch2* pFilter, float* pFrameOut, const float* pFrameIn)
{
ma_biquad_process_pcm_frame_f32(&pFilter->bq, pFrameOut, pFrameIn);
}
MA_API ma_result ma_notch2_process_pcm_frames(ma_notch2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
if (pFilter == NULL) {
return MA_INVALID_ARGS;
}
return ma_biquad_process_pcm_frames(&pFilter->bq, pFramesOut, pFramesIn, frameCount);
}
MA_API ma_uint32 ma_notch2_get_latency(const ma_notch2* pFilter)
{
if (pFilter == NULL) {
return 0;
}
return ma_biquad_get_latency(&pFilter->bq);
}
/**************************************************************************************************************************************************************
Peaking EQ Filter
**************************************************************************************************************************************************************/
MA_API ma_peak2_config ma_peak2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency)
{
ma_peak2_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.gainDB = gainDB;
config.q = q;
config.frequency = frequency;
if (config.q == 0) {
config.q = 0.707107;
}
return config;
}
static MA_INLINE ma_biquad_config ma_peak2__get_biquad_config(const ma_peak2_config* pConfig)
{
ma_biquad_config bqConfig;
double q;
double w;
double s;
double c;
double a;
double A;
MA_ASSERT(pConfig != NULL);
q = pConfig->q;
w = 2 * MA_PI_D * pConfig->frequency / pConfig->sampleRate;
s = ma_sind(w);
c = ma_cosd(w);
a = s / (2*q);
A = ma_powd(10, (pConfig->gainDB / 40));
bqConfig.b0 = 1 + (a * A);
bqConfig.b1 = -2 * c;
bqConfig.b2 = 1 - (a * A);
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return MA_SUCCESS;
}
MA_API ma_result ma_peak2_init(const ma_peak2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_peak2* pFilter)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_peak2_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_peak2_init_preallocated(pConfig, pHeap, pFilter);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pFilter->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
return MA_SUCCESS;
}
MA_API void ma_peak2_uninit(ma_peak2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pFilter == NULL) {
return;
}
ma_biquad_uninit(&pFilter->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
}
MA_API ma_result ma_peak2_reinit(const ma_peak2_config* pConfig, ma_peak2* pFilter)
{
ma_result result;
ma_biquad_config bqConfig;
if (pFilter == NULL || pConfig == NULL) {
return MA_INVALID_ARGS;
}
bqConfig = ma_peak2__get_biquad_config(pConfig);
result = ma_biquad_reinit(&bqConfig, &pFilter->bq);
if (result != MA_SUCCESS) {
return result;
}
return MA_SUCCESS;
}
static MA_INLINE void ma_peak2_process_pcm_frame_s16(ma_peak2* pFilter, ma_int16* pFrameOut, const ma_int16* pFrameIn)
{
ma_biquad_process_pcm_frame_s16(&pFilter->bq, pFrameOut, pFrameIn);
}
static MA_INLINE void ma_peak2_process_pcm_frame_f32(ma_peak2* pFilter, float* pFrameOut, const float* pFrameIn)
{
ma_biquad_process_pcm_frame_f32(&pFilter->bq, pFrameOut, pFrameIn);
}
MA_API ma_result ma_peak2_process_pcm_frames(ma_peak2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
if (pFilter == NULL) {
return MA_INVALID_ARGS;
}
return ma_biquad_process_pcm_frames(&pFilter->bq, pFramesOut, pFramesIn, frameCount);
}
MA_API ma_uint32 ma_peak2_get_latency(const ma_peak2* pFilter)
{
if (pFilter == NULL) {
return 0;
}
return ma_biquad_get_latency(&pFilter->bq);
}
/**************************************************************************************************************************************************************
Low Shelf Filter
**************************************************************************************************************************************************************/
MA_API ma_loshelf2_config ma_loshelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency)
{
ma_loshelf2_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.gainDB = gainDB;
config.shelfSlope = shelfSlope;
config.frequency = frequency;
return config;
}
static MA_INLINE ma_biquad_config ma_loshelf2__get_biquad_config(const ma_loshelf2_config* pConfig)
{
ma_biquad_config bqConfig;
double w;
double s;
double c;
double A;
double S;
double a;
double sqrtA;
MA_ASSERT(pConfig != NULL);
w = 2 * MA_PI_D * pConfig->frequency / pConfig->sampleRate;
s = ma_sind(w);
c = ma_cosd(w);
A = ma_powd(10, (pConfig->gainDB / 40));
S = pConfig->shelfSlope;
a = s/2 * ma_sqrtd((A + 1/A) * (1/S - 1) + 2);
sqrtA = 2*ma_sqrtd(A)*a;
bqConfig.b0 = A * ((A + 1) - (A - 1)*c + sqrtA);
bqConfig.b1 = 2 * A * ((A - 1) - (A + 1)*c);
bqConfig.b2 = A * ((A + 1) - (A - 1)*c - sqrtA);
bqConfig.a0 = (A + 1) + (A - 1)*c + sqrtA;
bqConfig.a1 = -2 * ((A - 1) + (A + 1)*c);
bqConfig.a2 = (A + 1) + (A - 1)*c - sqrtA;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
return MA_SUCCESS;
}
MA_API ma_result ma_loshelf2_init(const ma_loshelf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_loshelf2* pFilter)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_loshelf2_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_loshelf2_init_preallocated(pConfig, pHeap, pFilter);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pFilter->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
return MA_SUCCESS;
}
MA_API void ma_loshelf2_uninit(ma_loshelf2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pFilter == NULL) {
return;
}
ma_biquad_uninit(&pFilter->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
}
MA_API ma_result ma_loshelf2_reinit(const ma_loshelf2_config* pConfig, ma_loshelf2* pFilter)
{
ma_result result;
ma_biquad_config bqConfig;
if (pFilter == NULL || pConfig == NULL) {
return MA_INVALID_ARGS;
}
bqConfig = ma_loshelf2__get_biquad_config(pConfig);
result = ma_biquad_reinit(&bqConfig, &pFilter->bq);
if (result != MA_SUCCESS) {
return result;
}
return MA_SUCCESS;
}
static MA_INLINE void ma_loshelf2_process_pcm_frame_s16(ma_loshelf2* pFilter, ma_int16* pFrameOut, const ma_int16* pFrameIn)
{
ma_biquad_process_pcm_frame_s16(&pFilter->bq, pFrameOut, pFrameIn);
}
static MA_INLINE void ma_loshelf2_process_pcm_frame_f32(ma_loshelf2* pFilter, float* pFrameOut, const float* pFrameIn)
{
ma_biquad_process_pcm_frame_f32(&pFilter->bq, pFrameOut, pFrameIn);
}
MA_API ma_result ma_loshelf2_process_pcm_frames(ma_loshelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
if (pFilter == NULL) {
return MA_INVALID_ARGS;
}
return ma_biquad_process_pcm_frames(&pFilter->bq, pFramesOut, pFramesIn, frameCount);
}
MA_API ma_uint32 ma_loshelf2_get_latency(const ma_loshelf2* pFilter)
{
if (pFilter == NULL) {
return 0;
}
return ma_biquad_get_latency(&pFilter->bq);
}
/**************************************************************************************************************************************************************
High Shelf Filter
**************************************************************************************************************************************************************/
MA_API ma_hishelf2_config ma_hishelf2_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double shelfSlope, double frequency)
{
ma_hishelf2_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.gainDB = gainDB;
config.shelfSlope = shelfSlope;
config.frequency = frequency;
return config;
}
static MA_INLINE ma_biquad_config ma_hishelf2__get_biquad_config(const ma_hishelf2_config* pConfig)
{
ma_biquad_config bqConfig;
double w;
double s;
double c;
double A;
double S;
double a;
double sqrtA;
MA_ASSERT(pConfig != NULL);
w = 2 * MA_PI_D * pConfig->frequency / pConfig->sampleRate;
s = ma_sind(w);
c = ma_cosd(w);
A = ma_powd(10, (pConfig->gainDB / 40));
S = pConfig->shelfSlope;
a = s/2 * ma_sqrtd((A + 1/A) * (1/S - 1) + 2);
sqrtA = 2*ma_sqrtd(A)*a;
bqConfig.b0 = A * ((A + 1) + (A - 1)*c + sqrtA);
bqConfig.b1 = -2 * A * ((A - 1) + (A + 1)*c);
bqConfig.b2 = A * ((A + 1) + (A - 1)*c - sqrtA);
bqConfig.a0 = (A + 1) - (A - 1)*c + sqrtA;
bqConfig.a1 = 2 * ((A - 1) - (A + 1)*c);
bqConfig.a2 = (A + 1) - (A - 1)*c - sqrtA;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
return MA_SUCCESS;
}
MA_API ma_result ma_hishelf2_init(const ma_hishelf2_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hishelf2* pFilter)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_hishelf2_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_hishelf2_init_preallocated(pConfig, pHeap, pFilter);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pFilter->bq._ownsHeap = MA_TRUE; /* <-- This will cause the biquad to take ownership of the heap and free it when it's uninitialized. */
return MA_SUCCESS;
}
MA_API void ma_hishelf2_uninit(ma_hishelf2* pFilter, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pFilter == NULL) {
return;
}
ma_biquad_uninit(&pFilter->bq, pAllocationCallbacks); /* <-- This will free the heap allocation. */
}
MA_API ma_result ma_hishelf2_reinit(const ma_hishelf2_config* pConfig, ma_hishelf2* pFilter)
{
ma_result result;
ma_biquad_config bqConfig;
if (pFilter == NULL || pConfig == NULL) {
return MA_INVALID_ARGS;
}
bqConfig = ma_hishelf2__get_biquad_config(pConfig);
result = ma_biquad_reinit(&bqConfig, &pFilter->bq);
if (result != MA_SUCCESS) {
return result;
}
return MA_SUCCESS;
}
static MA_INLINE void ma_hishelf2_process_pcm_frame_s16(ma_hishelf2* pFilter, ma_int16* pFrameOut, const ma_int16* pFrameIn)
{
ma_biquad_process_pcm_frame_s16(&pFilter->bq, pFrameOut, pFrameIn);
}
static MA_INLINE void ma_hishelf2_process_pcm_frame_f32(ma_hishelf2* pFilter, float* pFrameOut, const float* pFrameIn)
{
ma_biquad_process_pcm_frame_f32(&pFilter->bq, pFrameOut, pFrameIn);
}
MA_API ma_result ma_hishelf2_process_pcm_frames(ma_hishelf2* pFilter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
if (pFilter == NULL) {
return MA_INVALID_ARGS;
}
return ma_biquad_process_pcm_frames(&pFilter->bq, pFramesOut, pFramesIn, frameCount);
}
MA_API ma_uint32 ma_hishelf2_get_latency(const ma_hishelf2* pFilter)
{
if (pFilter == NULL) {
return 0;
}
return ma_biquad_get_latency(&pFilter->bq);
}
/*
Delay
*/
MA_API ma_delay_config ma_delay_config_init(ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 delayInFrames, float decay)
{
ma_delay_config config;
MA_ZERO_OBJECT(&config);
config.channels = channels;
config.sampleRate = sampleRate;
config.delayInFrames = delayInFrames;
config.delayStart = (decay == 0) ? MA_TRUE : MA_FALSE; /* Delay the start if it looks like we're not configuring an echo. */
config.wet = 1;
config.dry = 1;
config.decay = decay;
return config;
}
MA_API ma_result ma_delay_init(const ma_delay_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_delay* pDelay)
{
if (pDelay == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pDelay);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->decay < 0 || pConfig->decay > 1) {
return MA_INVALID_ARGS;
}
pDelay->config = *pConfig;
pDelay->bufferSizeInFrames = pConfig->delayInFrames;
pDelay->cursor = 0;
pDelay->pBuffer = (float*)ma_malloc((size_t)(pDelay->bufferSizeInFrames * ma_get_bytes_per_frame(ma_format_f32, pConfig->channels)), pAllocationCallbacks);
if (pDelay->pBuffer == NULL) {
return MA_OUT_OF_MEMORY;
}
ma_silence_pcm_frames(pDelay->pBuffer, pDelay->bufferSizeInFrames, ma_format_f32, pConfig->channels);
return MA_SUCCESS;
}
MA_API void ma_delay_uninit(ma_delay* pDelay, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pDelay == NULL) {
return;
}
ma_free(pDelay->pBuffer, pAllocationCallbacks);
}
MA_API ma_result ma_delay_process_pcm_frames(ma_delay* pDelay, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount)
{
ma_uint32 iFrame;
ma_uint32 iChannel;
float* pFramesOutF32 = (float*)pFramesOut;
const float* pFramesInF32 = (const float*)pFramesIn;
if (pDelay == NULL || pFramesOut == NULL || pFramesIn == NULL) {
return MA_INVALID_ARGS;
}
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannel = 0; iChannel < pDelay->config.channels; iChannel += 1) {
ma_uint32 iBuffer = (pDelay->cursor * pDelay->config.channels) + iChannel;
if (pDelay->config.delayStart) {
/* Delayed start. */
/* Read */
pFramesOutF32[iChannel] = pDelay->pBuffer[iBuffer] * pDelay->config.wet;
/* Feedback */
pDelay->pBuffer[iBuffer] = (pDelay->pBuffer[iBuffer] * pDelay->config.decay) + (pFramesInF32[iChannel] * pDelay->config.dry);
} else {
/* Immediate start */
/* Feedback */
pDelay->pBuffer[iBuffer] = (pDelay->pBuffer[iBuffer] * pDelay->config.decay) + (pFramesInF32[iChannel] * pDelay->config.dry);
/* Read */
pFramesOutF32[iChannel] = pDelay->pBuffer[iBuffer] * pDelay->config.wet;
}
}
pDelay->cursor = (pDelay->cursor + 1) % pDelay->bufferSizeInFrames;
pFramesOutF32 += pDelay->config.channels;
pFramesInF32 += pDelay->config.channels;
}
return MA_SUCCESS;
}
MA_API void ma_delay_set_wet(ma_delay* pDelay, float value)
{
if (pDelay == NULL) {
return;
}
pDelay->config.wet = value;
}
MA_API float ma_delay_get_wet(const ma_delay* pDelay)
{
if (pDelay == NULL) {
return 0;
}
return pDelay->config.wet;
}
MA_API void ma_delay_set_dry(ma_delay* pDelay, float value)
{
if (pDelay == NULL) {
return;
}
pDelay->config.dry = value;
}
MA_API float ma_delay_get_dry(const ma_delay* pDelay)
{
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
pGainer->pNewGains = (float*)ma_offset_ptr(pHeap, heapLayout.newGainsOffset);
pGainer->config = *pConfig;
pGainer->t = (ma_uint32)-1; /* No interpolation by default. */
for (iChannel = 0; iChannel < pConfig->channels; iChannel += 1) {
pGainer->pOldGains[iChannel] = 1;
pGainer->pNewGains[iChannel] = 1;
}
return MA_SUCCESS;
}
MA_API ma_result ma_gainer_init(const ma_gainer_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_gainer* pGainer)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_gainer_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result; /* Failed to retrieve the size of the heap allocation. */
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_gainer_init_preallocated(pConfig, pHeap, pGainer);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pGainer->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
MA_API void ma_gainer_uninit(ma_gainer* pGainer, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pGainer == NULL) {
return;
}
if (pGainer->_ownsHeap) {
ma_free(pGainer->_pHeap, pAllocationCallbacks);
}
}
static float ma_gainer_calculate_current_gain(const ma_gainer* pGainer, ma_uint32 channel)
{
float a = (float)pGainer->t / pGainer->config.smoothTimeInFrames;
return ma_mix_f32_fast(pGainer->pOldGains[channel], pGainer->pNewGains[channel], a);
}
MA_API ma_result ma_gainer_process_pcm_frames(ma_gainer* pGainer, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
ma_uint64 iFrame;
ma_uint32 iChannel;
float* pFramesOutF32 = (float*)pFramesOut;
const float* pFramesInF32 = (const float*)pFramesIn;
if (pGainer == NULL) {
return MA_INVALID_ARGS;
}
if (pGainer->t >= pGainer->config.smoothTimeInFrames) {
/* Fast path. No gain calculation required. */
ma_copy_and_apply_volume_factor_per_channel_f32(pFramesOutF32, pFramesInF32, frameCount, pGainer->config.channels, pGainer->pNewGains);
/* Now that some frames have been processed we need to make sure future changes to the gain are interpolated. */
if (pGainer->t == (ma_uint32)-1) {
pGainer->t = pGainer->config.smoothTimeInFrames;
}
} else {
/* Slow path. Need to interpolate the gain for each channel individually. */
/* We can allow the input and output buffers to be null in which case we'll just update the internal timer. */
if (pFramesOut != NULL && pFramesIn != NULL) {
float a = (float)pGainer->t / pGainer->config.smoothTimeInFrames;
float d = 1.0f / pGainer->config.smoothTimeInFrames;
ma_uint32 channelCount = pGainer->config.channels;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannel = 0; iChannel < channelCount; iChannel += 1) {
pFramesOutF32[iChannel] = pFramesInF32[iChannel] * ma_mix_f32_fast(pGainer->pOldGains[iChannel], pGainer->pNewGains[iChannel], a);
}
pFramesOutF32 += channelCount;
pFramesInF32 += channelCount;
a += d;
if (a > 1) {
a = 1;
}
}
}
pGainer->t = (ma_uint32)ma_min(pGainer->t + frameCount, pGainer->config.smoothTimeInFrames);
#if 0 /* Reference implementation. */
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
/* We can allow the input and output buffers to be null in which case we'll just update the internal timer. */
if (pFramesOut != NULL && pFramesIn != NULL) {
for (iChannel = 0; iChannel < pGainer->config.channels; iChannel += 1) {
pFramesOutF32[iFrame*pGainer->config.channels + iChannel] = pFramesInF32[iFrame*pGainer->config.channels + iChannel] * ma_gainer_calculate_current_gain(pGainer, iChannel);
}
}
/* Move interpolation time forward, but don't go beyond our smoothing time. */
pGainer->t = ma_min(pGainer->t + 1, pGainer->config.smoothTimeInFrames);
}
#endif
}
return MA_SUCCESS;
}
static void ma_gainer_set_gain_by_index(ma_gainer* pGainer, float newGain, ma_uint32 iChannel)
{
pGainer->pOldGains[iChannel] = ma_gainer_calculate_current_gain(pGainer, iChannel);
pGainer->pNewGains[iChannel] = newGain;
}
static void ma_gainer_reset_smoothing_time(ma_gainer* pGainer)
{
if (pGainer->t == (ma_uint32)-1) {
pGainer->t = pGainer->config.smoothTimeInFrames; /* No smoothing required for initial gains setting. */
} else {
pGainer->t = 0;
}
}
MA_API ma_result ma_gainer_set_gain(ma_gainer* pGainer, float newGain)
{
ma_uint32 iChannel;
if (pGainer == NULL) {
return MA_INVALID_ARGS;
}
for (iChannel = 0; iChannel < pGainer->config.channels; iChannel += 1) {
ma_gainer_set_gain_by_index(pGainer, newGain, iChannel);
}
/* The smoothing time needs to be reset to ensure we always interpolate by the configured smoothing time, but only if it's not the first setting. */
ma_gainer_reset_smoothing_time(pGainer);
return MA_SUCCESS;
}
MA_API ma_result ma_gainer_set_gains(ma_gainer* pGainer, float* pNewGains)
{
ma_uint32 iChannel;
if (pGainer == NULL || pNewGains == NULL) {
return MA_INVALID_ARGS;
}
for (iChannel = 0; iChannel < pGainer->config.channels; iChannel += 1) {
ma_gainer_set_gain_by_index(pGainer, pNewGains[iChannel], iChannel);
}
/* The smoothing time needs to be reset to ensure we always interpolate by the configured smoothing time, but only if it's not the first setting. */
ma_gainer_reset_smoothing_time(pGainer);
return MA_SUCCESS;
}
MA_API ma_panner_config ma_panner_config_init(ma_format format, ma_uint32 channels)
{
ma_panner_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.mode = ma_pan_mode_balance; /* Set to balancing mode by default because it's consistent with other audio engines and most likely what the caller is expecting. */
config.pan = 0;
return config;
}
MA_API ma_result ma_panner_init(const ma_panner_config* pConfig, ma_panner* pPanner)
{
if (pPanner == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pPanner);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
pPanner->format = pConfig->format;
pPanner->channels = pConfig->channels;
pPanner->mode = pConfig->mode;
pPanner->pan = pConfig->pan;
return MA_SUCCESS;
}
static void ma_stereo_balance_pcm_frames_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, float pan)
{
ma_uint64 iFrame;
if (pan > 0) {
float factor = 1.0f - pan;
if (pFramesOut == pFramesIn) {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
pFramesOut[iFrame*2 + 0] = pFramesIn[iFrame*2 + 0] * factor;
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
pFramesOut[iFrame*2 + 0] = pFramesIn[iFrame*2 + 0] * factor;
pFramesOut[iFrame*2 + 1] = pFramesIn[iFrame*2 + 1];
}
}
} else {
float factor = 1.0f + pan;
if (pFramesOut == pFramesIn) {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
pFramesOut[iFrame*2 + 1] = pFramesIn[iFrame*2 + 1] * factor;
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
pFramesOut[iFrame*2 + 0] = pFramesIn[iFrame*2 + 0];
pFramesOut[iFrame*2 + 1] = pFramesIn[iFrame*2 + 1] * factor;
}
}
}
}
static void ma_stereo_balance_pcm_frames(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_format format, float pan)
{
if (pan == 0) {
/* Fast path. No panning required. */
if (pFramesOut == pFramesIn) {
/* No-op */
} else {
ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, format, 2);
}
return;
}
switch (format) {
case ma_format_f32: ma_stereo_balance_pcm_frames_f32((float*)pFramesOut, (float*)pFramesIn, frameCount, pan); break;
/* Unknown format. Just copy. */
default:
{
ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, format, 2);
} break;
}
}
static void ma_stereo_pan_pcm_frames_f32(float* pFramesOut, const float* pFramesIn, ma_uint64 frameCount, float pan)
{
ma_uint64 iFrame;
if (pan > 0) {
float factorL0 = 1.0f - pan;
float factorL1 = 0.0f + pan;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float sample0 = (pFramesIn[iFrame*2 + 0] * factorL0);
float sample1 = (pFramesIn[iFrame*2 + 0] * factorL1) + pFramesIn[iFrame*2 + 1];
pFramesOut[iFrame*2 + 0] = sample0;
pFramesOut[iFrame*2 + 1] = sample1;
}
} else {
float factorR0 = 0.0f - pan;
float factorR1 = 1.0f + pan;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float sample0 = pFramesIn[iFrame*2 + 0] + (pFramesIn[iFrame*2 + 1] * factorR0);
float sample1 = (pFramesIn[iFrame*2 + 1] * factorR1);
pFramesOut[iFrame*2 + 0] = sample0;
pFramesOut[iFrame*2 + 1] = sample1;
}
}
}
static void ma_stereo_pan_pcm_frames(void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount, ma_format format, float pan)
{
if (pan == 0) {
/* Fast path. No panning required. */
if (pFramesOut == pFramesIn) {
/* No-op */
} else {
ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, format, 2);
}
return;
}
switch (format) {
case ma_format_f32: ma_stereo_pan_pcm_frames_f32((float*)pFramesOut, (float*)pFramesIn, frameCount, pan); break;
/* Unknown format. Just copy. */
default:
{
ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, format, 2);
} break;
}
}
MA_API ma_result ma_panner_process_pcm_frames(ma_panner* pPanner, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
if (pPanner == NULL || pFramesOut == NULL || pFramesIn == NULL) {
return MA_INVALID_ARGS;
}
if (pPanner->channels == 2) {
/* Stereo case. For now assume channel 0 is left and channel right is 1, but should probably add support for a channel map. */
if (pPanner->mode == ma_pan_mode_balance) {
ma_stereo_balance_pcm_frames(pFramesOut, pFramesIn, frameCount, pPanner->format, pPanner->pan);
} else {
ma_stereo_pan_pcm_frames(pFramesOut, pFramesIn, frameCount, pPanner->format, pPanner->pan);
}
} else {
if (pPanner->channels == 1) {
/* Panning has no effect on mono streams. */
ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, pPanner->format, pPanner->channels);
} else {
/* For now we're not going to support non-stereo set ups. Not sure how I want to handle this case just yet. */
ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, pPanner->format, pPanner->channels);
}
}
return MA_SUCCESS;
}
MA_API void ma_panner_set_mode(ma_panner* pPanner, ma_pan_mode mode)
{
if (pPanner == NULL) {
return;
}
pPanner->mode = mode;
}
MA_API ma_pan_mode ma_panner_get_mode(const ma_panner* pPanner)
{
if (pPanner == NULL) {
return ma_pan_mode_balance;
}
return pPanner->mode;
}
MA_API void ma_panner_set_pan(ma_panner* pPanner, float pan)
{
if (pPanner == NULL) {
return;
}
pPanner->pan = ma_clamp(pan, -1.0f, 1.0f);
}
MA_API float ma_panner_get_pan(const ma_panner* pPanner)
{
if (pPanner == NULL) {
return 0;
}
return pPanner->pan;
}
MA_API ma_fader_config ma_fader_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
{
ma_fader_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
return config;
}
MA_API ma_result ma_fader_init(const ma_fader_config* pConfig, ma_fader* pFader)
{
if (pFader == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pFader);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
/* Only f32 is supported for now. */
if (pConfig->format != ma_format_f32) {
return MA_INVALID_ARGS;
}
pFader->config = *pConfig;
pFader->volumeBeg = 1;
pFader->volumeEnd = 1;
pFader->lengthInFrames = 0;
pFader->cursorInFrames = 0;
return MA_SUCCESS;
}
MA_API ma_result ma_fader_process_pcm_frames(ma_fader* pFader, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
if (pFader == NULL) {
return MA_INVALID_ARGS;
}
/*
For now we need to clamp frameCount so that the cursor never overflows 32-bits. This is required for
the conversion to a float which we use for the linear interpolation. This might be changed later.
*/
if (frameCount + pFader->cursorInFrames > UINT_MAX) {
frameCount = UINT_MAX - pFader->cursorInFrames;
}
/* Optimized path if volumeBeg and volumeEnd are equal. */
if (pFader->volumeBeg == pFader->volumeEnd) {
if (pFader->volumeBeg == 1) {
/* Straight copy. */
ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, pFader->config.format, pFader->config.channels);
} else {
/* Copy with volume. */
ma_copy_and_apply_volume_and_clip_pcm_frames(pFramesOut, pFramesIn, frameCount, pFader->config.format, pFader->config.channels, pFader->volumeEnd);
}
} else {
/* Slower path. Volumes are different, so may need to do an interpolation. */
if (pFader->cursorInFrames >= pFader->lengthInFrames) {
/* Fast path. We've gone past the end of the fade period so just apply the end volume to all samples. */
ma_copy_and_apply_volume_and_clip_pcm_frames(pFramesOut, pFramesIn, frameCount, pFader->config.format, pFader->config.channels, pFader->volumeEnd);
} else {
/* Slow path. This is where we do the actual fading. */
ma_uint64 iFrame;
ma_uint32 iChannel;
/* For now we only support f32. Support for other formats will be added later. */
if (pFader->config.format == ma_format_f32) {
const float* pFramesInF32 = (const float*)pFramesIn;
/* */ float* pFramesOutF32 = ( float*)pFramesOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float a = (ma_uint32)ma_min(pFader->cursorInFrames + iFrame, pFader->lengthInFrames) / (float)((ma_uint32)pFader->lengthInFrames); /* Safe cast due to the frameCount clamp at the top of this function. */
float volume = ma_mix_f32_fast(pFader->volumeBeg, pFader->volumeEnd, a);
for (iChannel = 0; iChannel < pFader->config.channels; iChannel += 1) {
pFramesOutF32[iFrame*pFader->config.channels + iChannel] = pFramesInF32[iFrame*pFader->config.channels + iChannel] * volume;
}
}
} else {
return MA_NOT_IMPLEMENTED;
}
}
}
pFader->cursorInFrames += frameCount;
return MA_SUCCESS;
}
MA_API void ma_fader_get_data_format(const ma_fader* pFader, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate)
{
if (pFader == NULL) {
return;
}
if (pFormat != NULL) {
*pFormat = pFader->config.format;
}
if (pChannels != NULL) {
*pChannels = pFader->config.channels;
}
if (pSampleRate != NULL) {
*pSampleRate = pFader->config.sampleRate;
}
}
MA_API void ma_fader_set_fade(ma_fader* pFader, float volumeBeg, float volumeEnd, ma_uint64 lengthInFrames)
{
if (pFader == NULL) {
return;
}
/* If the volume is negative, use current volume. */
if (volumeBeg < 0) {
volumeBeg = ma_fader_get_current_volume(pFader);
}
/*
The length needs to be clamped to 32-bits due to how we convert it to a float for linear
interpolation reasons. I might change this requirement later, but for now it's not important.
*/
if (lengthInFrames > UINT_MAX) {
lengthInFrames = UINT_MAX;
}
pFader->volumeBeg = volumeBeg;
pFader->volumeEnd = volumeEnd;
pFader->lengthInFrames = lengthInFrames;
pFader->cursorInFrames = 0; /* Reset cursor. */
}
MA_API float ma_fader_get_current_volume(ma_fader* pFader)
{
if (pFader == NULL) {
return 0.0f;
}
/* The current volume depends on the position of the cursor. */
if (pFader->cursorInFrames == 0) {
return pFader->volumeBeg;
} else if (pFader->cursorInFrames >= pFader->lengthInFrames) {
return pFader->volumeEnd;
} else {
/* The cursor is somewhere inside the fading period. We can figure this out with a simple linear interpoluation between volumeBeg and volumeEnd based on our cursor position. */
return ma_mix_f32_fast(pFader->volumeBeg, pFader->volumeEnd, (ma_uint32)pFader->cursorInFrames / (float)((ma_uint32)pFader->lengthInFrames)); /* Safe cast to uint32 because we clamp it in ma_fader_process_pcm_frames(). */
}
}
MA_API ma_vec3f ma_vec3f_init_3f(float x, float y, float z)
{
ma_vec3f v;
v.x = x;
v.y = y;
v.z = z;
return v;
}
MA_API ma_vec3f ma_vec3f_sub(ma_vec3f a, ma_vec3f b)
{
return ma_vec3f_init_3f(
a.x - b.x,
a.y - b.y,
a.z - b.z
);
}
MA_API ma_vec3f ma_vec3f_neg(ma_vec3f a)
{
return ma_vec3f_init_3f(
-a.x,
-a.y,
-a.z
);
}
MA_API float ma_vec3f_dot(ma_vec3f a, ma_vec3f b)
{
return a.x*b.x + a.y*b.y + a.z*b.z;
}
MA_API float ma_vec3f_len2(ma_vec3f v)
{
return ma_vec3f_dot(v, v);
}
MA_API float ma_vec3f_len(ma_vec3f v)
{
return (float)ma_sqrtd(ma_vec3f_len2(v));
}
MA_API float ma_vec3f_dist(ma_vec3f a, ma_vec3f b)
{
return ma_vec3f_len(ma_vec3f_sub(a, b));
}
MA_API ma_vec3f ma_vec3f_normalize(ma_vec3f v)
{
float f;
float l = ma_vec3f_len(v);
if (l == 0) {
return ma_vec3f_init_3f(0, 0, 0);
}
f = 1 / l;
v.x *= f;
v.y *= f;
v.z *= f;
return v;
}
MA_API ma_vec3f ma_vec3f_cross(ma_vec3f a, ma_vec3f b)
{
return ma_vec3f_init_3f(
a.y*b.z - a.z*b.y,
a.z*b.x - a.x*b.z,
a.x*b.y - a.y*b.x
);
}
static void ma_channel_map_apply_f32(float* pFramesOut, const ma_channel* pChannelMapOut, ma_uint32 channelsOut, const float* pFramesIn, const ma_channel* pChannelMapIn, ma_uint32 channelsIn, ma_uint64 frameCount, ma_channel_mix_mode mode, ma_mono_ex...
static ma_bool32 ma_is_spatial_channel_position(ma_channel channelPosition);
#ifndef MA_DEFAULT_SPEED_OF_SOUND
#define MA_DEFAULT_SPEED_OF_SOUND 343.3f
#endif
/*
These vectors represent the direction that speakers are facing from the center point. They're used
for panning in the spatializer. Must be normalized.
*/
static ma_vec3f g_maChannelDirections[MA_CHANNEL_POSITION_COUNT] = {
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_NONE */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_MONO */
{-0.7071f, 0.0f, -0.7071f }, /* MA_CHANNEL_FRONT_LEFT */
{+0.7071f, 0.0f, -0.7071f }, /* MA_CHANNEL_FRONT_RIGHT */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_FRONT_CENTER */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_LFE */
{-0.7071f, 0.0f, +0.7071f }, /* MA_CHANNEL_BACK_LEFT */
{+0.7071f, 0.0f, +0.7071f }, /* MA_CHANNEL_BACK_RIGHT */
{-0.3162f, 0.0f, -0.9487f }, /* MA_CHANNEL_FRONT_LEFT_CENTER */
{+0.3162f, 0.0f, -0.9487f }, /* MA_CHANNEL_FRONT_RIGHT_CENTER */
{ 0.0f, 0.0f, +1.0f }, /* MA_CHANNEL_BACK_CENTER */
{-1.0f, 0.0f, 0.0f }, /* MA_CHANNEL_SIDE_LEFT */
{+1.0f, 0.0f, 0.0f }, /* MA_CHANNEL_SIDE_RIGHT */
{ 0.0f, +1.0f, 0.0f }, /* MA_CHANNEL_TOP_CENTER */
{-0.5774f, +0.5774f, -0.5774f }, /* MA_CHANNEL_TOP_FRONT_LEFT */
{ 0.0f, +0.7071f, -0.7071f }, /* MA_CHANNEL_TOP_FRONT_CENTER */
{+0.5774f, +0.5774f, -0.5774f }, /* MA_CHANNEL_TOP_FRONT_RIGHT */
{-0.5774f, +0.5774f, +0.5774f }, /* MA_CHANNEL_TOP_BACK_LEFT */
{ 0.0f, +0.7071f, +0.7071f }, /* MA_CHANNEL_TOP_BACK_CENTER */
{+0.5774f, +0.5774f, +0.5774f }, /* MA_CHANNEL_TOP_BACK_RIGHT */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_0 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_1 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_2 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_3 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_4 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_5 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_6 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_7 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_8 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_9 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_10 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_11 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_12 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_13 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_14 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_15 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_16 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_17 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_18 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_19 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_20 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_21 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_22 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_23 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_24 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_25 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_26 */
{ 0.0f, 0.0f, -1.0f }, /* MA_CHANNEL_AUX_27 */
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
return MA_SUCCESS;
}
MA_API void ma_spatializer_uninit(ma_spatializer* pSpatializer, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pSpatializer == NULL) {
return;
}
ma_gainer_uninit(&pSpatializer->gainer, pAllocationCallbacks);
if (pSpatializer->_ownsHeap) {
ma_free(pSpatializer->_pHeap, pAllocationCallbacks);
}
}
static float ma_calculate_angular_gain(ma_vec3f dirA, ma_vec3f dirB, float coneInnerAngleInRadians, float coneOuterAngleInRadians, float coneOuterGain)
{
/*
Angular attenuation.
Unlike distance gain, the math for this is not specified by the OpenAL spec so we'll just go ahead and figure
this out for ourselves at the expense of possibly being inconsistent with other implementations.
To do cone attenuation, I'm just using the same math that we'd use to implement a basic spotlight in OpenGL. We
just need to get the direction from the source to the listener and then do a dot product against that and the
direction of the spotlight. Then we just compare that dot product against the cosine of the inner and outer
angles. If the dot product is greater than the the outer angle, we just use coneOuterGain. If it's less than
the inner angle, we just use a gain of 1. Otherwise we linearly interpolate between 1 and coneOuterGain.
*/
if (coneInnerAngleInRadians < 6.283185f) {
float angularGain = 1;
float cutoffInner = (float)ma_cosd(coneInnerAngleInRadians*0.5f);
float cutoffOuter = (float)ma_cosd(coneOuterAngleInRadians*0.5f);
float d;
d = ma_vec3f_dot(dirA, dirB);
if (d > cutoffInner) {
/* It's inside the inner angle. */
angularGain = 1;
} else {
/* It's outside the inner angle. */
if (d > cutoffOuter) {
/* It's between the inner and outer angle. We need to linearly interpolate between 1 and coneOuterGain. */
angularGain = ma_mix_f32(coneOuterGain, 1, (d - cutoffOuter) / (cutoffInner - cutoffOuter));
} else {
/* It's outside the outer angle. */
angularGain = coneOuterGain;
}
}
/*printf("d = %f; cutoffInner = %f; cutoffOuter = %f; angularGain = %f\n", d, cutoffInner, cutoffOuter, angularGain);*/
return angularGain;
} else {
/* Inner angle is 360 degrees so no need to do any attenuation. */
return 1;
}
}
MA_API ma_result ma_spatializer_process_pcm_frames(ma_spatializer* pSpatializer, ma_spatializer_listener* pListener, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
ma_channel* pChannelMapIn = pSpatializer->pChannelMapIn;
ma_channel* pChannelMapOut = pListener->config.pChannelMapOut;
if (pSpatializer == NULL) {
return MA_INVALID_ARGS;
}
/* If we're not spatializing we need to run an optimized path. */
if (c89atomic_load_i32(&pSpatializer->attenuationModel) == ma_attenuation_model_none) {
if (ma_spatializer_listener_is_enabled(pListener)) {
/* No attenuation is required, but we'll need to do some channel conversion. */
if (pSpatializer->channelsIn == pSpatializer->channelsOut) {
ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, ma_format_f32, pSpatializer->channelsIn);
} else {
ma_channel_map_apply_f32((float*)pFramesOut, pChannelMapOut, pSpatializer->channelsOut, (const float*)pFramesIn, pChannelMapIn, pSpatializer->channelsIn, frameCount, ma_channel_mix_mode_rectangular, ma_mono_expansion_mode_default); ...
}
} else {
/* The listener is disabled. Output silence. */
ma_silence_pcm_frames(pFramesOut, frameCount, ma_format_f32, pSpatializer->channelsOut);
}
/*
We're not doing attenuation so don't bother with doppler for now. I'm not sure if this is
the correct thinking so might need to review this later.
*/
pSpatializer->dopplerPitch = 1;
} else {
/*
Let's first determine which listener the sound is closest to. Need to keep in mind that we
might not have a world or any listeners, in which case we just spatializer based on the
listener being positioned at the origin (0, 0, 0).
*/
ma_vec3f relativePosNormalized;
ma_vec3f relativePos; /* The position relative to the listener. */
ma_vec3f relativeDir; /* The direction of the sound, relative to the listener. */
ma_vec3f listenerVel; /* The volocity of the listener. For doppler pitch calculation. */
float speedOfSound;
float distance = 0;
float gain = 1;
ma_uint32 iChannel;
const ma_uint32 channelsOut = pSpatializer->channelsOut;
const ma_uint32 channelsIn = pSpatializer->channelsIn;
float minDistance = ma_spatializer_get_min_distance(pSpatializer);
float maxDistance = ma_spatializer_get_max_distance(pSpatializer);
float rolloff = ma_spatializer_get_rolloff(pSpatializer);
float dopplerFactor = ma_spatializer_get_doppler_factor(pSpatializer);
/*
We'll need the listener velocity for doppler pitch calculations. The speed of sound is
defined by the listener, so we'll grab that here too.
*/
if (pListener != NULL) {
listenerVel = pListener->velocity;
speedOfSound = pListener->config.speedOfSound;
} else {
listenerVel = ma_vec3f_init_3f(0, 0, 0);
speedOfSound = MA_DEFAULT_SPEED_OF_SOUND;
}
if (pListener == NULL || ma_spatializer_get_positioning(pSpatializer) == ma_positioning_relative) {
/* There's no listener or we're using relative positioning. */
relativePos = pSpatializer->position;
relativeDir = pSpatializer->direction;
} else {
/*
We've found a listener and we're using absolute positioning. We need to transform the
sound's position and direction so that it's relative to listener. Later on we'll use
this for determining the factors to apply to each channel to apply the panning effect.
*/
ma_spatializer_get_relative_position_and_direction(pSpatializer, pListener, &relativePos, &relativeDir);
}
distance = ma_vec3f_len(relativePos);
/* We've gathered the data, so now we can apply some spatialization. */
switch (ma_spatializer_get_attenuation_model(pSpatializer)) {
case ma_attenuation_model_inverse:
{
gain = ma_attenuation_inverse(distance, minDistance, maxDistance, rolloff);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
/*
We're supporting angular gain on the listener as well for those who want to reduce the volume of sounds that
are positioned behind the listener. On default settings, this will have no effect.
*/
if (pListener != NULL && pListener->config.coneInnerAngleInRadians < 6.283185f) {
ma_vec3f listenerDirection;
float listenerInnerAngle;
float listenerOuterAngle;
float listenerOuterGain;
if (pListener->config.handedness == ma_handedness_right) {
listenerDirection = ma_vec3f_init_3f(0, 0, -1);
} else {
listenerDirection = ma_vec3f_init_3f(0, 0, +1);
}
listenerInnerAngle = pListener->config.coneInnerAngleInRadians;
listenerOuterAngle = pListener->config.coneOuterAngleInRadians;
listenerOuterGain = pListener->config.coneOuterGain;
gain *= ma_calculate_angular_gain(listenerDirection, relativePosNormalized, listenerInnerAngle, listenerOuterAngle, listenerOuterGain);
}
} else {
/* The sound is right on top of the listener. Don't do any angular attenuation. */
}
/* Clamp the gain. */
gain = ma_clamp(gain, ma_spatializer_get_min_gain(pSpatializer), ma_spatializer_get_max_gain(pSpatializer));
/*
Panning. This is where we'll apply the gain and convert to the output channel count. We have an optimized path for
when we're converting to a mono stream. In that case we don't really need to do any panning - we just apply the
gain to the final output.
*/
/*printf("distance=%f; gain=%f\n", distance, gain);*/
/* We must have a valid channel map here to ensure we spatialize properly. */
MA_ASSERT(pChannelMapOut != NULL);
/*
We're not converting to mono so we'll want to apply some panning. This is where the feeling of something being
to the left, right, infront or behind the listener is calculated. I'm just using a basic model here. Note that
the code below is not based on any specific algorithm. I'm just implementing this off the top of my head and
seeing how it goes. There might be better ways to do this.
To determine the direction of the sound relative to a speaker I'm using dot products. Each speaker is given a
direction. For example, the left channel in a stereo system will be -1 on the X axis and the right channel will
be +1 on the X axis. A dot product is performed against the direction vector of the channel and the normalized
position of the sound.
*/
for (iChannel = 0; iChannel < channelsOut; iChannel += 1) {
pSpatializer->pNewChannelGainsOut[iChannel] = gain;
}
/*
Convert to our output channel count. If the listener is disabled we just output silence here. We cannot ignore
the whole section of code here because we need to update some internal spatialization state.
*/
if (ma_spatializer_listener_is_enabled(pListener)) {
ma_channel_map_apply_f32((float*)pFramesOut, pChannelMapOut, channelsOut, (const float*)pFramesIn, pChannelMapIn, channelsIn, frameCount, ma_channel_mix_mode_rectangular, ma_mono_expansion_mode_default);
} else {
ma_silence_pcm_frames(pFramesOut, frameCount, ma_format_f32, pSpatializer->channelsOut);
}
/*
Calculate our per-channel gains. We do this based on the normalized relative position of the sound and it's
relation to the direction of the channel.
*/
if (distance > 0) {
ma_vec3f unitPos = relativePos;
float distanceInv = 1/distance;
unitPos.x *= distanceInv;
unitPos.y *= distanceInv;
unitPos.z *= distanceInv;
for (iChannel = 0; iChannel < channelsOut; iChannel += 1) {
ma_channel channelOut;
float d;
float dMin;
channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannel);
if (ma_is_spatial_channel_position(channelOut)) {
d = ma_mix_f32_fast(1, ma_vec3f_dot(unitPos, ma_get_channel_direction(channelOut)), ma_spatializer_get_directional_attenuation_factor(pSpatializer));
} else {
d = 1; /* It's not a spatial channel so there's no real notion of direction. */
}
/*
In my testing, if the panning effect is too aggressive it makes spatialization feel uncomfortable.
The "dMin" variable below is used to control the aggressiveness of the panning effect. When set to
0, panning will be most extreme and any sounds that are positioned on the opposite side of the
speaker will be completely silent from that speaker. Not only does this feel uncomfortable, it
doesn't even remotely represent the real world at all because sounds that come from your right side
are still clearly audible from your left side. Setting "dMin" to 1 will result in no panning at
all, which is also not ideal. By setting it to something greater than 0, the spatialization effect
becomes much less dramatic and a lot more bearable.
Summary: 0 = more extreme panning; 1 = no panning.
*/
dMin = 0.2f; /* TODO: Consider making this configurable. */
/*
At this point, "d" will be positive if the sound is on the same side as the channel and negative if
it's on the opposite side. It will be in the range of -1..1. There's two ways I can think of to
calculate a panning value. The first is to simply convert it to 0..1, however this has a problem
which I'm not entirely happy with. Considering a stereo system, when a sound is positioned right
in front of the listener it'll result in each speaker getting a gain of 0.5. I don't know if I like
the idea of having a scaling factor of 0.5 being applied to a sound when it's sitting right in front
of the listener. I would intuitively expect that to be played at full volume, or close to it.
The second idea I think of is to only apply a reduction in gain when the sound is on the opposite
side of the speaker. That is, reduce the gain only when the dot product is negative. The problem
with this is that there will not be any attenuation as the sound sweeps around the 180 degrees
where the dot product is positive. The idea with this option is that you leave the gain at 1 when
the sound is being played on the same side as the speaker and then you just reduce the volume when
the sound is on the other side.
The summarize, I think the first option should give a better sense of spatialization, but the second
option is better for preserving the sound's power.
UPDATE: In my testing, I find the first option to sound better. You can feel the sense of space a
bit better, but you can also hear the reduction in volume when it's right in front.
*/
#if 1
{
/*
Scale the dot product from -1..1 to 0..1. Will result in a sound directly in front losing power
by being played at 0.5 gain.
*/
d = (d + 1) * 0.5f; /* -1..1 to 0..1 */
d = ma_max(d, dMin);
pSpatializer->pNewChannelGainsOut[iChannel] *= d;
}
#else
{
/*
Only reduce the volume of the sound if it's on the opposite side. This path keeps the volume more
consistent, but comes at the expense of a worse sense of space and positioning.
*/
if (d < 0) {
d += 1; /* Move into the positive range. */
d = ma_max(d, dMin);
channelGainsOut[iChannel] *= d;
}
}
#endif
}
} else {
/* Assume the sound is right on top of us. Don't do any panning. */
}
/* Now we need to apply the volume to each channel. This needs to run through the gainer to ensure we get a smooth volume transition. */
ma_gainer_set_gains(&pSpatializer->gainer, pSpatializer->pNewChannelGainsOut);
ma_gainer_process_pcm_frames(&pSpatializer->gainer, pFramesOut, pFramesOut, frameCount);
/*
Before leaving we'll want to update our doppler pitch so that the caller can apply some
pitch shifting if they desire. Note that we need to negate the relative position here
because the doppler calculation needs to be source-to-listener, but ours is listener-to-
source.
*/
if (dopplerFactor > 0) {
pSpatializer->dopplerPitch = ma_doppler_pitch(ma_vec3f_sub(pListener->position, pSpatializer->position), pSpatializer->velocity, listenerVel, speedOfSound, dopplerFactor);
} else {
pSpatializer->dopplerPitch = 1;
}
}
return MA_SUCCESS;
}
MA_API ma_uint32 ma_spatializer_get_input_channels(const ma_spatializer* pSpatializer)
{
if (pSpatializer == NULL) {
return 0;
}
return pSpatializer->channelsIn;
}
MA_API ma_uint32 ma_spatializer_get_output_channels(const ma_spatializer* pSpatializer)
{
if (pSpatializer == NULL) {
return 0;
}
return pSpatializer->channelsOut;
}
MA_API void ma_spatializer_set_attenuation_model(ma_spatializer* pSpatializer, ma_attenuation_model attenuationModel)
{
if (pSpatializer == NULL) {
return;
}
c89atomic_exchange_i32(&pSpatializer->attenuationModel, attenuationModel);
}
MA_API ma_attenuation_model ma_spatializer_get_attenuation_model(const ma_spatializer* pSpatializer)
{
if (pSpatializer == NULL) {
return ma_attenuation_model_none;
}
return (ma_attenuation_model)c89atomic_load_i32(&pSpatializer->attenuationModel);
}
MA_API void ma_spatializer_set_positioning(ma_spatializer* pSpatializer, ma_positioning positioning)
{
if (pSpatializer == NULL) {
return;
}
c89atomic_exchange_i32(&pSpatializer->positioning, positioning);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
size_t lpfOffset;
} ma_linear_resampler_heap_layout;
static void ma_linear_resampler_adjust_timer_for_new_rate(ma_linear_resampler* pResampler, ma_uint32 oldSampleRateOut, ma_uint32 newSampleRateOut)
{
/*
So what's happening here? Basically we need to adjust the fractional component of the time advance based on the new rate. The old time advance will
be based on the old sample rate, but we are needing to adjust it to that it's based on the new sample rate.
*/
ma_uint32 oldRateTimeWhole = pResampler->inTimeFrac / oldSampleRateOut; /* <-- This should almost never be anything other than 0, but leaving it here to make this more general and robust just in case. */
ma_uint32 oldRateTimeFract = pResampler->inTimeFrac % oldSampleRateOut;
pResampler->inTimeFrac =
(oldRateTimeWhole * newSampleRateOut) +
((oldRateTimeFract * newSampleRateOut) / oldSampleRateOut);
/* Make sure the fractional part is less than the output sample rate. */
pResampler->inTimeInt += pResampler->inTimeFrac / pResampler->config.sampleRateOut;
pResampler->inTimeFrac = pResampler->inTimeFrac % pResampler->config.sampleRateOut;
}
static ma_result ma_linear_resampler_set_rate_internal(ma_linear_resampler* pResampler, void* pHeap, ma_linear_resampler_heap_layout* pHeapLayout, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_bool32 isResamplerAlreadyInitialized)
{
ma_result result;
ma_uint32 gcf;
ma_uint32 lpfSampleRate;
double lpfCutoffFrequency;
ma_lpf_config lpfConfig;
ma_uint32 oldSampleRateOut; /* Required for adjusting time advance down the bottom. */
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
if (sampleRateIn == 0 || sampleRateOut == 0) {
return MA_INVALID_ARGS;
}
oldSampleRateOut = pResampler->config.sampleRateOut;
pResampler->config.sampleRateIn = sampleRateIn;
pResampler->config.sampleRateOut = sampleRateOut;
/* Simplify the sample rate. */
gcf = ma_gcf_u32(pResampler->config.sampleRateIn, pResampler->config.sampleRateOut);
pResampler->config.sampleRateIn /= gcf;
pResampler->config.sampleRateOut /= gcf;
/* Always initialize the low-pass filter, even when the order is 0. */
if (pResampler->config.lpfOrder > MA_MAX_FILTER_ORDER) {
return MA_INVALID_ARGS;
}
lpfSampleRate = (ma_uint32)(ma_max(pResampler->config.sampleRateIn, pResampler->config.sampleRateOut));
lpfCutoffFrequency = ( double)(ma_min(pResampler->config.sampleRateIn, pResampler->config.sampleRateOut) * 0.5 * pResampler->config.lpfNyquistFactor);
lpfConfig = ma_lpf_config_init(pResampler->config.format, pResampler->config.channels, lpfSampleRate, lpfCutoffFrequency, pResampler->config.lpfOrder);
/*
If the resampler is alreay initialized we don't want to do a fresh initialization of the low-pass filter because it will result in the cached frames
getting cleared. Instead we re-initialize the filter which will maintain any cached frames.
*/
if (isResamplerAlreadyInitialized) {
result = ma_lpf_reinit(&lpfConfig, &pResampler->lpf);
} else {
result = ma_lpf_init_preallocated(&lpfConfig, ma_offset_ptr(pHeap, pHeapLayout->lpfOffset), &pResampler->lpf);
}
if (result != MA_SUCCESS) {
return result;
}
pResampler->inAdvanceInt = pResampler->config.sampleRateIn / pResampler->config.sampleRateOut;
pResampler->inAdvanceFrac = pResampler->config.sampleRateIn % pResampler->config.sampleRateOut;
/* Our timer was based on the old rate. We need to adjust it so that it's based on the new rate. */
ma_linear_resampler_adjust_timer_for_new_rate(pResampler, oldSampleRateOut, pResampler->config.sampleRateOut);
return MA_SUCCESS;
}
static ma_result ma_linear_resampler_get_heap_layout(const ma_linear_resampler_config* pConfig, ma_linear_resampler_heap_layout* pHeapLayout)
{
MA_ASSERT(pHeapLayout != NULL);
MA_ZERO_OBJECT(pHeapLayout);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->format != ma_format_f32 && pConfig->format != ma_format_s16) {
return MA_INVALID_ARGS;
}
if (pConfig->channels == 0) {
return MA_INVALID_ARGS;
}
pHeapLayout->sizeInBytes = 0;
/* x0 */
pHeapLayout->x0Offset = pHeapLayout->sizeInBytes;
if (pConfig->format == ma_format_f32) {
pHeapLayout->sizeInBytes += sizeof(float) * pConfig->channels;
} else {
pHeapLayout->sizeInBytes += sizeof(ma_int16) * pConfig->channels;
}
/* x1 */
pHeapLayout->x1Offset = pHeapLayout->sizeInBytes;
if (pConfig->format == ma_format_f32) {
pHeapLayout->sizeInBytes += sizeof(float) * pConfig->channels;
} else {
pHeapLayout->sizeInBytes += sizeof(ma_int16) * pConfig->channels;
}
/* LPF */
pHeapLayout->lpfOffset = pHeapLayout->sizeInBytes;
{
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
return MA_SUCCESS;
}
MA_API ma_result ma_linear_resampler_get_heap_size(const ma_linear_resampler_config* pConfig, size_t* pHeapSizeInBytes)
{
ma_result result;
ma_linear_resampler_heap_layout heapLayout;
if (pHeapSizeInBytes == NULL) {
return MA_INVALID_ARGS;
}
*pHeapSizeInBytes = 0;
result = ma_linear_resampler_get_heap_layout(pConfig, &heapLayout);
if (result != MA_SUCCESS) {
return result;
}
*pHeapSizeInBytes = heapLayout.sizeInBytes;
return MA_SUCCESS;
}
MA_API ma_result ma_linear_resampler_init_preallocated(const ma_linear_resampler_config* pConfig, void* pHeap, ma_linear_resampler* pResampler)
{
ma_result result;
ma_linear_resampler_heap_layout heapLayout;
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pResampler);
result = ma_linear_resampler_get_heap_layout(pConfig, &heapLayout);
if (result != MA_SUCCESS) {
return result;
}
pResampler->config = *pConfig;
pResampler->_pHeap = pHeap;
MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
if (pConfig->format == ma_format_f32) {
pResampler->x0.f32 = (float*)ma_offset_ptr(pHeap, heapLayout.x0Offset);
pResampler->x1.f32 = (float*)ma_offset_ptr(pHeap, heapLayout.x1Offset);
} else {
pResampler->x0.s16 = (ma_int16*)ma_offset_ptr(pHeap, heapLayout.x0Offset);
pResampler->x1.s16 = (ma_int16*)ma_offset_ptr(pHeap, heapLayout.x1Offset);
}
/* Setting the rate will set up the filter and time advances for us. */
result = ma_linear_resampler_set_rate_internal(pResampler, pHeap, &heapLayout, pConfig->sampleRateIn, pConfig->sampleRateOut, /* isResamplerAlreadyInitialized = */ MA_FALSE);
if (result != MA_SUCCESS) {
return result;
}
pResampler->inTimeInt = 1; /* Set this to one to force an input sample to always be loaded for the first output frame. */
pResampler->inTimeFrac = 0;
return MA_SUCCESS;
}
MA_API ma_result ma_linear_resampler_init(const ma_linear_resampler_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_linear_resampler* pResampler)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_linear_resampler_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_linear_resampler_init_preallocated(pConfig, pHeap, pResampler);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pResampler->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
MA_API void ma_linear_resampler_uninit(ma_linear_resampler* pResampler, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pResampler == NULL) {
return;
}
ma_lpf_uninit(&pResampler->lpf, pAllocationCallbacks);
if (pResampler->_ownsHeap) {
ma_free(pResampler->_pHeap, pAllocationCallbacks);
}
}
static MA_INLINE ma_int16 ma_linear_resampler_mix_s16(ma_int16 x, ma_int16 y, ma_int32 a, const ma_int32 shift)
{
ma_int32 b;
ma_int32 c;
ma_int32 r;
MA_ASSERT(a <= (1<<shift));
b = x * ((1<<shift) - a);
c = y * a;
r = b + c;
return (ma_int16)(r >> shift);
}
static void ma_linear_resampler_interpolate_frame_s16(ma_linear_resampler* pResampler, ma_int16* MA_RESTRICT pFrameOut)
{
ma_uint32 c;
ma_uint32 a;
const ma_uint32 channels = pResampler->config.channels;
const ma_uint32 shift = 12;
MA_ASSERT(pResampler != NULL);
MA_ASSERT(pFrameOut != NULL);
a = (pResampler->inTimeFrac << shift) / pResampler->config.sampleRateOut;
MA_ASSUME(channels > 0);
for (c = 0; c < channels; c += 1) {
ma_int16 s = ma_linear_resampler_mix_s16(pResampler->x0.s16[c], pResampler->x1.s16[c], a, shift);
pFrameOut[c] = s;
}
}
static void ma_linear_resampler_interpolate_frame_f32(ma_linear_resampler* pResampler, float* MA_RESTRICT pFrameOut)
{
ma_uint32 c;
float a;
const ma_uint32 channels = pResampler->config.channels;
MA_ASSERT(pResampler != NULL);
MA_ASSERT(pFrameOut != NULL);
a = (float)pResampler->inTimeFrac / pResampler->config.sampleRateOut;
MA_ASSUME(channels > 0);
for (c = 0; c < channels; c += 1) {
float s = ma_mix_f32_fast(pResampler->x0.f32[c], pResampler->x1.f32[c], a);
pFrameOut[c] = s;
}
}
static ma_result ma_linear_resampler_process_pcm_frames_s16_downsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
const ma_int16* pFramesInS16;
/* */ ma_int16* pFramesOutS16;
ma_uint64 frameCountIn;
ma_uint64 frameCountOut;
ma_uint64 framesProcessedIn;
ma_uint64 framesProcessedOut;
MA_ASSERT(pResampler != NULL);
MA_ASSERT(pFrameCountIn != NULL);
MA_ASSERT(pFrameCountOut != NULL);
pFramesInS16 = (const ma_int16*)pFramesIn;
pFramesOutS16 = ( ma_int16*)pFramesOut;
frameCountIn = *pFrameCountIn;
frameCountOut = *pFrameCountOut;
framesProcessedIn = 0;
framesProcessedOut = 0;
while (framesProcessedOut < frameCountOut) {
/* Before interpolating we need to load the buffers. When doing this we need to ensure we run every input sample through the filter. */
while (pResampler->inTimeInt > 0 && frameCountIn > framesProcessedIn) {
ma_uint32 iChannel;
if (pFramesInS16 != NULL) {
for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel];
pResampler->x1.s16[iChannel] = pFramesInS16[iChannel];
}
pFramesInS16 += pResampler->config.channels;
} else {
for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel];
pResampler->x1.s16[iChannel] = 0;
}
}
/* Filter. */
ma_lpf_process_pcm_frame_s16(&pResampler->lpf, pResampler->x1.s16, pResampler->x1.s16);
framesProcessedIn += 1;
pResampler->inTimeInt -= 1;
}
if (pResampler->inTimeInt > 0) {
break; /* Ran out of input data. */
}
/* Getting here means the frames have been loaded and filtered and we can generate the next output frame. */
if (pFramesOutS16 != NULL) {
MA_ASSERT(pResampler->inTimeInt == 0);
ma_linear_resampler_interpolate_frame_s16(pResampler, pFramesOutS16);
pFramesOutS16 += pResampler->config.channels;
}
framesProcessedOut += 1;
/* Advance time forward. */
pResampler->inTimeInt += pResampler->inAdvanceInt;
pResampler->inTimeFrac += pResampler->inAdvanceFrac;
if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) {
pResampler->inTimeFrac -= pResampler->config.sampleRateOut;
pResampler->inTimeInt += 1;
}
}
*pFrameCountIn = framesProcessedIn;
*pFrameCountOut = framesProcessedOut;
return MA_SUCCESS;
}
static ma_result ma_linear_resampler_process_pcm_frames_s16_upsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
const ma_int16* pFramesInS16;
/* */ ma_int16* pFramesOutS16;
ma_uint64 frameCountIn;
ma_uint64 frameCountOut;
ma_uint64 framesProcessedIn;
ma_uint64 framesProcessedOut;
MA_ASSERT(pResampler != NULL);
MA_ASSERT(pFrameCountIn != NULL);
MA_ASSERT(pFrameCountOut != NULL);
pFramesInS16 = (const ma_int16*)pFramesIn;
pFramesOutS16 = ( ma_int16*)pFramesOut;
frameCountIn = *pFrameCountIn;
frameCountOut = *pFrameCountOut;
framesProcessedIn = 0;
framesProcessedOut = 0;
while (framesProcessedOut < frameCountOut) {
/* Before interpolating we need to load the buffers. */
while (pResampler->inTimeInt > 0 && frameCountIn > framesProcessedIn) {
ma_uint32 iChannel;
if (pFramesInS16 != NULL) {
for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel];
pResampler->x1.s16[iChannel] = pFramesInS16[iChannel];
}
pFramesInS16 += pResampler->config.channels;
} else {
for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
pResampler->x0.s16[iChannel] = pResampler->x1.s16[iChannel];
pResampler->x1.s16[iChannel] = 0;
}
}
framesProcessedIn += 1;
pResampler->inTimeInt -= 1;
}
if (pResampler->inTimeInt > 0) {
break; /* Ran out of input data. */
}
/* Getting here means the frames have been loaded and we can generate the next output frame. */
if (pFramesOutS16 != NULL) {
MA_ASSERT(pResampler->inTimeInt == 0);
ma_linear_resampler_interpolate_frame_s16(pResampler, pFramesOutS16);
/* Filter. */
ma_lpf_process_pcm_frame_s16(&pResampler->lpf, pFramesOutS16, pFramesOutS16);
pFramesOutS16 += pResampler->config.channels;
}
framesProcessedOut += 1;
/* Advance time forward. */
pResampler->inTimeInt += pResampler->inAdvanceInt;
pResampler->inTimeFrac += pResampler->inAdvanceFrac;
if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) {
pResampler->inTimeFrac -= pResampler->config.sampleRateOut;
pResampler->inTimeInt += 1;
}
}
*pFrameCountIn = framesProcessedIn;
*pFrameCountOut = framesProcessedOut;
return MA_SUCCESS;
}
static ma_result ma_linear_resampler_process_pcm_frames_s16(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
MA_ASSERT(pResampler != NULL);
if (pResampler->config.sampleRateIn > pResampler->config.sampleRateOut) {
return ma_linear_resampler_process_pcm_frames_s16_downsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
} else {
return ma_linear_resampler_process_pcm_frames_s16_upsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
}
}
static ma_result ma_linear_resampler_process_pcm_frames_f32_downsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
const float* pFramesInF32;
/* */ float* pFramesOutF32;
ma_uint64 frameCountIn;
ma_uint64 frameCountOut;
ma_uint64 framesProcessedIn;
ma_uint64 framesProcessedOut;
MA_ASSERT(pResampler != NULL);
MA_ASSERT(pFrameCountIn != NULL);
MA_ASSERT(pFrameCountOut != NULL);
pFramesInF32 = (const float*)pFramesIn;
pFramesOutF32 = ( float*)pFramesOut;
frameCountIn = *pFrameCountIn;
frameCountOut = *pFrameCountOut;
framesProcessedIn = 0;
framesProcessedOut = 0;
while (framesProcessedOut < frameCountOut) {
/* Before interpolating we need to load the buffers. When doing this we need to ensure we run every input sample through the filter. */
while (pResampler->inTimeInt > 0 && frameCountIn > framesProcessedIn) {
ma_uint32 iChannel;
if (pFramesInF32 != NULL) {
for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel];
pResampler->x1.f32[iChannel] = pFramesInF32[iChannel];
}
pFramesInF32 += pResampler->config.channels;
} else {
for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel];
pResampler->x1.f32[iChannel] = 0;
}
}
/* Filter. */
ma_lpf_process_pcm_frame_f32(&pResampler->lpf, pResampler->x1.f32, pResampler->x1.f32);
framesProcessedIn += 1;
pResampler->inTimeInt -= 1;
}
if (pResampler->inTimeInt > 0) {
break; /* Ran out of input data. */
}
/* Getting here means the frames have been loaded and filtered and we can generate the next output frame. */
if (pFramesOutF32 != NULL) {
MA_ASSERT(pResampler->inTimeInt == 0);
ma_linear_resampler_interpolate_frame_f32(pResampler, pFramesOutF32);
pFramesOutF32 += pResampler->config.channels;
}
framesProcessedOut += 1;
/* Advance time forward. */
pResampler->inTimeInt += pResampler->inAdvanceInt;
pResampler->inTimeFrac += pResampler->inAdvanceFrac;
if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) {
pResampler->inTimeFrac -= pResampler->config.sampleRateOut;
pResampler->inTimeInt += 1;
}
}
*pFrameCountIn = framesProcessedIn;
*pFrameCountOut = framesProcessedOut;
return MA_SUCCESS;
}
static ma_result ma_linear_resampler_process_pcm_frames_f32_upsample(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
const float* pFramesInF32;
/* */ float* pFramesOutF32;
ma_uint64 frameCountIn;
ma_uint64 frameCountOut;
ma_uint64 framesProcessedIn;
ma_uint64 framesProcessedOut;
MA_ASSERT(pResampler != NULL);
MA_ASSERT(pFrameCountIn != NULL);
MA_ASSERT(pFrameCountOut != NULL);
pFramesInF32 = (const float*)pFramesIn;
pFramesOutF32 = ( float*)pFramesOut;
frameCountIn = *pFrameCountIn;
frameCountOut = *pFrameCountOut;
framesProcessedIn = 0;
framesProcessedOut = 0;
while (framesProcessedOut < frameCountOut) {
/* Before interpolating we need to load the buffers. */
while (pResampler->inTimeInt > 0 && frameCountIn > framesProcessedIn) {
ma_uint32 iChannel;
if (pFramesInF32 != NULL) {
for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel];
pResampler->x1.f32[iChannel] = pFramesInF32[iChannel];
}
pFramesInF32 += pResampler->config.channels;
} else {
for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
pResampler->x0.f32[iChannel] = pResampler->x1.f32[iChannel];
pResampler->x1.f32[iChannel] = 0;
}
}
framesProcessedIn += 1;
pResampler->inTimeInt -= 1;
}
if (pResampler->inTimeInt > 0) {
break; /* Ran out of input data. */
}
/* Getting here means the frames have been loaded and we can generate the next output frame. */
if (pFramesOutF32 != NULL) {
MA_ASSERT(pResampler->inTimeInt == 0);
ma_linear_resampler_interpolate_frame_f32(pResampler, pFramesOutF32);
/* Filter. */
ma_lpf_process_pcm_frame_f32(&pResampler->lpf, pFramesOutF32, pFramesOutF32);
pFramesOutF32 += pResampler->config.channels;
}
framesProcessedOut += 1;
/* Advance time forward. */
pResampler->inTimeInt += pResampler->inAdvanceInt;
pResampler->inTimeFrac += pResampler->inAdvanceFrac;
if (pResampler->inTimeFrac >= pResampler->config.sampleRateOut) {
pResampler->inTimeFrac -= pResampler->config.sampleRateOut;
pResampler->inTimeInt += 1;
}
}
*pFrameCountIn = framesProcessedIn;
*pFrameCountOut = framesProcessedOut;
return MA_SUCCESS;
}
static ma_result ma_linear_resampler_process_pcm_frames_f32(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
MA_ASSERT(pResampler != NULL);
if (pResampler->config.sampleRateIn > pResampler->config.sampleRateOut) {
return ma_linear_resampler_process_pcm_frames_f32_downsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
} else {
return ma_linear_resampler_process_pcm_frames_f32_upsample(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
}
}
MA_API ma_result ma_linear_resampler_process_pcm_frames(ma_linear_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
/* */ if (pResampler->config.format == ma_format_s16) {
return ma_linear_resampler_process_pcm_frames_s16(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
} else if (pResampler->config.format == ma_format_f32) {
return ma_linear_resampler_process_pcm_frames_f32(pResampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
} else {
/* Should never get here. Getting here means the format is not supported and you didn't check the return value of ma_linear_resampler_init(). */
MA_ASSERT(MA_FALSE);
return MA_INVALID_ARGS;
}
}
MA_API ma_result ma_linear_resampler_set_rate(ma_linear_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
{
return ma_linear_resampler_set_rate_internal(pResampler, NULL, NULL, sampleRateIn, sampleRateOut, /* isResamplerAlreadyInitialized = */ MA_TRUE);
}
MA_API ma_result ma_linear_resampler_set_rate_ratio(ma_linear_resampler* pResampler, float ratioInOut)
{
ma_uint32 n;
ma_uint32 d;
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
if (ratioInOut <= 0) {
return MA_INVALID_ARGS;
}
d = 1000;
n = (ma_uint32)(ratioInOut * d);
if (n == 0) {
return MA_INVALID_ARGS; /* Ratio too small. */
}
MA_ASSERT(n != 0);
return ma_linear_resampler_set_rate(pResampler, n, d);
}
MA_API ma_uint64 ma_linear_resampler_get_input_latency(const ma_linear_resampler* pResampler)
{
if (pResampler == NULL) {
return 0;
}
return 1 + ma_lpf_get_latency(&pResampler->lpf);
}
MA_API ma_uint64 ma_linear_resampler_get_output_latency(const ma_linear_resampler* pResampler)
{
if (pResampler == NULL) {
return 0;
}
return ma_linear_resampler_get_input_latency(pResampler) * pResampler->config.sampleRateOut / pResampler->config.sampleRateIn;
}
MA_API ma_result ma_linear_resampler_get_required_input_frame_count(const ma_linear_resampler* pResampler, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount)
{
ma_uint64 inputFrameCount;
if (pInputFrameCount == NULL) {
return MA_INVALID_ARGS;
}
*pInputFrameCount = 0;
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
if (outputFrameCount == 0) {
return MA_SUCCESS;
}
/* Any whole input frames are consumed before the first output frame is generated. */
inputFrameCount = pResampler->inTimeInt;
outputFrameCount -= 1;
/* The rest of the output frames can be calculated in constant time. */
inputFrameCount += outputFrameCount * pResampler->inAdvanceInt;
inputFrameCount += (pResampler->inTimeFrac + (outputFrameCount * pResampler->inAdvanceFrac)) / pResampler->config.sampleRateOut;
*pInputFrameCount = inputFrameCount;
return MA_SUCCESS;
}
MA_API ma_result ma_linear_resampler_get_expected_output_frame_count(const ma_linear_resampler* pResampler, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount)
{
ma_uint64 outputFrameCount;
ma_uint64 preliminaryInputFrameCountFromFrac;
ma_uint64 preliminaryInputFrameCount;
if (pOutputFrameCount == NULL) {
return MA_INVALID_ARGS;
}
*pOutputFrameCount = 0;
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
/*
The first step is to get a preliminary output frame count. This will either be exactly equal to what we need, or less by 1. We need to
determine how many input frames will be consumed by this value. If it's greater than our original input frame count it means we won't
be able to generate an extra frame because we will have run out of input data. Otherwise we will have enough input for the generation
of an extra output frame. This add-by-one logic is necessary due to how the data loading logic works when processing frames.
*/
outputFrameCount = (inputFrameCount * pResampler->config.sampleRateOut) / pResampler->config.sampleRateIn;
/*
We need to determine how many *whole* input frames will have been processed to generate our preliminary output frame count. This is
used in the logic below to determine whether or not we need to add an extra output frame.
*/
preliminaryInputFrameCountFromFrac = (pResampler->inTimeFrac + outputFrameCount*pResampler->inAdvanceFrac) / pResampler->config.sampleRateOut;
preliminaryInputFrameCount = (pResampler->inTimeInt + outputFrameCount*pResampler->inAdvanceInt ) + preliminaryInputFrameCountFromFrac;
/*
If the total number of *whole* input frames that would be required to generate our preliminary output frame count is greather than
the amount of whole input frames we have available as input we need to *not* add an extra output frame as there won't be enough data
to actually process. Otherwise we need to add the extra output frame.
*/
if (preliminaryInputFrameCount <= inputFrameCount) {
outputFrameCount += 1;
}
*pOutputFrameCount = outputFrameCount;
return MA_SUCCESS;
}
MA_API ma_result ma_linear_resampler_reset(ma_linear_resampler* pResampler)
{
ma_uint32 iChannel;
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
/* Timers need to be cleared back to zero. */
pResampler->inTimeInt = 1; /* Set this to one to force an input sample to always be loaded for the first output frame. */
pResampler->inTimeFrac = 0;
/* Cached samples need to be cleared. */
if (pResampler->config.format == ma_format_f32) {
for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
pResampler->x0.f32[iChannel] = 0;
pResampler->x1.f32[iChannel] = 0;
}
} else {
for (iChannel = 0; iChannel < pResampler->config.channels; iChannel += 1) {
pResampler->x0.s16[iChannel] = 0;
pResampler->x1.s16[iChannel] = 0;
}
}
/* The low pass filter needs to have it's cache reset. */
ma_lpf_clear_cache(&pResampler->lpf);
return MA_SUCCESS;
}
/* Linear resampler backend vtable. */
static ma_linear_resampler_config ma_resampling_backend_get_config__linear(const ma_resampler_config* pConfig)
{
ma_linear_resampler_config linearConfig;
linearConfig = ma_linear_resampler_config_init(pConfig->format, pConfig->channels, pConfig->sampleRateIn, pConfig->sampleRateOut);
linearConfig.lpfOrder = pConfig->linear.lpfOrder;
return linearConfig;
}
static ma_result ma_resampling_backend_get_heap_size__linear(void* pUserData, const ma_resampler_config* pConfig, size_t* pHeapSizeInBytes)
{
ma_linear_resampler_config linearConfig;
(void)pUserData;
linearConfig = ma_resampling_backend_get_config__linear(pConfig);
return ma_linear_resampler_get_heap_size(&linearConfig, pHeapSizeInBytes);
}
static ma_result ma_resampling_backend_init__linear(void* pUserData, const ma_resampler_config* pConfig, void* pHeap, ma_resampling_backend** ppBackend)
{
ma_resampler* pResampler = (ma_resampler*)pUserData;
ma_result result;
ma_linear_resampler_config linearConfig;
(void)pUserData;
linearConfig = ma_resampling_backend_get_config__linear(pConfig);
result = ma_linear_resampler_init_preallocated(&linearConfig, pHeap, &pResampler->state.linear);
if (result != MA_SUCCESS) {
return result;
}
*ppBackend = &pResampler->state.linear;
return MA_SUCCESS;
}
static void ma_resampling_backend_uninit__linear(void* pUserData, ma_resampling_backend* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
{
(void)pUserData;
ma_linear_resampler_uninit((ma_linear_resampler*)pBackend, pAllocationCallbacks);
}
static ma_result ma_resampling_backend_process__linear(void* pUserData, ma_resampling_backend* pBackend, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
(void)pUserData;
return ma_linear_resampler_process_pcm_frames((ma_linear_resampler*)pBackend, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
}
static ma_result ma_resampling_backend_set_rate__linear(void* pUserData, ma_resampling_backend* pBackend, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
{
(void)pUserData;
return ma_linear_resampler_set_rate((ma_linear_resampler*)pBackend, sampleRateIn, sampleRateOut);
}
static ma_uint64 ma_resampling_backend_get_input_latency__linear(void* pUserData, const ma_resampling_backend* pBackend)
{
(void)pUserData;
return ma_linear_resampler_get_input_latency((const ma_linear_resampler*)pBackend);
}
static ma_uint64 ma_resampling_backend_get_output_latency__linear(void* pUserData, const ma_resampling_backend* pBackend)
{
(void)pUserData;
return ma_linear_resampler_get_output_latency((const ma_linear_resampler*)pBackend);
}
static ma_result ma_resampling_backend_get_required_input_frame_count__linear(void* pUserData, const ma_resampling_backend* pBackend, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount)
{
(void)pUserData;
return ma_linear_resampler_get_required_input_frame_count((const ma_linear_resampler*)pBackend, outputFrameCount, pInputFrameCount);
}
static ma_result ma_resampling_backend_get_expected_output_frame_count__linear(void* pUserData, const ma_resampling_backend* pBackend, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount)
{
(void)pUserData;
return ma_linear_resampler_get_expected_output_frame_count((const ma_linear_resampler*)pBackend, inputFrameCount, pOutputFrameCount);
}
static ma_result ma_resampling_backend_reset__linear(void* pUserData, ma_resampling_backend* pBackend)
{
(void)pUserData;
return ma_linear_resampler_reset((ma_linear_resampler*)pBackend);
}
static ma_resampling_backend_vtable g_ma_linear_resampler_vtable =
{
ma_resampling_backend_get_heap_size__linear,
ma_resampling_backend_init__linear,
ma_resampling_backend_uninit__linear,
ma_resampling_backend_process__linear,
ma_resampling_backend_set_rate__linear,
ma_resampling_backend_get_input_latency__linear,
ma_resampling_backend_get_output_latency__linear,
ma_resampling_backend_get_required_input_frame_count__linear,
ma_resampling_backend_get_expected_output_frame_count__linear,
ma_resampling_backend_reset__linear
};
MA_API ma_resampler_config ma_resampler_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut, ma_resample_algorithm algorithm)
{
ma_resampler_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRateIn = sampleRateIn;
config.sampleRateOut = sampleRateOut;
config.algorithm = algorithm;
/* Linear. */
config.linear.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER);
return config;
}
static ma_result ma_resampler_get_vtable(const ma_resampler_config* pConfig, ma_resampler* pResampler, ma_resampling_backend_vtable** ppVTable, void** ppUserData)
{
MA_ASSERT(pConfig != NULL);
MA_ASSERT(ppVTable != NULL);
MA_ASSERT(ppUserData != NULL);
/* Safety. */
*ppVTable = NULL;
*ppUserData = NULL;
switch (pConfig->algorithm)
{
case ma_resample_algorithm_linear:
{
*ppVTable = &g_ma_linear_resampler_vtable;
*ppUserData = pResampler;
} break;
case ma_resample_algorithm_custom:
{
*ppVTable = pConfig->pBackendVTable;
*ppUserData = pConfig->pBackendUserData;
} break;
default: return MA_INVALID_ARGS;
}
return MA_SUCCESS;
}
MA_API ma_result ma_resampler_get_heap_size(const ma_resampler_config* pConfig, size_t* pHeapSizeInBytes)
{
ma_result result;
ma_resampling_backend_vtable* pVTable;
void* pVTableUserData;
if (pHeapSizeInBytes == NULL) {
return MA_INVALID_ARGS;
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}
if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onInit == NULL) {
return MA_NOT_IMPLEMENTED; /* onInit not implemented. */
}
result = pResampler->pBackendVTable->onInit(pResampler->pBackendUserData, pConfig, pHeap, &pResampler->pBackend);
if (result != MA_SUCCESS) {
return result;
}
return MA_SUCCESS;
}
MA_API ma_result ma_resampler_init(const ma_resampler_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_resampler* pResampler)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_resampler_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_resampler_init_preallocated(pConfig, pHeap, pResampler);
if (result != MA_SUCCESS) {
return result;
}
pResampler->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
MA_API void ma_resampler_uninit(ma_resampler* pResampler, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pResampler == NULL) {
return;
}
if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onUninit == NULL) {
return;
}
pResampler->pBackendVTable->onUninit(pResampler->pBackendUserData, pResampler->pBackend, pAllocationCallbacks);
if (pResampler->_ownsHeap) {
ma_free(pResampler->_pHeap, pAllocationCallbacks);
}
}
MA_API ma_result ma_resampler_process_pcm_frames(ma_resampler* pResampler, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
if (pFrameCountOut == NULL && pFrameCountIn == NULL) {
return MA_INVALID_ARGS;
}
if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onProcess == NULL) {
return MA_NOT_IMPLEMENTED;
}
return pResampler->pBackendVTable->onProcess(pResampler->pBackendUserData, pResampler->pBackend, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
}
MA_API ma_result ma_resampler_set_rate(ma_resampler* pResampler, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
{
ma_result result;
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
if (sampleRateIn == 0 || sampleRateOut == 0) {
return MA_INVALID_ARGS;
}
if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onSetRate == NULL) {
return MA_NOT_IMPLEMENTED;
}
result = pResampler->pBackendVTable->onSetRate(pResampler->pBackendUserData, pResampler->pBackend, sampleRateIn, sampleRateOut);
if (result != MA_SUCCESS) {
return result;
}
pResampler->sampleRateIn = sampleRateIn;
pResampler->sampleRateOut = sampleRateOut;
return MA_SUCCESS;
}
MA_API ma_result ma_resampler_set_rate_ratio(ma_resampler* pResampler, float ratio)
{
ma_uint32 n;
ma_uint32 d;
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
if (ratio <= 0) {
return MA_INVALID_ARGS;
}
d = 1000;
n = (ma_uint32)(ratio * d);
if (n == 0) {
return MA_INVALID_ARGS; /* Ratio too small. */
}
MA_ASSERT(n != 0);
return ma_resampler_set_rate(pResampler, n, d);
}
MA_API ma_uint64 ma_resampler_get_input_latency(const ma_resampler* pResampler)
{
if (pResampler == NULL) {
return 0;
}
if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onGetInputLatency == NULL) {
return 0;
}
return pResampler->pBackendVTable->onGetInputLatency(pResampler->pBackendUserData, pResampler->pBackend);
}
MA_API ma_uint64 ma_resampler_get_output_latency(const ma_resampler* pResampler)
{
if (pResampler == NULL) {
return 0;
}
if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onGetOutputLatency == NULL) {
return 0;
}
return pResampler->pBackendVTable->onGetOutputLatency(pResampler->pBackendUserData, pResampler->pBackend);
}
MA_API ma_result ma_resampler_get_required_input_frame_count(const ma_resampler* pResampler, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount)
{
if (pInputFrameCount == NULL) {
return MA_INVALID_ARGS;
}
*pInputFrameCount = 0;
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onGetRequiredInputFrameCount == NULL) {
return MA_NOT_IMPLEMENTED;
}
return pResampler->pBackendVTable->onGetRequiredInputFrameCount(pResampler->pBackendUserData, pResampler->pBackend, outputFrameCount, pInputFrameCount);
}
MA_API ma_result ma_resampler_get_expected_output_frame_count(const ma_resampler* pResampler, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount)
{
if (pOutputFrameCount == NULL) {
return MA_INVALID_ARGS;
}
*pOutputFrameCount = 0;
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onGetExpectedOutputFrameCount == NULL) {
return MA_NOT_IMPLEMENTED;
}
return pResampler->pBackendVTable->onGetExpectedOutputFrameCount(pResampler->pBackendUserData, pResampler->pBackend, inputFrameCount, pOutputFrameCount);
}
MA_API ma_result ma_resampler_reset(ma_resampler* pResampler)
{
if (pResampler == NULL) {
return MA_INVALID_ARGS;
}
if (pResampler->pBackendVTable == NULL || pResampler->pBackendVTable->onReset == NULL) {
return MA_NOT_IMPLEMENTED;
}
return pResampler->pBackendVTable->onReset(pResampler->pBackendUserData, pResampler->pBackend);
}
/**************************************************************************************************************************************************************
Channel Conversion
**************************************************************************************************************************************************************/
#ifndef MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT
#define MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT 12
#endif
#define MA_PLANE_LEFT 0
#define MA_PLANE_RIGHT 1
#define MA_PLANE_FRONT 2
#define MA_PLANE_BACK 3
#define MA_PLANE_BOTTOM 4
#define MA_PLANE_TOP 5
static float g_maChannelPlaneRatios[MA_CHANNEL_POSITION_COUNT][6] = {
{ 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_NONE */
{ 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_MONO */
{ 0.5f, 0.0f, 0.5f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_LEFT */
{ 0.0f, 0.5f, 0.5f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_RIGHT */
{ 0.0f, 0.0f, 1.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_CENTER */
{ 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_LFE */
{ 0.5f, 0.0f, 0.0f, 0.5f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_LEFT */
{ 0.0f, 0.5f, 0.0f, 0.5f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_RIGHT */
{ 0.25f, 0.0f, 0.75f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_LEFT_CENTER */
{ 0.0f, 0.25f, 0.75f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_FRONT_RIGHT_CENTER */
{ 0.0f, 0.0f, 0.0f, 1.0f, 0.0f, 0.0f}, /* MA_CHANNEL_BACK_CENTER */
{ 1.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f}, /* MA_CHANNEL_SIDE_LEFT */
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
ma_channel channelOut;
/* Default to MA_CHANNEL_INDEX_NULL so that if a mapping is not found it'll be set appropriately. */
pShuffleTable[iChannelOut] = MA_CHANNEL_INDEX_NULL;
channelOut = ma_channel_map_get_channel(pChannelMapOut, channelCountOut, iChannelOut);
for (iChannelIn = 0; iChannelIn < channelCountIn; iChannelIn += 1) {
ma_channel channelIn;
channelIn = ma_channel_map_get_channel(pChannelMapIn, channelCountIn, iChannelIn);
if (channelOut == channelIn) {
pShuffleTable[iChannelOut] = (ma_uint8)iChannelIn;
break;
}
/*
Getting here means the channels don't exactly match, but we are going to support some
relaxed matching for practicality. If, for example, there are two stereo channel maps,
but one uses front left/right and the other uses side left/right, it makes logical
sense to just map these. The way we'll do it is we'll check if there is a logical
corresponding mapping, and if so, apply it, but we will *not* break from the loop,
thereby giving the loop a chance to find an exact match later which will take priority.
*/
switch (channelOut)
{
/* Left channels. */
case MA_CHANNEL_FRONT_LEFT:
case MA_CHANNEL_SIDE_LEFT:
{
switch (channelIn) {
case MA_CHANNEL_FRONT_LEFT:
case MA_CHANNEL_SIDE_LEFT:
{
pShuffleTable[iChannelOut] = (ma_uint8)iChannelIn;
} break;
}
} break;
/* Right channels. */
case MA_CHANNEL_FRONT_RIGHT:
case MA_CHANNEL_SIDE_RIGHT:
{
switch (channelIn) {
case MA_CHANNEL_FRONT_RIGHT:
case MA_CHANNEL_SIDE_RIGHT:
{
pShuffleTable[iChannelOut] = (ma_uint8)iChannelIn;
} break;
}
} break;
default: break;
}
}
}
return MA_SUCCESS;
}
static void ma_channel_map_apply_shuffle_table_u8(ma_uint8* pFramesOut, ma_uint32 channelsOut, const ma_uint8* pFramesIn, ma_uint32 channelsIn, ma_uint64 frameCount, const ma_uint8* pShuffleTable)
{
ma_uint64 iFrame;
ma_uint32 iChannelOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
ma_uint8 iChannelIn = pShuffleTable[iChannelOut];
if (iChannelIn < channelsIn) { /* For safety, and to deal with MA_CHANNEL_INDEX_NULL. */
pFramesOut[iChannelOut] = pFramesIn[iChannelIn];
} else {
pFramesOut[iChannelOut] = 0;
}
}
pFramesOut += channelsOut;
pFramesIn += channelsIn;
}
}
static void ma_channel_map_apply_shuffle_table_s16(ma_int16* pFramesOut, ma_uint32 channelsOut, const ma_int16* pFramesIn, ma_uint32 channelsIn, ma_uint64 frameCount, const ma_uint8* pShuffleTable)
{
ma_uint64 iFrame;
ma_uint32 iChannelOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
ma_uint8 iChannelIn = pShuffleTable[iChannelOut];
if (iChannelIn < channelsIn) { /* For safety, and to deal with MA_CHANNEL_INDEX_NULL. */
pFramesOut[iChannelOut] = pFramesIn[iChannelIn];
} else {
pFramesOut[iChannelOut] = 0;
}
}
pFramesOut += channelsOut;
pFramesIn += channelsIn;
}
}
static void ma_channel_map_apply_shuffle_table_s24(ma_uint8* pFramesOut, ma_uint32 channelsOut, const ma_uint8* pFramesIn, ma_uint32 channelsIn, ma_uint64 frameCount, const ma_uint8* pShuffleTable)
{
ma_uint64 iFrame;
ma_uint32 iChannelOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
ma_uint8 iChannelIn = pShuffleTable[iChannelOut];
if (iChannelIn < channelsIn) { /* For safety, and to deal with MA_CHANNEL_INDEX_NULL. */
pFramesOut[iChannelOut*3 + 0] = pFramesIn[iChannelIn*3 + 0];
pFramesOut[iChannelOut*3 + 1] = pFramesIn[iChannelIn*3 + 1];
pFramesOut[iChannelOut*3 + 2] = pFramesIn[iChannelIn*3 + 2];
} else {
pFramesOut[iChannelOut*3 + 0] = 0;
} pFramesOut[iChannelOut*3 + 1] = 0;
} pFramesOut[iChannelOut*3 + 2] = 0;
pFramesOut += channelsOut*3;
pFramesIn += channelsIn*3;
}
}
static void ma_channel_map_apply_shuffle_table_s32(ma_int32* pFramesOut, ma_uint32 channelsOut, const ma_int32* pFramesIn, ma_uint32 channelsIn, ma_uint64 frameCount, const ma_uint8* pShuffleTable)
{
ma_uint64 iFrame;
ma_uint32 iChannelOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
ma_uint8 iChannelIn = pShuffleTable[iChannelOut];
if (iChannelIn < channelsIn) { /* For safety, and to deal with MA_CHANNEL_INDEX_NULL. */
pFramesOut[iChannelOut] = pFramesIn[iChannelIn];
} else {
pFramesOut[iChannelOut] = 0;
}
}
pFramesOut += channelsOut;
pFramesIn += channelsIn;
}
}
static void ma_channel_map_apply_shuffle_table_f32(float* pFramesOut, ma_uint32 channelsOut, const float* pFramesIn, ma_uint32 channelsIn, ma_uint64 frameCount, const ma_uint8* pShuffleTable)
{
ma_uint64 iFrame;
ma_uint32 iChannelOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
ma_uint8 iChannelIn = pShuffleTable[iChannelOut];
if (iChannelIn < channelsIn) { /* For safety, and to deal with MA_CHANNEL_INDEX_NULL. */
pFramesOut[iChannelOut] = pFramesIn[iChannelIn];
} else {
pFramesOut[iChannelOut] = 0;
}
}
pFramesOut += channelsOut;
pFramesIn += channelsIn;
}
}
static ma_result ma_channel_map_apply_shuffle_table(void* pFramesOut, ma_uint32 channelsOut, const void* pFramesIn, ma_uint32 channelsIn, ma_uint64 frameCount, const ma_uint8* pShuffleTable, ma_format format)
{
if (pFramesOut == NULL || pFramesIn == NULL || channelsOut == 0 || pShuffleTable == NULL) {
return MA_INVALID_ARGS;
}
switch (format)
{
case ma_format_u8:
{
ma_channel_map_apply_shuffle_table_u8((ma_uint8*)pFramesOut, channelsOut, (const ma_uint8*)pFramesIn, channelsIn, frameCount, pShuffleTable);
} break;
case ma_format_s16:
{
ma_channel_map_apply_shuffle_table_s16((ma_int16*)pFramesOut, channelsOut, (const ma_int16*)pFramesIn, channelsIn, frameCount, pShuffleTable);
} break;
case ma_format_s24:
{
ma_channel_map_apply_shuffle_table_s24((ma_uint8*)pFramesOut, channelsOut, (const ma_uint8*)pFramesIn, channelsIn, frameCount, pShuffleTable);
} break;
case ma_format_s32:
{
ma_channel_map_apply_shuffle_table_s32((ma_int32*)pFramesOut, channelsOut, (const ma_int32*)pFramesIn, channelsIn, frameCount, pShuffleTable);
} break;
case ma_format_f32:
{
ma_channel_map_apply_shuffle_table_f32((float*)pFramesOut, channelsOut, (const float*)pFramesIn, channelsIn, frameCount, pShuffleTable);
} break;
default: return MA_INVALID_ARGS; /* Unknown format. */
}
return MA_SUCCESS;
}
static ma_result ma_channel_map_apply_mono_out_f32(float* pFramesOut, const float* pFramesIn, const ma_channel* pChannelMapIn, ma_uint32 channelsIn, ma_uint64 frameCount)
{
ma_uint64 iFrame;
ma_uint32 iChannelIn;
ma_uint32 accumulationCount;
if (pFramesOut == NULL || pFramesIn == NULL || channelsIn == 0) {
return MA_INVALID_ARGS;
}
/* In this case the output stream needs to be the average of all channels, ignoring NONE. */
/* A quick pre-processing step to get the accumulation counter since we're ignoring NONE channels. */
accumulationCount = 0;
for (iChannelIn = 0; iChannelIn < channelsIn; iChannelIn += 1) {
if (ma_channel_map_get_channel(pChannelMapIn, channelsIn, iChannelIn) != MA_CHANNEL_NONE) {
accumulationCount += 1;
}
}
if (accumulationCount > 0) { /* <-- Prevent a division by zero. */
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float accumulation = 0;
for (iChannelIn = 0; iChannelIn < channelsIn; iChannelIn += 1) {
ma_channel channelIn = ma_channel_map_get_channel(pChannelMapIn, channelsIn, iChannelIn);
if (channelIn != MA_CHANNEL_NONE) {
accumulation += pFramesIn[iChannelIn];
}
}
pFramesOut[0] = accumulation / accumulationCount;
pFramesOut += 1;
pFramesIn += channelsIn;
}
} else {
ma_silence_pcm_frames(pFramesOut, frameCount, ma_format_f32, 1);
}
return MA_SUCCESS;
}
static ma_result ma_channel_map_apply_mono_in_f32(float* pFramesOut, const ma_channel* pChannelMapOut, ma_uint32 channelsOut, const float* pFramesIn, ma_uint64 frameCount, ma_mono_expansion_mode monoExpansionMode)
{
ma_uint64 iFrame;
ma_uint32 iChannelOut;
if (pFramesOut == NULL || channelsOut == 0 || pFramesIn == NULL) {
return MA_INVALID_ARGS;
}
/* Note that the MA_CHANNEL_NONE channel must be ignored in all cases. */
switch (monoExpansionMode)
{
case ma_mono_expansion_mode_average:
{
float weight;
ma_uint32 validChannelCount = 0;
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
if (channelOut != MA_CHANNEL_NONE) {
validChannelCount += 1;
}
}
weight = 1.0f / validChannelCount;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
if (channelOut != MA_CHANNEL_NONE) {
pFramesOut[iChannelOut] = pFramesIn[0] * weight;
}
}
pFramesOut += channelsOut;
pFramesIn += 1;
}
} break;
case ma_mono_expansion_mode_stereo_only:
{
if (channelsOut >= 2) {
ma_uint32 iChannelLeft = (ma_uint32)-1;
ma_uint32 iChannelRight = (ma_uint32)-1;
/*
We first need to find our stereo channels. We prefer front-left and front-right, but
if they're not available, we'll also try side-left and side-right. If neither are
available we'll fall through to the default case below.
*/
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
if (channelOut == MA_CHANNEL_SIDE_LEFT) {
iChannelLeft = iChannelOut;
}
if (channelOut == MA_CHANNEL_SIDE_RIGHT) {
iChannelRight = iChannelOut;
}
}
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
if (channelOut == MA_CHANNEL_FRONT_LEFT) {
iChannelLeft = iChannelOut;
}
if (channelOut == MA_CHANNEL_FRONT_RIGHT) {
iChannelRight = iChannelOut;
}
}
if (iChannelLeft != (ma_uint32)-1 && iChannelRight != (ma_uint32)-1) {
/* We found our stereo channels so we can duplicate the signal across those channels. */
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
if (channelOut != MA_CHANNEL_NONE) {
if (iChannelOut == iChannelLeft || iChannelOut == iChannelRight) {
pFramesOut[iChannelOut] = pFramesIn[0];
} else {
pFramesOut[iChannelOut] = 0.0f;
}
}
}
pFramesOut += channelsOut;
pFramesIn += 1;
}
break; /* Get out of the switch. */
} else {
/* Fallthrough. Does not have left and right channels. */
goto default_handler;
}
} else {
/* Fallthrough. Does not have stereo channels. */
goto default_handler;
}
}; /* Fallthrough. See comments above. */
case ma_mono_expansion_mode_duplicate:
default:
{
default_handler:
{
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
if (channelOut != MA_CHANNEL_NONE) {
pFramesOut[iChannelOut] = pFramesIn[0];
}
}
pFramesOut += channelsOut;
pFramesIn += 1;
}
}
} break;
}
return MA_SUCCESS;
}
static void ma_channel_map_apply_f32(float* pFramesOut, const ma_channel* pChannelMapOut, ma_uint32 channelsOut, const float* pFramesIn, const ma_channel* pChannelMapIn, ma_uint32 channelsIn, ma_uint64 frameCount, ma_channel_mix_mode mode, ma_mono_ex...
{
ma_channel_conversion_path conversionPath = ma_channel_map_get_conversion_path(pChannelMapIn, channelsIn, pChannelMapOut, channelsOut, mode);
/* Optimized Path: Passthrough */
if (conversionPath == ma_channel_conversion_path_passthrough) {
ma_copy_pcm_frames(pFramesOut, pFramesIn, frameCount, ma_format_f32, channelsOut);
return;
}
/* Special Path: Mono Output. */
if (conversionPath == ma_channel_conversion_path_mono_out) {
ma_channel_map_apply_mono_out_f32(pFramesOut, pFramesIn, pChannelMapIn, channelsIn, frameCount);
return;
}
/* Special Path: Mono Input. */
if (conversionPath == ma_channel_conversion_path_mono_in) {
ma_channel_map_apply_mono_in_f32(pFramesOut, pChannelMapOut, channelsOut, pFramesIn, frameCount, monoExpansionMode);
return;
}
/* Getting here means we aren't running on an optimized conversion path. */
if (channelsOut <= MA_MAX_CHANNELS) {
ma_result result;
if (mode == ma_channel_mix_mode_simple) {
ma_channel shuffleTable[MA_MAX_CHANNELS];
result = ma_channel_map_build_shuffle_table(pChannelMapIn, channelsIn, pChannelMapOut, channelsOut, shuffleTable);
if (result != MA_SUCCESS) {
return;
}
result = ma_channel_map_apply_shuffle_table(pFramesOut, channelsOut, pFramesIn, channelsIn, frameCount, shuffleTable, ma_format_f32);
if (result != MA_SUCCESS) {
return;
}
} else {
ma_uint32 iFrame;
ma_uint32 iChannelOut;
ma_uint32 iChannelIn;
float weights[32][32]; /* Do not use MA_MAX_CHANNELS here! */
/*
If we have a small enough number of channels, pre-compute the weights. Otherwise we'll just need to
fall back to a slower path because otherwise we'll run out of stack space.
*/
if (channelsIn <= ma_countof(weights) && channelsOut <= ma_countof(weights)) {
/* Pre-compute weights. */
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
for (iChannelIn = 0; iChannelIn < channelsIn; iChannelIn += 1) {
ma_channel channelIn = ma_channel_map_get_channel(pChannelMapIn, channelsIn, iChannelIn);
weights[iChannelOut][iChannelIn] = ma_calculate_channel_position_rectangular_weight(channelOut, channelIn);
}
}
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
float accumulation = 0;
for (iChannelIn = 0; iChannelIn < channelsIn; iChannelIn += 1) {
accumulation += pFramesIn[iChannelIn] * weights[iChannelOut][iChannelIn];
}
pFramesOut[iChannelOut] = accumulation;
}
pFramesOut += channelsOut;
pFramesIn += channelsIn;
}
} else {
/* Cannot pre-compute weights because not enough room in stack-allocated buffer. */
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelOut = 0; iChannelOut < channelsOut; iChannelOut += 1) {
float accumulation = 0;
ma_channel channelOut = ma_channel_map_get_channel(pChannelMapOut, channelsOut, iChannelOut);
for (iChannelIn = 0; iChannelIn < channelsIn; iChannelIn += 1) {
ma_channel channelIn = ma_channel_map_get_channel(pChannelMapIn, channelsIn, iChannelIn);
accumulation += pFramesIn[iChannelIn] * ma_calculate_channel_position_rectangular_weight(channelOut, channelIn);
}
pFramesOut[iChannelOut] = accumulation;
}
pFramesOut += channelsOut;
pFramesIn += channelsIn;
}
}
}
} else {
/* Fall back to silence. If you hit this, what are you doing with so many channels?! */
ma_silence_pcm_frames(pFramesOut, frameCount, ma_format_f32, channelsOut);
}
}
typedef struct
{
size_t sizeInBytes;
size_t channelMapInOffset;
size_t channelMapOutOffset;
size_t shuffleTableOffset;
size_t weightsOffset;
} ma_channel_converter_heap_layout;
static ma_channel_conversion_path ma_channel_converter_config_get_conversion_path(const ma_channel_converter_config* pConfig)
{
return ma_channel_map_get_conversion_path(pConfig->pChannelMapIn, pConfig->channelsIn, pConfig->pChannelMapOut, pConfig->channelsOut, pConfig->mixingMode);
}
static ma_result ma_channel_converter_get_heap_layout(const ma_channel_converter_config* pConfig, ma_channel_converter_heap_layout* pHeapLayout)
{
ma_channel_conversion_path conversionPath;
MA_ASSERT(pHeapLayout != NULL);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->channelsIn == 0 || pConfig->channelsOut == 0) {
return MA_INVALID_ARGS;
}
if (!ma_channel_map_is_valid(pConfig->pChannelMapIn, pConfig->channelsIn)) {
return MA_INVALID_ARGS;
}
if (!ma_channel_map_is_valid(pConfig->pChannelMapOut, pConfig->channelsOut)) {
return MA_INVALID_ARGS;
}
pHeapLayout->sizeInBytes = 0;
/* Input channel map. Only need to allocate this if we have an input channel map (otherwise default channel map is assumed). */
pHeapLayout->channelMapInOffset = pHeapLayout->sizeInBytes;
if (pConfig->pChannelMapIn != NULL) {
pHeapLayout->sizeInBytes += sizeof(ma_channel) * pConfig->channelsIn;
}
/* Output channel map. Only need to allocate this if we have an output channel map (otherwise default channel map is assumed). */
pHeapLayout->channelMapOutOffset = pHeapLayout->sizeInBytes;
if (pConfig->pChannelMapOut != NULL) {
pHeapLayout->sizeInBytes += sizeof(ma_channel) * pConfig->channelsOut;
}
/* Alignment for the next section. */
pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
/* Whether or not we use weights of a shuffle table depends on the channel map themselves and the algorithm we've chosen. */
conversionPath = ma_channel_converter_config_get_conversion_path(pConfig);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
pConverter->weights.f32[iChannelIn][iChannelOut] = weight;
}
} else {
if (pConverter->weights.s16[iChannelIn][iChannelOut] == 0) {
pConverter->weights.s16[iChannelIn][iChannelOut] = ma_channel_converter_float_to_fixed(weight);
}
}
}
}
}
}
}
} break;
}
}
return MA_SUCCESS;
}
MA_API ma_result ma_channel_converter_init(const ma_channel_converter_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_channel_converter* pConverter)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_channel_converter_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_channel_converter_init_preallocated(pConfig, pHeap, pConverter);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pConverter->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
MA_API void ma_channel_converter_uninit(ma_channel_converter* pConverter, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pConverter == NULL) {
return;
}
if (pConverter->_ownsHeap) {
ma_free(pConverter->_pHeap, pAllocationCallbacks);
}
}
static ma_result ma_channel_converter_process_pcm_frames__passthrough(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
MA_ASSERT(pConverter != NULL);
MA_ASSERT(pFramesOut != NULL);
MA_ASSERT(pFramesIn != NULL);
ma_copy_memory_64(pFramesOut, pFramesIn, frameCount * ma_get_bytes_per_frame(pConverter->format, pConverter->channelsOut));
return MA_SUCCESS;
}
static ma_result ma_channel_converter_process_pcm_frames__shuffle(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
MA_ASSERT(pConverter != NULL);
MA_ASSERT(pFramesOut != NULL);
MA_ASSERT(pFramesIn != NULL);
MA_ASSERT(pConverter->channelsIn == pConverter->channelsOut);
return ma_channel_map_apply_shuffle_table(pFramesOut, pConverter->channelsOut, pFramesIn, pConverter->channelsIn, frameCount, pConverter->pShuffleTable, pConverter->format);
}
static ma_result ma_channel_converter_process_pcm_frames__mono_in(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
ma_uint64 iFrame;
MA_ASSERT(pConverter != NULL);
MA_ASSERT(pFramesOut != NULL);
MA_ASSERT(pFramesIn != NULL);
MA_ASSERT(pConverter->channelsIn == 1);
switch (pConverter->format)
{
case ma_format_u8:
{
/* */ ma_uint8* pFramesOutU8 = ( ma_uint8*)pFramesOut;
const ma_uint8* pFramesInU8 = (const ma_uint8*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
pFramesOutU8[iFrame*pConverter->channelsOut + iChannel] = pFramesInU8[iFrame];
}
}
} break;
case ma_format_s16:
{
/* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
if (pConverter->channelsOut == 2) {
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
pFramesOutS16[iFrame*2 + 0] = pFramesInS16[iFrame];
pFramesOutS16[iFrame*2 + 1] = pFramesInS16[iFrame];
}
} else {
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
pFramesOutS16[iFrame*pConverter->channelsOut + iChannel] = pFramesInS16[iFrame];
}
}
}
} break;
case ma_format_s24:
{
/* */ ma_uint8* pFramesOutS24 = ( ma_uint8*)pFramesOut;
const ma_uint8* pFramesInS24 = (const ma_uint8*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
ma_uint64 iSampleOut = iFrame*pConverter->channelsOut + iChannel;
ma_uint64 iSampleIn = iFrame;
pFramesOutS24[iSampleOut*3 + 0] = pFramesInS24[iSampleIn*3 + 0];
pFramesOutS24[iSampleOut*3 + 1] = pFramesInS24[iSampleIn*3 + 1];
pFramesOutS24[iSampleOut*3 + 2] = pFramesInS24[iSampleIn*3 + 2];
}
}
} break;
case ma_format_s32:
{
/* */ ma_int32* pFramesOutS32 = ( ma_int32*)pFramesOut;
const ma_int32* pFramesInS32 = (const ma_int32*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
pFramesOutS32[iFrame*pConverter->channelsOut + iChannel] = pFramesInS32[iFrame];
}
}
} break;
case ma_format_f32:
{
/* */ float* pFramesOutF32 = ( float*)pFramesOut;
const float* pFramesInF32 = (const float*)pFramesIn;
if (pConverter->channelsOut == 2) {
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
pFramesOutF32[iFrame*2 + 0] = pFramesInF32[iFrame];
pFramesOutF32[iFrame*2 + 1] = pFramesInF32[iFrame];
}
} else {
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < pConverter->channelsOut; iChannel += 1) {
pFramesOutF32[iFrame*pConverter->channelsOut + iChannel] = pFramesInF32[iFrame];
}
}
}
} break;
default: return MA_INVALID_OPERATION; /* Unknown format. */
}
return MA_SUCCESS;
}
static ma_result ma_channel_converter_process_pcm_frames__mono_out(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
ma_uint64 iFrame;
ma_uint32 iChannel;
MA_ASSERT(pConverter != NULL);
MA_ASSERT(pFramesOut != NULL);
MA_ASSERT(pFramesIn != NULL);
MA_ASSERT(pConverter->channelsOut == 1);
switch (pConverter->format)
{
case ma_format_u8:
{
/* */ ma_uint8* pFramesOutU8 = ( ma_uint8*)pFramesOut;
const ma_uint8* pFramesInU8 = (const ma_uint8*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
ma_int32 t = 0;
for (iChannel = 0; iChannel < pConverter->channelsIn; iChannel += 1) {
t += ma_pcm_sample_u8_to_s16_no_scale(pFramesInU8[iFrame*pConverter->channelsIn + iChannel]);
}
pFramesOutU8[iFrame] = ma_clip_u8(t / pConverter->channelsOut);
}
} break;
case ma_format_s16:
{
/* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
ma_int32 t = 0;
for (iChannel = 0; iChannel < pConverter->channelsIn; iChannel += 1) {
t += pFramesInS16[iFrame*pConverter->channelsIn + iChannel];
}
pFramesOutS16[iFrame] = (ma_int16)(t / pConverter->channelsIn);
}
} break;
case ma_format_s24:
{
/* */ ma_uint8* pFramesOutS24 = ( ma_uint8*)pFramesOut;
const ma_uint8* pFramesInS24 = (const ma_uint8*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
ma_int64 t = 0;
for (iChannel = 0; iChannel < pConverter->channelsIn; iChannel += 1) {
t += ma_pcm_sample_s24_to_s32_no_scale(&pFramesInS24[(iFrame*pConverter->channelsIn + iChannel)*3]);
}
ma_pcm_sample_s32_to_s24_no_scale(t / pConverter->channelsIn, &pFramesOutS24[iFrame*3]);
}
} break;
case ma_format_s32:
{
/* */ ma_int32* pFramesOutS32 = ( ma_int32*)pFramesOut;
const ma_int32* pFramesInS32 = (const ma_int32*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
ma_int64 t = 0;
for (iChannel = 0; iChannel < pConverter->channelsIn; iChannel += 1) {
t += pFramesInS32[iFrame*pConverter->channelsIn + iChannel];
}
pFramesOutS32[iFrame] = (ma_int32)(t / pConverter->channelsIn);
}
} break;
case ma_format_f32:
{
/* */ float* pFramesOutF32 = ( float*)pFramesOut;
const float* pFramesInF32 = (const float*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; ++iFrame) {
float t = 0;
for (iChannel = 0; iChannel < pConverter->channelsIn; iChannel += 1) {
t += pFramesInF32[iFrame*pConverter->channelsIn + iChannel];
}
pFramesOutF32[iFrame] = t / pConverter->channelsIn;
}
} break;
default: return MA_INVALID_OPERATION; /* Unknown format. */
}
return MA_SUCCESS;
}
static ma_result ma_channel_converter_process_pcm_frames__weights(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
ma_uint32 iFrame;
ma_uint32 iChannelIn;
ma_uint32 iChannelOut;
MA_ASSERT(pConverter != NULL);
MA_ASSERT(pFramesOut != NULL);
MA_ASSERT(pFramesIn != NULL);
/* This is the more complicated case. Each of the output channels is accumulated with 0 or more input channels. */
/* Clear. */
ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->format, pConverter->channelsOut));
/* Accumulate. */
switch (pConverter->format)
{
case ma_format_u8:
{
/* */ ma_uint8* pFramesOutU8 = ( ma_uint8*)pFramesOut;
const ma_uint8* pFramesInU8 = (const ma_uint8*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
ma_int16 u8_O = ma_pcm_sample_u8_to_s16_no_scale(pFramesOutU8[iFrame*pConverter->channelsOut + iChannelOut]);
ma_int16 u8_I = ma_pcm_sample_u8_to_s16_no_scale(pFramesInU8 [iFrame*pConverter->channelsIn + iChannelIn ]);
ma_int32 s = (ma_int32)ma_clamp(u8_O + ((u8_I * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT), -128, 127);
pFramesOutU8[iFrame*pConverter->channelsOut + iChannelOut] = ma_clip_u8((ma_int16)s);
}
}
}
} break;
case ma_format_s16:
{
/* */ ma_int16* pFramesOutS16 = ( ma_int16*)pFramesOut;
const ma_int16* pFramesInS16 = (const ma_int16*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
ma_int32 s = pFramesOutS16[iFrame*pConverter->channelsOut + iChannelOut];
s += (pFramesInS16[iFrame*pConverter->channelsIn + iChannelIn] * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT;
pFramesOutS16[iFrame*pConverter->channelsOut + iChannelOut] = (ma_int16)ma_clamp(s, -32768, 32767);
}
}
}
} break;
case ma_format_s24:
{
/* */ ma_uint8* pFramesOutS24 = ( ma_uint8*)pFramesOut;
const ma_uint8* pFramesInS24 = (const ma_uint8*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
ma_int64 s24_O = ma_pcm_sample_s24_to_s32_no_scale(&pFramesOutS24[(iFrame*pConverter->channelsOut + iChannelOut)*3]);
ma_int64 s24_I = ma_pcm_sample_s24_to_s32_no_scale(&pFramesInS24 [(iFrame*pConverter->channelsIn + iChannelIn )*3]);
ma_int64 s24 = (ma_int32)ma_clamp(s24_O + ((s24_I * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT), -8388608, 8388607);
ma_pcm_sample_s32_to_s24_no_scale(s24, &pFramesOutS24[(iFrame*pConverter->channelsOut + iChannelOut)*3]);
}
}
}
} break;
case ma_format_s32:
{
/* */ ma_int32* pFramesOutS32 = ( ma_int32*)pFramesOut;
const ma_int32* pFramesInS32 = (const ma_int32*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
ma_int64 s = pFramesOutS32[iFrame*pConverter->channelsOut + iChannelOut];
s += ((ma_int64)pFramesInS32[iFrame*pConverter->channelsIn + iChannelIn] * pConverter->weights.s16[iChannelIn][iChannelOut]) >> MA_CHANNEL_CONVERTER_FIXED_POINT_SHIFT;
pFramesOutS32[iFrame*pConverter->channelsOut + iChannelOut] = ma_clip_s32(s);
}
}
}
} break;
case ma_format_f32:
{
/* */ float* pFramesOutF32 = ( float*)pFramesOut;
const float* pFramesInF32 = (const float*)pFramesIn;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannelIn = 0; iChannelIn < pConverter->channelsIn; ++iChannelIn) {
for (iChannelOut = 0; iChannelOut < pConverter->channelsOut; ++iChannelOut) {
pFramesOutF32[iFrame*pConverter->channelsOut + iChannelOut] += pFramesInF32[iFrame*pConverter->channelsIn + iChannelIn] * pConverter->weights.f32[iChannelIn][iChannelOut];
}
}
}
} break;
default: return MA_INVALID_OPERATION; /* Unknown format. */
}
return MA_SUCCESS;
}
MA_API ma_result ma_channel_converter_process_pcm_frames(ma_channel_converter* pConverter, void* pFramesOut, const void* pFramesIn, ma_uint64 frameCount)
{
if (pConverter == NULL) {
return MA_INVALID_ARGS;
}
if (pFramesOut == NULL) {
return MA_INVALID_ARGS;
}
if (pFramesIn == NULL) {
ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->format, pConverter->channelsOut));
return MA_SUCCESS;
}
switch (pConverter->conversionPath)
{
case ma_channel_conversion_path_passthrough: return ma_channel_converter_process_pcm_frames__passthrough(pConverter, pFramesOut, pFramesIn, frameCount);
case ma_channel_conversion_path_mono_out: return ma_channel_converter_process_pcm_frames__mono_out(pConverter, pFramesOut, pFramesIn, frameCount);
case ma_channel_conversion_path_mono_in: return ma_channel_converter_process_pcm_frames__mono_in(pConverter, pFramesOut, pFramesIn, frameCount);
case ma_channel_conversion_path_shuffle: return ma_channel_converter_process_pcm_frames__shuffle(pConverter, pFramesOut, pFramesIn, frameCount);
case ma_channel_conversion_path_weights:
default:
{
return ma_channel_converter_process_pcm_frames__weights(pConverter, pFramesOut, pFramesIn, frameCount);
}
}
}
MA_API ma_result ma_channel_converter_get_input_channel_map(const ma_channel_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap)
{
if (pConverter == NULL || pChannelMap == NULL) {
return MA_INVALID_ARGS;
}
ma_channel_map_copy_or_default(pChannelMap, channelMapCap, pConverter->pChannelMapIn, pConverter->channelsIn);
return MA_SUCCESS;
}
MA_API ma_result ma_channel_converter_get_output_channel_map(const ma_channel_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap)
{
if (pConverter == NULL || pChannelMap == NULL) {
return MA_INVALID_ARGS;
}
ma_channel_map_copy_or_default(pChannelMap, channelMapCap, pConverter->pChannelMapOut, pConverter->channelsOut);
return MA_SUCCESS;
}
/**************************************************************************************************************************************************************
Data Conversion
**************************************************************************************************************************************************************/
MA_API ma_data_converter_config ma_data_converter_config_init_default()
{
ma_data_converter_config config;
MA_ZERO_OBJECT(&config);
config.ditherMode = ma_dither_mode_none;
config.resampling.algorithm = ma_resample_algorithm_linear;
config.allowDynamicSampleRate = MA_FALSE; /* Disable dynamic sample rates by default because dynamic rate adjustments should be quite rare and it allows an optimization for cases when the in and out sample rates are the same. */
/* Linear resampling defaults. */
config.resampling.linear.lpfOrder = 1;
return config;
}
MA_API ma_data_converter_config ma_data_converter_config_init(ma_format formatIn, ma_format formatOut, ma_uint32 channelsIn, ma_uint32 channelsOut, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
{
ma_data_converter_config config = ma_data_converter_config_init_default();
config.formatIn = formatIn;
config.formatOut = formatOut;
config.channelsIn = channelsIn;
config.channelsOut = channelsOut;
config.sampleRateIn = sampleRateIn;
config.sampleRateOut = sampleRateOut;
return config;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
/* Channel routing not required. */
if (pConverter->hasResampler) {
pConverter->executionPath = ma_data_converter_execution_path_resample_only;
} else {
pConverter->executionPath = ma_data_converter_execution_path_format_only;
}
}
}
}
return MA_SUCCESS;
}
MA_API ma_result ma_data_converter_init(const ma_data_converter_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_converter* pConverter)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_data_converter_get_heap_size(pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_data_converter_init_preallocated(pConfig, pHeap, pConverter);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
pConverter->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
MA_API void ma_data_converter_uninit(ma_data_converter* pConverter, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pConverter == NULL) {
return;
}
if (pConverter->hasResampler) {
ma_resampler_uninit(&pConverter->resampler, pAllocationCallbacks);
}
ma_channel_converter_uninit(&pConverter->channelConverter, pAllocationCallbacks);
if (pConverter->_ownsHeap) {
ma_free(pConverter->_pHeap, pAllocationCallbacks);
}
}
static ma_result ma_data_converter_process_pcm_frames__passthrough(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
ma_uint64 frameCountIn;
ma_uint64 frameCountOut;
ma_uint64 frameCount;
MA_ASSERT(pConverter != NULL);
frameCountIn = 0;
if (pFrameCountIn != NULL) {
frameCountIn = *pFrameCountIn;
}
frameCountOut = 0;
if (pFrameCountOut != NULL) {
frameCountOut = *pFrameCountOut;
}
frameCount = ma_min(frameCountIn, frameCountOut);
if (pFramesOut != NULL) {
if (pFramesIn != NULL) {
ma_copy_memory_64(pFramesOut, pFramesIn, frameCount * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
} else {
ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
}
}
if (pFrameCountIn != NULL) {
*pFrameCountIn = frameCount;
}
if (pFrameCountOut != NULL) {
*pFrameCountOut = frameCount;
}
return MA_SUCCESS;
}
static ma_result ma_data_converter_process_pcm_frames__format_only(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
ma_uint64 frameCountIn;
ma_uint64 frameCountOut;
ma_uint64 frameCount;
MA_ASSERT(pConverter != NULL);
frameCountIn = 0;
if (pFrameCountIn != NULL) {
frameCountIn = *pFrameCountIn;
}
frameCountOut = 0;
if (pFrameCountOut != NULL) {
frameCountOut = *pFrameCountOut;
}
frameCount = ma_min(frameCountIn, frameCountOut);
if (pFramesOut != NULL) {
if (pFramesIn != NULL) {
ma_convert_pcm_frames_format(pFramesOut, pConverter->formatOut, pFramesIn, pConverter->formatIn, frameCount, pConverter->channelsIn, pConverter->ditherMode);
} else {
ma_zero_memory_64(pFramesOut, frameCount * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
}
}
if (pFrameCountIn != NULL) {
*pFrameCountIn = frameCount;
}
if (pFrameCountOut != NULL) {
*pFrameCountOut = frameCount;
}
return MA_SUCCESS;
}
static ma_result ma_data_converter_process_pcm_frames__resample_with_format_conversion(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
ma_result result = MA_SUCCESS;
ma_uint64 frameCountIn;
ma_uint64 frameCountOut;
ma_uint64 framesProcessedIn;
ma_uint64 framesProcessedOut;
MA_ASSERT(pConverter != NULL);
frameCountIn = 0;
if (pFrameCountIn != NULL) {
frameCountIn = *pFrameCountIn;
}
frameCountOut = 0;
if (pFrameCountOut != NULL) {
frameCountOut = *pFrameCountOut;
}
framesProcessedIn = 0;
framesProcessedOut = 0;
while (framesProcessedOut < frameCountOut) {
ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
const ma_uint32 tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->resampler.format, pConverter->resampler.channels);
const void* pFramesInThisIteration;
/* */ void* pFramesOutThisIteration;
ma_uint64 frameCountInThisIteration;
ma_uint64 frameCountOutThisIteration;
if (pFramesIn != NULL) {
pFramesInThisIteration = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pConverter->formatIn, pConverter->channelsIn));
} else {
pFramesInThisIteration = NULL;
}
if (pFramesOut != NULL) {
pFramesOutThisIteration = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
} else {
pFramesOutThisIteration = NULL;
}
/* Do a pre format conversion if necessary. */
if (pConverter->hasPreFormatConversion) {
ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
const ma_uint32 tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->resampler.format, pConverter->resampler.channels);
frameCountInThisIteration = (frameCountIn - framesProcessedIn);
if (frameCountInThisIteration > tempBufferInCap) {
frameCountInThisIteration = tempBufferInCap;
}
if (pConverter->hasPostFormatConversion) {
if (frameCountInThisIteration > tempBufferOutCap) {
frameCountInThisIteration = tempBufferOutCap;
}
}
if (pFramesInThisIteration != NULL) {
ma_convert_pcm_frames_format(pTempBufferIn, pConverter->resampler.format, pFramesInThisIteration, pConverter->formatIn, frameCountInThisIteration, pConverter->channelsIn, pConverter->ditherMode);
} else {
MA_ZERO_MEMORY(pTempBufferIn, sizeof(pTempBufferIn));
}
frameCountOutThisIteration = (frameCountOut - framesProcessedOut);
if (pConverter->hasPostFormatConversion) {
/* Both input and output conversion required. Output to the temp buffer. */
if (frameCountOutThisIteration > tempBufferOutCap) {
frameCountOutThisIteration = tempBufferOutCap;
}
result = ma_resampler_process_pcm_frames(&pConverter->resampler, pTempBufferIn, &frameCountInThisIteration, pTempBufferOut, &frameCountOutThisIteration);
} else {
/* Only pre-format required. Output straight to the output buffer. */
result = ma_resampler_process_pcm_frames(&pConverter->resampler, pTempBufferIn, &frameCountInThisIteration, pFramesOutThisIteration, &frameCountOutThisIteration);
}
if (result != MA_SUCCESS) {
break;
}
} else {
/* No pre-format required. Just read straight from the input buffer. */
MA_ASSERT(pConverter->hasPostFormatConversion == MA_TRUE);
frameCountInThisIteration = (frameCountIn - framesProcessedIn);
frameCountOutThisIteration = (frameCountOut - framesProcessedOut);
if (frameCountOutThisIteration > tempBufferOutCap) {
frameCountOutThisIteration = tempBufferOutCap;
}
result = ma_resampler_process_pcm_frames(&pConverter->resampler, pFramesInThisIteration, &frameCountInThisIteration, pTempBufferOut, &frameCountOutThisIteration);
if (result != MA_SUCCESS) {
break;
}
}
/* If we are doing a post format conversion we need to do that now. */
if (pConverter->hasPostFormatConversion) {
if (pFramesOutThisIteration != NULL) {
ma_convert_pcm_frames_format(pFramesOutThisIteration, pConverter->formatOut, pTempBufferOut, pConverter->resampler.format, frameCountOutThisIteration, pConverter->resampler.channels, pConverter->ditherMode);
}
}
framesProcessedIn += frameCountInThisIteration;
framesProcessedOut += frameCountOutThisIteration;
MA_ASSERT(framesProcessedIn <= frameCountIn);
MA_ASSERT(framesProcessedOut <= frameCountOut);
if (frameCountOutThisIteration == 0) {
break; /* Consumed all of our input data. */
}
}
if (pFrameCountIn != NULL) {
*pFrameCountIn = framesProcessedIn;
}
if (pFrameCountOut != NULL) {
*pFrameCountOut = framesProcessedOut;
}
return result;
}
static ma_result ma_data_converter_process_pcm_frames__resample_only(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
MA_ASSERT(pConverter != NULL);
if (pConverter->hasPreFormatConversion == MA_FALSE && pConverter->hasPostFormatConversion == MA_FALSE) {
/* Neither pre- nor post-format required. This is simple case where only resampling is required. */
return ma_resampler_process_pcm_frames(&pConverter->resampler, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
} else {
/* Format conversion required. */
return ma_data_converter_process_pcm_frames__resample_with_format_conversion(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
}
}
static ma_result ma_data_converter_process_pcm_frames__channels_only(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
ma_result result;
ma_uint64 frameCountIn;
ma_uint64 frameCountOut;
ma_uint64 frameCount;
MA_ASSERT(pConverter != NULL);
frameCountIn = 0;
if (pFrameCountIn != NULL) {
frameCountIn = *pFrameCountIn;
}
frameCountOut = 0;
if (pFrameCountOut != NULL) {
frameCountOut = *pFrameCountOut;
}
frameCount = ma_min(frameCountIn, frameCountOut);
if (pConverter->hasPreFormatConversion == MA_FALSE && pConverter->hasPostFormatConversion == MA_FALSE) {
/* No format conversion required. */
result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pFramesOut, pFramesIn, frameCount);
if (result != MA_SUCCESS) {
return result;
}
} else {
/* Format conversion required. */
ma_uint64 framesProcessed = 0;
while (framesProcessed < frameCount) {
ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
const ma_uint32 tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsOut);
const void* pFramesInThisIteration;
/* */ void* pFramesOutThisIteration;
ma_uint64 frameCountThisIteration;
if (pFramesIn != NULL) {
pFramesInThisIteration = ma_offset_ptr(pFramesIn, framesProcessed * ma_get_bytes_per_frame(pConverter->formatIn, pConverter->channelsIn));
} else {
pFramesInThisIteration = NULL;
}
if (pFramesOut != NULL) {
pFramesOutThisIteration = ma_offset_ptr(pFramesOut, framesProcessed * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
} else {
pFramesOutThisIteration = NULL;
}
/* Do a pre format conversion if necessary. */
if (pConverter->hasPreFormatConversion) {
ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
const ma_uint32 tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsIn);
frameCountThisIteration = (frameCount - framesProcessed);
if (frameCountThisIteration > tempBufferInCap) {
frameCountThisIteration = tempBufferInCap;
}
if (pConverter->hasPostFormatConversion) {
if (frameCountThisIteration > tempBufferOutCap) {
frameCountThisIteration = tempBufferOutCap;
}
}
if (pFramesInThisIteration != NULL) {
ma_convert_pcm_frames_format(pTempBufferIn, pConverter->channelConverter.format, pFramesInThisIteration, pConverter->formatIn, frameCountThisIteration, pConverter->channelsIn, pConverter->ditherMode);
} else {
MA_ZERO_MEMORY(pTempBufferIn, sizeof(pTempBufferIn));
}
if (pConverter->hasPostFormatConversion) {
/* Both input and output conversion required. Output to the temp buffer. */
result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pTempBufferOut, pTempBufferIn, frameCountThisIteration);
} else {
/* Only pre-format required. Output straight to the output buffer. */
result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pFramesOutThisIteration, pTempBufferIn, frameCountThisIteration);
}
if (result != MA_SUCCESS) {
break;
}
} else {
/* No pre-format required. Just read straight from the input buffer. */
MA_ASSERT(pConverter->hasPostFormatConversion == MA_TRUE);
frameCountThisIteration = (frameCount - framesProcessed);
if (frameCountThisIteration > tempBufferOutCap) {
frameCountThisIteration = tempBufferOutCap;
}
result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pTempBufferOut, pFramesInThisIteration, frameCountThisIteration);
if (result != MA_SUCCESS) {
break;
}
}
/* If we are doing a post format conversion we need to do that now. */
if (pConverter->hasPostFormatConversion) {
if (pFramesOutThisIteration != NULL) {
ma_convert_pcm_frames_format(pFramesOutThisIteration, pConverter->formatOut, pTempBufferOut, pConverter->channelConverter.format, frameCountThisIteration, pConverter->channelConverter.channelsOut, pConverter->ditherMode);
}
}
framesProcessed += frameCountThisIteration;
}
}
if (pFrameCountIn != NULL) {
*pFrameCountIn = frameCount;
}
if (pFrameCountOut != NULL) {
*pFrameCountOut = frameCount;
}
return MA_SUCCESS;
}
static ma_result ma_data_converter_process_pcm_frames__resample_first(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
ma_result result;
ma_uint64 frameCountIn;
ma_uint64 frameCountOut;
ma_uint64 framesProcessedIn;
ma_uint64 framesProcessedOut;
ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format. */
ma_uint64 tempBufferInCap;
ma_uint8 pTempBufferMid[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format, channel converter input format. */
ma_uint64 tempBufferMidCap;
ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In channel converter output format. */
ma_uint64 tempBufferOutCap;
MA_ASSERT(pConverter != NULL);
MA_ASSERT(pConverter->resampler.format == pConverter->channelConverter.format);
MA_ASSERT(pConverter->resampler.channels == pConverter->channelConverter.channelsIn);
MA_ASSERT(pConverter->resampler.channels < pConverter->channelConverter.channelsOut);
frameCountIn = 0;
if (pFrameCountIn != NULL) {
frameCountIn = *pFrameCountIn;
}
frameCountOut = 0;
if (pFrameCountOut != NULL) {
frameCountOut = *pFrameCountOut;
}
framesProcessedIn = 0;
framesProcessedOut = 0;
tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->resampler.format, pConverter->resampler.channels);
tempBufferMidCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->resampler.format, pConverter->resampler.channels);
tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsOut);
while (framesProcessedOut < frameCountOut) {
ma_uint64 frameCountInThisIteration;
ma_uint64 frameCountOutThisIteration;
const void* pRunningFramesIn = NULL;
void* pRunningFramesOut = NULL;
const void* pResampleBufferIn;
void* pChannelsBufferOut;
if (pFramesIn != NULL) {
pRunningFramesIn = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pConverter->formatIn, pConverter->channelsIn));
}
if (pFramesOut != NULL) {
pRunningFramesOut = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
}
/* Run input data through the resampler and output it to the temporary buffer. */
frameCountInThisIteration = (frameCountIn - framesProcessedIn);
if (pConverter->hasPreFormatConversion) {
if (frameCountInThisIteration > tempBufferInCap) {
frameCountInThisIteration = tempBufferInCap;
}
}
frameCountOutThisIteration = (frameCountOut - framesProcessedOut);
if (frameCountOutThisIteration > tempBufferMidCap) {
frameCountOutThisIteration = tempBufferMidCap;
}
/* We can't read more frames than can fit in the output buffer. */
if (pConverter->hasPostFormatConversion) {
if (frameCountOutThisIteration > tempBufferOutCap) {
frameCountOutThisIteration = tempBufferOutCap;
}
}
/* We need to ensure we don't try to process too many input frames that we run out of room in the output buffer. If this happens we'll end up glitching. */
/*
We need to try to predict how many input frames will be required for the resampler. If the
resampler can tell us, we'll use that. Otherwise we'll need to make a best guess. The further
off we are from this, the more wasted format conversions we'll end up doing.
*/
#if 1
{
ma_uint64 requiredInputFrameCount;
result = ma_resampler_get_required_input_frame_count(&pConverter->resampler, frameCountOutThisIteration, &requiredInputFrameCount);
if (result != MA_SUCCESS) {
/* Fall back to a best guess. */
requiredInputFrameCount = (frameCountOutThisIteration * pConverter->resampler.sampleRateIn) / pConverter->resampler.sampleRateOut;
}
if (frameCountInThisIteration > requiredInputFrameCount) {
frameCountInThisIteration = requiredInputFrameCount;
}
}
#endif
if (pConverter->hasPreFormatConversion) {
if (pFramesIn != NULL) {
ma_convert_pcm_frames_format(pTempBufferIn, pConverter->resampler.format, pRunningFramesIn, pConverter->formatIn, frameCountInThisIteration, pConverter->channelsIn, pConverter->ditherMode);
pResampleBufferIn = pTempBufferIn;
} else {
pResampleBufferIn = NULL;
}
} else {
pResampleBufferIn = pRunningFramesIn;
}
result = ma_resampler_process_pcm_frames(&pConverter->resampler, pResampleBufferIn, &frameCountInThisIteration, pTempBufferMid, &frameCountOutThisIteration);
if (result != MA_SUCCESS) {
return result;
}
/*
The input data has been resampled so now we need to run it through the channel converter. The input data is always contained in pTempBufferMid. We only need to do
this part if we have an output buffer.
*/
if (pFramesOut != NULL) {
if (pConverter->hasPostFormatConversion) {
pChannelsBufferOut = pTempBufferOut;
} else {
pChannelsBufferOut = pRunningFramesOut;
}
result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pChannelsBufferOut, pTempBufferMid, frameCountOutThisIteration);
if (result != MA_SUCCESS) {
return result;
}
/* Finally we do post format conversion. */
if (pConverter->hasPostFormatConversion) {
ma_convert_pcm_frames_format(pRunningFramesOut, pConverter->formatOut, pChannelsBufferOut, pConverter->channelConverter.format, frameCountOutThisIteration, pConverter->channelConverter.channelsOut, pConverter->ditherMode);
}
}
framesProcessedIn += frameCountInThisIteration;
framesProcessedOut += frameCountOutThisIteration;
MA_ASSERT(framesProcessedIn <= frameCountIn);
MA_ASSERT(framesProcessedOut <= frameCountOut);
if (frameCountOutThisIteration == 0) {
break; /* Consumed all of our input data. */
}
}
if (pFrameCountIn != NULL) {
*pFrameCountIn = framesProcessedIn;
}
if (pFrameCountOut != NULL) {
*pFrameCountOut = framesProcessedOut;
}
return MA_SUCCESS;
}
static ma_result ma_data_converter_process_pcm_frames__channels_first(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
ma_result result;
ma_uint64 frameCountIn;
ma_uint64 frameCountOut;
ma_uint64 framesProcessedIn;
ma_uint64 framesProcessedOut;
ma_uint8 pTempBufferIn[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format. */
ma_uint64 tempBufferInCap;
ma_uint8 pTempBufferMid[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In resampler format, channel converter input format. */
ma_uint64 tempBufferMidCap;
ma_uint8 pTempBufferOut[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In channel converter output format. */
ma_uint64 tempBufferOutCap;
MA_ASSERT(pConverter != NULL);
MA_ASSERT(pConverter->resampler.format == pConverter->channelConverter.format);
MA_ASSERT(pConverter->resampler.channels == pConverter->channelConverter.channelsOut);
MA_ASSERT(pConverter->resampler.channels <= pConverter->channelConverter.channelsIn);
frameCountIn = 0;
if (pFrameCountIn != NULL) {
frameCountIn = *pFrameCountIn;
}
frameCountOut = 0;
if (pFrameCountOut != NULL) {
frameCountOut = *pFrameCountOut;
}
framesProcessedIn = 0;
framesProcessedOut = 0;
tempBufferInCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsIn);
tempBufferMidCap = sizeof(pTempBufferIn) / ma_get_bytes_per_frame(pConverter->channelConverter.format, pConverter->channelConverter.channelsOut);
tempBufferOutCap = sizeof(pTempBufferOut) / ma_get_bytes_per_frame(pConverter->resampler.format, pConverter->resampler.channels);
while (framesProcessedOut < frameCountOut) {
ma_uint64 frameCountInThisIteration;
ma_uint64 frameCountOutThisIteration;
const void* pRunningFramesIn = NULL;
void* pRunningFramesOut = NULL;
const void* pChannelsBufferIn;
void* pResampleBufferOut;
if (pFramesIn != NULL) {
pRunningFramesIn = ma_offset_ptr(pFramesIn, framesProcessedIn * ma_get_bytes_per_frame(pConverter->formatIn, pConverter->channelsIn));
}
if (pFramesOut != NULL) {
pRunningFramesOut = ma_offset_ptr(pFramesOut, framesProcessedOut * ma_get_bytes_per_frame(pConverter->formatOut, pConverter->channelsOut));
}
/*
Before doing any processing we need to determine how many frames we should try processing
this iteration, for both input and output. The resampler requires us to perform format and
channel conversion before passing any data into it. If we get our input count wrong, we'll
end up peforming redundant pre-processing. This isn't the end of the world, but it does
result in some inefficiencies proportionate to how far our estimates are off.
If the resampler has a means to calculate exactly how much we'll need, we'll use that.
Otherwise we'll make a best guess. In order to do this, we'll need to calculate the output
frame count first.
*/
frameCountOutThisIteration = (frameCountOut - framesProcessedOut);
if (frameCountOutThisIteration > tempBufferMidCap) {
frameCountOutThisIteration = tempBufferMidCap;
}
if (pConverter->hasPostFormatConversion) {
if (frameCountOutThisIteration > tempBufferOutCap) {
frameCountOutThisIteration = tempBufferOutCap;
}
}
/* Now that we have the output frame count we can determine the input frame count. */
frameCountInThisIteration = (frameCountIn - framesProcessedIn);
if (pConverter->hasPreFormatConversion) {
if (frameCountInThisIteration > tempBufferInCap) {
frameCountInThisIteration = tempBufferInCap;
}
}
if (frameCountInThisIteration > tempBufferMidCap) {
frameCountInThisIteration = tempBufferMidCap;
}
#if 1
{
ma_uint64 requiredInputFrameCount;
result = ma_resampler_get_required_input_frame_count(&pConverter->resampler, frameCountOutThisIteration, &requiredInputFrameCount);
if (result != MA_SUCCESS) {
/* Fall back to a best guess. */
requiredInputFrameCount = (frameCountOutThisIteration * pConverter->resampler.sampleRateIn) / pConverter->resampler.sampleRateOut;
}
if (frameCountInThisIteration > requiredInputFrameCount) {
frameCountInThisIteration = requiredInputFrameCount;
}
}
#endif
/* Pre format conversion. */
if (pConverter->hasPreFormatConversion) {
if (pRunningFramesIn != NULL) {
ma_convert_pcm_frames_format(pTempBufferIn, pConverter->channelConverter.format, pRunningFramesIn, pConverter->formatIn, frameCountInThisIteration, pConverter->channelsIn, pConverter->ditherMode);
pChannelsBufferIn = pTempBufferIn;
} else {
pChannelsBufferIn = NULL;
}
} else {
pChannelsBufferIn = pRunningFramesIn;
}
/* Channel conversion. */
result = ma_channel_converter_process_pcm_frames(&pConverter->channelConverter, pTempBufferMid, pChannelsBufferIn, frameCountInThisIteration);
if (result != MA_SUCCESS) {
return result;
}
/* Resampling. */
if (pConverter->hasPostFormatConversion) {
pResampleBufferOut = pTempBufferOut;
} else {
pResampleBufferOut = pRunningFramesOut;
}
result = ma_resampler_process_pcm_frames(&pConverter->resampler, pTempBufferMid, &frameCountInThisIteration, pResampleBufferOut, &frameCountOutThisIteration);
if (result != MA_SUCCESS) {
return result;
}
/* Post format conversion. */
if (pConverter->hasPostFormatConversion) {
if (pRunningFramesOut != NULL) {
ma_convert_pcm_frames_format(pRunningFramesOut, pConverter->formatOut, pResampleBufferOut, pConverter->resampler.format, frameCountOutThisIteration, pConverter->channelsOut, pConverter->ditherMode);
}
}
framesProcessedIn += frameCountInThisIteration;
framesProcessedOut += frameCountOutThisIteration;
MA_ASSERT(framesProcessedIn <= frameCountIn);
MA_ASSERT(framesProcessedOut <= frameCountOut);
if (frameCountOutThisIteration == 0) {
break; /* Consumed all of our input data. */
}
}
if (pFrameCountIn != NULL) {
*pFrameCountIn = framesProcessedIn;
}
if (pFrameCountOut != NULL) {
*pFrameCountOut = framesProcessedOut;
}
return MA_SUCCESS;
}
MA_API ma_result ma_data_converter_process_pcm_frames(ma_data_converter* pConverter, const void* pFramesIn, ma_uint64* pFrameCountIn, void* pFramesOut, ma_uint64* pFrameCountOut)
{
if (pConverter == NULL) {
return MA_INVALID_ARGS;
}
switch (pConverter->executionPath)
{
case ma_data_converter_execution_path_passthrough: return ma_data_converter_process_pcm_frames__passthrough(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
case ma_data_converter_execution_path_format_only: return ma_data_converter_process_pcm_frames__format_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
case ma_data_converter_execution_path_channels_only: return ma_data_converter_process_pcm_frames__channels_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
case ma_data_converter_execution_path_resample_only: return ma_data_converter_process_pcm_frames__resample_only(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
case ma_data_converter_execution_path_resample_first: return ma_data_converter_process_pcm_frames__resample_first(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
case ma_data_converter_execution_path_channels_first: return ma_data_converter_process_pcm_frames__channels_first(pConverter, pFramesIn, pFrameCountIn, pFramesOut, pFrameCountOut);
default: return MA_INVALID_OPERATION; /* Should never hit this. */
}
}
MA_API ma_result ma_data_converter_set_rate(ma_data_converter* pConverter, ma_uint32 sampleRateIn, ma_uint32 sampleRateOut)
{
if (pConverter == NULL) {
return MA_INVALID_ARGS;
}
if (pConverter->hasResampler == MA_FALSE) {
return MA_INVALID_OPERATION; /* Dynamic resampling not enabled. */
}
return ma_resampler_set_rate(&pConverter->resampler, sampleRateIn, sampleRateOut);
}
MA_API ma_result ma_data_converter_set_rate_ratio(ma_data_converter* pConverter, float ratioInOut)
{
if (pConverter == NULL) {
return MA_INVALID_ARGS;
}
if (pConverter->hasResampler == MA_FALSE) {
return MA_INVALID_OPERATION; /* Dynamic resampling not enabled. */
}
return ma_resampler_set_rate_ratio(&pConverter->resampler, ratioInOut);
}
MA_API ma_uint64 ma_data_converter_get_input_latency(const ma_data_converter* pConverter)
{
if (pConverter == NULL) {
return 0;
}
if (pConverter->hasResampler) {
return ma_resampler_get_input_latency(&pConverter->resampler);
}
return 0; /* No latency without a resampler. */
}
MA_API ma_uint64 ma_data_converter_get_output_latency(const ma_data_converter* pConverter)
{
if (pConverter == NULL) {
return 0;
}
if (pConverter->hasResampler) {
return ma_resampler_get_output_latency(&pConverter->resampler);
}
return 0; /* No latency without a resampler. */
}
MA_API ma_result ma_data_converter_get_required_input_frame_count(const ma_data_converter* pConverter, ma_uint64 outputFrameCount, ma_uint64* pInputFrameCount)
{
if (pInputFrameCount == NULL) {
return MA_INVALID_ARGS;
}
*pInputFrameCount = 0;
if (pConverter == NULL) {
return MA_INVALID_ARGS;
}
if (pConverter->hasResampler) {
return ma_resampler_get_required_input_frame_count(&pConverter->resampler, outputFrameCount, pInputFrameCount);
} else {
*pInputFrameCount = outputFrameCount; /* 1:1 */
return MA_SUCCESS;
}
}
MA_API ma_result ma_data_converter_get_expected_output_frame_count(const ma_data_converter* pConverter, ma_uint64 inputFrameCount, ma_uint64* pOutputFrameCount)
{
if (pOutputFrameCount == NULL) {
return MA_INVALID_ARGS;
}
*pOutputFrameCount = 0;
if (pConverter == NULL) {
return MA_INVALID_ARGS;
}
if (pConverter->hasResampler) {
return ma_resampler_get_expected_output_frame_count(&pConverter->resampler, inputFrameCount, pOutputFrameCount);
} else {
*pOutputFrameCount = inputFrameCount; /* 1:1 */
return MA_SUCCESS;
}
}
MA_API ma_result ma_data_converter_get_input_channel_map(const ma_data_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap)
{
if (pConverter == NULL || pChannelMap == NULL) {
return MA_INVALID_ARGS;
}
if (pConverter->hasChannelConverter) {
ma_channel_converter_get_output_channel_map(&pConverter->channelConverter, pChannelMap, channelMapCap);
} else {
ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pConverter->channelsOut);
}
return MA_SUCCESS;
}
MA_API ma_result ma_data_converter_get_output_channel_map(const ma_data_converter* pConverter, ma_channel* pChannelMap, size_t channelMapCap)
{
if (pConverter == NULL || pChannelMap == NULL) {
return MA_INVALID_ARGS;
}
if (pConverter->hasChannelConverter) {
ma_channel_converter_get_input_channel_map(&pConverter->channelConverter, pChannelMap, channelMapCap);
} else {
ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pConverter->channelsIn);
}
return MA_SUCCESS;
}
MA_API ma_result ma_data_converter_reset(ma_data_converter* pConverter)
{
if (pConverter == NULL) {
return MA_INVALID_ARGS;
}
/* There's nothing to do if we're not resampling. */
if (pConverter->hasResampler == MA_FALSE) {
return MA_SUCCESS;
}
return ma_resampler_reset(&pConverter->resampler);
}
/**************************************************************************************************************************************************************
Channel Maps
**************************************************************************************************************************************************************/
static ma_channel ma_channel_map_init_standard_channel(ma_standard_channel_map standardChannelMap, ma_uint32 channelCount, ma_uint32 channelIndex);
MA_API ma_channel ma_channel_map_get_channel(const ma_channel* pChannelMap, ma_uint32 channelCount, ma_uint32 channelIndex)
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
return MA_TRUE;
}
MA_API ma_bool32 ma_channel_map_is_equal(const ma_channel* pChannelMapA, const ma_channel* pChannelMapB, ma_uint32 channels)
{
ma_uint32 iChannel;
if (pChannelMapA == pChannelMapB) {
return MA_TRUE;
}
for (iChannel = 0; iChannel < channels; ++iChannel) {
if (ma_channel_map_get_channel(pChannelMapA, channels, iChannel) != ma_channel_map_get_channel(pChannelMapB, channels, iChannel)) {
return MA_FALSE;
}
}
return MA_TRUE;
}
MA_API ma_bool32 ma_channel_map_is_blank(const ma_channel* pChannelMap, ma_uint32 channels)
{
ma_uint32 iChannel;
/* A null channel map is equivalent to the default channel map. */
if (pChannelMap == NULL) {
return MA_FALSE;
}
for (iChannel = 0; iChannel < channels; ++iChannel) {
if (pChannelMap[iChannel] != MA_CHANNEL_NONE) {
return MA_FALSE;
}
}
return MA_TRUE;
}
MA_API ma_bool32 ma_channel_map_contains_channel_position(ma_uint32 channels, const ma_channel* pChannelMap, ma_channel channelPosition)
{
ma_uint32 iChannel;
for (iChannel = 0; iChannel < channels; ++iChannel) {
if (ma_channel_map_get_channel(pChannelMap, channels, iChannel) == channelPosition) {
return MA_TRUE;
}
}
return MA_FALSE;
}
/**************************************************************************************************************************************************************
Conversion Helpers
**************************************************************************************************************************************************************/
MA_API ma_uint64 ma_convert_frames(void* pOut, ma_uint64 frameCountOut, ma_format formatOut, ma_uint32 channelsOut, ma_uint32 sampleRateOut, const void* pIn, ma_uint64 frameCountIn, ma_format formatIn, ma_uint32 channelsIn, ma_uint32 sampleRateIn)
{
ma_data_converter_config config;
config = ma_data_converter_config_init(formatIn, formatOut, channelsIn, channelsOut, sampleRateIn, sampleRateOut);
config.resampling.linear.lpfOrder = ma_min(MA_DEFAULT_RESAMPLER_LPF_ORDER, MA_MAX_FILTER_ORDER);
return ma_convert_frames_ex(pOut, frameCountOut, pIn, frameCountIn, &config);
}
MA_API ma_uint64 ma_convert_frames_ex(void* pOut, ma_uint64 frameCountOut, const void* pIn, ma_uint64 frameCountIn, const ma_data_converter_config* pConfig)
{
ma_result result;
ma_data_converter converter;
if (frameCountIn == 0 || pConfig == NULL) {
return 0;
}
result = ma_data_converter_init(pConfig, NULL, &converter);
if (result != MA_SUCCESS) {
return 0; /* Failed to initialize the data converter. */
}
if (pOut == NULL) {
result = ma_data_converter_get_expected_output_frame_count(&converter, frameCountIn, &frameCountOut);
if (result != MA_SUCCESS) {
if (result == MA_NOT_IMPLEMENTED) {
/* No way to calculate the number of frames, so we'll need to brute force it and loop. */
frameCountOut = 0;
while (frameCountIn > 0) {
ma_uint64 framesProcessedIn = frameCountIn;
ma_uint64 framesProcessedOut = 0xFFFFFFFF;
result = ma_data_converter_process_pcm_frames(&converter, pIn, &framesProcessedIn, NULL, &framesProcessedOut);
if (result != MA_SUCCESS) {
break;
}
frameCountIn -= framesProcessedIn;
}
}
}
} else {
result = ma_data_converter_process_pcm_frames(&converter, pIn, &frameCountIn, pOut, &frameCountOut);
if (result != MA_SUCCESS) {
frameCountOut = 0;
}
}
ma_data_converter_uninit(&converter, NULL);
return frameCountOut;
}
/**************************************************************************************************************************************************************
Ring Buffer
**************************************************************************************************************************************************************/
static MA_INLINE ma_uint32 ma_rb__extract_offset_in_bytes(ma_uint32 encodedOffset)
{
return encodedOffset & 0x7FFFFFFF;
}
static MA_INLINE ma_uint32 ma_rb__extract_offset_loop_flag(ma_uint32 encodedOffset)
{
return encodedOffset & 0x80000000;
}
static MA_INLINE void* ma_rb__get_read_ptr(ma_rb* pRB)
{
MA_ASSERT(pRB != NULL);
return ma_offset_ptr(pRB->pBuffer, ma_rb__extract_offset_in_bytes(c89atomic_load_32(&pRB->encodedReadOffset)));
}
static MA_INLINE void* ma_rb__get_write_ptr(ma_rb* pRB)
{
MA_ASSERT(pRB != NULL);
return ma_offset_ptr(pRB->pBuffer, ma_rb__extract_offset_in_bytes(c89atomic_load_32(&pRB->encodedWriteOffset)));
}
static MA_INLINE ma_uint32 ma_rb__construct_offset(ma_uint32 offsetInBytes, ma_uint32 offsetLoopFlag)
{
return offsetLoopFlag | offsetInBytes;
}
static MA_INLINE void ma_rb__deconstruct_offset(ma_uint32 encodedOffset, ma_uint32* pOffsetInBytes, ma_uint32* pOffsetLoopFlag)
{
MA_ASSERT(pOffsetInBytes != NULL);
MA_ASSERT(pOffsetLoopFlag != NULL);
*pOffsetInBytes = ma_rb__extract_offset_in_bytes(encodedOffset);
*pOffsetLoopFlag = ma_rb__extract_offset_loop_flag(encodedOffset);
}
MA_API ma_result ma_rb_init_ex(size_t subbufferSizeInBytes, size_t subbufferCount, size_t subbufferStrideInBytes, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_rb* pRB)
{
ma_result result;
const ma_uint32 maxSubBufferSize = 0x7FFFFFFF - (MA_SIMD_ALIGNMENT-1);
if (pRB == NULL) {
return MA_INVALID_ARGS;
}
if (subbufferSizeInBytes == 0 || subbufferCount == 0) {
return MA_INVALID_ARGS;
}
if (subbufferSizeInBytes > maxSubBufferSize) {
return MA_INVALID_ARGS; /* Maximum buffer size is ~2GB. The most significant bit is a flag for use internally. */
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
return dist;
}
MA_API ma_uint32 ma_rb_available_write(ma_rb* pRB)
{
if (pRB == NULL) {
return 0;
}
return (ma_uint32)(ma_rb_get_subbuffer_size(pRB) - ma_rb_pointer_distance(pRB));
}
MA_API size_t ma_rb_get_subbuffer_size(ma_rb* pRB)
{
if (pRB == NULL) {
return 0;
}
return pRB->subbufferSizeInBytes;
}
MA_API size_t ma_rb_get_subbuffer_stride(ma_rb* pRB)
{
if (pRB == NULL) {
return 0;
}
if (pRB->subbufferStrideInBytes == 0) {
return (size_t)pRB->subbufferSizeInBytes;
}
return (size_t)pRB->subbufferStrideInBytes;
}
MA_API size_t ma_rb_get_subbuffer_offset(ma_rb* pRB, size_t subbufferIndex)
{
if (pRB == NULL) {
return 0;
}
return subbufferIndex * ma_rb_get_subbuffer_stride(pRB);
}
MA_API void* ma_rb_get_subbuffer_ptr(ma_rb* pRB, size_t subbufferIndex, void* pBuffer)
{
if (pRB == NULL) {
return NULL;
}
return ma_offset_ptr(pBuffer, ma_rb_get_subbuffer_offset(pRB, subbufferIndex));
}
static MA_INLINE ma_uint32 ma_pcm_rb_get_bpf(ma_pcm_rb* pRB)
{
MA_ASSERT(pRB != NULL);
return ma_get_bytes_per_frame(pRB->format, pRB->channels);
}
MA_API ma_result ma_pcm_rb_init_ex(ma_format format, ma_uint32 channels, ma_uint32 subbufferSizeInFrames, ma_uint32 subbufferCount, ma_uint32 subbufferStrideInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallba...
{
ma_uint32 bpf;
ma_result result;
if (pRB == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pRB);
bpf = ma_get_bytes_per_frame(format, channels);
if (bpf == 0) {
return MA_INVALID_ARGS;
}
result = ma_rb_init_ex(subbufferSizeInFrames*bpf, subbufferCount, subbufferStrideInFrames*bpf, pOptionalPreallocatedBuffer, pAllocationCallbacks, &pRB->rb);
if (result != MA_SUCCESS) {
return result;
}
pRB->format = format;
pRB->channels = channels;
return MA_SUCCESS;
}
MA_API ma_result ma_pcm_rb_init(ma_format format, ma_uint32 channels, ma_uint32 bufferSizeInFrames, void* pOptionalPreallocatedBuffer, const ma_allocation_callbacks* pAllocationCallbacks, ma_pcm_rb* pRB)
{
return ma_pcm_rb_init_ex(format, channels, bufferSizeInFrames, 1, 0, pOptionalPreallocatedBuffer, pAllocationCallbacks, pRB);
}
MA_API void ma_pcm_rb_uninit(ma_pcm_rb* pRB)
{
if (pRB == NULL) {
return;
}
ma_rb_uninit(&pRB->rb);
}
MA_API void ma_pcm_rb_reset(ma_pcm_rb* pRB)
{
if (pRB == NULL) {
return;
}
ma_rb_reset(&pRB->rb);
}
MA_API ma_result ma_pcm_rb_acquire_read(ma_pcm_rb* pRB, ma_uint32* pSizeInFrames, void** ppBufferOut)
{
size_t sizeInBytes;
ma_result result;
if (pRB == NULL || pSizeInFrames == NULL) {
return MA_INVALID_ARGS;
}
sizeInBytes = *pSizeInFrames * ma_pcm_rb_get_bpf(pRB);
result = ma_rb_acquire_read(&pRB->rb, &sizeInBytes, ppBufferOut);
if (result != MA_SUCCESS) {
return result;
}
*pSizeInFrames = (ma_uint32)(sizeInBytes / (size_t)ma_pcm_rb_get_bpf(pRB));
return MA_SUCCESS;
}
MA_API ma_result ma_pcm_rb_commit_read(ma_pcm_rb* pRB, ma_uint32 sizeInFrames)
{
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
{
if (pRB == NULL) {
return 0;
}
return ma_rb_available_read(&pRB->rb) / ma_pcm_rb_get_bpf(pRB);
}
MA_API ma_uint32 ma_pcm_rb_available_write(ma_pcm_rb* pRB)
{
if (pRB == NULL) {
return 0;
}
return ma_rb_available_write(&pRB->rb) / ma_pcm_rb_get_bpf(pRB);
}
MA_API ma_uint32 ma_pcm_rb_get_subbuffer_size(ma_pcm_rb* pRB)
{
if (pRB == NULL) {
return 0;
}
return (ma_uint32)(ma_rb_get_subbuffer_size(&pRB->rb) / ma_pcm_rb_get_bpf(pRB));
}
MA_API ma_uint32 ma_pcm_rb_get_subbuffer_stride(ma_pcm_rb* pRB)
{
if (pRB == NULL) {
return 0;
}
return (ma_uint32)(ma_rb_get_subbuffer_stride(&pRB->rb) / ma_pcm_rb_get_bpf(pRB));
}
MA_API ma_uint32 ma_pcm_rb_get_subbuffer_offset(ma_pcm_rb* pRB, ma_uint32 subbufferIndex)
{
if (pRB == NULL) {
return 0;
}
return (ma_uint32)(ma_rb_get_subbuffer_offset(&pRB->rb, subbufferIndex) / ma_pcm_rb_get_bpf(pRB));
}
MA_API void* ma_pcm_rb_get_subbuffer_ptr(ma_pcm_rb* pRB, ma_uint32 subbufferIndex, void* pBuffer)
{
if (pRB == NULL) {
return NULL;
}
return ma_rb_get_subbuffer_ptr(&pRB->rb, subbufferIndex, pBuffer);
}
MA_API ma_result ma_duplex_rb_init(ma_format captureFormat, ma_uint32 captureChannels, ma_uint32 sampleRate, ma_uint32 captureInternalSampleRate, ma_uint32 captureInternalPeriodSizeInFrames, const ma_allocation_callbacks* pAllocationCallbacks, ma_dup...
{
ma_result result;
ma_uint32 sizeInFrames;
sizeInFrames = (ma_uint32)ma_calculate_frame_count_after_resampling(sampleRate, captureInternalSampleRate, captureInternalPeriodSizeInFrames * 5);
if (sizeInFrames == 0) {
return MA_INVALID_ARGS;
}
result = ma_pcm_rb_init(captureFormat, captureChannels, sizeInFrames, NULL, pAllocationCallbacks, &pRB->rb);
if (result != MA_SUCCESS) {
return result;
}
/* Seek forward a bit so we have a bit of a buffer in case of desyncs. */
ma_pcm_rb_seek_write((ma_pcm_rb*)pRB, captureInternalPeriodSizeInFrames * 2);
return MA_SUCCESS;
}
MA_API ma_result ma_duplex_rb_uninit(ma_duplex_rb* pRB)
{
ma_pcm_rb_uninit((ma_pcm_rb*)pRB);
return MA_SUCCESS;
}
/**************************************************************************************************************************************************************
Miscellaneous Helpers
**************************************************************************************************************************************************************/
MA_API const char* ma_result_description(ma_result result)
{
switch (result)
{
case MA_SUCCESS: return "No error";
case MA_ERROR: return "Unknown error";
case MA_INVALID_ARGS: return "Invalid argument";
case MA_INVALID_OPERATION: return "Invalid operation";
case MA_OUT_OF_MEMORY: return "Out of memory";
case MA_OUT_OF_RANGE: return "Out of range";
case MA_ACCESS_DENIED: return "Permission denied";
case MA_DOES_NOT_EXIST: return "Resource does not exist";
case MA_ALREADY_EXISTS: return "Resource already exists";
case MA_TOO_MANY_OPEN_FILES: return "Too many open files";
case MA_INVALID_FILE: return "Invalid file";
case MA_TOO_BIG: return "Too large";
case MA_PATH_TOO_LONG: return "Path too long";
case MA_NAME_TOO_LONG: return "Name too long";
case MA_NOT_DIRECTORY: return "Not a directory";
case MA_IS_DIRECTORY: return "Is a directory";
case MA_DIRECTORY_NOT_EMPTY: return "Directory not empty";
case MA_AT_END: return "At end";
case MA_NO_SPACE: return "No space available";
case MA_BUSY: return "Device or resource busy";
case MA_IO_ERROR: return "Input/output error";
case MA_INTERRUPT: return "Interrupted";
case MA_UNAVAILABLE: return "Resource unavailable";
case MA_ALREADY_IN_USE: return "Resource already in use";
case MA_BAD_ADDRESS: return "Bad address";
case MA_BAD_SEEK: return "Illegal seek";
case MA_BAD_PIPE: return "Broken pipe";
case MA_DEADLOCK: return "Deadlock";
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pDataSourceBase);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
pDataSourceBase->vtable = pConfig->vtable;
pDataSourceBase->rangeBegInFrames = 0;
pDataSourceBase->rangeEndInFrames = ~((ma_uint64)0);
pDataSourceBase->loopBegInFrames = 0;
pDataSourceBase->loopEndInFrames = ~((ma_uint64)0);
pDataSourceBase->pCurrent = pDataSource; /* Always read from ourself by default. */
pDataSourceBase->pNext = NULL;
pDataSourceBase->onGetNext = NULL;
return MA_SUCCESS;
}
MA_API void ma_data_source_uninit(ma_data_source* pDataSource)
{
if (pDataSource == NULL) {
return;
}
/*
This is placeholder in case we need this later. Data sources need to call this in their
uninitialization routine to ensure things work later on if something is added here.
*/
}
static ma_result ma_data_source_resolve_current(ma_data_source* pDataSource, ma_data_source** ppCurrentDataSource)
{
ma_data_source_base* pCurrentDataSource = (ma_data_source_base*)pDataSource;
MA_ASSERT(pDataSource != NULL);
MA_ASSERT(ppCurrentDataSource != NULL);
if (pCurrentDataSource->pCurrent == NULL) {
/*
The current data source is NULL. If we're using this in the context of a chain we need to return NULL
here so that we don't end up looping. Otherwise we just return the data source itself.
*/
if (pCurrentDataSource->pNext != NULL || pCurrentDataSource->onGetNext != NULL) {
pCurrentDataSource = NULL;
} else {
pCurrentDataSource = (ma_data_source_base*)pDataSource; /* Not being used in a chain. Make sure we just always read from the data source itself at all times. */
}
} else {
pCurrentDataSource = (ma_data_source_base*)pCurrentDataSource->pCurrent;
}
*ppCurrentDataSource = pCurrentDataSource;
return MA_SUCCESS;
}
static ma_result ma_data_source_read_pcm_frames_within_range(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
ma_result result;
ma_uint64 framesRead = 0;
ma_bool32 loop = ma_data_source_is_looping(pDataSource);
if (pDataSourceBase == NULL) {
return MA_AT_END;
}
if (frameCount == 0) {
return MA_INVALID_ARGS;
}
if ((pDataSourceBase->vtable->flags & MA_DATA_SOURCE_SELF_MANAGED_RANGE_AND_LOOP_POINT) != 0 || (pDataSourceBase->rangeEndInFrames == ~((ma_uint64)0) && (pDataSourceBase->loopEndInFrames == ~((ma_uint64)0) || loop == MA_FALSE))) {
/* Either the data source is self-managing the range, or no range is set - just read like normal. The data source itself will tell us when the end is reached. */
result = pDataSourceBase->vtable->onRead(pDataSourceBase, pFramesOut, frameCount, &framesRead);
} else {
/* Need to clamp to within the range. */
ma_uint64 cursor;
result = ma_data_source_get_cursor_in_pcm_frames(pDataSourceBase, &cursor);
if (result != MA_SUCCESS) {
/* Failed to retrieve the cursor. Cannot read within a range or loop points. Just read like normal - this may happen for things like noise data sources where it doesn't really matter. */
result = pDataSourceBase->vtable->onRead(pDataSourceBase, pFramesOut, frameCount, &framesRead);
} else {
ma_uint64 rangeEnd;
/* We have the cursor. We need to make sure we don't read beyond our range. */
rangeEnd = pDataSourceBase->rangeEndInFrames;
/* If looping, make sure we're within range. */
if (loop) {
if (pDataSourceBase->loopEndInFrames != ~((ma_uint64)0)) {
rangeEnd = ma_min(rangeEnd, pDataSourceBase->rangeBegInFrames + pDataSourceBase->loopEndInFrames);
}
}
if (frameCount > (rangeEnd - cursor) && rangeEnd != ~((ma_uint64)0)) {
frameCount = (rangeEnd - cursor);
}
/*
If the cursor is sitting on the end of the range the frame count will be set to 0 which can
result in MA_INVALID_ARGS. In this case, we don't want to try reading, but instead return
MA_AT_END so the higher level function can know about it.
*/
if (frameCount > 0) {
result = pDataSourceBase->vtable->onRead(pDataSourceBase, pFramesOut, frameCount, &framesRead);
} else {
result = MA_AT_END; /* The cursor is sitting on the end of the range which means we're at the end. */
}
}
}
if (pFramesRead != NULL) {
*pFramesRead = framesRead;
}
/* We need to make sure MA_AT_END is returned if we hit the end of the range. */
if (result == MA_SUCCESS && framesRead == 0) {
result = MA_AT_END;
}
return result;
}
MA_API ma_result ma_data_source_read_pcm_frames(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
ma_result result = MA_SUCCESS;
ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
ma_data_source_base* pCurrentDataSource;
void* pRunningFramesOut = pFramesOut;
ma_uint64 totalFramesProcessed = 0;
ma_format format;
ma_uint32 channels;
ma_uint32 emptyLoopCounter = 0; /* Keeps track of how many times 0 frames have been read. For infinite loop detection of sounds with no audio data. */
ma_bool32 loop;
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
if (frameCount == 0) {
return MA_INVALID_ARGS;
}
if (pDataSourceBase == NULL) {
return MA_INVALID_ARGS;
}
loop = ma_data_source_is_looping(pDataSource);
/*
We need to know the data format so we can advance the output buffer as we read frames. If this
fails, chaining will not work and we'll just read as much as we can from the current source.
*/
if (ma_data_source_get_data_format(pDataSource, &format, &channels, NULL, NULL, 0) != MA_SUCCESS) {
result = ma_data_source_resolve_current(pDataSource, (ma_data_source**)&pCurrentDataSource);
if (result != MA_SUCCESS) {
return result;
}
return ma_data_source_read_pcm_frames_within_range(pCurrentDataSource, pFramesOut, frameCount, pFramesRead);
}
/*
Looping is a bit of a special case. When the `loop` argument is true, chaining will not work and
only the current data source will be read from.
*/
/* Keep reading until we've read as many frames as possible. */
while (totalFramesProcessed < frameCount) {
ma_uint64 framesProcessed;
ma_uint64 framesRemaining = frameCount - totalFramesProcessed;
/* We need to resolve the data source that we'll actually be reading from. */
result = ma_data_source_resolve_current(pDataSource, (ma_data_source**)&pCurrentDataSource);
if (result != MA_SUCCESS) {
break;
}
if (pCurrentDataSource == NULL) {
break;
}
result = ma_data_source_read_pcm_frames_within_range(pCurrentDataSource, pRunningFramesOut, framesRemaining, &framesProcessed);
totalFramesProcessed += framesProcessed;
/*
If we encounted an error from the read callback, make sure it's propagated to the caller. The caller may need to know whether or not MA_BUSY is returned which is
not necessarily considered an error.
*/
if (result != MA_SUCCESS && result != MA_AT_END) {
break;
}
/*
We can determine if we've reached the end by checking if ma_data_source_read_pcm_frames_within_range() returned
MA_AT_END. To loop back to the start, all we need to do is seek back to the first frame.
*/
if (result == MA_AT_END) {
/*
The result needs to be reset back to MA_SUCCESS (from MA_AT_END) so that we don't
accidentally return MA_AT_END when data has been read in prior loop iterations. at the
end of this function, the result will be checked for MA_SUCCESS, and if the total
number of frames processed is 0, will be explicitly set to MA_AT_END.
*/
result = MA_SUCCESS;
/*
We reached the end. If we're looping, we just loop back to the start of the current
data source. If we're not looping we need to check if we have another in the chain, and
if so, switch to it.
*/
if (loop) {
if (framesProcessed == 0) {
emptyLoopCounter += 1;
if (emptyLoopCounter > 1) {
break; /* Infinite loop detected. Get out. */
}
} else {
emptyLoopCounter = 0;
}
result = ma_data_source_seek_to_pcm_frame(pCurrentDataSource, pCurrentDataSource->loopBegInFrames);
if (result != MA_SUCCESS) {
break; /* Failed to loop. Abort. */
}
/* Don't return MA_AT_END for looping sounds. */
result = MA_SUCCESS;
} else {
if (pCurrentDataSource->pNext != NULL) {
pDataSourceBase->pCurrent = pCurrentDataSource->pNext;
} else if (pCurrentDataSource->onGetNext != NULL) {
pDataSourceBase->pCurrent = pCurrentDataSource->onGetNext(pCurrentDataSource);
if (pDataSourceBase->pCurrent == NULL) {
break; /* Our callback did not return a next data source. We're done. */
}
} else {
/* Reached the end of the chain. We're done. */
break;
}
/* The next data source needs to be rewound to ensure data is read in looping scenarios. */
result = ma_data_source_seek_to_pcm_frame(pDataSourceBase->pCurrent, 0);
if (result != MA_SUCCESS) {
break;
}
}
}
if (pRunningFramesOut != NULL) {
pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesProcessed * ma_get_bytes_per_frame(format, channels));
}
}
if (pFramesRead != NULL) {
*pFramesRead = totalFramesProcessed;
}
MA_ASSERT(!(result == MA_AT_END && totalFramesProcessed > 0)); /* We should never be returning MA_AT_END if we read some data. */
if (result == MA_SUCCESS && totalFramesProcessed == 0) {
result = MA_AT_END;
}
return result;
}
MA_API ma_result ma_data_source_seek_pcm_frames(ma_data_source* pDataSource, ma_uint64 frameCount, ma_uint64* pFramesSeeked)
{
return ma_data_source_read_pcm_frames(pDataSource, NULL, frameCount, pFramesSeeked);
}
MA_API ma_result ma_data_source_seek_to_pcm_frame(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
if (pDataSourceBase == NULL) {
return MA_SUCCESS;
}
if (pDataSourceBase->vtable->onSeek == NULL) {
return MA_NOT_IMPLEMENTED;
}
if (frameIndex > pDataSourceBase->rangeEndInFrames) {
return MA_INVALID_OPERATION; /* Trying to seek to far forward. */
}
return pDataSourceBase->vtable->onSeek(pDataSource, pDataSourceBase->rangeBegInFrames + frameIndex);
}
MA_API ma_result ma_data_source_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
ma_result result;
ma_format format;
ma_uint32 channels;
ma_uint32 sampleRate;
/* Initialize to defaults for safety just in case the data source does not implement this callback. */
if (pFormat != NULL) {
*pFormat = ma_format_unknown;
}
if (pChannels != NULL) {
*pChannels = 0;
}
if (pSampleRate != NULL) {
*pSampleRate = 0;
}
if (pChannelMap != NULL) {
MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
}
if (pDataSourceBase == NULL) {
return MA_INVALID_ARGS;
}
if (pDataSourceBase->vtable->onGetDataFormat == NULL) {
return MA_NOT_IMPLEMENTED;
}
result = pDataSourceBase->vtable->onGetDataFormat(pDataSource, &format, &channels, &sampleRate, pChannelMap, channelMapCap);
if (result != MA_SUCCESS) {
return result;
}
if (pFormat != NULL) {
*pFormat = format;
}
if (pChannels != NULL) {
*pChannels = channels;
}
if (pSampleRate != NULL) {
*pSampleRate = sampleRate;
}
/* Channel map was passed in directly to the callback. This is safe due to the channelMapCap parameter. */
return MA_SUCCESS;
}
MA_API ma_result ma_data_source_get_cursor_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pCursor)
{
ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
ma_result result;
ma_uint64 cursor;
if (pCursor == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = 0;
if (pDataSourceBase == NULL) {
return MA_SUCCESS;
}
if (pDataSourceBase->vtable->onGetCursor == NULL) {
return MA_NOT_IMPLEMENTED;
}
result = pDataSourceBase->vtable->onGetCursor(pDataSourceBase, &cursor);
if (result != MA_SUCCESS) {
return result;
}
/* The cursor needs to be made relative to the start of the range. */
if (cursor < pDataSourceBase->rangeBegInFrames) { /* Safety check so we don't return some huge number. */
*pCursor = 0;
} else {
*pCursor = cursor - pDataSourceBase->rangeBegInFrames;
}
return MA_SUCCESS;
}
MA_API ma_result ma_data_source_get_length_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLength)
{
ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
if (pLength == NULL) {
return MA_INVALID_ARGS;
}
*pLength = 0;
if (pDataSourceBase == NULL) {
return MA_INVALID_ARGS;
}
/*
If we have a range defined we'll use that to determine the length. This is one of rare times
where we'll actually trust the caller. If they've set the range, I think it's mostly safe to
assume they've set it based on some higher level knowledge of the structure of the sound bank.
*/
if (pDataSourceBase->rangeEndInFrames != ~((ma_uint64)0)) {
*pLength = pDataSourceBase->rangeEndInFrames - pDataSourceBase->rangeBegInFrames;
return MA_SUCCESS;
}
/*
Getting here means a range is not defined so we'll need to get the data source itself to tell
us the length.
*/
if (pDataSourceBase->vtable->onGetLength == NULL) {
return MA_NOT_IMPLEMENTED;
}
return pDataSourceBase->vtable->onGetLength(pDataSource, pLength);
}
MA_API ma_result ma_data_source_get_cursor_in_seconds(ma_data_source* pDataSource, float* pCursor)
{
ma_result result;
ma_uint64 cursorInPCMFrames;
ma_uint32 sampleRate;
if (pCursor == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = 0;
result = ma_data_source_get_cursor_in_pcm_frames(pDataSource, &cursorInPCMFrames);
if (result != MA_SUCCESS) {
return result;
}
result = ma_data_source_get_data_format(pDataSource, NULL, NULL, &sampleRate, NULL, 0);
if (result != MA_SUCCESS) {
return result;
}
*pCursor = cursorInPCMFrames / (float)sampleRate;
return MA_SUCCESS;
}
MA_API ma_result ma_data_source_get_length_in_seconds(ma_data_source* pDataSource, float* pLength)
{
ma_result result;
ma_uint64 lengthInPCMFrames;
ma_uint32 sampleRate;
if (pLength == NULL) {
return MA_INVALID_ARGS;
}
*pLength = 0;
result = ma_data_source_get_length_in_pcm_frames(pDataSource, &lengthInPCMFrames);
if (result != MA_SUCCESS) {
return result;
}
result = ma_data_source_get_data_format(pDataSource, NULL, NULL, &sampleRate, NULL, 0);
if (result != MA_SUCCESS) {
return result;
}
*pLength = lengthInPCMFrames / (float)sampleRate;
return MA_SUCCESS;
}
MA_API ma_result ma_data_source_set_looping(ma_data_source* pDataSource, ma_bool32 isLooping)
{
ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
c89atomic_exchange_32(&pDataSourceBase->isLooping, isLooping);
/* If there's no callback for this just treat it as a successful no-op. */
if (pDataSourceBase->vtable->onSetLooping == NULL) {
return MA_SUCCESS;
}
return pDataSourceBase->vtable->onSetLooping(pDataSource, isLooping);
}
MA_API ma_bool32 ma_data_source_is_looping(const ma_data_source* pDataSource)
{
const ma_data_source_base* pDataSourceBase = (const ma_data_source_base*)pDataSource;
if (pDataSource == NULL) {
return MA_FALSE;
}
return c89atomic_load_32(&pDataSourceBase->isLooping);
}
MA_API ma_result ma_data_source_set_range_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 rangeBegInFrames, ma_uint64 rangeEndInFrames)
{
ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
ma_result result;
ma_uint64 cursor;
ma_uint64 loopBegAbsolute;
ma_uint64 loopEndAbsolute;
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
if (rangeEndInFrames < rangeBegInFrames) {
return MA_INVALID_ARGS; /* The end of the range must come after the beginning. */
}
/*
The loop points need to be updated. We'll be storing the loop points relative to the range. We'll update
these so that they maintain their absolute positioning. The loop points will then be clamped to the range.
*/
loopBegAbsolute = pDataSourceBase->loopBegInFrames + pDataSourceBase->rangeBegInFrames;
loopEndAbsolute = pDataSourceBase->loopEndInFrames + ((pDataSourceBase->loopEndInFrames != ~((ma_uint64)0)) ? pDataSourceBase->rangeBegInFrames : 0);
pDataSourceBase->rangeBegInFrames = rangeBegInFrames;
pDataSourceBase->rangeEndInFrames = rangeEndInFrames;
/* Make the loop points relative again, and make sure they're clamped to within the range. */
if (loopBegAbsolute > pDataSourceBase->rangeBegInFrames) {
pDataSourceBase->loopBegInFrames = loopBegAbsolute - pDataSourceBase->rangeBegInFrames;
} else {
pDataSourceBase->loopBegInFrames = 0;
}
if (pDataSourceBase->loopBegInFrames > pDataSourceBase->rangeEndInFrames) {
pDataSourceBase->loopBegInFrames = pDataSourceBase->rangeEndInFrames;
}
/* Only need to update the loop end point if it's not -1. */
if (loopEndAbsolute != ~((ma_uint64)0)) {
if (loopEndAbsolute > pDataSourceBase->rangeBegInFrames) {
pDataSourceBase->loopEndInFrames = loopEndAbsolute - pDataSourceBase->rangeBegInFrames;
} else {
pDataSourceBase->loopEndInFrames = 0;
}
if (pDataSourceBase->loopEndInFrames > pDataSourceBase->rangeEndInFrames && pDataSourceBase->loopEndInFrames) {
pDataSourceBase->loopEndInFrames = pDataSourceBase->rangeEndInFrames;
}
}
/* If the new range is past the current cursor position we need to seek to it. */
result = ma_data_source_get_cursor_in_pcm_frames(pDataSource, &cursor);
if (result == MA_SUCCESS) {
/* Seek to within range. Note that our seek positions here are relative to the new range. */
if (cursor < rangeBegInFrames) {
ma_data_source_seek_to_pcm_frame(pDataSource, 0);
} else if (cursor > rangeEndInFrames) {
ma_data_source_seek_to_pcm_frame(pDataSource, rangeEndInFrames - rangeBegInFrames);
}
} else {
/* We failed to get the cursor position. Probably means the data source has no notion of a cursor such a noise data source. Just pretend the seeking worked. */
}
return MA_SUCCESS;
}
MA_API void ma_data_source_get_range_in_pcm_frames(const ma_data_source* pDataSource, ma_uint64* pRangeBegInFrames, ma_uint64* pRangeEndInFrames)
{
const ma_data_source_base* pDataSourceBase = (const ma_data_source_base*)pDataSource;
if (pDataSource == NULL) {
return;
}
if (pRangeBegInFrames != NULL) {
*pRangeBegInFrames = pDataSourceBase->rangeBegInFrames;
}
if (pRangeEndInFrames != NULL) {
*pRangeEndInFrames = pDataSourceBase->rangeEndInFrames;
}
}
MA_API ma_result ma_data_source_set_loop_point_in_pcm_frames(ma_data_source* pDataSource, ma_uint64 loopBegInFrames, ma_uint64 loopEndInFrames)
{
ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
if (loopEndInFrames < loopBegInFrames) {
return MA_INVALID_ARGS; /* The end of the loop point must come after the beginning. */
}
if (loopEndInFrames > pDataSourceBase->rangeEndInFrames && loopEndInFrames != ~((ma_uint64)0)) {
return MA_INVALID_ARGS; /* The end of the loop point must not go beyond the range. */
}
pDataSourceBase->loopBegInFrames = loopBegInFrames;
pDataSourceBase->loopEndInFrames = loopEndInFrames;
/* The end cannot exceed the range. */
if (pDataSourceBase->loopEndInFrames > (pDataSourceBase->rangeEndInFrames - pDataSourceBase->rangeBegInFrames) && pDataSourceBase->loopEndInFrames != ~((ma_uint64)0)) {
pDataSourceBase->loopEndInFrames = (pDataSourceBase->rangeEndInFrames - pDataSourceBase->rangeBegInFrames);
}
return MA_SUCCESS;
}
MA_API void ma_data_source_get_loop_point_in_pcm_frames(const ma_data_source* pDataSource, ma_uint64* pLoopBegInFrames, ma_uint64* pLoopEndInFrames)
{
const ma_data_source_base* pDataSourceBase = (const ma_data_source_base*)pDataSource;
if (pDataSource == NULL) {
return;
}
if (pLoopBegInFrames != NULL) {
*pLoopBegInFrames = pDataSourceBase->loopBegInFrames;
}
if (pLoopEndInFrames != NULL) {
*pLoopEndInFrames = pDataSourceBase->loopEndInFrames;
}
}
MA_API ma_result ma_data_source_set_current(ma_data_source* pDataSource, ma_data_source* pCurrentDataSource)
{
ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
pDataSourceBase->pCurrent = pCurrentDataSource;
return MA_SUCCESS;
}
MA_API ma_data_source* ma_data_source_get_current(const ma_data_source* pDataSource)
{
const ma_data_source_base* pDataSourceBase = (const ma_data_source_base*)pDataSource;
if (pDataSource == NULL) {
return NULL;
}
return pDataSourceBase->pCurrent;
}
MA_API ma_result ma_data_source_set_next(ma_data_source* pDataSource, ma_data_source* pNextDataSource)
{
ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
pDataSourceBase->pNext = pNextDataSource;
return MA_SUCCESS;
}
MA_API ma_data_source* ma_data_source_get_next(const ma_data_source* pDataSource)
{
const ma_data_source_base* pDataSourceBase = (const ma_data_source_base*)pDataSource;
if (pDataSource == NULL) {
return NULL;
}
return pDataSourceBase->pNext;
}
MA_API ma_result ma_data_source_set_next_callback(ma_data_source* pDataSource, ma_data_source_get_next_proc onGetNext)
{
ma_data_source_base* pDataSourceBase = (ma_data_source_base*)pDataSource;
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
pDataSourceBase->onGetNext = onGetNext;
return MA_SUCCESS;
}
MA_API ma_data_source_get_next_proc ma_data_source_get_next_callback(const ma_data_source* pDataSource)
{
const ma_data_source_base* pDataSourceBase = (const ma_data_source_base*)pDataSource;
if (pDataSource == NULL) {
return NULL;
}
return pDataSourceBase->onGetNext;
}
static ma_result ma_audio_buffer_ref__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
ma_audio_buffer_ref* pAudioBufferRef = (ma_audio_buffer_ref*)pDataSource;
ma_uint64 framesRead = ma_audio_buffer_ref_read_pcm_frames(pAudioBufferRef, pFramesOut, frameCount, MA_FALSE);
if (pFramesRead != NULL) {
*pFramesRead = framesRead;
}
if (framesRead < frameCount || framesRead == 0) {
return MA_AT_END;
}
return MA_SUCCESS;
}
static ma_result ma_audio_buffer_ref__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
return ma_audio_buffer_ref_seek_to_pcm_frame((ma_audio_buffer_ref*)pDataSource, frameIndex);
}
static ma_result ma_audio_buffer_ref__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
ma_audio_buffer_ref* pAudioBufferRef = (ma_audio_buffer_ref*)pDataSource;
*pFormat = pAudioBufferRef->format;
*pChannels = pAudioBufferRef->channels;
*pSampleRate = pAudioBufferRef->sampleRate;
ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pAudioBufferRef->channels);
return MA_SUCCESS;
}
static ma_result ma_audio_buffer_ref__data_source_on_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
ma_audio_buffer_ref* pAudioBufferRef = (ma_audio_buffer_ref*)pDataSource;
*pCursor = pAudioBufferRef->cursor;
return MA_SUCCESS;
}
static ma_result ma_audio_buffer_ref__data_source_on_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
{
ma_audio_buffer_ref* pAudioBufferRef = (ma_audio_buffer_ref*)pDataSource;
*pLength = pAudioBufferRef->sizeInFrames;
return MA_SUCCESS;
}
static ma_data_source_vtable g_ma_audio_buffer_ref_data_source_vtable =
{
ma_audio_buffer_ref__data_source_on_read,
ma_audio_buffer_ref__data_source_on_seek,
ma_audio_buffer_ref__data_source_on_get_data_format,
ma_audio_buffer_ref__data_source_on_get_cursor,
ma_audio_buffer_ref__data_source_on_get_length,
NULL, /* onSetLooping */
0
};
MA_API ma_result ma_audio_buffer_ref_init(ma_format format, ma_uint32 channels, const void* pData, ma_uint64 sizeInFrames, ma_audio_buffer_ref* pAudioBufferRef)
{
ma_result result;
ma_data_source_config dataSourceConfig;
if (pAudioBufferRef == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pAudioBufferRef);
dataSourceConfig = ma_data_source_config_init();
dataSourceConfig.vtable = &g_ma_audio_buffer_ref_data_source_vtable;
result = ma_data_source_init(&dataSourceConfig, &pAudioBufferRef->ds);
if (result != MA_SUCCESS) {
return result;
}
pAudioBufferRef->format = format;
pAudioBufferRef->channels = channels;
pAudioBufferRef->sampleRate = 0; /* TODO: Version 0.12. Set this to sampleRate. */
pAudioBufferRef->cursor = 0;
pAudioBufferRef->sizeInFrames = sizeInFrames;
pAudioBufferRef->pData = pData;
return MA_SUCCESS;
}
MA_API void ma_audio_buffer_ref_uninit(ma_audio_buffer_ref* pAudioBufferRef)
{
if (pAudioBufferRef == NULL) {
return;
}
ma_data_source_uninit(&pAudioBufferRef->ds);
}
MA_API ma_result ma_audio_buffer_ref_set_data(ma_audio_buffer_ref* pAudioBufferRef, const void* pData, ma_uint64 sizeInFrames)
{
if (pAudioBufferRef == NULL) {
return MA_INVALID_ARGS;
}
pAudioBufferRef->cursor = 0;
pAudioBufferRef->sizeInFrames = sizeInFrames;
pAudioBufferRef->pData = pData;
return MA_SUCCESS;
}
MA_API ma_uint64 ma_audio_buffer_ref_read_pcm_frames(ma_audio_buffer_ref* pAudioBufferRef, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop)
{
ma_uint64 totalFramesRead = 0;
if (pAudioBufferRef == NULL) {
return 0;
}
if (frameCount == 0) {
return 0;
}
while (totalFramesRead < frameCount) {
ma_uint64 framesAvailable = pAudioBufferRef->sizeInFrames - pAudioBufferRef->cursor;
ma_uint64 framesRemaining = frameCount - totalFramesRead;
ma_uint64 framesToRead;
framesToRead = framesRemaining;
if (framesToRead > framesAvailable) {
framesToRead = framesAvailable;
}
if (pFramesOut != NULL) {
ma_copy_pcm_frames(ma_offset_ptr(pFramesOut, totalFramesRead * ma_get_bytes_per_frame(pAudioBufferRef->format, pAudioBufferRef->channels)), ma_offset_ptr(pAudioBufferRef->pData, pAudioBufferRef->cursor * ma_get_bytes_per_frame(pAudioBuffe...
}
totalFramesRead += framesToRead;
pAudioBufferRef->cursor += framesToRead;
if (pAudioBufferRef->cursor == pAudioBufferRef->sizeInFrames) {
if (loop) {
pAudioBufferRef->cursor = 0;
} else {
break; /* We've reached the end and we're not looping. Done. */
}
}
MA_ASSERT(pAudioBufferRef->cursor < pAudioBufferRef->sizeInFrames);
}
return totalFramesRead;
}
MA_API ma_result ma_audio_buffer_ref_seek_to_pcm_frame(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameIndex)
{
if (pAudioBufferRef == NULL) {
return MA_INVALID_ARGS;
}
if (frameIndex > pAudioBufferRef->sizeInFrames) {
return MA_INVALID_ARGS;
}
pAudioBufferRef->cursor = (size_t)frameIndex;
return MA_SUCCESS;
}
MA_API ma_result ma_audio_buffer_ref_map(ma_audio_buffer_ref* pAudioBufferRef, void** ppFramesOut, ma_uint64* pFrameCount)
{
ma_uint64 framesAvailable;
ma_uint64 frameCount = 0;
if (ppFramesOut != NULL) {
*ppFramesOut = NULL; /* Safety. */
}
if (pFrameCount != NULL) {
frameCount = *pFrameCount;
*pFrameCount = 0; /* Safety. */
}
if (pAudioBufferRef == NULL || ppFramesOut == NULL || pFrameCount == NULL) {
return MA_INVALID_ARGS;
}
framesAvailable = pAudioBufferRef->sizeInFrames - pAudioBufferRef->cursor;
if (frameCount > framesAvailable) {
frameCount = framesAvailable;
}
*ppFramesOut = ma_offset_ptr(pAudioBufferRef->pData, pAudioBufferRef->cursor * ma_get_bytes_per_frame(pAudioBufferRef->format, pAudioBufferRef->channels));
*pFrameCount = frameCount;
return MA_SUCCESS;
}
MA_API ma_result ma_audio_buffer_ref_unmap(ma_audio_buffer_ref* pAudioBufferRef, ma_uint64 frameCount)
{
ma_uint64 framesAvailable;
if (pAudioBufferRef == NULL) {
return MA_INVALID_ARGS;
}
framesAvailable = pAudioBufferRef->sizeInFrames - pAudioBufferRef->cursor;
if (frameCount > framesAvailable) {
return MA_INVALID_ARGS; /* The frame count was too big. This should never happen in an unmapping. Need to make sure the caller is aware of this. */
}
pAudioBufferRef->cursor += frameCount;
if (pAudioBufferRef->cursor == pAudioBufferRef->sizeInFrames) {
return MA_AT_END; /* Successful. Need to tell the caller that the end has been reached so that it can loop if desired. */
} else {
return MA_SUCCESS;
}
}
MA_API ma_bool32 ma_audio_buffer_ref_at_end(const ma_audio_buffer_ref* pAudioBufferRef)
{
if (pAudioBufferRef == NULL) {
return MA_FALSE;
}
return pAudioBufferRef->cursor == pAudioBufferRef->sizeInFrames;
}
MA_API ma_result ma_audio_buffer_ref_get_cursor_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pCursor)
{
if (pCursor == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = 0;
if (pAudioBufferRef == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = pAudioBufferRef->cursor;
return MA_SUCCESS;
}
MA_API ma_result ma_audio_buffer_ref_get_length_in_pcm_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pLength)
{
if (pLength == NULL) {
return MA_INVALID_ARGS;
}
*pLength = 0;
if (pAudioBufferRef == NULL) {
return MA_INVALID_ARGS;
}
*pLength = pAudioBufferRef->sizeInFrames;
return MA_SUCCESS;
}
MA_API ma_result ma_audio_buffer_ref_get_available_frames(const ma_audio_buffer_ref* pAudioBufferRef, ma_uint64* pAvailableFrames)
{
if (pAvailableFrames == NULL) {
return MA_INVALID_ARGS;
}
*pAvailableFrames = 0;
if (pAudioBufferRef == NULL) {
return MA_INVALID_ARGS;
}
if (pAudioBufferRef->sizeInFrames <= pAudioBufferRef->cursor) {
*pAvailableFrames = 0;
} else {
*pAvailableFrames = pAudioBufferRef->sizeInFrames - pAudioBufferRef->cursor;
}
return MA_SUCCESS;
}
MA_API ma_audio_buffer_config ma_audio_buffer_config_init(ma_format format, ma_uint32 channels, ma_uint64 sizeInFrames, const void* pData, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_audio_buffer_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = 0; /* TODO: Version 0.12. Set this to sampleRate. */
config.sizeInFrames = sizeInFrames;
config.pData = pData;
ma_allocation_callbacks_init_copy(&config.allocationCallbacks, pAllocationCallbacks);
return config;
}
static ma_result ma_audio_buffer_init_ex(const ma_audio_buffer_config* pConfig, ma_bool32 doCopy, ma_audio_buffer* pAudioBuffer)
{
ma_result result;
if (pAudioBuffer == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_MEMORY(pAudioBuffer, sizeof(*pAudioBuffer) - sizeof(pAudioBuffer->_pExtraData)); /* Safety. Don't overwrite the extra data. */
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->sizeInFrames == 0) {
return MA_INVALID_ARGS; /* Not allowing buffer sizes of 0 frames. */
}
result = ma_audio_buffer_ref_init(pConfig->format, pConfig->channels, NULL, 0, &pAudioBuffer->ref);
if (result != MA_SUCCESS) {
return result;
}
/* TODO: Version 0.12. Set this in ma_audio_buffer_ref_init() instead of here. */
pAudioBuffer->ref.sampleRate = pConfig->sampleRate;
ma_allocation_callbacks_init_copy(&pAudioBuffer->allocationCallbacks, &pConfig->allocationCallbacks);
if (doCopy) {
ma_uint64 allocationSizeInBytes;
void* pData;
allocationSizeInBytes = pConfig->sizeInFrames * ma_get_bytes_per_frame(pConfig->format, pConfig->channels);
if (allocationSizeInBytes > MA_SIZE_MAX) {
return MA_OUT_OF_MEMORY; /* Too big. */
}
pData = ma_malloc((size_t)allocationSizeInBytes, &pAudioBuffer->allocationCallbacks); /* Safe cast to size_t. */
if (pData == NULL) {
return MA_OUT_OF_MEMORY;
}
if (pConfig->pData != NULL) {
ma_copy_pcm_frames(pData, pConfig->pData, pConfig->sizeInFrames, pConfig->format, pConfig->channels);
} else {
ma_silence_pcm_frames(pData, pConfig->sizeInFrames, pConfig->format, pConfig->channels);
}
ma_audio_buffer_ref_set_data(&pAudioBuffer->ref, pData, pConfig->sizeInFrames);
pAudioBuffer->ownsData = MA_TRUE;
} else {
ma_audio_buffer_ref_set_data(&pAudioBuffer->ref, pConfig->pData, pConfig->sizeInFrames);
pAudioBuffer->ownsData = MA_FALSE;
}
return MA_SUCCESS;
}
static void ma_audio_buffer_uninit_ex(ma_audio_buffer* pAudioBuffer, ma_bool32 doFree)
{
if (pAudioBuffer == NULL) {
return;
}
if (pAudioBuffer->ownsData && pAudioBuffer->ref.pData != &pAudioBuffer->_pExtraData[0]) {
ma_free((void*)pAudioBuffer->ref.pData, &pAudioBuffer->allocationCallbacks); /* Naugty const cast, but OK in this case since we've guarded it with the ownsData check. */
}
if (doFree) {
ma_free(pAudioBuffer, &pAudioBuffer->allocationCallbacks);
}
ma_audio_buffer_ref_uninit(&pAudioBuffer->ref);
}
MA_API ma_result ma_audio_buffer_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer)
{
return ma_audio_buffer_init_ex(pConfig, MA_FALSE, pAudioBuffer);
}
MA_API ma_result ma_audio_buffer_init_copy(const ma_audio_buffer_config* pConfig, ma_audio_buffer* pAudioBuffer)
{
return ma_audio_buffer_init_ex(pConfig, MA_TRUE, pAudioBuffer);
}
MA_API ma_result ma_audio_buffer_alloc_and_init(const ma_audio_buffer_config* pConfig, ma_audio_buffer** ppAudioBuffer)
{
ma_result result;
ma_audio_buffer* pAudioBuffer;
ma_audio_buffer_config innerConfig; /* We'll be making some changes to the config, so need to make a copy. */
ma_uint64 allocationSizeInBytes;
if (ppAudioBuffer == NULL) {
return MA_INVALID_ARGS;
}
*ppAudioBuffer = NULL; /* Safety. */
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
innerConfig = *pConfig;
ma_allocation_callbacks_init_copy(&innerConfig.allocationCallbacks, &pConfig->allocationCallbacks);
allocationSizeInBytes = sizeof(*pAudioBuffer) - sizeof(pAudioBuffer->_pExtraData) + (pConfig->sizeInFrames * ma_get_bytes_per_frame(pConfig->format, pConfig->channels));
if (allocationSizeInBytes > MA_SIZE_MAX) {
return MA_OUT_OF_MEMORY; /* Too big. */
}
pAudioBuffer = (ma_audio_buffer*)ma_malloc((size_t)allocationSizeInBytes, &innerConfig.allocationCallbacks); /* Safe cast to size_t. */
if (pAudioBuffer == NULL) {
return MA_OUT_OF_MEMORY;
}
if (pConfig->pData != NULL) {
ma_copy_pcm_frames(&pAudioBuffer->_pExtraData[0], pConfig->pData, pConfig->sizeInFrames, pConfig->format, pConfig->channels);
} else {
ma_silence_pcm_frames(&pAudioBuffer->_pExtraData[0], pConfig->sizeInFrames, pConfig->format, pConfig->channels);
}
innerConfig.pData = &pAudioBuffer->_pExtraData[0];
result = ma_audio_buffer_init_ex(&innerConfig, MA_FALSE, pAudioBuffer);
if (result != MA_SUCCESS) {
ma_free(pAudioBuffer, &innerConfig.allocationCallbacks);
return result;
}
*ppAudioBuffer = pAudioBuffer;
return MA_SUCCESS;
}
MA_API void ma_audio_buffer_uninit(ma_audio_buffer* pAudioBuffer)
{
ma_audio_buffer_uninit_ex(pAudioBuffer, MA_FALSE);
}
MA_API void ma_audio_buffer_uninit_and_free(ma_audio_buffer* pAudioBuffer)
{
ma_audio_buffer_uninit_ex(pAudioBuffer, MA_TRUE);
}
MA_API ma_uint64 ma_audio_buffer_read_pcm_frames(ma_audio_buffer* pAudioBuffer, void* pFramesOut, ma_uint64 frameCount, ma_bool32 loop)
{
if (pAudioBuffer == NULL) {
return 0;
}
return ma_audio_buffer_ref_read_pcm_frames(&pAudioBuffer->ref, pFramesOut, frameCount, loop);
}
MA_API ma_result ma_audio_buffer_seek_to_pcm_frame(ma_audio_buffer* pAudioBuffer, ma_uint64 frameIndex)
{
if (pAudioBuffer == NULL) {
return MA_INVALID_ARGS;
}
return ma_audio_buffer_ref_seek_to_pcm_frame(&pAudioBuffer->ref, frameIndex);
}
MA_API ma_result ma_audio_buffer_map(ma_audio_buffer* pAudioBuffer, void** ppFramesOut, ma_uint64* pFrameCount)
{
if (ppFramesOut != NULL) {
*ppFramesOut = NULL; /* Safety. */
}
if (pAudioBuffer == NULL) {
if (pFrameCount != NULL) {
*pFrameCount = 0;
}
return MA_INVALID_ARGS;
}
return ma_audio_buffer_ref_map(&pAudioBuffer->ref, ppFramesOut, pFrameCount);
}
MA_API ma_result ma_audio_buffer_unmap(ma_audio_buffer* pAudioBuffer, ma_uint64 frameCount)
{
if (pAudioBuffer == NULL) {
return MA_INVALID_ARGS;
}
return ma_audio_buffer_ref_unmap(&pAudioBuffer->ref, frameCount);
}
MA_API ma_bool32 ma_audio_buffer_at_end(const ma_audio_buffer* pAudioBuffer)
{
if (pAudioBuffer == NULL) {
return MA_FALSE;
}
return ma_audio_buffer_ref_at_end(&pAudioBuffer->ref);
}
MA_API ma_result ma_audio_buffer_get_cursor_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pCursor)
{
if (pAudioBuffer == NULL) {
return MA_INVALID_ARGS;
}
return ma_audio_buffer_ref_get_cursor_in_pcm_frames(&pAudioBuffer->ref, pCursor);
}
MA_API ma_result ma_audio_buffer_get_length_in_pcm_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pLength)
{
if (pAudioBuffer == NULL) {
return MA_INVALID_ARGS;
}
return ma_audio_buffer_ref_get_length_in_pcm_frames(&pAudioBuffer->ref, pLength);
}
MA_API ma_result ma_audio_buffer_get_available_frames(const ma_audio_buffer* pAudioBuffer, ma_uint64* pAvailableFrames)
{
if (pAvailableFrames == NULL) {
return MA_INVALID_ARGS;
}
*pAvailableFrames = 0;
if (pAudioBuffer == NULL) {
return MA_INVALID_ARGS;
}
return ma_audio_buffer_ref_get_available_frames(&pAudioBuffer->ref, pAvailableFrames);
}
MA_API ma_result ma_paged_audio_buffer_data_init(ma_format format, ma_uint32 channels, ma_paged_audio_buffer_data* pData)
{
if (pData == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pData);
pData->format = format;
pData->channels = channels;
pData->pTail = &pData->head;
return MA_SUCCESS;
}
MA_API void ma_paged_audio_buffer_data_uninit(ma_paged_audio_buffer_data* pData, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_paged_audio_buffer_page* pPage;
if (pData == NULL) {
return;
}
/* All pages need to be freed. */
pPage = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pData->head.pNext);
while (pPage != NULL) {
ma_paged_audio_buffer_page* pNext = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pPage->pNext);
ma_free(pPage, pAllocationCallbacks);
pPage = pNext;
}
}
MA_API ma_paged_audio_buffer_page* ma_paged_audio_buffer_data_get_head(ma_paged_audio_buffer_data* pData)
{
if (pData == NULL) {
return NULL;
}
return &pData->head;
}
MA_API ma_paged_audio_buffer_page* ma_paged_audio_buffer_data_get_tail(ma_paged_audio_buffer_data* pData)
{
if (pData == NULL) {
return NULL;
}
return pData->pTail;
}
MA_API ma_result ma_paged_audio_buffer_data_get_length_in_pcm_frames(ma_paged_audio_buffer_data* pData, ma_uint64* pLength)
{
ma_paged_audio_buffer_page* pPage;
if (pLength == NULL) {
return MA_INVALID_ARGS;
}
*pLength = 0;
if (pData == NULL) {
return MA_INVALID_ARGS;
}
/* Calculate the length from the linked list. */
for (pPage = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pData->head.pNext); pPage != NULL; pPage = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pPage->pNext)) {
*pLength += pPage->sizeInFrames;
}
return MA_SUCCESS;
}
MA_API ma_result ma_paged_audio_buffer_data_allocate_page(ma_paged_audio_buffer_data* pData, ma_uint64 pageSizeInFrames, const void* pInitialData, const ma_allocation_callbacks* pAllocationCallbacks, ma_paged_audio_buffer_page** ppPage)
{
ma_paged_audio_buffer_page* pPage;
ma_uint64 allocationSize;
if (ppPage == NULL) {
return MA_INVALID_ARGS;
}
*ppPage = NULL;
if (pData == NULL) {
return MA_INVALID_ARGS;
}
allocationSize = sizeof(*pPage) + (pageSizeInFrames * ma_get_bytes_per_frame(pData->format, pData->channels));
if (allocationSize > MA_SIZE_MAX) {
return MA_OUT_OF_MEMORY; /* Too big. */
}
pPage = (ma_paged_audio_buffer_page*)ma_malloc((size_t)allocationSize, pAllocationCallbacks); /* Safe cast to size_t. */
if (pPage == NULL) {
return MA_OUT_OF_MEMORY;
}
pPage->pNext = NULL;
pPage->sizeInFrames = pageSizeInFrames;
if (pInitialData != NULL) {
ma_copy_pcm_frames(pPage->pAudioData, pInitialData, pageSizeInFrames, pData->format, pData->channels);
}
*ppPage = pPage;
return MA_SUCCESS;
}
MA_API ma_result ma_paged_audio_buffer_data_free_page(ma_paged_audio_buffer_data* pData, ma_paged_audio_buffer_page* pPage, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pData == NULL || pPage == NULL) {
return MA_INVALID_ARGS;
}
/* It's assumed the page is not attached to the list. */
ma_free(pPage, pAllocationCallbacks);
return MA_SUCCESS;
}
MA_API ma_result ma_paged_audio_buffer_data_append_page(ma_paged_audio_buffer_data* pData, ma_paged_audio_buffer_page* pPage)
{
if (pData == NULL || pPage == NULL) {
return MA_INVALID_ARGS;
}
/* This function assumes the page has been filled with audio data by this point. As soon as we append, the page will be available for reading. */
/* First thing to do is update the tail. */
for (;;) {
ma_paged_audio_buffer_page* pOldTail = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pData->pTail);
ma_paged_audio_buffer_page* pNewTail = pPage;
if (c89atomic_compare_exchange_weak_ptr((volatile void**)&pData->pTail, (void**)&pOldTail, pNewTail)) {
/* Here is where we append the page to the list. After this, the page is attached to the list and ready to be read from. */
c89atomic_exchange_ptr(&pOldTail->pNext, pPage);
break; /* Done. */
}
}
return MA_SUCCESS;
}
MA_API ma_result ma_paged_audio_buffer_data_allocate_and_append_page(ma_paged_audio_buffer_data* pData, ma_uint32 pageSizeInFrames, const void* pInitialData, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_result result;
ma_paged_audio_buffer_page* pPage;
result = ma_paged_audio_buffer_data_allocate_page(pData, pageSizeInFrames, pInitialData, pAllocationCallbacks, &pPage);
if (result != MA_SUCCESS) {
return result;
}
return ma_paged_audio_buffer_data_append_page(pData, pPage); /* <-- Should never fail. */
}
MA_API ma_paged_audio_buffer_config ma_paged_audio_buffer_config_init(ma_paged_audio_buffer_data* pData)
{
ma_paged_audio_buffer_config config;
MA_ZERO_OBJECT(&config);
config.pData = pData;
return config;
}
static ma_result ma_paged_audio_buffer__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
return ma_paged_audio_buffer_read_pcm_frames((ma_paged_audio_buffer*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
static ma_result ma_paged_audio_buffer__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
return ma_paged_audio_buffer_seek_to_pcm_frame((ma_paged_audio_buffer*)pDataSource, frameIndex);
}
static ma_result ma_paged_audio_buffer__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
ma_paged_audio_buffer* pPagedAudioBuffer = (ma_paged_audio_buffer*)pDataSource;
*pFormat = pPagedAudioBuffer->pData->format;
*pChannels = pPagedAudioBuffer->pData->channels;
*pSampleRate = 0; /* There is no notion of a sample rate with audio buffers. */
ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pPagedAudioBuffer->pData->channels);
return MA_SUCCESS;
}
static ma_result ma_paged_audio_buffer__data_source_on_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
return ma_paged_audio_buffer_get_cursor_in_pcm_frames((ma_paged_audio_buffer*)pDataSource, pCursor);
}
static ma_result ma_paged_audio_buffer__data_source_on_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
{
return ma_paged_audio_buffer_get_length_in_pcm_frames((ma_paged_audio_buffer*)pDataSource, pLength);
}
static ma_data_source_vtable g_ma_paged_audio_buffer_data_source_vtable =
{
ma_paged_audio_buffer__data_source_on_read,
ma_paged_audio_buffer__data_source_on_seek,
ma_paged_audio_buffer__data_source_on_get_data_format,
ma_paged_audio_buffer__data_source_on_get_cursor,
ma_paged_audio_buffer__data_source_on_get_length,
NULL, /* onSetLooping */
0
};
MA_API ma_result ma_paged_audio_buffer_init(const ma_paged_audio_buffer_config* pConfig, ma_paged_audio_buffer* pPagedAudioBuffer)
{
ma_result result;
ma_data_source_config dataSourceConfig;
if (pPagedAudioBuffer == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pPagedAudioBuffer);
/* A config is required for the format and channel count. */
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->pData == NULL) {
return MA_INVALID_ARGS; /* No underlying data specified. */
}
dataSourceConfig = ma_data_source_config_init();
dataSourceConfig.vtable = &g_ma_paged_audio_buffer_data_source_vtable;
result = ma_data_source_init(&dataSourceConfig, &pPagedAudioBuffer->ds);
if (result != MA_SUCCESS) {
return result;
}
pPagedAudioBuffer->pData = pConfig->pData;
pPagedAudioBuffer->pCurrent = ma_paged_audio_buffer_data_get_head(pConfig->pData);
pPagedAudioBuffer->relativeCursor = 0;
pPagedAudioBuffer->absoluteCursor = 0;
return MA_SUCCESS;
}
MA_API void ma_paged_audio_buffer_uninit(ma_paged_audio_buffer* pPagedAudioBuffer)
{
if (pPagedAudioBuffer == NULL) {
return;
}
/* Nothing to do. The data needs to be deleted separately. */
}
MA_API ma_result ma_paged_audio_buffer_read_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
ma_result result = MA_SUCCESS;
ma_uint64 totalFramesRead = 0;
ma_format format;
ma_uint32 channels;
if (pPagedAudioBuffer == NULL) {
return MA_INVALID_ARGS;
}
format = pPagedAudioBuffer->pData->format;
channels = pPagedAudioBuffer->pData->channels;
while (totalFramesRead < frameCount) {
/* Read from the current page. The buffer should never be in a state where this is NULL. */
ma_uint64 framesRemainingInCurrentPage;
ma_uint64 framesRemainingToRead = frameCount - totalFramesRead;
ma_uint64 framesToReadThisIteration;
MA_ASSERT(pPagedAudioBuffer->pCurrent != NULL);
framesRemainingInCurrentPage = pPagedAudioBuffer->pCurrent->sizeInFrames - pPagedAudioBuffer->relativeCursor;
framesToReadThisIteration = ma_min(framesRemainingInCurrentPage, framesRemainingToRead);
ma_copy_pcm_frames(ma_offset_pcm_frames_ptr(pFramesOut, totalFramesRead, format, channels), ma_offset_pcm_frames_ptr(pPagedAudioBuffer->pCurrent->pAudioData, pPagedAudioBuffer->relativeCursor, format, channels), framesToReadThisIteration, for...
totalFramesRead += framesToReadThisIteration;
pPagedAudioBuffer->absoluteCursor += framesToReadThisIteration;
pPagedAudioBuffer->relativeCursor += framesToReadThisIteration;
/* Move to the next page if necessary. If there's no more pages, we need to return MA_AT_END. */
MA_ASSERT(pPagedAudioBuffer->relativeCursor <= pPagedAudioBuffer->pCurrent->sizeInFrames);
if (pPagedAudioBuffer->relativeCursor == pPagedAudioBuffer->pCurrent->sizeInFrames) {
/* We reached the end of the page. Need to move to the next. If there's no more pages, we're done. */
ma_paged_audio_buffer_page* pNext = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pPagedAudioBuffer->pCurrent->pNext);
if (pNext == NULL) {
result = MA_AT_END;
break; /* We've reached the end. */
} else {
pPagedAudioBuffer->pCurrent = pNext;
pPagedAudioBuffer->relativeCursor = 0;
}
}
}
if (pFramesRead != NULL) {
*pFramesRead = totalFramesRead;
}
return result;
}
MA_API ma_result ma_paged_audio_buffer_seek_to_pcm_frame(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64 frameIndex)
{
if (pPagedAudioBuffer == NULL) {
return MA_INVALID_ARGS;
}
if (frameIndex == pPagedAudioBuffer->absoluteCursor) {
return MA_SUCCESS; /* Nothing to do. */
}
if (frameIndex < pPagedAudioBuffer->absoluteCursor) {
/* Moving backwards. Need to move the cursor back to the start, and then move forward. */
pPagedAudioBuffer->pCurrent = ma_paged_audio_buffer_data_get_head(pPagedAudioBuffer->pData);
pPagedAudioBuffer->absoluteCursor = 0;
pPagedAudioBuffer->relativeCursor = 0;
/* Fall through to the forward seeking section below. */
}
if (frameIndex > pPagedAudioBuffer->absoluteCursor) {
/* Moving forward. */
ma_paged_audio_buffer_page* pPage;
ma_uint64 runningCursor = 0;
for (pPage = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&ma_paged_audio_buffer_data_get_head(pPagedAudioBuffer->pData)->pNext); pPage != NULL; pPage = (ma_paged_audio_buffer_page*)c89atomic_load_ptr(&pPage->pNext)) {
ma_uint64 pageRangeBeg = runningCursor;
ma_uint64 pageRangeEnd = pageRangeBeg + pPage->sizeInFrames;
if (frameIndex >= pageRangeBeg) {
if (frameIndex < pageRangeEnd || (frameIndex == pageRangeEnd && pPage == (ma_paged_audio_buffer_page*)c89atomic_load_ptr(ma_paged_audio_buffer_data_get_tail(pPagedAudioBuffer->pData)))) { /* A small edge case - allow seeking to the v...
/* We found the page. */
pPagedAudioBuffer->pCurrent = pPage;
pPagedAudioBuffer->absoluteCursor = frameIndex;
pPagedAudioBuffer->relativeCursor = frameIndex - pageRangeBeg;
return MA_SUCCESS;
}
}
runningCursor = pageRangeEnd;
}
/* Getting here means we tried seeking too far forward. Don't change any state. */
return MA_BAD_SEEK;
}
return MA_SUCCESS;
}
MA_API ma_result ma_paged_audio_buffer_get_cursor_in_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64* pCursor)
{
if (pCursor == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = 0; /* Safety. */
if (pPagedAudioBuffer == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = pPagedAudioBuffer->absoluteCursor;
return MA_SUCCESS;
}
MA_API ma_result ma_paged_audio_buffer_get_length_in_pcm_frames(ma_paged_audio_buffer* pPagedAudioBuffer, ma_uint64* pLength)
{
return ma_paged_audio_buffer_data_get_length_in_pcm_frames(pPagedAudioBuffer->pData, pLength);
}
/**************************************************************************************************************************************************************
VFS
**************************************************************************************************************************************************************/
MA_API ma_result ma_vfs_open(ma_vfs* pVFS, const char* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
{
ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
if (pFile == NULL) {
return MA_INVALID_ARGS;
}
*pFile = NULL;
if (pVFS == NULL || pFilePath == NULL || openMode == 0) {
return MA_INVALID_ARGS;
}
if (pCallbacks->onOpen == NULL) {
return MA_NOT_IMPLEMENTED;
}
return pCallbacks->onOpen(pVFS, pFilePath, openMode, pFile);
}
MA_API ma_result ma_vfs_open_w(ma_vfs* pVFS, const wchar_t* pFilePath, ma_uint32 openMode, ma_vfs_file* pFile)
{
ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
if (pFile == NULL) {
return MA_INVALID_ARGS;
}
*pFile = NULL;
if (pVFS == NULL || pFilePath == NULL || openMode == 0) {
return MA_INVALID_ARGS;
}
if (pCallbacks->onOpenW == NULL) {
return MA_NOT_IMPLEMENTED;
}
return pCallbacks->onOpenW(pVFS, pFilePath, openMode, pFile);
}
MA_API ma_result ma_vfs_close(ma_vfs* pVFS, ma_vfs_file file)
{
ma_vfs_callbacks* pCallbacks = (ma_vfs_callbacks*)pVFS;
if (pVFS == NULL || file == NULL) {
return MA_INVALID_ARGS;
}
if (pCallbacks->onClose == NULL) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
{
drwav_cue cue;
drwav_smpl smpl;
drwav_acid acid;
drwav_inst inst;
drwav_bext bext;
drwav_list_label_or_note labelOrNote;
drwav_list_labelled_cue_region labelledCueRegion;
drwav_list_info_text infoText;
drwav_unknown_metadata unknown;
} data;
} drwav_metadata;
typedef struct
{
drwav_read_proc onRead;
drwav_write_proc onWrite;
drwav_seek_proc onSeek;
void* pUserData;
drwav_allocation_callbacks allocationCallbacks;
drwav_container container;
drwav_fmt fmt;
drwav_uint32 sampleRate;
drwav_uint16 channels;
drwav_uint16 bitsPerSample;
drwav_uint16 translatedFormatTag;
drwav_uint64 totalPCMFrameCount;
drwav_uint64 dataChunkDataSize;
drwav_uint64 dataChunkDataPos;
drwav_uint64 bytesRemaining;
drwav_uint64 readCursorInPCMFrames;
drwav_uint64 dataChunkDataSizeTargetWrite;
drwav_bool32 isSequentialWrite;
drwav_metadata_type allowedMetadataTypes;
drwav_metadata* pMetadata;
drwav_uint32 metadataCount;
drwav__memory_stream memoryStream;
drwav__memory_stream_write memoryStreamWrite;
struct
{
drwav_uint32 bytesRemainingInBlock;
drwav_uint16 predictor[2];
drwav_int32 delta[2];
drwav_int32 cachedFrames[4];
drwav_uint32 cachedFrameCount;
drwav_int32 prevFrames[2][2];
} msadpcm;
struct
{
drwav_uint32 bytesRemainingInBlock;
drwav_int32 predictor[2];
drwav_int32 stepIndex[2];
drwav_int32 cachedFrames[16];
drwav_uint32 cachedFrameCount;
} ima;
} drwav;
DRWAV_API drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_chunk_proc onChunk, void* pReadSeekUserData, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_with_metadata(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_write_sequential_pcm_frames(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_write_with_metadata(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks, drwav_metadata* pMetadata,...
DRWAV_API drwav_uint64 drwav_target_write_size_bytes(const drwav_data_format* pFormat, drwav_uint64 totalFrameCount, drwav_metadata* pMetadata, drwav_uint32 metadataCount);
DRWAV_API drwav_metadata* drwav_take_ownership_of_metadata(drwav* pWav);
DRWAV_API drwav_result drwav_uninit(drwav* pWav);
DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut);
DRWAV_API drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut);
DRWAV_API drwav_uint64 drwav_read_pcm_frames_le(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut);
DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut);
DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex);
DRWAV_API drwav_result drwav_get_cursor_in_pcm_frames(drwav* pWav, drwav_uint64* pCursor);
DRWAV_API drwav_result drwav_get_length_in_pcm_frames(drwav* pWav, drwav_uint64* pLength);
DRWAV_API size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData);
DRWAV_API drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData);
DRWAV_API drwav_uint64 drwav_write_pcm_frames_le(drwav* pWav, drwav_uint64 framesToWrite, const void* pData);
DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 framesToWrite, const void* pData);
#ifndef DR_WAV_NO_CONVERSION_API
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut);
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16le(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut);
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16be(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut);
DRWAV_API void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
DRWAV_API void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
DRWAV_API void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount);
DRWAV_API void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount);
DRWAV_API void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount);
DRWAV_API void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
DRWAV_API void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount);
DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut);
DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut);
DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32be(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut);
DRWAV_API void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
DRWAV_API void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount);
DRWAV_API void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
DRWAV_API void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount);
DRWAV_API void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount);
DRWAV_API void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
DRWAV_API void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount);
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut);
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut);
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32be(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut);
DRWAV_API void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
DRWAV_API void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount);
DRWAV_API void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
DRWAV_API void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount);
DRWAV_API void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount);
DRWAV_API void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
DRWAV_API void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount);
#endif
#ifndef DR_WAV_NO_STDIO
DRWAV_API drwav_bool32 drwav_init_file(drwav* pWav, const char* filename, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_file_w(drwav* pWav, const wchar_t* filename, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_file_ex_w(drwav* pWav, const wchar_t* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_file_with_metadata(drwav* pWav, const char* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_file_with_metadata_w(drwav* pWav, const wchar_t* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_file_write_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_file_write_sequential_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks);
#endif
DRWAV_API drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_memory_with_metadata(drwav* pWav, const void* data, size_t dataSize, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_bool32 drwav_init_memory_write_sequential_pcm_frames(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks);
#ifndef DR_WAV_NO_CONVERSION_API
DRWAV_API drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAl...
DRWAV_API float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocati...
DRWAV_API drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAl...
#ifndef DR_WAV_NO_STDIO
DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
#endif
DRWAV_API drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks);
#endif
DRWAV_API void drwav_free(void* p, const drwav_allocation_callbacks* pAllocationCallbacks);
DRWAV_API drwav_uint16 drwav_bytes_to_u16(const drwav_uint8* data);
DRWAV_API drwav_int16 drwav_bytes_to_s16(const drwav_uint8* data);
DRWAV_API drwav_uint32 drwav_bytes_to_u32(const drwav_uint8* data);
DRWAV_API drwav_int32 drwav_bytes_to_s32(const drwav_uint8* data);
DRWAV_API drwav_uint64 drwav_bytes_to_u64(const drwav_uint8* data);
DRWAV_API drwav_int64 drwav_bytes_to_s64(const drwav_uint8* data);
DRWAV_API float drwav_bytes_to_f32(const drwav_uint8* data);
DRWAV_API drwav_bool32 drwav_guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]);
DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b);
#ifdef __cplusplus
}
#endif
#endif
/* dr_wav_h end */
#endif /* MA_NO_WAV */
#if !defined(MA_NO_FLAC) && !defined(MA_NO_DECODING)
/* dr_flac_h begin */
#ifndef dr_flac_h
#define dr_flac_h
#ifdef __cplusplus
extern "C" {
#endif
#define DRFLAC_STRINGIFY(x) #x
#define DRFLAC_XSTRINGIFY(x) DRFLAC_STRINGIFY(x)
#define DRFLAC_VERSION_MAJOR 0
#define DRFLAC_VERSION_MINOR 12
#define DRFLAC_VERSION_REVISION 38
#define DRFLAC_VERSION_STRING DRFLAC_XSTRINGIFY(DRFLAC_VERSION_MAJOR) "." DRFLAC_XSTRINGIFY(DRFLAC_VERSION_MINOR) "." DRFLAC_XSTRINGIFY(DRFLAC_VERSION_REVISION)
#include <stddef.h>
typedef signed char drflac_int8;
typedef unsigned char drflac_uint8;
typedef signed short drflac_int16;
typedef unsigned short drflac_uint16;
typedef signed int drflac_int32;
typedef unsigned int drflac_uint32;
#if defined(_MSC_VER) && !defined(__clang__)
typedef signed __int64 drflac_int64;
typedef unsigned __int64 drflac_uint64;
#else
#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
#pragma GCC diagnostic push
#pragma GCC diagnostic ignored "-Wlong-long"
#if defined(__clang__)
#pragma GCC diagnostic ignored "-Wc++11-long-long"
#endif
#endif
typedef signed long long drflac_int64;
typedef unsigned long long drflac_uint64;
#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6)))
#pragma GCC diagnostic pop
#endif
#endif
#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined(_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__)
typedef drflac_uint64 drflac_uintptr;
#else
typedef drflac_uint32 drflac_uintptr;
#endif
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
drflac_uint32 commentCount;
const void* pComments;
} vorbis_comment;
struct
{
char catalog[128];
drflac_uint64 leadInSampleCount;
drflac_bool32 isCD;
drflac_uint8 trackCount;
const void* pTrackData;
} cuesheet;
struct
{
drflac_uint32 type;
drflac_uint32 mimeLength;
const char* mime;
drflac_uint32 descriptionLength;
const char* description;
drflac_uint32 width;
drflac_uint32 height;
drflac_uint32 colorDepth;
drflac_uint32 indexColorCount;
drflac_uint32 pictureDataSize;
const drflac_uint8* pPictureData;
} picture;
} data;
} drflac_metadata;
typedef size_t (* drflac_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead);
typedef drflac_bool32 (* drflac_seek_proc)(void* pUserData, int offset, drflac_seek_origin origin);
typedef void (* drflac_meta_proc)(void* pUserData, drflac_metadata* pMetadata);
typedef struct
{
void* pUserData;
void* (* onMalloc)(size_t sz, void* pUserData);
void* (* onRealloc)(void* p, size_t sz, void* pUserData);
void (* onFree)(void* p, void* pUserData);
} drflac_allocation_callbacks;
typedef struct
{
const drflac_uint8* data;
size_t dataSize;
size_t currentReadPos;
} drflac__memory_stream;
typedef struct
{
drflac_read_proc onRead;
drflac_seek_proc onSeek;
void* pUserData;
size_t unalignedByteCount;
drflac_cache_t unalignedCache;
drflac_uint32 nextL2Line;
drflac_uint32 consumedBits;
drflac_cache_t cacheL2[DR_FLAC_BUFFER_SIZE/sizeof(drflac_cache_t)];
drflac_cache_t cache;
drflac_uint16 crc16;
drflac_cache_t crc16Cache;
drflac_uint32 crc16CacheIgnoredBytes;
} drflac_bs;
typedef struct
{
drflac_uint8 subframeType;
drflac_uint8 wastedBitsPerSample;
drflac_uint8 lpcOrder;
drflac_int32* pSamplesS32;
} drflac_subframe;
typedef struct
{
drflac_uint64 pcmFrameNumber;
drflac_uint32 flacFrameNumber;
drflac_uint32 sampleRate;
drflac_uint16 blockSizeInPCMFrames;
drflac_uint8 channelAssignment;
drflac_uint8 bitsPerSample;
drflac_uint8 crc8;
} drflac_frame_header;
typedef struct
{
drflac_frame_header header;
drflac_uint32 pcmFramesRemaining;
drflac_subframe subframes[8];
} drflac_frame;
typedef struct
{
drflac_meta_proc onMeta;
void* pUserDataMD;
drflac_allocation_callbacks allocationCallbacks;
drflac_uint32 sampleRate;
drflac_uint8 channels;
drflac_uint8 bitsPerSample;
drflac_uint16 maxBlockSizeInPCMFrames;
drflac_uint64 totalPCMFrameCount;
drflac_container container;
drflac_uint32 seekpointCount;
drflac_frame currentFLACFrame;
drflac_uint64 currentPCMFrame;
drflac_uint64 firstFLACFramePosInBytes;
drflac__memory_stream memoryStream;
drflac_int32* pDecodedSamples;
drflac_seekpoint* pSeekpoints;
void* _oggbs;
drflac_bool32 _noSeekTableSeek : 1;
drflac_bool32 _noBinarySearchSeek : 1;
drflac_bool32 _noBruteForceSeek : 1;
drflac_bs bs;
drflac_uint8 pExtraData[1];
} drflac;
DRFLAC_API drflac* drflac_open(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API drflac* drflac_open_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API drflac* drflac_open_with_metadata(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API drflac* drflac_open_with_metadata_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API void drflac_close(drflac* pFlac);
DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s32(drflac* pFlac, drflac_uint64 framesToRead, drflac_int32* pBufferOut);
DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s16(drflac* pFlac, drflac_uint64 framesToRead, drflac_int16* pBufferOut);
DRFLAC_API drflac_uint64 drflac_read_pcm_frames_f32(drflac* pFlac, drflac_uint64 framesToRead, float* pBufferOut);
DRFLAC_API drflac_bool32 drflac_seek_to_pcm_frame(drflac* pFlac, drflac_uint64 pcmFrameIndex);
#ifndef DR_FLAC_NO_STDIO
DRFLAC_API drflac* drflac_open_file(const char* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API drflac* drflac_open_file_w(const wchar_t* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API drflac* drflac_open_file_with_metadata(const char* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API drflac* drflac_open_file_with_metadata_w(const wchar_t* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
#endif
DRFLAC_API drflac* drflac_open_memory(const void* pData, size_t dataSize, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API drflac* drflac_open_memory_with_metadata(const void* pData, size_t dataSize, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API drflac_int32* drflac_open_and_read_pcm_frames_s32(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pA...
DRFLAC_API drflac_int16* drflac_open_and_read_pcm_frames_s16(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pA...
DRFLAC_API float* drflac_open_and_read_pcm_frames_f32(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocati...
#ifndef DR_FLAC_NO_STDIO
DRFLAC_API drflac_int32* drflac_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API drflac_int16* drflac_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API float* drflac_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
#endif
DRFLAC_API drflac_int32* drflac_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API drflac_int16* drflac_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API float* drflac_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks);
DRFLAC_API void drflac_free(void* p, const drflac_allocation_callbacks* pAllocationCallbacks);
typedef struct
{
drflac_uint32 countRemaining;
const char* pRunningData;
} drflac_vorbis_comment_iterator;
DRFLAC_API void drflac_init_vorbis_comment_iterator(drflac_vorbis_comment_iterator* pIter, drflac_uint32 commentCount, const void* pComments);
DRFLAC_API const char* drflac_next_vorbis_comment(drflac_vorbis_comment_iterator* pIter, drflac_uint32* pCommentLengthOut);
typedef struct
{
drflac_uint32 countRemaining;
const char* pRunningData;
} drflac_cuesheet_track_iterator;
#pragma pack(4)
typedef struct
{
drflac_uint64 offset;
drflac_uint8 index;
drflac_uint8 reserved[3];
} drflac_cuesheet_track_index;
#pragma pack()
typedef struct
{
drflac_uint64 offset;
drflac_uint8 trackNumber;
char ISRC[12];
drflac_bool8 isAudio;
drflac_bool8 preEmphasis;
drflac_uint8 indexCount;
const drflac_cuesheet_track_index* pIndexPoints;
} drflac_cuesheet_track;
DRFLAC_API void drflac_init_cuesheet_track_iterator(drflac_cuesheet_track_iterator* pIter, drflac_uint32 trackCount, const void* pTrackData);
DRFLAC_API drflac_bool32 drflac_next_cuesheet_track(drflac_cuesheet_track_iterator* pIter, drflac_cuesheet_track* pCuesheetTrack);
#ifdef __cplusplus
}
#endif
#endif
/* dr_flac_h end */
#endif /* MA_NO_FLAC */
#if !defined(MA_NO_MP3) && !defined(MA_NO_DECODING)
/* dr_mp3_h begin */
#ifndef dr_mp3_h
#define dr_mp3_h
#ifdef __cplusplus
extern "C" {
#endif
#define DRMP3_STRINGIFY(x) #x
#define DRMP3_XSTRINGIFY(x) DRMP3_STRINGIFY(x)
#define DRMP3_VERSION_MAJOR 0
#define DRMP3_VERSION_MINOR 6
#define DRMP3_VERSION_REVISION 33
#define DRMP3_VERSION_STRING DRMP3_XSTRINGIFY(DRMP3_VERSION_MAJOR) "." DRMP3_XSTRINGIFY(DRMP3_VERSION_MINOR) "." DRMP3_XSTRINGIFY(DRMP3_VERSION_REVISION)
#include <stddef.h>
typedef signed char drmp3_int8;
typedef unsigned char drmp3_uint8;
typedef signed short drmp3_int16;
typedef unsigned short drmp3_uint16;
typedef signed int drmp3_int32;
typedef unsigned int drmp3_uint32;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
#define DRMP3_NO_SPACE -18
#define DRMP3_BUSY -19
#define DRMP3_IO_ERROR -20
#define DRMP3_INTERRUPT -21
#define DRMP3_UNAVAILABLE -22
#define DRMP3_ALREADY_IN_USE -23
#define DRMP3_BAD_ADDRESS -24
#define DRMP3_BAD_SEEK -25
#define DRMP3_BAD_PIPE -26
#define DRMP3_DEADLOCK -27
#define DRMP3_TOO_MANY_LINKS -28
#define DRMP3_NOT_IMPLEMENTED -29
#define DRMP3_NO_MESSAGE -30
#define DRMP3_BAD_MESSAGE -31
#define DRMP3_NO_DATA_AVAILABLE -32
#define DRMP3_INVALID_DATA -33
#define DRMP3_TIMEOUT -34
#define DRMP3_NO_NETWORK -35
#define DRMP3_NOT_UNIQUE -36
#define DRMP3_NOT_SOCKET -37
#define DRMP3_NO_ADDRESS -38
#define DRMP3_BAD_PROTOCOL -39
#define DRMP3_PROTOCOL_UNAVAILABLE -40
#define DRMP3_PROTOCOL_NOT_SUPPORTED -41
#define DRMP3_PROTOCOL_FAMILY_NOT_SUPPORTED -42
#define DRMP3_ADDRESS_FAMILY_NOT_SUPPORTED -43
#define DRMP3_SOCKET_NOT_SUPPORTED -44
#define DRMP3_CONNECTION_RESET -45
#define DRMP3_ALREADY_CONNECTED -46
#define DRMP3_NOT_CONNECTED -47
#define DRMP3_CONNECTION_REFUSED -48
#define DRMP3_NO_HOST -49
#define DRMP3_IN_PROGRESS -50
#define DRMP3_CANCELLED -51
#define DRMP3_MEMORY_ALREADY_MAPPED -52
#define DRMP3_AT_END -53
#define DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME 1152
#define DRMP3_MAX_SAMPLES_PER_FRAME (DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME*2)
#ifdef _MSC_VER
#define DRMP3_INLINE __forceinline
#elif defined(__GNUC__)
#if defined(__STRICT_ANSI__)
#define DRMP3_GNUC_INLINE_HINT __inline__
#else
#define DRMP3_GNUC_INLINE_HINT inline
#endif
#if (__GNUC__ > 3 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 2)) || defined(__clang__)
#define DRMP3_INLINE DRMP3_GNUC_INLINE_HINT __attribute__((always_inline))
#else
#define DRMP3_INLINE DRMP3_GNUC_INLINE_HINT
#endif
#elif defined(__WATCOMC__)
#define DRMP3_INLINE __inline
#else
#define DRMP3_INLINE
#endif
DRMP3_API void drmp3_version(drmp3_uint32* pMajor, drmp3_uint32* pMinor, drmp3_uint32* pRevision);
DRMP3_API const char* drmp3_version_string(void);
typedef struct
{
int frame_bytes, channels, hz, layer, bitrate_kbps;
} drmp3dec_frame_info;
typedef struct
{
float mdct_overlap[2][9*32], qmf_state[15*2*32];
int reserv, free_format_bytes;
drmp3_uint8 header[4], reserv_buf[511];
} drmp3dec;
DRMP3_API void drmp3dec_init(drmp3dec *dec);
DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info);
DRMP3_API void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, size_t num_samples);
typedef enum
{
drmp3_seek_origin_start,
drmp3_seek_origin_current
} drmp3_seek_origin;
typedef struct
{
drmp3_uint64 seekPosInBytes;
drmp3_uint64 pcmFrameIndex;
drmp3_uint16 mp3FramesToDiscard;
drmp3_uint16 pcmFramesToDiscard;
} drmp3_seek_point;
typedef size_t (* drmp3_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead);
typedef drmp3_bool32 (* drmp3_seek_proc)(void* pUserData, int offset, drmp3_seek_origin origin);
typedef struct
{
void* pUserData;
void* (* onMalloc)(size_t sz, void* pUserData);
void* (* onRealloc)(void* p, size_t sz, void* pUserData);
void (* onFree)(void* p, void* pUserData);
} drmp3_allocation_callbacks;
typedef struct
{
drmp3_uint32 channels;
drmp3_uint32 sampleRate;
} drmp3_config;
typedef struct
{
drmp3dec decoder;
drmp3dec_frame_info frameInfo;
drmp3_uint32 channels;
drmp3_uint32 sampleRate;
drmp3_read_proc onRead;
drmp3_seek_proc onSeek;
void* pUserData;
drmp3_allocation_callbacks allocationCallbacks;
drmp3_uint32 mp3FrameChannels;
drmp3_uint32 mp3FrameSampleRate;
drmp3_uint32 pcmFramesConsumedInMP3Frame;
drmp3_uint32 pcmFramesRemainingInMP3Frame;
drmp3_uint8 pcmFrames[sizeof(float)*DRMP3_MAX_SAMPLES_PER_FRAME];
drmp3_uint64 currentPCMFrame;
drmp3_uint64 streamCursor;
drmp3_seek_point* pSeekPoints;
drmp3_uint32 seekPointCount;
size_t dataSize;
size_t dataCapacity;
size_t dataConsumed;
drmp3_uint8* pData;
drmp3_bool32 atEnd : 1;
struct
{
const drmp3_uint8* pData;
size_t dataSize;
size_t currentReadPos;
} memory;
} drmp3;
DRMP3_API drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_allocation_callbacks* pAllocationCallbacks);
DRMP3_API drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_allocation_callbacks* pAllocationCallbacks);
#ifndef DR_MP3_NO_STDIO
DRMP3_API drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks);
DRMP3_API drmp3_bool32 drmp3_init_file_w(drmp3* pMP3, const wchar_t* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks);
#endif
DRMP3_API void drmp3_uninit(drmp3* pMP3);
DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut);
DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_s16(drmp3* pMP3, drmp3_uint64 framesToRead, drmp3_int16* pBufferOut);
DRMP3_API drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex);
DRMP3_API drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3);
DRMP3_API drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3);
DRMP3_API drmp3_bool32 drmp3_get_mp3_and_pcm_frame_count(drmp3* pMP3, drmp3_uint64* pMP3FrameCount, drmp3_uint64* pPCMFrameCount);
DRMP3_API drmp3_bool32 drmp3_calculate_seek_points(drmp3* pMP3, drmp3_uint32* pSeekPointCount, drmp3_seek_point* pSeekPoints);
DRMP3_API drmp3_bool32 drmp3_bind_seek_table(drmp3* pMP3, drmp3_uint32 seekPointCount, drmp3_seek_point* pSeekPoints);
DRMP3_API float* drmp3_open_and_read_pcm_frames_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
DRMP3_API drmp3_int16* drmp3_open_and_read_pcm_frames_s16(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
DRMP3_API float* drmp3_open_memory_and_read_pcm_frames_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
DRMP3_API drmp3_int16* drmp3_open_memory_and_read_pcm_frames_s16(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
#ifndef DR_MP3_NO_STDIO
DRMP3_API float* drmp3_open_file_and_read_pcm_frames_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
DRMP3_API drmp3_int16* drmp3_open_file_and_read_pcm_frames_s16(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks);
#endif
DRMP3_API void* drmp3_malloc(size_t sz, const drmp3_allocation_callbacks* pAllocationCallbacks);
DRMP3_API void drmp3_free(void* p, const drmp3_allocation_callbacks* pAllocationCallbacks);
#ifdef __cplusplus
}
#endif
#endif
/* dr_mp3_h end */
#endif /* MA_NO_MP3 */
/**************************************************************************************************************************************************************
Decoding
**************************************************************************************************************************************************************/
#ifndef MA_NO_DECODING
static ma_result ma_decoder_read_bytes(ma_decoder* pDecoder, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead)
{
MA_ASSERT(pDecoder != NULL);
return pDecoder->onRead(pDecoder, pBufferOut, bytesToRead, pBytesRead);
}
static ma_result ma_decoder_seek_bytes(ma_decoder* pDecoder, ma_int64 byteOffset, ma_seek_origin origin)
{
MA_ASSERT(pDecoder != NULL);
return pDecoder->onSeek(pDecoder, byteOffset, origin);
}
static ma_result ma_decoder_tell_bytes(ma_decoder* pDecoder, ma_int64* pCursor)
{
MA_ASSERT(pDecoder != NULL);
if (pDecoder->onTell == NULL) {
return MA_NOT_IMPLEMENTED;
}
return pDecoder->onTell(pDecoder, pCursor);
}
MA_API ma_decoding_backend_config ma_decoding_backend_config_init(ma_format preferredFormat, ma_uint32 seekPointCount)
{
ma_decoding_backend_config config;
MA_ZERO_OBJECT(&config);
config.preferredFormat = preferredFormat;
config.seekPointCount = seekPointCount;
return config;
}
MA_API ma_decoder_config ma_decoder_config_init(ma_format outputFormat, ma_uint32 outputChannels, ma_uint32 outputSampleRate)
{
ma_decoder_config config;
MA_ZERO_OBJECT(&config);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
ma_channel internalChannelMap[MA_MAX_CHANNELS];
MA_ASSERT(pDecoder != NULL);
MA_ASSERT(pConfig != NULL);
result = ma_data_source_get_data_format(pDecoder->pBackend, &internalFormat, &internalChannels, &internalSampleRate, internalChannelMap, ma_countof(internalChannelMap));
if (result != MA_SUCCESS) {
return result; /* Failed to retrieve the internal data format. */
}
/* Make sure we're not asking for too many channels. */
if (pConfig->channels > MA_MAX_CHANNELS) {
return MA_INVALID_ARGS;
}
/* The internal channels should have already been validated at a higher level, but we'll do it again explicitly here for safety. */
if (internalChannels > MA_MAX_CHANNELS) {
return MA_INVALID_ARGS;
}
/* Output format. */
if (pConfig->format == ma_format_unknown) {
pDecoder->outputFormat = internalFormat;
} else {
pDecoder->outputFormat = pConfig->format;
}
if (pConfig->channels == 0) {
pDecoder->outputChannels = internalChannels;
} else {
pDecoder->outputChannels = pConfig->channels;
}
if (pConfig->sampleRate == 0) {
pDecoder->outputSampleRate = internalSampleRate;
} else {
pDecoder->outputSampleRate = pConfig->sampleRate;
}
converterConfig = ma_data_converter_config_init(
internalFormat, pDecoder->outputFormat,
internalChannels, pDecoder->outputChannels,
internalSampleRate, pDecoder->outputSampleRate
);
converterConfig.pChannelMapIn = internalChannelMap;
converterConfig.pChannelMapOut = pConfig->pChannelMap;
converterConfig.channelMixMode = pConfig->channelMixMode;
converterConfig.ditherMode = pConfig->ditherMode;
converterConfig.allowDynamicSampleRate = MA_FALSE; /* Never allow dynamic sample rate conversion. Setting this to true will disable passthrough optimizations. */
converterConfig.resampling = pConfig->resampling;
result = ma_data_converter_init(&converterConfig, &pDecoder->allocationCallbacks, &pDecoder->converter);
if (result != MA_SUCCESS) {
return result;
}
/*
Now that we have the decoder we need to determine whether or not we need a heap-allocated cache. We'll
need this if the data converter does not support calculation of the required input frame count. To
determine support for this we'll just run a test.
*/
{
ma_uint64 unused;
result = ma_data_converter_get_required_input_frame_count(&pDecoder->converter, 1, &unused);
if (result != MA_SUCCESS) {
/*
We were unable to calculate the required input frame count which means we'll need to use
a heap-allocated cache.
*/
ma_uint64 inputCacheCapSizeInBytes;
pDecoder->inputCacheCap = MA_DATA_CONVERTER_STACK_BUFFER_SIZE / ma_get_bytes_per_frame(internalFormat, internalChannels);
/* Not strictly necessary, but keeping here for safety in case we change the default value of pDecoder->inputCacheCap. */
inputCacheCapSizeInBytes = pDecoder->inputCacheCap * ma_get_bytes_per_frame(internalFormat, internalChannels);
if (inputCacheCapSizeInBytes > MA_SIZE_MAX) {
ma_data_converter_uninit(&pDecoder->converter, &pDecoder->allocationCallbacks);
return MA_OUT_OF_MEMORY;
}
pDecoder->pInputCache = ma_malloc((size_t)inputCacheCapSizeInBytes, &pDecoder->allocationCallbacks); /* Safe cast to size_t. */
if (pDecoder->pInputCache == NULL) {
ma_data_converter_uninit(&pDecoder->converter, &pDecoder->allocationCallbacks);
return MA_OUT_OF_MEMORY;
}
}
}
return MA_SUCCESS;
}
static ma_result ma_decoder_internal_on_read__custom(void* pUserData, void* pBufferOut, size_t bytesToRead, size_t* pBytesRead)
{
ma_decoder* pDecoder = (ma_decoder*)pUserData;
MA_ASSERT(pDecoder != NULL);
return ma_decoder_read_bytes(pDecoder, pBufferOut, bytesToRead, pBytesRead);
}
static ma_result ma_decoder_internal_on_seek__custom(void* pUserData, ma_int64 offset, ma_seek_origin origin)
{
ma_decoder* pDecoder = (ma_decoder*)pUserData;
MA_ASSERT(pDecoder != NULL);
return ma_decoder_seek_bytes(pDecoder, offset, origin);
}
static ma_result ma_decoder_internal_on_tell__custom(void* pUserData, ma_int64* pCursor)
{
ma_decoder* pDecoder = (ma_decoder*)pUserData;
MA_ASSERT(pDecoder != NULL);
return ma_decoder_tell_bytes(pDecoder, pCursor);
}
static ma_result ma_decoder_init_from_vtable(const ma_decoding_backend_vtable* pVTable, void* pVTableUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
ma_result result;
ma_decoding_backend_config backendConfig;
ma_data_source* pBackend;
MA_ASSERT(pVTable != NULL);
MA_ASSERT(pConfig != NULL);
MA_ASSERT(pDecoder != NULL);
if (pVTable->onInit == NULL) {
return MA_NOT_IMPLEMENTED;
}
backendConfig = ma_decoding_backend_config_init(pConfig->format, pConfig->seekPointCount);
result = pVTable->onInit(pVTableUserData, ma_decoder_internal_on_read__custom, ma_decoder_internal_on_seek__custom, ma_decoder_internal_on_tell__custom, pDecoder, &backendConfig, &pDecoder->allocationCallbacks, &pBackend);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
static ma_result ma_decoder_init_custom__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
ma_result result = MA_NO_BACKEND;
size_t ivtable;
MA_ASSERT(pConfig != NULL);
MA_ASSERT(pDecoder != NULL);
if (pConfig->ppCustomBackendVTables == NULL) {
return MA_NO_BACKEND;
}
/* The order each backend is listed is what defines the priority. */
for (ivtable = 0; ivtable < pConfig->customBackendCount; ivtable += 1) {
const ma_decoding_backend_vtable* pVTable = pConfig->ppCustomBackendVTables[ivtable];
if (pVTable != NULL && pVTable->onInit != NULL) {
result = ma_decoder_init_from_vtable(pVTable, pConfig->pCustomBackendUserData, pConfig, pDecoder);
if (result == MA_SUCCESS) {
return MA_SUCCESS;
} else {
/* Initialization failed. Move on to the next one, but seek back to the start first so the next vtable starts from the first byte of the file. */
result = ma_decoder_seek_bytes(pDecoder, 0, ma_seek_origin_start);
if (result != MA_SUCCESS) {
return result; /* Failed to seek back to the start. */
}
}
} else {
/* No vtable. */
}
}
/* Getting here means we couldn't find a backend. */
return MA_NO_BACKEND;
}
/* WAV */
#ifdef dr_wav_h
#define MA_HAS_WAV
typedef struct
{
ma_data_source_base ds;
ma_read_proc onRead;
ma_seek_proc onSeek;
ma_tell_proc onTell;
void* pReadSeekTellUserData;
ma_format format; /* Can be f32, s16 or s32. */
#if !defined(MA_NO_WAV)
drwav dr;
#endif
} ma_wav;
MA_API ma_result ma_wav_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav);
MA_API ma_result ma_wav_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav);
MA_API ma_result ma_wav_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav);
MA_API ma_result ma_wav_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav);
MA_API void ma_wav_uninit(ma_wav* pWav, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_wav_read_pcm_frames(ma_wav* pWav, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
MA_API ma_result ma_wav_seek_to_pcm_frame(ma_wav* pWav, ma_uint64 frameIndex);
MA_API ma_result ma_wav_get_data_format(ma_wav* pWav, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
MA_API ma_result ma_wav_get_cursor_in_pcm_frames(ma_wav* pWav, ma_uint64* pCursor);
MA_API ma_result ma_wav_get_length_in_pcm_frames(ma_wav* pWav, ma_uint64* pLength);
static ma_result ma_wav_ds_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
return ma_wav_read_pcm_frames((ma_wav*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
static ma_result ma_wav_ds_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
return ma_wav_seek_to_pcm_frame((ma_wav*)pDataSource, frameIndex);
}
static ma_result ma_wav_ds_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
return ma_wav_get_data_format((ma_wav*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
static ma_result ma_wav_ds_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
return ma_wav_get_cursor_in_pcm_frames((ma_wav*)pDataSource, pCursor);
}
static ma_result ma_wav_ds_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
{
return ma_wav_get_length_in_pcm_frames((ma_wav*)pDataSource, pLength);
}
static ma_data_source_vtable g_ma_wav_ds_vtable =
{
ma_wav_ds_read,
ma_wav_ds_seek,
ma_wav_ds_get_data_format,
ma_wav_ds_get_cursor,
ma_wav_ds_get_length,
NULL, /* onSetLooping */
0
};
#if !defined(MA_NO_WAV)
static drwav_allocation_callbacks drwav_allocation_callbacks_from_miniaudio(const ma_allocation_callbacks* pAllocationCallbacks)
{
drwav_allocation_callbacks callbacks;
if (pAllocationCallbacks != NULL) {
callbacks.onMalloc = pAllocationCallbacks->onMalloc;
callbacks.onRealloc = pAllocationCallbacks->onRealloc;
callbacks.onFree = pAllocationCallbacks->onFree;
callbacks.pUserData = pAllocationCallbacks->pUserData;
} else {
callbacks.onMalloc = ma__malloc_default;
callbacks.onRealloc = ma__realloc_default;
callbacks.onFree = ma__free_default;
callbacks.pUserData = NULL;
}
return callbacks;
}
static size_t ma_wav_dr_callback__read(void* pUserData, void* pBufferOut, size_t bytesToRead)
{
ma_wav* pWav = (ma_wav*)pUserData;
ma_result result;
size_t bytesRead;
MA_ASSERT(pWav != NULL);
result = pWav->onRead(pWav->pReadSeekTellUserData, pBufferOut, bytesToRead, &bytesRead);
(void)result;
return bytesRead;
}
static drwav_bool32 ma_wav_dr_callback__seek(void* pUserData, int offset, drwav_seek_origin origin)
{
ma_wav* pWav = (ma_wav*)pUserData;
ma_result result;
ma_seek_origin maSeekOrigin;
MA_ASSERT(pWav != NULL);
maSeekOrigin = ma_seek_origin_start;
if (origin == drwav_seek_origin_current) {
maSeekOrigin = ma_seek_origin_current;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
(void)pAllocationCallbacks;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_wav_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_wav* pWav)
{
ma_result result;
result = ma_wav_init_internal(pConfig, pWav);
if (result != MA_SUCCESS) {
return result;
}
#if !defined(MA_NO_WAV)
{
drwav_allocation_callbacks wavAllocationCallbacks = drwav_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
drwav_bool32 wavResult;
wavResult = drwav_init_memory(&pWav->dr, pData, dataSize, &wavAllocationCallbacks);
if (wavResult != MA_TRUE) {
return MA_INVALID_FILE;
}
return MA_SUCCESS;
}
#else
{
/* wav is disabled. */
(void)pData;
(void)dataSize;
(void)pAllocationCallbacks;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API void ma_wav_uninit(ma_wav* pWav, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pWav == NULL) {
return;
}
(void)pAllocationCallbacks;
#if !defined(MA_NO_WAV)
{
drwav_uninit(&pWav->dr);
}
#else
{
/* wav is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
}
#endif
ma_data_source_uninit(&pWav->ds);
}
MA_API ma_result ma_wav_read_pcm_frames(ma_wav* pWav, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
if (frameCount == 0) {
return MA_INVALID_ARGS;
}
if (pWav == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_WAV)
{
/* We always use floating point format. */
ma_result result = MA_SUCCESS; /* Must be initialized to MA_SUCCESS. */
ma_uint64 totalFramesRead = 0;
ma_format format;
ma_wav_get_data_format(pWav, &format, NULL, NULL, NULL, 0);
switch (format)
{
case ma_format_f32:
{
totalFramesRead = drwav_read_pcm_frames_f32(&pWav->dr, frameCount, (float*)pFramesOut);
} break;
case ma_format_s16:
{
totalFramesRead = drwav_read_pcm_frames_s16(&pWav->dr, frameCount, (drwav_int16*)pFramesOut);
} break;
case ma_format_s32:
{
totalFramesRead = drwav_read_pcm_frames_s32(&pWav->dr, frameCount, (drwav_int32*)pFramesOut);
} break;
/* Fallback to a raw read. */
case ma_format_unknown: return MA_INVALID_OPERATION; /* <-- this should never be hit because initialization would just fall back to a supported format. */
default:
{
totalFramesRead = drwav_read_pcm_frames(&pWav->dr, frameCount, pFramesOut);
} break;
}
/* In the future we'll update dr_wav to return MA_AT_END for us. */
if (totalFramesRead == 0) {
result = MA_AT_END;
}
if (pFramesRead != NULL) {
*pFramesRead = totalFramesRead;
}
if (result == MA_SUCCESS && totalFramesRead == 0) {
result = MA_AT_END;
}
return result;
}
#else
{
/* wav is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
(void)pFramesOut;
(void)frameCount;
(void)pFramesRead;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_wav_seek_to_pcm_frame(ma_wav* pWav, ma_uint64 frameIndex)
{
if (pWav == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_WAV)
{
drwav_bool32 wavResult;
wavResult = drwav_seek_to_pcm_frame(&pWav->dr, frameIndex);
if (wavResult != DRWAV_TRUE) {
return MA_ERROR;
}
return MA_SUCCESS;
}
#else
{
/* wav is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
(void)frameIndex;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_wav_get_data_format(ma_wav* pWav, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
/* Defaults for safety. */
if (pFormat != NULL) {
*pFormat = ma_format_unknown;
}
if (pChannels != NULL) {
*pChannels = 0;
}
if (pSampleRate != NULL) {
*pSampleRate = 0;
}
if (pChannelMap != NULL) {
MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
}
if (pWav == NULL) {
return MA_INVALID_OPERATION;
}
if (pFormat != NULL) {
*pFormat = pWav->format;
}
#if !defined(MA_NO_WAV)
{
if (pChannels != NULL) {
*pChannels = pWav->dr.channels;
}
if (pSampleRate != NULL) {
*pSampleRate = pWav->dr.sampleRate;
}
if (pChannelMap != NULL) {
ma_channel_map_init_standard(ma_standard_channel_map_microsoft, pChannelMap, channelMapCap, pWav->dr.channels);
}
return MA_SUCCESS;
}
#else
{
/* wav is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_wav_get_cursor_in_pcm_frames(ma_wav* pWav, ma_uint64* pCursor)
{
if (pCursor == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = 0; /* Safety. */
if (pWav == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_WAV)
{
drwav_result wavResult = drwav_get_cursor_in_pcm_frames(&pWav->dr, pCursor);
if (wavResult != DRWAV_SUCCESS) {
return (ma_result)wavResult; /* dr_wav result codes map to miniaudio's. */
}
return MA_SUCCESS;
}
#else
{
/* wav is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_wav_get_length_in_pcm_frames(ma_wav* pWav, ma_uint64* pLength)
{
if (pLength == NULL) {
return MA_INVALID_ARGS;
}
*pLength = 0; /* Safety. */
if (pWav == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_WAV)
{
drwav_result wavResult = drwav_get_length_in_pcm_frames(&pWav->dr, pLength);
if (wavResult != DRWAV_SUCCESS) {
return (ma_result)wavResult; /* dr_wav result codes map to miniaudio's. */
}
return MA_SUCCESS;
}
#else
{
/* wav is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
return MA_NOT_IMPLEMENTED;
}
#endif
}
static ma_result ma_decoding_backend_init__wav(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks...
{
ma_result result;
ma_wav* pWav;
(void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
/* For now we're just allocating the decoder backend on the heap. */
pWav = (ma_wav*)ma_malloc(sizeof(*pWav), pAllocationCallbacks);
if (pWav == NULL) {
return MA_OUT_OF_MEMORY;
}
result = ma_wav_init(onRead, onSeek, onTell, pReadSeekTellUserData, pConfig, pAllocationCallbacks, pWav);
if (result != MA_SUCCESS) {
ma_free(pWav, pAllocationCallbacks);
return result;
}
*ppBackend = pWav;
return MA_SUCCESS;
}
static ma_result ma_decoding_backend_init_file__wav(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
{
ma_result result;
ma_wav* pWav;
(void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
/* For now we're just allocating the decoder backend on the heap. */
pWav = (ma_wav*)ma_malloc(sizeof(*pWav), pAllocationCallbacks);
if (pWav == NULL) {
return MA_OUT_OF_MEMORY;
}
result = ma_wav_init_file(pFilePath, pConfig, pAllocationCallbacks, pWav);
if (result != MA_SUCCESS) {
ma_free(pWav, pAllocationCallbacks);
return result;
}
*ppBackend = pWav;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
result = ma_wav_init_memory(pData, dataSize, pConfig, pAllocationCallbacks, pWav);
if (result != MA_SUCCESS) {
ma_free(pWav, pAllocationCallbacks);
return result;
}
*ppBackend = pWav;
return MA_SUCCESS;
}
static void ma_decoding_backend_uninit__wav(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_wav* pWav = (ma_wav*)pBackend;
(void)pUserData;
ma_wav_uninit(pWav, pAllocationCallbacks);
ma_free(pWav, pAllocationCallbacks);
}
static ma_decoding_backend_vtable g_ma_decoding_backend_vtable_wav =
{
ma_decoding_backend_init__wav,
ma_decoding_backend_init_file__wav,
ma_decoding_backend_init_file_w__wav,
ma_decoding_backend_init_memory__wav,
ma_decoding_backend_uninit__wav
};
static ma_result ma_decoder_init_wav__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
return ma_decoder_init_from_vtable(&g_ma_decoding_backend_vtable_wav, NULL, pConfig, pDecoder);
}
#endif /* dr_wav_h */
/* FLAC */
#ifdef dr_flac_h
#define MA_HAS_FLAC
typedef struct
{
ma_data_source_base ds;
ma_read_proc onRead;
ma_seek_proc onSeek;
ma_tell_proc onTell;
void* pReadSeekTellUserData;
ma_format format; /* Can be f32, s16 or s32. */
#if !defined(MA_NO_FLAC)
drflac* dr;
#endif
} ma_flac;
MA_API ma_result ma_flac_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac);
MA_API ma_result ma_flac_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac);
MA_API ma_result ma_flac_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac);
MA_API ma_result ma_flac_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac);
MA_API void ma_flac_uninit(ma_flac* pFlac, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_flac_read_pcm_frames(ma_flac* pFlac, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
MA_API ma_result ma_flac_seek_to_pcm_frame(ma_flac* pFlac, ma_uint64 frameIndex);
MA_API ma_result ma_flac_get_data_format(ma_flac* pFlac, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
MA_API ma_result ma_flac_get_cursor_in_pcm_frames(ma_flac* pFlac, ma_uint64* pCursor);
MA_API ma_result ma_flac_get_length_in_pcm_frames(ma_flac* pFlac, ma_uint64* pLength);
static ma_result ma_flac_ds_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
return ma_flac_read_pcm_frames((ma_flac*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
static ma_result ma_flac_ds_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
return ma_flac_seek_to_pcm_frame((ma_flac*)pDataSource, frameIndex);
}
static ma_result ma_flac_ds_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
return ma_flac_get_data_format((ma_flac*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
static ma_result ma_flac_ds_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
return ma_flac_get_cursor_in_pcm_frames((ma_flac*)pDataSource, pCursor);
}
static ma_result ma_flac_ds_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
{
return ma_flac_get_length_in_pcm_frames((ma_flac*)pDataSource, pLength);
}
static ma_data_source_vtable g_ma_flac_ds_vtable =
{
ma_flac_ds_read,
ma_flac_ds_seek,
ma_flac_ds_get_data_format,
ma_flac_ds_get_cursor,
ma_flac_ds_get_length,
NULL, /* onSetLooping */
0
};
#if !defined(MA_NO_FLAC)
static drflac_allocation_callbacks drflac_allocation_callbacks_from_miniaudio(const ma_allocation_callbacks* pAllocationCallbacks)
{
drflac_allocation_callbacks callbacks;
if (pAllocationCallbacks != NULL) {
callbacks.onMalloc = pAllocationCallbacks->onMalloc;
callbacks.onRealloc = pAllocationCallbacks->onRealloc;
callbacks.onFree = pAllocationCallbacks->onFree;
callbacks.pUserData = pAllocationCallbacks->pUserData;
} else {
callbacks.onMalloc = ma__malloc_default;
callbacks.onRealloc = ma__realloc_default;
callbacks.onFree = ma__free_default;
callbacks.pUserData = NULL;
}
return callbacks;
}
static size_t ma_flac_dr_callback__read(void* pUserData, void* pBufferOut, size_t bytesToRead)
{
ma_flac* pFlac = (ma_flac*)pUserData;
ma_result result;
size_t bytesRead;
MA_ASSERT(pFlac != NULL);
result = pFlac->onRead(pFlac->pReadSeekTellUserData, pBufferOut, bytesToRead, &bytesRead);
(void)result;
return bytesRead;
}
static drflac_bool32 ma_flac_dr_callback__seek(void* pUserData, int offset, drflac_seek_origin origin)
{
ma_flac* pFlac = (ma_flac*)pUserData;
ma_result result;
ma_seek_origin maSeekOrigin;
MA_ASSERT(pFlac != NULL);
maSeekOrigin = ma_seek_origin_start;
if (origin == drflac_seek_origin_current) {
maSeekOrigin = ma_seek_origin_current;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
(void)pFilePath;
(void)pAllocationCallbacks;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_flac_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_flac* pFlac)
{
ma_result result;
result = ma_flac_init_internal(pConfig, pFlac);
if (result != MA_SUCCESS) {
return result;
}
#if !defined(MA_NO_FLAC)
{
drflac_allocation_callbacks flacAllocationCallbacks = drflac_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
pFlac->dr = drflac_open_memory(pData, dataSize, &flacAllocationCallbacks);
if (pFlac->dr == NULL) {
return MA_INVALID_FILE;
}
return MA_SUCCESS;
}
#else
{
/* flac is disabled. */
(void)pData;
(void)dataSize;
(void)pAllocationCallbacks;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API void ma_flac_uninit(ma_flac* pFlac, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pFlac == NULL) {
return;
}
(void)pAllocationCallbacks;
#if !defined(MA_NO_FLAC)
{
drflac_close(pFlac->dr);
}
#else
{
/* flac is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
}
#endif
ma_data_source_uninit(&pFlac->ds);
}
MA_API ma_result ma_flac_read_pcm_frames(ma_flac* pFlac, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
if (frameCount == 0) {
return MA_INVALID_ARGS;
}
if (pFlac == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_FLAC)
{
/* We always use floating point format. */
ma_result result = MA_SUCCESS; /* Must be initialized to MA_SUCCESS. */
ma_uint64 totalFramesRead = 0;
ma_format format;
ma_flac_get_data_format(pFlac, &format, NULL, NULL, NULL, 0);
switch (format)
{
case ma_format_f32:
{
totalFramesRead = drflac_read_pcm_frames_f32(pFlac->dr, frameCount, (float*)pFramesOut);
} break;
case ma_format_s16:
{
totalFramesRead = drflac_read_pcm_frames_s16(pFlac->dr, frameCount, (drflac_int16*)pFramesOut);
} break;
case ma_format_s32:
{
totalFramesRead = drflac_read_pcm_frames_s32(pFlac->dr, frameCount, (drflac_int32*)pFramesOut);
} break;
case ma_format_u8:
case ma_format_s24:
case ma_format_unknown:
default:
{
return MA_INVALID_OPERATION;
};
}
/* In the future we'll update dr_flac to return MA_AT_END for us. */
if (totalFramesRead == 0) {
result = MA_AT_END;
}
if (pFramesRead != NULL) {
*pFramesRead = totalFramesRead;
}
if (result == MA_SUCCESS && totalFramesRead == 0) {
result = MA_AT_END;
}
return result;
}
#else
{
/* flac is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
(void)pFramesOut;
(void)frameCount;
(void)pFramesRead;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_flac_seek_to_pcm_frame(ma_flac* pFlac, ma_uint64 frameIndex)
{
if (pFlac == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_FLAC)
{
drflac_bool32 flacResult;
flacResult = drflac_seek_to_pcm_frame(pFlac->dr, frameIndex);
if (flacResult != DRFLAC_TRUE) {
return MA_ERROR;
}
return MA_SUCCESS;
}
#else
{
/* flac is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
(void)frameIndex;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_flac_get_data_format(ma_flac* pFlac, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
/* Defaults for safety. */
if (pFormat != NULL) {
*pFormat = ma_format_unknown;
}
if (pChannels != NULL) {
*pChannels = 0;
}
if (pSampleRate != NULL) {
*pSampleRate = 0;
}
if (pChannelMap != NULL) {
MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
}
if (pFlac == NULL) {
return MA_INVALID_OPERATION;
}
if (pFormat != NULL) {
*pFormat = pFlac->format;
}
#if !defined(MA_NO_FLAC)
{
if (pChannels != NULL) {
*pChannels = pFlac->dr->channels;
}
if (pSampleRate != NULL) {
*pSampleRate = pFlac->dr->sampleRate;
}
if (pChannelMap != NULL) {
ma_channel_map_init_standard(ma_standard_channel_map_microsoft, pChannelMap, channelMapCap, pFlac->dr->channels);
}
return MA_SUCCESS;
}
#else
{
/* flac is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_flac_get_cursor_in_pcm_frames(ma_flac* pFlac, ma_uint64* pCursor)
{
if (pCursor == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = 0; /* Safety. */
if (pFlac == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_FLAC)
{
*pCursor = pFlac->dr->currentPCMFrame;
return MA_SUCCESS;
}
#else
{
/* flac is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_flac_get_length_in_pcm_frames(ma_flac* pFlac, ma_uint64* pLength)
{
if (pLength == NULL) {
return MA_INVALID_ARGS;
}
*pLength = 0; /* Safety. */
if (pFlac == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_FLAC)
{
*pLength = pFlac->dr->totalPCMFrameCount;
return MA_SUCCESS;
}
#else
{
/* flac is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
return MA_NOT_IMPLEMENTED;
}
#endif
}
static ma_result ma_decoding_backend_init__flac(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallback...
{
ma_result result;
ma_flac* pFlac;
(void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
/* For now we're just allocating the decoder backend on the heap. */
pFlac = (ma_flac*)ma_malloc(sizeof(*pFlac), pAllocationCallbacks);
if (pFlac == NULL) {
return MA_OUT_OF_MEMORY;
}
result = ma_flac_init(onRead, onSeek, onTell, pReadSeekTellUserData, pConfig, pAllocationCallbacks, pFlac);
if (result != MA_SUCCESS) {
ma_free(pFlac, pAllocationCallbacks);
return result;
}
*ppBackend = pFlac;
return MA_SUCCESS;
}
static ma_result ma_decoding_backend_init_file__flac(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
{
ma_result result;
ma_flac* pFlac;
(void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
/* For now we're just allocating the decoder backend on the heap. */
pFlac = (ma_flac*)ma_malloc(sizeof(*pFlac), pAllocationCallbacks);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
result = ma_flac_init_memory(pData, dataSize, pConfig, pAllocationCallbacks, pFlac);
if (result != MA_SUCCESS) {
ma_free(pFlac, pAllocationCallbacks);
return result;
}
*ppBackend = pFlac;
return MA_SUCCESS;
}
static void ma_decoding_backend_uninit__flac(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_flac* pFlac = (ma_flac*)pBackend;
(void)pUserData;
ma_flac_uninit(pFlac, pAllocationCallbacks);
ma_free(pFlac, pAllocationCallbacks);
}
static ma_decoding_backend_vtable g_ma_decoding_backend_vtable_flac =
{
ma_decoding_backend_init__flac,
ma_decoding_backend_init_file__flac,
ma_decoding_backend_init_file_w__flac,
ma_decoding_backend_init_memory__flac,
ma_decoding_backend_uninit__flac
};
static ma_result ma_decoder_init_flac__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
return ma_decoder_init_from_vtable(&g_ma_decoding_backend_vtable_flac, NULL, pConfig, pDecoder);
}
#endif /* dr_flac_h */
/* MP3 */
#ifdef dr_mp3_h
#define MA_HAS_MP3
typedef struct
{
ma_data_source_base ds;
ma_read_proc onRead;
ma_seek_proc onSeek;
ma_tell_proc onTell;
void* pReadSeekTellUserData;
ma_format format; /* Can be f32 or s16. */
#if !defined(MA_NO_MP3)
drmp3 dr;
drmp3_uint32 seekPointCount;
drmp3_seek_point* pSeekPoints; /* Only used if seek table generation is used. */
#endif
} ma_mp3;
MA_API ma_result ma_mp3_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3);
MA_API ma_result ma_mp3_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3);
MA_API ma_result ma_mp3_init_file_w(const wchar_t* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3);
MA_API ma_result ma_mp3_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3);
MA_API void ma_mp3_uninit(ma_mp3* pMP3, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_mp3_read_pcm_frames(ma_mp3* pMP3, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
MA_API ma_result ma_mp3_seek_to_pcm_frame(ma_mp3* pMP3, ma_uint64 frameIndex);
MA_API ma_result ma_mp3_get_data_format(ma_mp3* pMP3, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
MA_API ma_result ma_mp3_get_cursor_in_pcm_frames(ma_mp3* pMP3, ma_uint64* pCursor);
MA_API ma_result ma_mp3_get_length_in_pcm_frames(ma_mp3* pMP3, ma_uint64* pLength);
static ma_result ma_mp3_ds_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
return ma_mp3_read_pcm_frames((ma_mp3*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
static ma_result ma_mp3_ds_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
return ma_mp3_seek_to_pcm_frame((ma_mp3*)pDataSource, frameIndex);
}
static ma_result ma_mp3_ds_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
return ma_mp3_get_data_format((ma_mp3*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
static ma_result ma_mp3_ds_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
return ma_mp3_get_cursor_in_pcm_frames((ma_mp3*)pDataSource, pCursor);
}
static ma_result ma_mp3_ds_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
{
return ma_mp3_get_length_in_pcm_frames((ma_mp3*)pDataSource, pLength);
}
static ma_data_source_vtable g_ma_mp3_ds_vtable =
{
ma_mp3_ds_read,
ma_mp3_ds_seek,
ma_mp3_ds_get_data_format,
ma_mp3_ds_get_cursor,
ma_mp3_ds_get_length,
NULL, /* onSetLooping */
0
};
#if !defined(MA_NO_MP3)
static drmp3_allocation_callbacks drmp3_allocation_callbacks_from_miniaudio(const ma_allocation_callbacks* pAllocationCallbacks)
{
drmp3_allocation_callbacks callbacks;
if (pAllocationCallbacks != NULL) {
callbacks.onMalloc = pAllocationCallbacks->onMalloc;
callbacks.onRealloc = pAllocationCallbacks->onRealloc;
callbacks.onFree = pAllocationCallbacks->onFree;
callbacks.pUserData = pAllocationCallbacks->pUserData;
} else {
callbacks.onMalloc = ma__malloc_default;
callbacks.onRealloc = ma__realloc_default;
callbacks.onFree = ma__free_default;
callbacks.pUserData = NULL;
}
return callbacks;
}
static size_t ma_mp3_dr_callback__read(void* pUserData, void* pBufferOut, size_t bytesToRead)
{
ma_mp3* pMP3 = (ma_mp3*)pUserData;
ma_result result;
size_t bytesRead;
MA_ASSERT(pMP3 != NULL);
result = pMP3->onRead(pMP3->pReadSeekTellUserData, pBufferOut, bytesToRead, &bytesRead);
(void)result;
return bytesRead;
}
static drmp3_bool32 ma_mp3_dr_callback__seek(void* pUserData, int offset, drmp3_seek_origin origin)
{
ma_mp3* pMP3 = (ma_mp3*)pUserData;
ma_result result;
ma_seek_origin maSeekOrigin;
MA_ASSERT(pMP3 != NULL);
maSeekOrigin = ma_seek_origin_start;
if (origin == drmp3_seek_origin_current) {
maSeekOrigin = ma_seek_origin_current;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
#endif
}
MA_API ma_result ma_mp3_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_mp3* pMP3)
{
ma_result result;
result = ma_mp3_init_internal(pConfig, pMP3);
if (result != MA_SUCCESS) {
return result;
}
#if !defined(MA_NO_MP3)
{
drmp3_allocation_callbacks mp3AllocationCallbacks = drmp3_allocation_callbacks_from_miniaudio(pAllocationCallbacks);
drmp3_bool32 mp3Result;
mp3Result = drmp3_init_memory(&pMP3->dr, pData, dataSize, &mp3AllocationCallbacks);
if (mp3Result != MA_TRUE) {
return MA_INVALID_FILE;
}
ma_mp3_generate_seek_table(pMP3, pConfig, pAllocationCallbacks);
return MA_SUCCESS;
}
#else
{
/* mp3 is disabled. */
(void)pData;
(void)dataSize;
(void)pAllocationCallbacks;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API void ma_mp3_uninit(ma_mp3* pMP3, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pMP3 == NULL) {
return;
}
#if !defined(MA_NO_MP3)
{
drmp3_uninit(&pMP3->dr);
}
#else
{
/* mp3 is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
}
#endif
/* Seek points need to be freed after the MP3 decoder has been uninitialized to ensure they're no longer being referenced. */
ma_free(pMP3->pSeekPoints, pAllocationCallbacks);
ma_data_source_uninit(&pMP3->ds);
}
MA_API ma_result ma_mp3_read_pcm_frames(ma_mp3* pMP3, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
if (frameCount == 0) {
return MA_INVALID_ARGS;
}
if (pMP3 == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_MP3)
{
/* We always use floating point format. */
ma_result result = MA_SUCCESS; /* Must be initialized to MA_SUCCESS. */
ma_uint64 totalFramesRead = 0;
ma_format format;
ma_mp3_get_data_format(pMP3, &format, NULL, NULL, NULL, 0);
switch (format)
{
case ma_format_f32:
{
totalFramesRead = drmp3_read_pcm_frames_f32(&pMP3->dr, frameCount, (float*)pFramesOut);
} break;
case ma_format_s16:
{
totalFramesRead = drmp3_read_pcm_frames_s16(&pMP3->dr, frameCount, (drmp3_int16*)pFramesOut);
} break;
case ma_format_u8:
case ma_format_s24:
case ma_format_s32:
case ma_format_unknown:
default:
{
return MA_INVALID_OPERATION;
};
}
/* In the future we'll update dr_mp3 to return MA_AT_END for us. */
if (totalFramesRead == 0) {
result = MA_AT_END;
}
if (pFramesRead != NULL) {
*pFramesRead = totalFramesRead;
}
return result;
}
#else
{
/* mp3 is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
(void)pFramesOut;
(void)frameCount;
(void)pFramesRead;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_mp3_seek_to_pcm_frame(ma_mp3* pMP3, ma_uint64 frameIndex)
{
if (pMP3 == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_MP3)
{
drmp3_bool32 mp3Result;
mp3Result = drmp3_seek_to_pcm_frame(&pMP3->dr, frameIndex);
if (mp3Result != DRMP3_TRUE) {
return MA_ERROR;
}
return MA_SUCCESS;
}
#else
{
/* mp3 is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
(void)frameIndex;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_mp3_get_data_format(ma_mp3* pMP3, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
/* Defaults for safety. */
if (pFormat != NULL) {
*pFormat = ma_format_unknown;
}
if (pChannels != NULL) {
*pChannels = 0;
}
if (pSampleRate != NULL) {
*pSampleRate = 0;
}
if (pChannelMap != NULL) {
MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
}
if (pMP3 == NULL) {
return MA_INVALID_OPERATION;
}
if (pFormat != NULL) {
*pFormat = pMP3->format;
}
#if !defined(MA_NO_MP3)
{
if (pChannels != NULL) {
*pChannels = pMP3->dr.channels;
}
if (pSampleRate != NULL) {
*pSampleRate = pMP3->dr.sampleRate;
}
if (pChannelMap != NULL) {
ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pMP3->dr.channels);
}
return MA_SUCCESS;
}
#else
{
/* mp3 is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_mp3_get_cursor_in_pcm_frames(ma_mp3* pMP3, ma_uint64* pCursor)
{
if (pCursor == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = 0; /* Safety. */
if (pMP3 == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_MP3)
{
*pCursor = pMP3->dr.currentPCMFrame;
return MA_SUCCESS;
}
#else
{
/* mp3 is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_mp3_get_length_in_pcm_frames(ma_mp3* pMP3, ma_uint64* pLength)
{
if (pLength == NULL) {
return MA_INVALID_ARGS;
}
*pLength = 0; /* Safety. */
if (pMP3 == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_MP3)
{
*pLength = drmp3_get_pcm_frame_count(&pMP3->dr);
return MA_SUCCESS;
}
#else
{
/* mp3 is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
return MA_NOT_IMPLEMENTED;
}
#endif
}
static ma_result ma_decoding_backend_init__mp3(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks...
{
ma_result result;
ma_mp3* pMP3;
(void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
/* For now we're just allocating the decoder backend on the heap. */
pMP3 = (ma_mp3*)ma_malloc(sizeof(*pMP3), pAllocationCallbacks);
if (pMP3 == NULL) {
return MA_OUT_OF_MEMORY;
}
result = ma_mp3_init(onRead, onSeek, onTell, pReadSeekTellUserData, pConfig, pAllocationCallbacks, pMP3);
if (result != MA_SUCCESS) {
ma_free(pMP3, pAllocationCallbacks);
return result;
}
*ppBackend = pMP3;
return MA_SUCCESS;
}
static ma_result ma_decoding_backend_init_file__mp3(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
{
ma_result result;
ma_mp3* pMP3;
(void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
/* For now we're just allocating the decoder backend on the heap. */
pMP3 = (ma_mp3*)ma_malloc(sizeof(*pMP3), pAllocationCallbacks);
if (pMP3 == NULL) {
return MA_OUT_OF_MEMORY;
}
result = ma_mp3_init_file(pFilePath, pConfig, pAllocationCallbacks, pMP3);
if (result != MA_SUCCESS) {
ma_free(pMP3, pAllocationCallbacks);
return result;
}
*ppBackend = pMP3;
return MA_SUCCESS;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
ma_mp3* pMP3;
(void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
/* For now we're just allocating the decoder backend on the heap. */
pMP3 = (ma_mp3*)ma_malloc(sizeof(*pMP3), pAllocationCallbacks);
if (pMP3 == NULL) {
return MA_OUT_OF_MEMORY;
}
result = ma_mp3_init_memory(pData, dataSize, pConfig, pAllocationCallbacks, pMP3);
if (result != MA_SUCCESS) {
ma_free(pMP3, pAllocationCallbacks);
return result;
}
*ppBackend = pMP3;
return MA_SUCCESS;
}
static void ma_decoding_backend_uninit__mp3(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_mp3* pMP3 = (ma_mp3*)pBackend;
(void)pUserData;
ma_mp3_uninit(pMP3, pAllocationCallbacks);
ma_free(pMP3, pAllocationCallbacks);
}
static ma_decoding_backend_vtable g_ma_decoding_backend_vtable_mp3 =
{
ma_decoding_backend_init__mp3,
ma_decoding_backend_init_file__mp3,
ma_decoding_backend_init_file_w__mp3,
ma_decoding_backend_init_memory__mp3,
ma_decoding_backend_uninit__mp3
};
static ma_result ma_decoder_init_mp3__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
return ma_decoder_init_from_vtable(&g_ma_decoding_backend_vtable_mp3, NULL, pConfig, pDecoder);
}
#endif /* dr_mp3_h */
/* Vorbis */
#ifdef STB_VORBIS_INCLUDE_STB_VORBIS_H
#define MA_HAS_VORBIS
/* The size in bytes of each chunk of data to read from the Vorbis stream. */
#define MA_VORBIS_DATA_CHUNK_SIZE 4096
typedef struct
{
ma_data_source_base ds;
ma_read_proc onRead;
ma_seek_proc onSeek;
ma_tell_proc onTell;
void* pReadSeekTellUserData;
ma_allocation_callbacks allocationCallbacks; /* Store the allocation callbacks within the structure because we may need to dynamically expand a buffer in ma_stbvorbis_read_pcm_frames() when using push mode. */
ma_format format; /* Only f32 is allowed with stb_vorbis. */
ma_uint32 channels;
ma_uint32 sampleRate;
ma_uint64 cursor;
#if !defined(MA_NO_VORBIS)
stb_vorbis* stb;
ma_bool32 usingPushMode;
struct
{
ma_uint8* pData;
size_t dataSize;
size_t dataCapacity;
ma_uint32 framesConsumed; /* The number of frames consumed in ppPacketData. */
ma_uint32 framesRemaining; /* The number of frames remaining in ppPacketData. */
float** ppPacketData;
} push;
#endif
} ma_stbvorbis;
MA_API ma_result ma_stbvorbis_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis);
MA_API ma_result ma_stbvorbis_init_file(const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis);
MA_API ma_result ma_stbvorbis_init_memory(const void* pData, size_t dataSize, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis);
MA_API void ma_stbvorbis_uninit(ma_stbvorbis* pVorbis, const ma_allocation_callbacks* pAllocationCallbacks);
MA_API ma_result ma_stbvorbis_read_pcm_frames(ma_stbvorbis* pVorbis, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead);
MA_API ma_result ma_stbvorbis_seek_to_pcm_frame(ma_stbvorbis* pVorbis, ma_uint64 frameIndex);
MA_API ma_result ma_stbvorbis_get_data_format(ma_stbvorbis* pVorbis, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap);
MA_API ma_result ma_stbvorbis_get_cursor_in_pcm_frames(ma_stbvorbis* pVorbis, ma_uint64* pCursor);
MA_API ma_result ma_stbvorbis_get_length_in_pcm_frames(ma_stbvorbis* pVorbis, ma_uint64* pLength);
static ma_result ma_stbvorbis_ds_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
return ma_stbvorbis_read_pcm_frames((ma_stbvorbis*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
static ma_result ma_stbvorbis_ds_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
return ma_stbvorbis_seek_to_pcm_frame((ma_stbvorbis*)pDataSource, frameIndex);
}
static ma_result ma_stbvorbis_ds_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
return ma_stbvorbis_get_data_format((ma_stbvorbis*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
static ma_result ma_stbvorbis_ds_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
return ma_stbvorbis_get_cursor_in_pcm_frames((ma_stbvorbis*)pDataSource, pCursor);
}
static ma_result ma_stbvorbis_ds_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
{
return ma_stbvorbis_get_length_in_pcm_frames((ma_stbvorbis*)pDataSource, pLength);
}
static ma_data_source_vtable g_ma_stbvorbis_ds_vtable =
{
ma_stbvorbis_ds_read,
ma_stbvorbis_ds_seek,
ma_stbvorbis_ds_get_data_format,
ma_stbvorbis_ds_get_cursor,
ma_stbvorbis_ds_get_length,
NULL, /* onSetLooping */
0
};
static ma_result ma_stbvorbis_init_internal(const ma_decoding_backend_config* pConfig, ma_stbvorbis* pVorbis)
{
ma_result result;
ma_data_source_config dataSourceConfig;
(void)pConfig;
if (pVorbis == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pVorbis);
pVorbis->format = ma_format_f32; /* Only supporting f32. */
dataSourceConfig = ma_data_source_config_init();
dataSourceConfig.vtable = &g_ma_stbvorbis_ds_vtable;
result = ma_data_source_init(&dataSourceConfig, &pVorbis->ds);
if (result != MA_SUCCESS) {
return result; /* Failed to initialize the base data source. */
}
return MA_SUCCESS;
}
#if !defined(MA_NO_VORBIS)
static ma_result ma_stbvorbis_post_init(ma_stbvorbis* pVorbis)
{
stb_vorbis_info info;
MA_ASSERT(pVorbis != NULL);
info = stb_vorbis_get_info(pVorbis->stb);
pVorbis->channels = info.channels;
pVorbis->sampleRate = info.sample_rate;
return MA_SUCCESS;
}
#endif
MA_API ma_result ma_stbvorbis_init(ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_stbvorbis* pVorbis)
{
ma_result result;
result = ma_stbvorbis_init_internal(pConfig, pVorbis);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
#if !defined(MA_NO_VORBIS)
{
(void)pAllocationCallbacks;
/* stb_vorbis uses an int as it's size specifier, restricting it to 32-bit even on 64-bit systems. *sigh*. */
if (dataSize > INT_MAX) {
return MA_TOO_BIG;
}
pVorbis->stb = stb_vorbis_open_memory((const unsigned char*)pData, (int)dataSize, NULL, NULL);
if (pVorbis->stb == NULL) {
return MA_INVALID_FILE;
}
pVorbis->usingPushMode = MA_FALSE;
result = ma_stbvorbis_post_init(pVorbis);
if (result != MA_SUCCESS) {
stb_vorbis_close(pVorbis->stb);
return result;
}
return MA_SUCCESS;
}
#else
{
/* vorbis is disabled. */
(void)pData;
(void)dataSize;
(void)pAllocationCallbacks;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API void ma_stbvorbis_uninit(ma_stbvorbis* pVorbis, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pVorbis == NULL) {
return;
}
#if !defined(MA_NO_VORBIS)
{
stb_vorbis_close(pVorbis->stb);
/* We'll have to clear some memory if we're using push mode. */
if (pVorbis->usingPushMode) {
ma_free(pVorbis->push.pData, pAllocationCallbacks);
}
}
#else
{
/* vorbis is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
}
#endif
ma_data_source_uninit(&pVorbis->ds);
}
MA_API ma_result ma_stbvorbis_read_pcm_frames(ma_stbvorbis* pVorbis, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
if (frameCount == 0) {
return MA_INVALID_ARGS;
}
if (pVorbis == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_VORBIS)
{
/* We always use floating point format. */
ma_result result = MA_SUCCESS; /* Must be initialized to MA_SUCCESS. */
ma_uint64 totalFramesRead = 0;
ma_format format;
ma_uint32 channels;
ma_stbvorbis_get_data_format(pVorbis, &format, &channels, NULL, NULL, 0);
if (format == ma_format_f32) {
/* We read differently depending on whether or not we're using push mode. */
if (pVorbis->usingPushMode) {
/* Push mode. This is the complex case. */
float* pFramesOutF32 = (float*)pFramesOut;
while (totalFramesRead < frameCount) {
/* The first thing to do is read from any already-cached frames. */
ma_uint32 framesToReadFromCache = (ma_uint32)ma_min(pVorbis->push.framesRemaining, (frameCount - totalFramesRead)); /* Safe cast because pVorbis->framesRemaining is 32-bit. */
/* The output pointer can be null in which case we just treate it as a seek. */
if (pFramesOut != NULL) {
ma_uint64 iFrame;
for (iFrame = 0; iFrame < framesToReadFromCache; iFrame += 1) {
ma_uint32 iChannel;
for (iChannel = 0; iChannel < pVorbis->channels; iChannel += 1) {
pFramesOutF32[iChannel] = pVorbis->push.ppPacketData[iChannel][pVorbis->push.framesConsumed + iFrame];
}
pFramesOutF32 += pVorbis->channels;
}
}
/* Update pointers and counters. */
pVorbis->push.framesConsumed += framesToReadFromCache;
pVorbis->push.framesRemaining -= framesToReadFromCache;
totalFramesRead += framesToReadFromCache;
/* Don't bother reading any more frames right now if we've just finished loading. */
if (totalFramesRead == frameCount) {
break;
}
MA_ASSERT(pVorbis->push.framesRemaining == 0);
/* Getting here means we've run out of cached frames. We'll need to load some more. */
for (;;) {
int samplesRead = 0;
int consumedDataSize;
/* We need to case dataSize to an int, so make sure we can do it safely. */
if (pVorbis->push.dataSize > INT_MAX) {
break; /* Too big. */
}
consumedDataSize = stb_vorbis_decode_frame_pushdata(pVorbis->stb, pVorbis->push.pData, (int)pVorbis->push.dataSize, NULL, &pVorbis->push.ppPacketData, &samplesRead);
if (consumedDataSize != 0) {
/* Successfully decoded a Vorbis frame. Consume the data. */
pVorbis->push.dataSize -= (size_t)consumedDataSize;
MA_MOVE_MEMORY(pVorbis->push.pData, ma_offset_ptr(pVorbis->push.pData, consumedDataSize), pVorbis->push.dataSize);
pVorbis->push.framesConsumed = 0;
pVorbis->push.framesRemaining = samplesRead;
break;
} else {
/* Not enough data. Read more. */
size_t bytesRead;
/* Expand the data buffer if necessary. */
if (pVorbis->push.dataCapacity == pVorbis->push.dataSize) {
size_t newCap = pVorbis->push.dataCapacity + MA_VORBIS_DATA_CHUNK_SIZE;
ma_uint8* pNewData;
pNewData = (ma_uint8*)ma_realloc(pVorbis->push.pData, newCap, &pVorbis->allocationCallbacks);
if (pNewData == NULL) {
result = MA_OUT_OF_MEMORY;
break;
}
pVorbis->push.pData = pNewData;
pVorbis->push.dataCapacity = newCap;
}
/* We should have enough room to load some data. */
result = pVorbis->onRead(pVorbis->pReadSeekTellUserData, ma_offset_ptr(pVorbis->push.pData, pVorbis->push.dataSize), (pVorbis->push.dataCapacity - pVorbis->push.dataSize), &bytesRead);
pVorbis->push.dataSize += bytesRead;
if (result != MA_SUCCESS) {
break; /* Failed to read any data. Get out. */
}
}
}
/* If we don't have a success code at this point it means we've encounted an error or the end of the file has been reached (probably the latter). */
if (result != MA_SUCCESS) {
break;
}
}
} else {
/* Pull mode. This is the simple case, but we still need to run in a loop because stb_vorbis loves using 32-bit instead of 64-bit. */
while (totalFramesRead < frameCount) {
ma_uint64 framesRemaining = (frameCount - totalFramesRead);
int framesRead;
if (framesRemaining > INT_MAX) {
framesRemaining = INT_MAX;
}
framesRead = stb_vorbis_get_samples_float_interleaved(pVorbis->stb, channels, (float*)ma_offset_pcm_frames_ptr(pFramesOut, totalFramesRead, format, channels), (int)framesRemaining * channels); /* Safe cast. */
totalFramesRead += framesRead;
if (framesRead < (int)framesRemaining) {
break; /* Nothing left to read. Get out. */
}
}
}
} else {
result = MA_INVALID_ARGS;
}
pVorbis->cursor += totalFramesRead;
if (totalFramesRead == 0) {
result = MA_AT_END;
}
if (pFramesRead != NULL) {
*pFramesRead = totalFramesRead;
}
if (result == MA_SUCCESS && totalFramesRead == 0) {
result = MA_AT_END;
}
return result;
}
#else
{
/* vorbis is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
(void)pFramesOut;
(void)frameCount;
(void)pFramesRead;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_stbvorbis_seek_to_pcm_frame(ma_stbvorbis* pVorbis, ma_uint64 frameIndex)
{
if (pVorbis == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_VORBIS)
{
/* Different seeking methods depending on whether or not we're using push mode. */
if (pVorbis->usingPushMode) {
/* Push mode. This is the complex case. */
ma_result result;
float buffer[4096];
/*
This is terribly inefficient because stb_vorbis does not have a good seeking solution with it's push API. Currently this just performs
a full decode right from the start of the stream. Later on I'll need to write a layer that goes through all of the Ogg pages until we
find the one containing the sample we need. Then we know exactly where to seek for stb_vorbis.
TODO: Use seeking logic documented for stb_vorbis_flush_pushdata().
*/
/* Seek to the start of the file to begin with. */
result = pVorbis->onSeek(pVorbis->pReadSeekTellUserData, 0, ma_seek_origin_start);
if (result != MA_SUCCESS) {
return result;
}
stb_vorbis_flush_pushdata(pVorbis->stb);
pVorbis->push.framesRemaining = 0;
pVorbis->push.dataSize = 0;
/* Move the cursor back to the start. We'll increment this in the loop below. */
pVorbis->cursor = 0;
while (pVorbis->cursor < frameIndex) {
ma_uint64 framesRead;
ma_uint64 framesToRead = ma_countof(buffer)/pVorbis->channels;
if (framesToRead > (frameIndex - pVorbis->cursor)) {
framesToRead = (frameIndex - pVorbis->cursor);
}
result = ma_stbvorbis_read_pcm_frames(pVorbis, buffer, framesToRead, &framesRead);
pVorbis->cursor += framesRead;
if (result != MA_SUCCESS) {
return result;
}
}
} else {
/* Pull mode. This is the simple case. */
int vorbisResult;
if (frameIndex > UINT_MAX) {
return MA_INVALID_ARGS; /* Trying to seek beyond the 32-bit maximum of stb_vorbis. */
}
vorbisResult = stb_vorbis_seek(pVorbis->stb, (unsigned int)frameIndex); /* Safe cast. */
if (vorbisResult == 0) {
return MA_ERROR; /* See failed. */
}
pVorbis->cursor = frameIndex;
}
return MA_SUCCESS;
}
#else
{
/* vorbis is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
(void)frameIndex;
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_stbvorbis_get_data_format(ma_stbvorbis* pVorbis, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
/* Defaults for safety. */
if (pFormat != NULL) {
*pFormat = ma_format_unknown;
}
if (pChannels != NULL) {
*pChannels = 0;
}
if (pSampleRate != NULL) {
*pSampleRate = 0;
}
if (pChannelMap != NULL) {
MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
}
if (pVorbis == NULL) {
return MA_INVALID_OPERATION;
}
if (pFormat != NULL) {
*pFormat = pVorbis->format;
}
#if !defined(MA_NO_VORBIS)
{
if (pChannels != NULL) {
*pChannels = pVorbis->channels;
}
if (pSampleRate != NULL) {
*pSampleRate = pVorbis->sampleRate;
}
if (pChannelMap != NULL) {
ma_channel_map_init_standard(ma_standard_channel_map_vorbis, pChannelMap, channelMapCap, pVorbis->channels);
}
return MA_SUCCESS;
}
#else
{
/* vorbis is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_stbvorbis_get_cursor_in_pcm_frames(ma_stbvorbis* pVorbis, ma_uint64* pCursor)
{
if (pCursor == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = 0; /* Safety. */
if (pVorbis == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_VORBIS)
{
*pCursor = pVorbis->cursor;
return MA_SUCCESS;
}
#else
{
/* vorbis is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
return MA_NOT_IMPLEMENTED;
}
#endif
}
MA_API ma_result ma_stbvorbis_get_length_in_pcm_frames(ma_stbvorbis* pVorbis, ma_uint64* pLength)
{
if (pLength == NULL) {
return MA_INVALID_ARGS;
}
*pLength = 0; /* Safety. */
if (pVorbis == NULL) {
return MA_INVALID_ARGS;
}
#if !defined(MA_NO_VORBIS)
{
if (pVorbis->usingPushMode) {
*pLength = 0; /* I don't know of a good way to determine this reliably with stb_vorbis and push mode. */
} else {
*pLength = stb_vorbis_stream_length_in_samples(pVorbis->stb);
}
return MA_SUCCESS;
}
#else
{
/* vorbis is disabled. Should never hit this since initialization would have failed. */
MA_ASSERT(MA_FALSE);
return MA_NOT_IMPLEMENTED;
}
#endif
}
static ma_result ma_decoding_backend_init__stbvorbis(void* pUserData, ma_read_proc onRead, ma_seek_proc onSeek, ma_tell_proc onTell, void* pReadSeekTellUserData, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCal...
{
ma_result result;
ma_stbvorbis* pVorbis;
(void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
/* For now we're just allocating the decoder backend on the heap. */
pVorbis = (ma_stbvorbis*)ma_malloc(sizeof(*pVorbis), pAllocationCallbacks);
if (pVorbis == NULL) {
return MA_OUT_OF_MEMORY;
}
result = ma_stbvorbis_init(onRead, onSeek, onTell, pReadSeekTellUserData, pConfig, pAllocationCallbacks, pVorbis);
if (result != MA_SUCCESS) {
ma_free(pVorbis, pAllocationCallbacks);
return result;
}
*ppBackend = pVorbis;
return MA_SUCCESS;
}
static ma_result ma_decoding_backend_init_file__stbvorbis(void* pUserData, const char* pFilePath, const ma_decoding_backend_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source** ppBackend)
{
ma_result result;
ma_stbvorbis* pVorbis;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
ma_stbvorbis* pVorbis;
(void)pUserData; /* For now not using pUserData, but once we start storing the vorbis decoder state within the ma_decoder structure this will be set to the decoder so we can avoid a malloc. */
/* For now we're just allocating the decoder backend on the heap. */
pVorbis = (ma_stbvorbis*)ma_malloc(sizeof(*pVorbis), pAllocationCallbacks);
if (pVorbis == NULL) {
return MA_OUT_OF_MEMORY;
}
result = ma_stbvorbis_init_memory(pData, dataSize, pConfig, pAllocationCallbacks, pVorbis);
if (result != MA_SUCCESS) {
ma_free(pVorbis, pAllocationCallbacks);
return result;
}
*ppBackend = pVorbis;
return MA_SUCCESS;
}
static void ma_decoding_backend_uninit__stbvorbis(void* pUserData, ma_data_source* pBackend, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_stbvorbis* pVorbis = (ma_stbvorbis*)pBackend;
(void)pUserData;
ma_stbvorbis_uninit(pVorbis, pAllocationCallbacks);
ma_free(pVorbis, pAllocationCallbacks);
}
static ma_decoding_backend_vtable g_ma_decoding_backend_vtable_stbvorbis =
{
ma_decoding_backend_init__stbvorbis,
ma_decoding_backend_init_file__stbvorbis,
NULL, /* onInitFileW() */
ma_decoding_backend_init_memory__stbvorbis,
ma_decoding_backend_uninit__stbvorbis
};
static ma_result ma_decoder_init_vorbis__internal(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
return ma_decoder_init_from_vtable(&g_ma_decoding_backend_vtable_stbvorbis, NULL, pConfig, pDecoder);
}
#endif /* STB_VORBIS_INCLUDE_STB_VORBIS_H */
static ma_result ma_decoder__init_allocation_callbacks(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
MA_ASSERT(pDecoder != NULL);
if (pConfig != NULL) {
return ma_allocation_callbacks_init_copy(&pDecoder->allocationCallbacks, &pConfig->allocationCallbacks);
} else {
pDecoder->allocationCallbacks = ma_allocation_callbacks_init_default();
return MA_SUCCESS;
}
}
static ma_result ma_decoder__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
return ma_decoder_read_pcm_frames((ma_decoder*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
static ma_result ma_decoder__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
return ma_decoder_seek_to_pcm_frame((ma_decoder*)pDataSource, frameIndex);
}
static ma_result ma_decoder__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
return ma_decoder_get_data_format((ma_decoder*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
static ma_result ma_decoder__data_source_on_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
return ma_decoder_get_cursor_in_pcm_frames((ma_decoder*)pDataSource, pCursor);
}
static ma_result ma_decoder__data_source_on_get_length(ma_data_source* pDataSource, ma_uint64* pLength)
{
return ma_decoder_get_length_in_pcm_frames((ma_decoder*)pDataSource, pLength);
}
static ma_data_source_vtable g_ma_decoder_data_source_vtable =
{
ma_decoder__data_source_on_read,
ma_decoder__data_source_on_seek,
ma_decoder__data_source_on_get_data_format,
ma_decoder__data_source_on_get_cursor,
ma_decoder__data_source_on_get_length,
NULL, /* onSetLooping */
0
};
static ma_result ma_decoder__preinit(ma_decoder_read_proc onRead, ma_decoder_seek_proc onSeek, ma_decoder_tell_proc onTell, void* pUserData, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
ma_result result;
ma_data_source_config dataSourceConfig;
MA_ASSERT(pConfig != NULL);
if (pDecoder == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pDecoder);
if (onRead == NULL || onSeek == NULL) {
return MA_INVALID_ARGS;
}
dataSourceConfig = ma_data_source_config_init();
dataSourceConfig.vtable = &g_ma_decoder_data_source_vtable;
result = ma_data_source_init(&dataSourceConfig, &pDecoder->ds);
if (result != MA_SUCCESS) {
return result;
}
pDecoder->onRead = onRead;
pDecoder->onSeek = onSeek;
pDecoder->onTell = onTell;
pDecoder->pUserData = pUserData;
result = ma_decoder__init_allocation_callbacks(pConfig, pDecoder);
if (result != MA_SUCCESS) {
ma_data_source_uninit(&pDecoder->ds);
return result;
}
return MA_SUCCESS;
}
static ma_result ma_decoder__postinit(const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
ma_result result;
result = ma_decoder__init_data_converter(pDecoder, pConfig);
/* If we failed post initialization we need to uninitialize the decoder before returning to prevent a memory leak. */
if (result != MA_SUCCESS) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
result = ma_decoder_init_mp3__internal(&config, pDecoder);
if (result != MA_SUCCESS) {
ma_decoder__on_seek_vfs(pDecoder, 0, ma_seek_origin_start);
}
}
#endif
}
/* If we still haven't got a result just use trial and error. Otherwise we can finish up. */
if (result != MA_SUCCESS) {
result = ma_decoder_init__internal(ma_decoder__on_read_vfs, ma_decoder__on_seek_vfs, NULL, &config, pDecoder);
} else {
result = ma_decoder__postinit(&config, pDecoder);
}
if (result != MA_SUCCESS) {
ma_vfs_or_default_close(pVFS, pDecoder->data.vfs.file);
return result;
}
return MA_SUCCESS;
}
MA_API ma_result ma_decoder_init_file(const char* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
return ma_decoder_init_vfs(NULL, pFilePath, pConfig, pDecoder);
}
MA_API ma_result ma_decoder_init_file_w(const wchar_t* pFilePath, const ma_decoder_config* pConfig, ma_decoder* pDecoder)
{
return ma_decoder_init_vfs_w(NULL, pFilePath, pConfig, pDecoder);
}
MA_API ma_result ma_decoder_uninit(ma_decoder* pDecoder)
{
if (pDecoder == NULL) {
return MA_INVALID_ARGS;
}
if (pDecoder->pBackend != NULL) {
if (pDecoder->pBackendVTable != NULL && pDecoder->pBackendVTable->onUninit != NULL) {
pDecoder->pBackendVTable->onUninit(pDecoder->pBackendUserData, pDecoder->pBackend, &pDecoder->allocationCallbacks);
}
}
if (pDecoder->onRead == ma_decoder__on_read_vfs) {
ma_vfs_or_default_close(pDecoder->data.vfs.pVFS, pDecoder->data.vfs.file);
pDecoder->data.vfs.file = NULL;
}
ma_data_converter_uninit(&pDecoder->converter, &pDecoder->allocationCallbacks);
ma_data_source_uninit(&pDecoder->ds);
if (pDecoder->pInputCache != NULL) {
ma_free(pDecoder->pInputCache, &pDecoder->allocationCallbacks);
}
return MA_SUCCESS;
}
MA_API ma_result ma_decoder_read_pcm_frames(ma_decoder* pDecoder, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
ma_result result = MA_SUCCESS;
ma_uint64 totalFramesReadOut;
void* pRunningFramesOut;
if (pFramesRead != NULL) {
*pFramesRead = 0; /* Safety. */
}
if (frameCount == 0) {
return MA_INVALID_ARGS;
}
if (pDecoder == NULL) {
return MA_INVALID_ARGS;
}
if (pDecoder->pBackend == NULL) {
return MA_INVALID_OPERATION;
}
/* Fast path. */
if (pDecoder->converter.isPassthrough) {
result = ma_data_source_read_pcm_frames(pDecoder->pBackend, pFramesOut, frameCount, &totalFramesReadOut);
} else {
/*
Getting here means we need to do data conversion. If we're seeking forward and are _not_ doing resampling we can run this in a fast path. If we're doing resampling we
need to run through each sample because we need to ensure it's internal cache is updated.
*/
if (pFramesOut == NULL && pDecoder->converter.hasResampler == MA_FALSE) {
result = ma_data_source_read_pcm_frames(pDecoder->pBackend, NULL, frameCount, &totalFramesReadOut);
} else {
/* Slow path. Need to run everything through the data converter. */
ma_format internalFormat;
ma_uint32 internalChannels;
totalFramesReadOut = 0;
pRunningFramesOut = pFramesOut;
result = ma_data_source_get_data_format(pDecoder->pBackend, &internalFormat, &internalChannels, NULL, NULL, 0);
if (result != MA_SUCCESS) {
return result; /* Failed to retrieve the internal format and channel count. */
}
/*
We run a different path depending on whether or not we are using a heap-allocated
intermediary buffer or not. If the data converter does not support the calculation of
the required number of input frames, we'll use the heap-allocated path. Otherwise we'll
use the stack-allocated path.
*/
if (pDecoder->pInputCache != NULL) {
/* We don't have a way of determining the required number of input frames, so need to persistently store input data in a cache. */
while (totalFramesReadOut < frameCount) {
ma_uint64 framesToReadThisIterationIn;
ma_uint64 framesToReadThisIterationOut;
/* If there's any data available in the cache, that needs to get processed first. */
if (pDecoder->inputCacheRemaining > 0) {
framesToReadThisIterationOut = (frameCount - totalFramesReadOut);
framesToReadThisIterationIn = framesToReadThisIterationOut;
if (framesToReadThisIterationIn > pDecoder->inputCacheRemaining) {
framesToReadThisIterationIn = pDecoder->inputCacheRemaining;
}
result = ma_data_converter_process_pcm_frames(&pDecoder->converter, ma_offset_pcm_frames_ptr(pDecoder->pInputCache, pDecoder->inputCacheConsumed, internalFormat, internalChannels), &framesToReadThisIterationIn, pRunningFramesO...
if (result != MA_SUCCESS) {
break;
}
pDecoder->inputCacheConsumed += framesToReadThisIterationIn;
pDecoder->inputCacheRemaining -= framesToReadThisIterationIn;
totalFramesReadOut += framesToReadThisIterationOut;
if (pRunningFramesOut != NULL) {
pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesToReadThisIterationOut * ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels));
}
if (framesToReadThisIterationIn == 0 && framesToReadThisIterationOut == 0) {
break; /* We're done. */
}
}
/* Getting here means there's no data in the cache and we need to fill it up from the data source. */
if (pDecoder->inputCacheRemaining == 0) {
pDecoder->inputCacheConsumed = 0;
result = ma_data_source_read_pcm_frames(pDecoder->pBackend, pDecoder->pInputCache, pDecoder->inputCacheCap, &pDecoder->inputCacheRemaining);
if (result != MA_SUCCESS) {
break;
}
}
}
} else {
/* We have a way of determining the required number of input frames so just use the stack. */
while (totalFramesReadOut < frameCount) {
ma_uint8 pIntermediaryBuffer[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* In internal format. */
ma_uint64 intermediaryBufferCap = sizeof(pIntermediaryBuffer) / ma_get_bytes_per_frame(internalFormat, internalChannels);
ma_uint64 framesToReadThisIterationIn;
ma_uint64 framesReadThisIterationIn;
ma_uint64 framesToReadThisIterationOut;
ma_uint64 framesReadThisIterationOut;
ma_uint64 requiredInputFrameCount;
framesToReadThisIterationOut = (frameCount - totalFramesReadOut);
framesToReadThisIterationIn = framesToReadThisIterationOut;
if (framesToReadThisIterationIn > intermediaryBufferCap) {
framesToReadThisIterationIn = intermediaryBufferCap;
}
ma_data_converter_get_required_input_frame_count(&pDecoder->converter, framesToReadThisIterationOut, &requiredInputFrameCount);
if (framesToReadThisIterationIn > requiredInputFrameCount) {
framesToReadThisIterationIn = requiredInputFrameCount;
}
if (requiredInputFrameCount > 0) {
result = ma_data_source_read_pcm_frames(pDecoder->pBackend, pIntermediaryBuffer, framesToReadThisIterationIn, &framesReadThisIterationIn);
} else {
framesReadThisIterationIn = 0;
}
/*
At this point we have our decoded data in input format and now we need to convert to output format. Note that even if we didn't read any
input frames, we still want to try processing frames because there may some output frames generated from cached input data.
*/
framesReadThisIterationOut = framesToReadThisIterationOut;
result = ma_data_converter_process_pcm_frames(&pDecoder->converter, pIntermediaryBuffer, &framesReadThisIterationIn, pRunningFramesOut, &framesReadThisIterationOut);
if (result != MA_SUCCESS) {
break;
}
totalFramesReadOut += framesReadThisIterationOut;
if (pRunningFramesOut != NULL) {
pRunningFramesOut = ma_offset_ptr(pRunningFramesOut, framesReadThisIterationOut * ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels));
}
if (framesReadThisIterationIn == 0 && framesReadThisIterationOut == 0) {
break; /* We're done. */
}
}
}
}
}
pDecoder->readPointerInPCMFrames += totalFramesReadOut;
if (pFramesRead != NULL) {
*pFramesRead = totalFramesReadOut;
}
if (result == MA_SUCCESS && totalFramesReadOut == 0) {
result = MA_AT_END;
}
return result;
}
MA_API ma_result ma_decoder_seek_to_pcm_frame(ma_decoder* pDecoder, ma_uint64 frameIndex)
{
if (pDecoder == NULL) {
return MA_INVALID_ARGS;
}
if (pDecoder->pBackend != NULL) {
ma_result result;
ma_uint64 internalFrameIndex;
ma_uint32 internalSampleRate;
ma_uint64 currentFrameIndex;
result = ma_data_source_get_data_format(pDecoder->pBackend, NULL, NULL, &internalSampleRate, NULL, 0);
if (result != MA_SUCCESS) {
return result; /* Failed to retrieve the internal sample rate. */
}
if (internalSampleRate == pDecoder->outputSampleRate) {
internalFrameIndex = frameIndex;
} else {
internalFrameIndex = ma_calculate_frame_count_after_resampling(internalSampleRate, pDecoder->outputSampleRate, frameIndex);
}
/* Only seek if we're requesting a different frame to what we're currently sitting on. */
ma_data_source_get_cursor_in_pcm_frames(pDecoder->pBackend, ¤tFrameIndex);
if (currentFrameIndex != internalFrameIndex) {
result = ma_data_source_seek_to_pcm_frame(pDecoder->pBackend, internalFrameIndex);
if (result == MA_SUCCESS) {
pDecoder->readPointerInPCMFrames = frameIndex;
}
/* Reset the data converter so that any cached data in the resampler is cleared. */
ma_data_converter_reset(&pDecoder->converter);
}
return result;
}
/* Should never get here, but if we do it means onSeekToPCMFrame was not set by the backend. */
return MA_INVALID_ARGS;
}
MA_API ma_result ma_decoder_get_data_format(ma_decoder* pDecoder, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
if (pDecoder == NULL) {
return MA_INVALID_ARGS;
}
if (pFormat != NULL) {
*pFormat = pDecoder->outputFormat;
}
if (pChannels != NULL) {
*pChannels = pDecoder->outputChannels;
}
if (pSampleRate != NULL) {
*pSampleRate = pDecoder->outputSampleRate;
}
if (pChannelMap != NULL) {
ma_data_converter_get_output_channel_map(&pDecoder->converter, pChannelMap, channelMapCap);
}
return MA_SUCCESS;
}
MA_API ma_result ma_decoder_get_cursor_in_pcm_frames(ma_decoder* pDecoder, ma_uint64* pCursor)
{
if (pCursor == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = 0;
if (pDecoder == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = pDecoder->readPointerInPCMFrames;
return MA_SUCCESS;
}
MA_API ma_result ma_decoder_get_length_in_pcm_frames(ma_decoder* pDecoder, ma_uint64* pLength)
{
if (pLength == NULL) {
return MA_INVALID_ARGS;
}
*pLength = 0;
if (pDecoder == NULL) {
return MA_INVALID_ARGS;
}
if (pDecoder->pBackend != NULL) {
ma_result result;
ma_uint64 internalLengthInPCMFrames;
ma_uint32 internalSampleRate;
result = ma_data_source_get_length_in_pcm_frames(pDecoder->pBackend, &internalLengthInPCMFrames);
if (result != MA_SUCCESS) {
return result; /* Failed to retrieve the internal length. */
}
result = ma_data_source_get_data_format(pDecoder->pBackend, NULL, NULL, &internalSampleRate, NULL, 0);
if (result != MA_SUCCESS) {
return result; /* Failed to retrieve the internal sample rate. */
}
if (internalSampleRate == pDecoder->outputSampleRate) {
*pLength = internalLengthInPCMFrames;
} else {
*pLength = ma_calculate_frame_count_after_resampling(pDecoder->outputSampleRate, internalSampleRate, internalLengthInPCMFrames);
}
return MA_SUCCESS;
} else {
return MA_NO_BACKEND;
}
}
MA_API ma_result ma_decoder_get_available_frames(ma_decoder* pDecoder, ma_uint64* pAvailableFrames)
{
ma_result result;
ma_uint64 totalFrameCount;
if (pAvailableFrames == NULL) {
return MA_INVALID_ARGS;
}
*pAvailableFrames = 0;
if (pDecoder == NULL) {
return MA_INVALID_ARGS;
}
result = ma_decoder_get_length_in_pcm_frames(pDecoder, &totalFrameCount);
if (result != MA_SUCCESS) {
return result;
}
if (totalFrameCount <= pDecoder->readPointerInPCMFrames) {
*pAvailableFrames = 0;
} else {
*pAvailableFrames = totalFrameCount - pDecoder->readPointerInPCMFrames;
}
return MA_SUCCESS;
}
static ma_result ma_decoder__full_decode_and_uninit(ma_decoder* pDecoder, ma_decoder_config* pConfigOut, ma_uint64* pFrameCountOut, void** ppPCMFramesOut)
{
ma_result result;
ma_uint64 totalFrameCount;
ma_uint64 bpf;
ma_uint64 dataCapInFrames;
void* pPCMFramesOut;
MA_ASSERT(pDecoder != NULL);
totalFrameCount = 0;
bpf = ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels);
/* The frame count is unknown until we try reading. Thus, we just run in a loop. */
dataCapInFrames = 0;
pPCMFramesOut = NULL;
for (;;) {
ma_uint64 frameCountToTryReading;
ma_uint64 framesJustRead;
/* Make room if there's not enough. */
if (totalFrameCount == dataCapInFrames) {
void* pNewPCMFramesOut;
ma_uint64 newDataCapInFrames = dataCapInFrames*2;
if (newDataCapInFrames == 0) {
newDataCapInFrames = 4096;
}
if ((newDataCapInFrames * bpf) > MA_SIZE_MAX) {
ma_free(pPCMFramesOut, &pDecoder->allocationCallbacks);
return MA_TOO_BIG;
}
pNewPCMFramesOut = (void*)ma_realloc(pPCMFramesOut, (size_t)(newDataCapInFrames * bpf), &pDecoder->allocationCallbacks);
if (pNewPCMFramesOut == NULL) {
ma_free(pPCMFramesOut, &pDecoder->allocationCallbacks);
return MA_OUT_OF_MEMORY;
}
dataCapInFrames = newDataCapInFrames;
pPCMFramesOut = pNewPCMFramesOut;
}
frameCountToTryReading = dataCapInFrames - totalFrameCount;
MA_ASSERT(frameCountToTryReading > 0);
result = ma_decoder_read_pcm_frames(pDecoder, (ma_uint8*)pPCMFramesOut + (totalFrameCount * bpf), frameCountToTryReading, &framesJustRead);
totalFrameCount += framesJustRead;
if (result != MA_SUCCESS) {
break;
}
if (framesJustRead < frameCountToTryReading) {
break;
}
}
if (pConfigOut != NULL) {
pConfigOut->format = pDecoder->outputFormat;
pConfigOut->channels = pDecoder->outputChannels;
pConfigOut->sampleRate = pDecoder->outputSampleRate;
}
if (ppPCMFramesOut != NULL) {
*ppPCMFramesOut = pPCMFramesOut;
} else {
ma_free(pPCMFramesOut, &pDecoder->allocationCallbacks);
}
if (pFrameCountOut != NULL) {
*pFrameCountOut = totalFrameCount;
}
ma_decoder_uninit(pDecoder);
return MA_SUCCESS;
}
MA_API ma_result ma_decode_from_vfs(ma_vfs* pVFS, const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut)
{
ma_result result;
ma_decoder_config config;
ma_decoder decoder;
if (pFrameCountOut != NULL) {
*pFrameCountOut = 0;
}
if (ppPCMFramesOut != NULL) {
*ppPCMFramesOut = NULL;
}
config = ma_decoder_config_init_copy(pConfig);
result = ma_decoder_init_vfs(pVFS, pFilePath, &config, &decoder);
if (result != MA_SUCCESS) {
return result;
}
result = ma_decoder__full_decode_and_uninit(&decoder, pConfig, pFrameCountOut, ppPCMFramesOut);
return result;
}
MA_API ma_result ma_decode_file(const char* pFilePath, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut)
{
return ma_decode_from_vfs(NULL, pFilePath, pConfig, pFrameCountOut, ppPCMFramesOut);
}
MA_API ma_result ma_decode_memory(const void* pData, size_t dataSize, ma_decoder_config* pConfig, ma_uint64* pFrameCountOut, void** ppPCMFramesOut)
{
ma_decoder_config config;
ma_decoder decoder;
ma_result result;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
MA_ASSERT(pEncoder != NULL);
result = pEncoder->onSeek(pEncoder, offset, (origin == drwav_seek_origin_start) ? ma_seek_origin_start : ma_seek_origin_current);
if (result != MA_SUCCESS) {
return DRWAV_FALSE;
} else {
return DRWAV_TRUE;
}
}
static ma_result ma_encoder__on_init_wav(ma_encoder* pEncoder)
{
drwav_data_format wavFormat;
drwav_allocation_callbacks allocationCallbacks;
drwav* pWav;
MA_ASSERT(pEncoder != NULL);
pWav = (drwav*)ma_malloc(sizeof(*pWav), &pEncoder->config.allocationCallbacks);
if (pWav == NULL) {
return MA_OUT_OF_MEMORY;
}
wavFormat.container = drwav_container_riff;
wavFormat.channels = pEncoder->config.channels;
wavFormat.sampleRate = pEncoder->config.sampleRate;
wavFormat.bitsPerSample = ma_get_bytes_per_sample(pEncoder->config.format) * 8;
if (pEncoder->config.format == ma_format_f32) {
wavFormat.format = DR_WAVE_FORMAT_IEEE_FLOAT;
} else {
wavFormat.format = DR_WAVE_FORMAT_PCM;
}
allocationCallbacks.pUserData = pEncoder->config.allocationCallbacks.pUserData;
allocationCallbacks.onMalloc = pEncoder->config.allocationCallbacks.onMalloc;
allocationCallbacks.onRealloc = pEncoder->config.allocationCallbacks.onRealloc;
allocationCallbacks.onFree = pEncoder->config.allocationCallbacks.onFree;
if (!drwav_init_write(pWav, &wavFormat, ma_encoder__internal_on_write_wav, ma_encoder__internal_on_seek_wav, pEncoder, &allocationCallbacks)) {
return MA_ERROR;
}
pEncoder->pInternalEncoder = pWav;
return MA_SUCCESS;
}
static void ma_encoder__on_uninit_wav(ma_encoder* pEncoder)
{
drwav* pWav;
MA_ASSERT(pEncoder != NULL);
pWav = (drwav*)pEncoder->pInternalEncoder;
MA_ASSERT(pWav != NULL);
drwav_uninit(pWav);
ma_free(pWav, &pEncoder->config.allocationCallbacks);
}
static ma_result ma_encoder__on_write_pcm_frames_wav(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount, ma_uint64* pFramesWritten)
{
drwav* pWav;
ma_uint64 framesWritten;
MA_ASSERT(pEncoder != NULL);
pWav = (drwav*)pEncoder->pInternalEncoder;
MA_ASSERT(pWav != NULL);
framesWritten = drwav_write_pcm_frames(pWav, frameCount, pFramesIn);
if (pFramesWritten != NULL) {
*pFramesWritten = framesWritten;
}
return MA_SUCCESS;
}
#endif
MA_API ma_encoder_config ma_encoder_config_init(ma_encoding_format encodingFormat, ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
{
ma_encoder_config config;
MA_ZERO_OBJECT(&config);
config.encodingFormat = encodingFormat;
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
return config;
}
MA_API ma_result ma_encoder_preinit(const ma_encoder_config* pConfig, ma_encoder* pEncoder)
{
ma_result result;
if (pEncoder == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pEncoder);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->format == ma_format_unknown || pConfig->channels == 0 || pConfig->sampleRate == 0) {
return MA_INVALID_ARGS;
}
pEncoder->config = *pConfig;
result = ma_allocation_callbacks_init_copy(&pEncoder->config.allocationCallbacks, &pConfig->allocationCallbacks);
if (result != MA_SUCCESS) {
return result;
}
return MA_SUCCESS;
}
MA_API ma_result ma_encoder_init__internal(ma_encoder_write_proc onWrite, ma_encoder_seek_proc onSeek, void* pUserData, ma_encoder* pEncoder)
{
ma_result result = MA_SUCCESS;
/* This assumes ma_encoder_preinit() has been called prior. */
MA_ASSERT(pEncoder != NULL);
if (onWrite == NULL || onSeek == NULL) {
return MA_INVALID_ARGS;
}
pEncoder->onWrite = onWrite;
pEncoder->onSeek = onSeek;
pEncoder->pUserData = pUserData;
switch (pEncoder->config.encodingFormat)
{
case ma_encoding_format_wav:
{
#if defined(MA_HAS_WAV)
pEncoder->onInit = ma_encoder__on_init_wav;
pEncoder->onUninit = ma_encoder__on_uninit_wav;
pEncoder->onWritePCMFrames = ma_encoder__on_write_pcm_frames_wav;
#else
result = MA_NO_BACKEND;
#endif
} break;
default:
{
result = MA_INVALID_ARGS;
} break;
}
/* Getting here means we should have our backend callbacks set up. */
if (result == MA_SUCCESS) {
result = pEncoder->onInit(pEncoder);
}
return result;
}
static ma_result ma_encoder__on_write_vfs(ma_encoder* pEncoder, const void* pBufferIn, size_t bytesToWrite, size_t* pBytesWritten)
{
return ma_vfs_or_default_write(pEncoder->data.vfs.pVFS, pEncoder->data.vfs.file, pBufferIn, bytesToWrite, pBytesWritten);
}
static ma_result ma_encoder__on_seek_vfs(ma_encoder* pEncoder, ma_int64 offset, ma_seek_origin origin)
{
return ma_vfs_or_default_seek(pEncoder->data.vfs.pVFS, pEncoder->data.vfs.file, offset, origin);
}
MA_API ma_result ma_encoder_init_vfs(ma_vfs* pVFS, const char* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
{
ma_result result;
ma_vfs_file file;
result = ma_encoder_preinit(pConfig, pEncoder);
if (result != MA_SUCCESS) {
return result;
}
/* Now open the file. If this fails we don't need to uninitialize the encoder. */
result = ma_vfs_or_default_open(pVFS, pFilePath, MA_OPEN_MODE_WRITE, &file);
if (result != MA_SUCCESS) {
return result;
}
pEncoder->data.vfs.pVFS = pVFS;
pEncoder->data.vfs.file = file;
result = ma_encoder_init__internal(ma_encoder__on_write_vfs, ma_encoder__on_seek_vfs, NULL, pEncoder);
if (result != MA_SUCCESS) {
ma_vfs_or_default_close(pVFS, file);
return result;
}
return MA_SUCCESS;
}
MA_API ma_result ma_encoder_init_vfs_w(ma_vfs* pVFS, const wchar_t* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
{
ma_result result;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
/* Now open the file. If this fails we don't need to uninitialize the encoder. */
result = ma_vfs_or_default_open_w(pVFS, pFilePath, MA_OPEN_MODE_WRITE, &file);
if (result != MA_SUCCESS) {
return result;
}
pEncoder->data.vfs.pVFS = pVFS;
pEncoder->data.vfs.file = file;
result = ma_encoder_init__internal(ma_encoder__on_write_vfs, ma_encoder__on_seek_vfs, NULL, pEncoder);
if (result != MA_SUCCESS) {
ma_vfs_or_default_close(pVFS, file);
return result;
}
return MA_SUCCESS;
}
MA_API ma_result ma_encoder_init_file(const char* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
{
return ma_encoder_init_vfs(NULL, pFilePath, pConfig, pEncoder);
}
MA_API ma_result ma_encoder_init_file_w(const wchar_t* pFilePath, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
{
return ma_encoder_init_vfs_w(NULL, pFilePath, pConfig, pEncoder);
}
MA_API ma_result ma_encoder_init(ma_encoder_write_proc onWrite, ma_encoder_seek_proc onSeek, void* pUserData, const ma_encoder_config* pConfig, ma_encoder* pEncoder)
{
ma_result result;
result = ma_encoder_preinit(pConfig, pEncoder);
if (result != MA_SUCCESS) {
return result;
}
return ma_encoder_init__internal(onWrite, onSeek, pUserData, pEncoder);
}
MA_API void ma_encoder_uninit(ma_encoder* pEncoder)
{
if (pEncoder == NULL) {
return;
}
if (pEncoder->onUninit) {
pEncoder->onUninit(pEncoder);
}
/* If we have a file handle, close it. */
if (pEncoder->onWrite == ma_encoder__on_write_vfs) {
ma_vfs_or_default_close(pEncoder->data.vfs.pVFS, pEncoder->data.vfs.file);
pEncoder->data.vfs.file = NULL;
}
}
MA_API ma_result ma_encoder_write_pcm_frames(ma_encoder* pEncoder, const void* pFramesIn, ma_uint64 frameCount, ma_uint64* pFramesWritten)
{
if (pFramesWritten != NULL) {
*pFramesWritten = 0;
}
if (pEncoder == NULL || pFramesIn == NULL) {
return MA_INVALID_ARGS;
}
return pEncoder->onWritePCMFrames(pEncoder, pFramesIn, frameCount, pFramesWritten);
}
#endif /* MA_NO_ENCODING */
/**************************************************************************************************************************************************************
Generation
**************************************************************************************************************************************************************/
#ifndef MA_NO_GENERATION
MA_API ma_waveform_config ma_waveform_config_init(ma_format format, ma_uint32 channels, ma_uint32 sampleRate, ma_waveform_type type, double amplitude, double frequency)
{
ma_waveform_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.sampleRate = sampleRate;
config.type = type;
config.amplitude = amplitude;
config.frequency = frequency;
return config;
}
static ma_result ma_waveform__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
return ma_waveform_read_pcm_frames((ma_waveform*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
static ma_result ma_waveform__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
return ma_waveform_seek_to_pcm_frame((ma_waveform*)pDataSource, frameIndex);
}
static ma_result ma_waveform__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
ma_waveform* pWaveform = (ma_waveform*)pDataSource;
*pFormat = pWaveform->config.format;
*pChannels = pWaveform->config.channels;
*pSampleRate = pWaveform->config.sampleRate;
ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pWaveform->config.channels);
return MA_SUCCESS;
}
static ma_result ma_waveform__data_source_on_get_cursor(ma_data_source* pDataSource, ma_uint64* pCursor)
{
ma_waveform* pWaveform = (ma_waveform*)pDataSource;
*pCursor = (ma_uint64)(pWaveform->time / pWaveform->advance);
return MA_SUCCESS;
}
static double ma_waveform__calculate_advance(ma_uint32 sampleRate, double frequency)
{
return (1.0 / (sampleRate / frequency));
}
static void ma_waveform__update_advance(ma_waveform* pWaveform)
{
pWaveform->advance = ma_waveform__calculate_advance(pWaveform->config.sampleRate, pWaveform->config.frequency);
}
static ma_data_source_vtable g_ma_waveform_data_source_vtable =
{
ma_waveform__data_source_on_read,
ma_waveform__data_source_on_seek,
ma_waveform__data_source_on_get_data_format,
ma_waveform__data_source_on_get_cursor,
NULL, /* onGetLength. There's no notion of a length in waveforms. */
NULL, /* onSetLooping */
0
};
MA_API ma_result ma_waveform_init(const ma_waveform_config* pConfig, ma_waveform* pWaveform)
{
ma_result result;
ma_data_source_config dataSourceConfig;
if (pWaveform == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pWaveform);
dataSourceConfig = ma_data_source_config_init();
dataSourceConfig.vtable = &g_ma_waveform_data_source_vtable;
result = ma_data_source_init(&dataSourceConfig, &pWaveform->ds);
if (result != MA_SUCCESS) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
static float ma_waveform_sine_f32(double time, double amplitude)
{
return (float)(ma_sind(MA_TAU_D * time) * amplitude);
}
static ma_int16 ma_waveform_sine_s16(double time, double amplitude)
{
return ma_pcm_sample_f32_to_s16(ma_waveform_sine_f32(time, amplitude));
}
static float ma_waveform_square_f32(double time, double amplitude)
{
double f = time - (ma_int64)time;
double r;
if (f < 0.5) {
r = amplitude;
} else {
r = -amplitude;
}
return (float)r;
}
static ma_int16 ma_waveform_square_s16(double time, double amplitude)
{
return ma_pcm_sample_f32_to_s16(ma_waveform_square_f32(time, amplitude));
}
static float ma_waveform_triangle_f32(double time, double amplitude)
{
double f = time - (ma_int64)time;
double r;
r = 2 * ma_abs(2 * (f - 0.5)) - 1;
return (float)(r * amplitude);
}
static ma_int16 ma_waveform_triangle_s16(double time, double amplitude)
{
return ma_pcm_sample_f32_to_s16(ma_waveform_triangle_f32(time, amplitude));
}
static float ma_waveform_sawtooth_f32(double time, double amplitude)
{
double f = time - (ma_int64)time;
double r;
r = 2 * (f - 0.5);
return (float)(r * amplitude);
}
static ma_int16 ma_waveform_sawtooth_s16(double time, double amplitude)
{
return ma_pcm_sample_f32_to_s16(ma_waveform_sawtooth_f32(time, amplitude));
}
static void ma_waveform_read_pcm_frames__sine(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
{
ma_uint64 iFrame;
ma_uint64 iChannel;
ma_uint32 bps = ma_get_bytes_per_sample(pWaveform->config.format);
ma_uint32 bpf = bps * pWaveform->config.channels;
MA_ASSERT(pWaveform != NULL);
MA_ASSERT(pFramesOut != NULL);
if (pWaveform->config.format == ma_format_f32) {
float* pFramesOutF32 = (float*)pFramesOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_waveform_sine_f32(pWaveform->time, pWaveform->config.amplitude);
pWaveform->time += pWaveform->advance;
for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
pFramesOutF32[iFrame*pWaveform->config.channels + iChannel] = s;
}
}
} else if (pWaveform->config.format == ma_format_s16) {
ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_int16 s = ma_waveform_sine_s16(pWaveform->time, pWaveform->config.amplitude);
pWaveform->time += pWaveform->advance;
for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
pFramesOutS16[iFrame*pWaveform->config.channels + iChannel] = s;
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_waveform_sine_f32(pWaveform->time, pWaveform->config.amplitude);
pWaveform->time += pWaveform->advance;
for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pWaveform->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
}
}
}
}
static void ma_waveform_read_pcm_frames__square(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
{
ma_uint64 iFrame;
ma_uint64 iChannel;
ma_uint32 bps = ma_get_bytes_per_sample(pWaveform->config.format);
ma_uint32 bpf = bps * pWaveform->config.channels;
MA_ASSERT(pWaveform != NULL);
MA_ASSERT(pFramesOut != NULL);
if (pWaveform->config.format == ma_format_f32) {
float* pFramesOutF32 = (float*)pFramesOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_waveform_square_f32(pWaveform->time, pWaveform->config.amplitude);
pWaveform->time += pWaveform->advance;
for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
pFramesOutF32[iFrame*pWaveform->config.channels + iChannel] = s;
}
}
} else if (pWaveform->config.format == ma_format_s16) {
ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_int16 s = ma_waveform_square_s16(pWaveform->time, pWaveform->config.amplitude);
pWaveform->time += pWaveform->advance;
for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
pFramesOutS16[iFrame*pWaveform->config.channels + iChannel] = s;
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_waveform_square_f32(pWaveform->time, pWaveform->config.amplitude);
pWaveform->time += pWaveform->advance;
for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pWaveform->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
}
}
}
}
static void ma_waveform_read_pcm_frames__triangle(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
{
ma_uint64 iFrame;
ma_uint64 iChannel;
ma_uint32 bps = ma_get_bytes_per_sample(pWaveform->config.format);
ma_uint32 bpf = bps * pWaveform->config.channels;
MA_ASSERT(pWaveform != NULL);
MA_ASSERT(pFramesOut != NULL);
if (pWaveform->config.format == ma_format_f32) {
float* pFramesOutF32 = (float*)pFramesOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_waveform_triangle_f32(pWaveform->time, pWaveform->config.amplitude);
pWaveform->time += pWaveform->advance;
for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
pFramesOutF32[iFrame*pWaveform->config.channels + iChannel] = s;
}
}
} else if (pWaveform->config.format == ma_format_s16) {
ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_int16 s = ma_waveform_triangle_s16(pWaveform->time, pWaveform->config.amplitude);
pWaveform->time += pWaveform->advance;
for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
pFramesOutS16[iFrame*pWaveform->config.channels + iChannel] = s;
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_waveform_triangle_f32(pWaveform->time, pWaveform->config.amplitude);
pWaveform->time += pWaveform->advance;
for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pWaveform->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
}
}
}
}
static void ma_waveform_read_pcm_frames__sawtooth(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount)
{
ma_uint64 iFrame;
ma_uint64 iChannel;
ma_uint32 bps = ma_get_bytes_per_sample(pWaveform->config.format);
ma_uint32 bpf = bps * pWaveform->config.channels;
MA_ASSERT(pWaveform != NULL);
MA_ASSERT(pFramesOut != NULL);
if (pWaveform->config.format == ma_format_f32) {
float* pFramesOutF32 = (float*)pFramesOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_waveform_sawtooth_f32(pWaveform->time, pWaveform->config.amplitude);
pWaveform->time += pWaveform->advance;
for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
pFramesOutF32[iFrame*pWaveform->config.channels + iChannel] = s;
}
}
} else if (pWaveform->config.format == ma_format_s16) {
ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_int16 s = ma_waveform_sawtooth_s16(pWaveform->time, pWaveform->config.amplitude);
pWaveform->time += pWaveform->advance;
for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
pFramesOutS16[iFrame*pWaveform->config.channels + iChannel] = s;
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_waveform_sawtooth_f32(pWaveform->time, pWaveform->config.amplitude);
pWaveform->time += pWaveform->advance;
for (iChannel = 0; iChannel < pWaveform->config.channels; iChannel += 1) {
ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pWaveform->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
}
}
}
}
MA_API ma_result ma_waveform_read_pcm_frames(ma_waveform* pWaveform, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
if (frameCount == 0) {
return MA_INVALID_ARGS;
}
if (pWaveform == NULL) {
return MA_INVALID_ARGS;
}
if (pFramesOut != NULL) {
switch (pWaveform->config.type)
{
case ma_waveform_type_sine:
{
ma_waveform_read_pcm_frames__sine(pWaveform, pFramesOut, frameCount);
} break;
case ma_waveform_type_square:
{
ma_waveform_read_pcm_frames__square(pWaveform, pFramesOut, frameCount);
} break;
case ma_waveform_type_triangle:
{
ma_waveform_read_pcm_frames__triangle(pWaveform, pFramesOut, frameCount);
} break;
case ma_waveform_type_sawtooth:
{
ma_waveform_read_pcm_frames__sawtooth(pWaveform, pFramesOut, frameCount);
} break;
default: return MA_INVALID_OPERATION; /* Unknown waveform type. */
}
} else {
pWaveform->time += pWaveform->advance * (ma_int64)frameCount; /* Cast to int64 required for VC6. Won't affect anything in practice. */
}
if (pFramesRead != NULL) {
*pFramesRead = frameCount;
}
return MA_SUCCESS;
}
MA_API ma_result ma_waveform_seek_to_pcm_frame(ma_waveform* pWaveform, ma_uint64 frameIndex)
{
if (pWaveform == NULL) {
return MA_INVALID_ARGS;
}
pWaveform->time = pWaveform->advance * (ma_int64)frameIndex; /* Casting for VC6. Won't be an issue in practice. */
return MA_SUCCESS;
}
MA_API ma_noise_config ma_noise_config_init(ma_format format, ma_uint32 channels, ma_noise_type type, ma_int32 seed, double amplitude)
{
ma_noise_config config;
MA_ZERO_OBJECT(&config);
config.format = format;
config.channels = channels;
config.type = type;
config.seed = seed;
config.amplitude = amplitude;
if (config.seed == 0) {
config.seed = MA_DEFAULT_LCG_SEED;
}
return config;
}
static ma_result ma_noise__data_source_on_read(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
return ma_noise_read_pcm_frames((ma_noise*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
static ma_result ma_noise__data_source_on_seek(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
/* No-op. Just pretend to be successful. */
(void)pDataSource;
(void)frameIndex;
return MA_SUCCESS;
}
static ma_result ma_noise__data_source_on_get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
ma_noise* pNoise = (ma_noise*)pDataSource;
*pFormat = pNoise->config.format;
*pChannels = pNoise->config.channels;
*pSampleRate = 0; /* There is no notion of sample rate with noise generation. */
ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pNoise->config.channels);
return MA_SUCCESS;
}
static ma_data_source_vtable g_ma_noise_data_source_vtable =
{
ma_noise__data_source_on_read,
ma_noise__data_source_on_seek, /* No-op for noise. */
ma_noise__data_source_on_get_data_format,
NULL, /* onGetCursor. No notion of a cursor for noise. */
NULL, /* onGetLength. No notion of a length for noise. */
NULL, /* onSetLooping */
0
};
#ifndef MA_PINK_NOISE_BIN_SIZE
#define MA_PINK_NOISE_BIN_SIZE 16
#endif
typedef struct
{
size_t sizeInBytes;
struct
{
size_t binOffset;
size_t accumulationOffset;
size_t counterOffset;
} pink;
struct
{
size_t accumulationOffset;
} brownian;
} ma_noise_heap_layout;
static ma_result ma_noise_get_heap_layout(const ma_noise_config* pConfig, ma_noise_heap_layout* pHeapLayout)
{
MA_ASSERT(pHeapLayout != NULL);
MA_ZERO_OBJECT(pHeapLayout);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->channels == 0) {
return MA_INVALID_ARGS;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
pNoise->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
MA_API void ma_noise_uninit(ma_noise* pNoise, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pNoise == NULL) {
return;
}
ma_data_source_uninit(&pNoise->ds);
if (pNoise->_ownsHeap) {
ma_free(pNoise->_pHeap, pAllocationCallbacks);
}
}
MA_API ma_result ma_noise_set_amplitude(ma_noise* pNoise, double amplitude)
{
if (pNoise == NULL) {
return MA_INVALID_ARGS;
}
pNoise->config.amplitude = amplitude;
return MA_SUCCESS;
}
MA_API ma_result ma_noise_set_seed(ma_noise* pNoise, ma_int32 seed)
{
if (pNoise == NULL) {
return MA_INVALID_ARGS;
}
pNoise->lcg.state = seed;
return MA_SUCCESS;
}
MA_API ma_result ma_noise_set_type(ma_noise* pNoise, ma_noise_type type)
{
if (pNoise == NULL) {
return MA_INVALID_ARGS;
}
pNoise->config.type = type;
return MA_SUCCESS;
}
static MA_INLINE float ma_noise_f32_white(ma_noise* pNoise)
{
return (float)(ma_lcg_rand_f64(&pNoise->lcg) * pNoise->config.amplitude);
}
static MA_INLINE ma_int16 ma_noise_s16_white(ma_noise* pNoise)
{
return ma_pcm_sample_f32_to_s16(ma_noise_f32_white(pNoise));
}
static MA_INLINE ma_uint64 ma_noise_read_pcm_frames__white(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount)
{
ma_uint64 iFrame;
ma_uint32 iChannel;
const ma_uint32 channels = pNoise->config.channels;
MA_ASSUME(channels > 0);
if (pNoise->config.format == ma_format_f32) {
float* pFramesOutF32 = (float*)pFramesOut;
if (pNoise->config.duplicateChannels) {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_noise_f32_white(pNoise);
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOutF32[iFrame*channels + iChannel] = s;
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOutF32[iFrame*channels + iChannel] = ma_noise_f32_white(pNoise);
}
}
}
} else if (pNoise->config.format == ma_format_s16) {
ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
if (pNoise->config.duplicateChannels) {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_int16 s = ma_noise_s16_white(pNoise);
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOutS16[iFrame*channels + iChannel] = s;
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOutS16[iFrame*channels + iChannel] = ma_noise_s16_white(pNoise);
}
}
}
} else {
const ma_uint32 bps = ma_get_bytes_per_sample(pNoise->config.format);
const ma_uint32 bpf = bps * channels;
if (pNoise->config.duplicateChannels) {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_noise_f32_white(pNoise);
for (iChannel = 0; iChannel < channels; iChannel += 1) {
ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannel = 0; iChannel < channels; iChannel += 1) {
float s = ma_noise_f32_white(pNoise);
ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
}
}
}
}
return frameCount;
}
static MA_INLINE unsigned int ma_tzcnt32(unsigned int x)
{
unsigned int n;
/* Special case for odd numbers since they should happen about half the time. */
if (x & 0x1) {
return 0;
}
if (x == 0) {
return sizeof(x) << 3;
}
n = 1;
if ((x & 0x0000FFFF) == 0) { x >>= 16; n += 16; }
if ((x & 0x000000FF) == 0) { x >>= 8; n += 8; }
if ((x & 0x0000000F) == 0) { x >>= 4; n += 4; }
if ((x & 0x00000003) == 0) { x >>= 2; n += 2; }
n -= x & 0x00000001;
return n;
}
/*
Pink noise generation based on Tonic (public domain) with modifications. https://github.com/TonicAudio/Tonic/blob/master/src/Tonic/Noise.h
This is basically _the_ reference for pink noise from what I've found: http://www.firstpr.com.au/dsp/pink-noise/
*/
static MA_INLINE float ma_noise_f32_pink(ma_noise* pNoise, ma_uint32 iChannel)
{
double result;
double binPrev;
double binNext;
unsigned int ibin;
ibin = ma_tzcnt32(pNoise->state.pink.counter[iChannel]) & (MA_PINK_NOISE_BIN_SIZE - 1);
binPrev = pNoise->state.pink.bin[iChannel][ibin];
binNext = ma_lcg_rand_f64(&pNoise->lcg);
pNoise->state.pink.bin[iChannel][ibin] = binNext;
pNoise->state.pink.accumulation[iChannel] += (binNext - binPrev);
pNoise->state.pink.counter[iChannel] += 1;
result = (ma_lcg_rand_f64(&pNoise->lcg) + pNoise->state.pink.accumulation[iChannel]);
result /= 10;
return (float)(result * pNoise->config.amplitude);
}
static MA_INLINE ma_int16 ma_noise_s16_pink(ma_noise* pNoise, ma_uint32 iChannel)
{
return ma_pcm_sample_f32_to_s16(ma_noise_f32_pink(pNoise, iChannel));
}
static MA_INLINE ma_uint64 ma_noise_read_pcm_frames__pink(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount)
{
ma_uint64 iFrame;
ma_uint32 iChannel;
const ma_uint32 channels = pNoise->config.channels;
MA_ASSUME(channels > 0);
if (pNoise->config.format == ma_format_f32) {
float* pFramesOutF32 = (float*)pFramesOut;
if (pNoise->config.duplicateChannels) {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_noise_f32_pink(pNoise, 0);
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOutF32[iFrame*channels + iChannel] = s;
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOutF32[iFrame*channels + iChannel] = ma_noise_f32_pink(pNoise, iChannel);
}
}
}
} else if (pNoise->config.format == ma_format_s16) {
ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
if (pNoise->config.duplicateChannels) {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_int16 s = ma_noise_s16_pink(pNoise, 0);
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOutS16[iFrame*channels + iChannel] = s;
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOutS16[iFrame*channels + iChannel] = ma_noise_s16_pink(pNoise, iChannel);
}
}
}
} else {
const ma_uint32 bps = ma_get_bytes_per_sample(pNoise->config.format);
const ma_uint32 bpf = bps * channels;
if (pNoise->config.duplicateChannels) {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_noise_f32_pink(pNoise, 0);
for (iChannel = 0; iChannel < channels; iChannel += 1) {
ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannel = 0; iChannel < channels; iChannel += 1) {
float s = ma_noise_f32_pink(pNoise, iChannel);
ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
}
}
}
}
return frameCount;
}
static MA_INLINE float ma_noise_f32_brownian(ma_noise* pNoise, ma_uint32 iChannel)
{
double result;
result = (ma_lcg_rand_f64(&pNoise->lcg) + pNoise->state.brownian.accumulation[iChannel]);
result /= 1.005; /* Don't escape the -1..1 range on average. */
pNoise->state.brownian.accumulation[iChannel] = result;
result /= 20;
return (float)(result * pNoise->config.amplitude);
}
static MA_INLINE ma_int16 ma_noise_s16_brownian(ma_noise* pNoise, ma_uint32 iChannel)
{
return ma_pcm_sample_f32_to_s16(ma_noise_f32_brownian(pNoise, iChannel));
}
static MA_INLINE ma_uint64 ma_noise_read_pcm_frames__brownian(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount)
{
ma_uint64 iFrame;
ma_uint32 iChannel;
const ma_uint32 channels = pNoise->config.channels;
MA_ASSUME(channels > 0);
if (pNoise->config.format == ma_format_f32) {
float* pFramesOutF32 = (float*)pFramesOut;
if (pNoise->config.duplicateChannels) {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_noise_f32_brownian(pNoise, 0);
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOutF32[iFrame*channels + iChannel] = s;
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOutF32[iFrame*channels + iChannel] = ma_noise_f32_brownian(pNoise, iChannel);
}
}
}
} else if (pNoise->config.format == ma_format_s16) {
ma_int16* pFramesOutS16 = (ma_int16*)pFramesOut;
if (pNoise->config.duplicateChannels) {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
ma_int16 s = ma_noise_s16_brownian(pNoise, 0);
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOutS16[iFrame*channels + iChannel] = s;
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannel = 0; iChannel < channels; iChannel += 1) {
pFramesOutS16[iFrame*channels + iChannel] = ma_noise_s16_brownian(pNoise, iChannel);
}
}
}
} else {
const ma_uint32 bps = ma_get_bytes_per_sample(pNoise->config.format);
const ma_uint32 bpf = bps * channels;
if (pNoise->config.duplicateChannels) {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
float s = ma_noise_f32_brownian(pNoise, 0);
for (iChannel = 0; iChannel < channels; iChannel += 1) {
ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
}
}
} else {
for (iFrame = 0; iFrame < frameCount; iFrame += 1) {
for (iChannel = 0; iChannel < channels; iChannel += 1) {
float s = ma_noise_f32_brownian(pNoise, iChannel);
ma_pcm_convert(ma_offset_ptr(pFramesOut, iFrame*bpf + iChannel*bps), pNoise->config.format, &s, ma_format_f32, 1, ma_dither_mode_none);
}
}
}
}
return frameCount;
}
MA_API ma_result ma_noise_read_pcm_frames(ma_noise* pNoise, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
ma_uint64 framesRead = 0;
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
if (frameCount == 0) {
return MA_INVALID_ARGS;
}
if (pNoise == NULL) {
return MA_INVALID_ARGS;
}
/* The output buffer is allowed to be NULL. Since we aren't tracking cursors or anything we can just do nothing and pretend to be successful. */
if (pFramesOut == NULL) {
framesRead = frameCount;
} else {
switch (pNoise->config.type) {
case ma_noise_type_white: framesRead = ma_noise_read_pcm_frames__white (pNoise, pFramesOut, frameCount); break;
case ma_noise_type_pink: framesRead = ma_noise_read_pcm_frames__pink (pNoise, pFramesOut, frameCount); break;
case ma_noise_type_brownian: framesRead = ma_noise_read_pcm_frames__brownian(pNoise, pFramesOut, frameCount); break;
default: return MA_INVALID_OPERATION; /* Unknown noise type. */
}
}
if (pFramesRead != NULL) {
*pFramesRead = framesRead;
}
return MA_SUCCESS;
}
#endif /* MA_NO_GENERATION */
#ifndef MA_NO_RESOURCE_MANAGER
#ifndef MA_RESOURCE_MANAGER_PAGE_SIZE_IN_MILLISECONDS
#define MA_RESOURCE_MANAGER_PAGE_SIZE_IN_MILLISECONDS 1000
#endif
#ifndef MA_JOB_TYPE_RESOURCE_MANAGER_QUEUE_CAPACITY
#define MA_JOB_TYPE_RESOURCE_MANAGER_QUEUE_CAPACITY 1024
#endif
MA_API ma_resource_manager_pipeline_notifications ma_resource_manager_pipeline_notifications_init(void)
{
ma_resource_manager_pipeline_notifications notifications;
MA_ZERO_OBJECT(¬ifications);
return notifications;
}
static void ma_resource_manager_pipeline_notifications_signal_all_notifications(const ma_resource_manager_pipeline_notifications* pPipelineNotifications)
{
if (pPipelineNotifications == NULL) {
return;
}
if (pPipelineNotifications->init.pNotification) { ma_async_notification_signal(pPipelineNotifications->init.pNotification); }
if (pPipelineNotifications->done.pNotification) { ma_async_notification_signal(pPipelineNotifications->done.pNotification); }
}
static void ma_resource_manager_pipeline_notifications_acquire_all_fences(const ma_resource_manager_pipeline_notifications* pPipelineNotifications)
{
if (pPipelineNotifications == NULL) {
return;
}
if (pPipelineNotifications->init.pFence != NULL) { ma_fence_acquire(pPipelineNotifications->init.pFence); }
if (pPipelineNotifications->done.pFence != NULL) { ma_fence_acquire(pPipelineNotifications->done.pFence); }
}
static void ma_resource_manager_pipeline_notifications_release_all_fences(const ma_resource_manager_pipeline_notifications* pPipelineNotifications)
{
if (pPipelineNotifications == NULL) {
return;
}
if (pPipelineNotifications->init.pFence != NULL) { ma_fence_release(pPipelineNotifications->init.pFence); }
if (pPipelineNotifications->done.pFence != NULL) { ma_fence_release(pPipelineNotifications->done.pFence); }
}
#ifndef MA_DEFAULT_HASH_SEED
#define MA_DEFAULT_HASH_SEED 42
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
static ma_result ma_resource_manager_data_buffer_init_connector(ma_resource_manager_data_buffer* pDataBuffer, const ma_resource_manager_data_source_config* pConfig, ma_async_notification* pInitNotification, ma_fence* pInitFence)
{
ma_result result;
MA_ASSERT(pDataBuffer != NULL);
MA_ASSERT(pConfig != NULL);
MA_ASSERT(pDataBuffer->isConnectorInitialized == MA_FALSE);
/* The underlying data buffer must be initialized before we'll be able to know how to initialize the backend. */
result = ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode);
if (result != MA_SUCCESS && result != MA_BUSY) {
return result; /* The data buffer is in an erroneous state. */
}
/*
We need to initialize either a ma_decoder or an ma_audio_buffer depending on whether or not the backing data is encoded or decoded. These act as the
"instance" to the data and are used to form the connection between underlying data buffer and the data source. If the data buffer is decoded, we can use
an ma_audio_buffer. This enables us to use memory mapping when mixing which saves us a bit of data movement overhead.
*/
switch (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode))
{
case ma_resource_manager_data_supply_type_encoded: /* Connector is a decoder. */
{
ma_decoder_config config;
config = ma_resource_manager__init_decoder_config(pDataBuffer->pResourceManager);
result = ma_decoder_init_memory(pDataBuffer->pNode->data.backend.encoded.pData, pDataBuffer->pNode->data.backend.encoded.sizeInBytes, &config, &pDataBuffer->connector.decoder);
} break;
case ma_resource_manager_data_supply_type_decoded: /* Connector is an audio buffer. */
{
ma_audio_buffer_config config;
config = ma_audio_buffer_config_init(pDataBuffer->pNode->data.backend.decoded.format, pDataBuffer->pNode->data.backend.decoded.channels, pDataBuffer->pNode->data.backend.decoded.totalFrameCount, pDataBuffer->pNode->data.backend.decoded.pD...
result = ma_audio_buffer_init(&config, &pDataBuffer->connector.buffer);
} break;
case ma_resource_manager_data_supply_type_decoded_paged: /* Connector is a paged audio buffer. */
{
ma_paged_audio_buffer_config config;
config = ma_paged_audio_buffer_config_init(&pDataBuffer->pNode->data.backend.decodedPaged.data);
result = ma_paged_audio_buffer_init(&config, &pDataBuffer->connector.pagedBuffer);
} break;
case ma_resource_manager_data_supply_type_unknown:
default:
{
/* Unknown data supply type. Should never happen. Need to post an error here. */
return MA_INVALID_ARGS;
};
}
/*
Initialization of the connector is when we can fire the init notification. This will give the application access to
the format/channels/rate of the data source.
*/
if (result == MA_SUCCESS) {
/*
Make sure the looping state is set before returning in order to handle the case where the
loop state was set on the data buffer before the connector was initialized.
*/
ma_data_source_set_range_in_pcm_frames(pDataBuffer, pConfig->rangeBegInPCMFrames, pConfig->rangeEndInPCMFrames);
ma_data_source_set_loop_point_in_pcm_frames(pDataBuffer, pConfig->loopPointBegInPCMFrames, pConfig->loopPointEndInPCMFrames);
ma_data_source_set_looping(pDataBuffer, pConfig->isLooping);
pDataBuffer->isConnectorInitialized = MA_TRUE;
if (pInitNotification != NULL) {
ma_async_notification_signal(pInitNotification);
}
if (pInitFence != NULL) {
ma_fence_release(pInitFence);
}
}
/* At this point the backend should be initialized. We do *not* want to set pDataSource->result here - that needs to be done at a higher level to ensure it's done as the last step. */
return result;
}
static ma_result ma_resource_manager_data_buffer_uninit_connector(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer* pDataBuffer)
{
MA_ASSERT(pResourceManager != NULL);
MA_ASSERT(pDataBuffer != NULL);
switch (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode))
{
case ma_resource_manager_data_supply_type_encoded: /* Connector is a decoder. */
{
ma_decoder_uninit(&pDataBuffer->connector.decoder);
} break;
case ma_resource_manager_data_supply_type_decoded: /* Connector is an audio buffer. */
{
ma_audio_buffer_uninit(&pDataBuffer->connector.buffer);
} break;
case ma_resource_manager_data_supply_type_decoded_paged: /* Connector is a paged audio buffer. */
{
ma_paged_audio_buffer_uninit(&pDataBuffer->connector.pagedBuffer);
} break;
case ma_resource_manager_data_supply_type_unknown:
default:
{
/* Unknown data supply type. Should never happen. Need to post an error here. */
return MA_INVALID_ARGS;
};
}
return MA_SUCCESS;
}
static ma_uint32 ma_resource_manager_data_buffer_node_next_execution_order(ma_resource_manager_data_buffer_node* pDataBufferNode)
{
MA_ASSERT(pDataBufferNode != NULL);
return c89atomic_fetch_add_32(&pDataBufferNode->executionCounter, 1);
}
static ma_result ma_resource_manager_data_buffer_node_init_supply_encoded(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode, const char* pFilePath, const wchar_t* pFilePathW)
{
ma_result result;
size_t dataSizeInBytes;
void* pData;
MA_ASSERT(pResourceManager != NULL);
MA_ASSERT(pDataBufferNode != NULL);
MA_ASSERT(pFilePath != NULL || pFilePathW != NULL);
result = ma_vfs_open_and_read_file_ex(pResourceManager->config.pVFS, pFilePath, pFilePathW, &pData, &dataSizeInBytes, &pResourceManager->config.allocationCallbacks);
if (result != MA_SUCCESS) {
if (pFilePath != NULL) {
ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_WARNING, "Failed to load file \"%s\". %s.\n", pFilePath, ma_result_description(result));
} else {
#if (defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L) || defined(_MSC_VER)
ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_WARNING, "Failed to load file \"%ls\". %s.\n", pFilePathW, ma_result_description(result));
#endif
}
return result;
}
pDataBufferNode->data.backend.encoded.pData = pData;
pDataBufferNode->data.backend.encoded.sizeInBytes = dataSizeInBytes;
ma_resource_manager_data_buffer_node_set_data_supply_type(pDataBufferNode, ma_resource_manager_data_supply_type_encoded); /* <-- Must be set last. */
return MA_SUCCESS;
}
static ma_result ma_resource_manager_data_buffer_node_init_supply_decoded(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode, const char* pFilePath, const wchar_t* pFilePathW, ma_uint32 flags, ma_decoder** pp...
{
ma_result result = MA_SUCCESS;
ma_decoder* pDecoder;
ma_uint64 totalFrameCount;
MA_ASSERT(pResourceManager != NULL);
MA_ASSERT(pDataBufferNode != NULL);
MA_ASSERT(ppDecoder != NULL);
MA_ASSERT(pFilePath != NULL || pFilePathW != NULL);
*ppDecoder = NULL; /* For safety. */
pDecoder = (ma_decoder*)ma_malloc(sizeof(*pDecoder), &pResourceManager->config.allocationCallbacks);
if (pDecoder == NULL) {
return MA_OUT_OF_MEMORY;
}
result = ma_resource_manager__init_decoder(pResourceManager, pFilePath, pFilePathW, pDecoder);
if (result != MA_SUCCESS) {
ma_free(pDecoder, &pResourceManager->config.allocationCallbacks);
return result;
}
/*
At this point we have the decoder and we now need to initialize the data supply. This will
be either a decoded buffer, or a decoded paged buffer. A regular buffer is just one big heap
allocated buffer, whereas a paged buffer is a linked list of paged-sized buffers. The latter
is used when the length of a sound is unknown until a full decode has been performed.
*/
if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_UNKNOWN_LENGTH) == 0) {
result = ma_decoder_get_length_in_pcm_frames(pDecoder, &totalFrameCount);
if (result != MA_SUCCESS) {
return result;
}
} else {
totalFrameCount = 0;
}
if (totalFrameCount > 0) {
/* It's a known length. The data supply is a regular decoded buffer. */
ma_uint64 dataSizeInBytes;
void* pData;
dataSizeInBytes = totalFrameCount * ma_get_bytes_per_frame(pDecoder->outputFormat, pDecoder->outputChannels);
if (dataSizeInBytes > MA_SIZE_MAX) {
ma_decoder_uninit(pDecoder);
ma_free(pDecoder, &pResourceManager->config.allocationCallbacks);
return MA_TOO_BIG;
}
pData = ma_malloc((size_t)dataSizeInBytes, &pResourceManager->config.allocationCallbacks);
if (pData == NULL) {
ma_decoder_uninit(pDecoder);
ma_free(pDecoder, &pResourceManager->config.allocationCallbacks);
return MA_OUT_OF_MEMORY;
}
/* The buffer needs to be initialized to silence in case the caller reads from it. */
ma_silence_pcm_frames(pData, totalFrameCount, pDecoder->outputFormat, pDecoder->outputChannels);
/* Data has been allocated and the data supply can now be initialized. */
pDataBufferNode->data.backend.decoded.pData = pData;
pDataBufferNode->data.backend.decoded.totalFrameCount = totalFrameCount;
pDataBufferNode->data.backend.decoded.format = pDecoder->outputFormat;
pDataBufferNode->data.backend.decoded.channels = pDecoder->outputChannels;
pDataBufferNode->data.backend.decoded.sampleRate = pDecoder->outputSampleRate;
pDataBufferNode->data.backend.decoded.decodedFrameCount = 0;
ma_resource_manager_data_buffer_node_set_data_supply_type(pDataBufferNode, ma_resource_manager_data_supply_type_decoded); /* <-- Must be set last. */
} else {
/*
It's an unknown length. The data supply is a paged decoded buffer. Setting this up is
actually easier than the non-paged decoded buffer because we just need to initialize
a ma_paged_audio_buffer object.
*/
result = ma_paged_audio_buffer_data_init(pDecoder->outputFormat, pDecoder->outputChannels, &pDataBufferNode->data.backend.decodedPaged.data);
if (result != MA_SUCCESS) {
ma_decoder_uninit(pDecoder);
ma_free(pDecoder, &pResourceManager->config.allocationCallbacks);
return result;
}
pDataBufferNode->data.backend.decodedPaged.sampleRate = pDecoder->outputSampleRate;
pDataBufferNode->data.backend.decodedPaged.decodedFrameCount = 0;
ma_resource_manager_data_buffer_node_set_data_supply_type(pDataBufferNode, ma_resource_manager_data_supply_type_decoded_paged); /* <-- Must be set last. */
}
*ppDecoder = pDecoder;
return MA_SUCCESS;
}
static ma_result ma_resource_manager_data_buffer_node_decode_next_page(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode, ma_decoder* pDecoder)
{
ma_result result = MA_SUCCESS;
ma_uint64 pageSizeInFrames;
ma_uint64 framesToTryReading;
ma_uint64 framesRead;
MA_ASSERT(pResourceManager != NULL);
MA_ASSERT(pDataBufferNode != NULL);
MA_ASSERT(pDecoder != NULL);
/* We need to know the size of a page in frames to know how many frames to decode. */
pageSizeInFrames = MA_RESOURCE_MANAGER_PAGE_SIZE_IN_MILLISECONDS * (pDecoder->outputSampleRate/1000);
framesToTryReading = pageSizeInFrames;
/*
Here is where we do the decoding of the next page. We'll run a slightly different path depending
on whether or not we're using a flat or paged buffer because the allocation of the page differs
between the two. For a flat buffer it's an offset to an already-allocated buffer. For a paged
buffer, we need to allocate a new page and attach it to the linked list.
*/
switch (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBufferNode))
{
case ma_resource_manager_data_supply_type_decoded:
{
/* The destination buffer is an offset to the existing buffer. Don't read more than we originally retrieved when we first initialized the decoder. */
void* pDst;
ma_uint64 framesRemaining = pDataBufferNode->data.backend.decoded.totalFrameCount - pDataBufferNode->data.backend.decoded.decodedFrameCount;
if (framesToTryReading > framesRemaining) {
framesToTryReading = framesRemaining;
}
if (framesToTryReading > 0) {
pDst = ma_offset_ptr(
pDataBufferNode->data.backend.decoded.pData,
pDataBufferNode->data.backend.decoded.decodedFrameCount * ma_get_bytes_per_frame(pDataBufferNode->data.backend.decoded.format, pDataBufferNode->data.backend.decoded.channels)
);
MA_ASSERT(pDst != NULL);
result = ma_decoder_read_pcm_frames(pDecoder, pDst, framesToTryReading, &framesRead);
if (framesRead > 0) {
pDataBufferNode->data.backend.decoded.decodedFrameCount += framesRead;
}
} else {
framesRead = 0;
}
} break;
case ma_resource_manager_data_supply_type_decoded_paged:
{
/* The destination buffer is a freshly allocated page. */
ma_paged_audio_buffer_page* pPage;
result = ma_paged_audio_buffer_data_allocate_page(&pDataBufferNode->data.backend.decodedPaged.data, framesToTryReading, NULL, &pResourceManager->config.allocationCallbacks, &pPage);
if (result != MA_SUCCESS) {
return result;
}
result = ma_decoder_read_pcm_frames(pDecoder, pPage->pAudioData, framesToTryReading, &framesRead);
if (framesRead > 0) {
pPage->sizeInFrames = framesRead;
result = ma_paged_audio_buffer_data_append_page(&pDataBufferNode->data.backend.decodedPaged.data, pPage);
if (result == MA_SUCCESS) {
pDataBufferNode->data.backend.decodedPaged.decodedFrameCount += framesRead;
} else {
/* Failed to append the page. Just abort and set the status to MA_AT_END. */
ma_paged_audio_buffer_data_free_page(&pDataBufferNode->data.backend.decodedPaged.data, pPage, &pResourceManager->config.allocationCallbacks);
result = MA_AT_END;
}
} else {
/* No frames were read. Free the page and just set the status to MA_AT_END. */
ma_paged_audio_buffer_data_free_page(&pDataBufferNode->data.backend.decodedPaged.data, pPage, &pResourceManager->config.allocationCallbacks);
result = MA_AT_END;
}
} break;
case ma_resource_manager_data_supply_type_encoded:
case ma_resource_manager_data_supply_type_unknown:
default:
{
/* Unexpected data supply type. */
ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_ERROR, "Unexpected data supply type (%d) when decoding page.", ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBufferNode));
return MA_ERROR;
};
}
if (result == MA_SUCCESS && framesRead == 0) {
result = MA_AT_END;
}
return result;
}
static ma_result ma_resource_manager_data_buffer_node_acquire_critical_section(ma_resource_manager* pResourceManager, const char* pFilePath, const wchar_t* pFilePathW, ma_uint32 hashedName32, ma_uint32 flags, const ma_resource_manager_data_supply* pE...
{
ma_result result = MA_SUCCESS;
ma_resource_manager_data_buffer_node* pDataBufferNode = NULL;
ma_resource_manager_data_buffer_node* pInsertPoint;
if (ppDataBufferNode != NULL) {
*ppDataBufferNode = NULL;
}
result = ma_resource_manager_data_buffer_node_insert_point(pResourceManager, hashedName32, &pInsertPoint);
if (result == MA_ALREADY_EXISTS) {
/* The node already exists. We just need to increment the reference count. */
pDataBufferNode = pInsertPoint;
result = ma_resource_manager_data_buffer_node_increment_ref(pResourceManager, pDataBufferNode, NULL);
if (result != MA_SUCCESS) {
return result; /* Should never happen. Failed to increment the reference count. */
}
result = MA_ALREADY_EXISTS;
goto done;
} else {
/*
The node does not already exist. We need to post a LOAD_DATA_BUFFER_NODE job here. This
needs to be done inside the critical section to ensure an uninitialization of the node
does not occur before initialization on another thread.
*/
pDataBufferNode = (ma_resource_manager_data_buffer_node*)ma_malloc(sizeof(*pDataBufferNode), &pResourceManager->config.allocationCallbacks);
if (pDataBufferNode == NULL) {
return MA_OUT_OF_MEMORY;
}
MA_ZERO_OBJECT(pDataBufferNode);
pDataBufferNode->hashedName32 = hashedName32;
pDataBufferNode->refCount = 1; /* Always set to 1 by default (this is our first reference). */
if (pExistingData == NULL) {
pDataBufferNode->data.type = ma_resource_manager_data_supply_type_unknown; /* <-- We won't know this until we start decoding. */
pDataBufferNode->result = MA_BUSY; /* Must be set to MA_BUSY before we leave the critical section, so might as well do it now. */
pDataBufferNode->isDataOwnedByResourceManager = MA_TRUE;
} else {
pDataBufferNode->data = *pExistingData;
pDataBufferNode->result = MA_SUCCESS; /* Not loading asynchronously, so just set the status */
pDataBufferNode->isDataOwnedByResourceManager = MA_FALSE;
}
result = ma_resource_manager_data_buffer_node_insert_at(pResourceManager, pDataBufferNode, pInsertPoint);
if (result != MA_SUCCESS) {
ma_free(pDataBufferNode, &pResourceManager->config.allocationCallbacks);
return result; /* Should never happen. Failed to insert the data buffer into the BST. */
}
/*
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
ma_resource_manager_data_buffer_bst_unlock(pResourceManager);
stage2:
if (result != MA_SUCCESS) {
return result;
}
/*
Here is where we need to free the node. We don't want to do this inside the critical section
above because we want to keep that as small as possible for multi-threaded efficiency.
*/
if (refCount == 0) {
if (ma_resource_manager_data_buffer_node_result(pDataBufferNode) == MA_BUSY) {
/* The sound is still loading. We need to delay the freeing of the node to a safe time. */
ma_job job;
/* We need to mark the node as unavailable for the sake of the resource manager worker threads. */
c89atomic_exchange_i32(&pDataBufferNode->result, MA_UNAVAILABLE);
job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER_NODE);
job.order = ma_resource_manager_data_buffer_node_next_execution_order(pDataBufferNode);
job.data.resourceManager.freeDataBufferNode.pResourceManager = pResourceManager;
job.data.resourceManager.freeDataBufferNode.pDataBufferNode = pDataBufferNode;
result = ma_resource_manager_post_job(pResourceManager, &job);
if (result != MA_SUCCESS) {
ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_ERROR, "Failed to post MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER_NODE job. %s.\n", ma_result_description(result));
return result;
}
/* If we don't support threading, process the job queue here. */
if (ma_resource_manager_is_threading_enabled(pResourceManager) == MA_FALSE) {
while (ma_resource_manager_data_buffer_node_result(pDataBufferNode) == MA_BUSY) {
result = ma_resource_manager_process_next_job(pResourceManager);
if (result == MA_NO_DATA_AVAILABLE || result == MA_CANCELLED) {
result = MA_SUCCESS;
break;
}
}
} else {
/* Threading is enabled. The job queue will deal with the rest of the cleanup from here. */
}
} else {
/* The sound isn't loading so we can just free the node here. */
ma_resource_manager_data_buffer_node_free(pResourceManager, pDataBufferNode);
}
}
return result;
}
static ma_uint32 ma_resource_manager_data_buffer_next_execution_order(ma_resource_manager_data_buffer* pDataBuffer)
{
MA_ASSERT(pDataBuffer != NULL);
return c89atomic_fetch_add_32(&pDataBuffer->executionCounter, 1);
}
static ma_result ma_resource_manager_data_buffer_cb__read_pcm_frames(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
return ma_resource_manager_data_buffer_read_pcm_frames((ma_resource_manager_data_buffer*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
static ma_result ma_resource_manager_data_buffer_cb__seek_to_pcm_frame(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
return ma_resource_manager_data_buffer_seek_to_pcm_frame((ma_resource_manager_data_buffer*)pDataSource, frameIndex);
}
static ma_result ma_resource_manager_data_buffer_cb__get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
return ma_resource_manager_data_buffer_get_data_format((ma_resource_manager_data_buffer*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
static ma_result ma_resource_manager_data_buffer_cb__get_cursor_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pCursor)
{
return ma_resource_manager_data_buffer_get_cursor_in_pcm_frames((ma_resource_manager_data_buffer*)pDataSource, pCursor);
}
static ma_result ma_resource_manager_data_buffer_cb__get_length_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLength)
{
return ma_resource_manager_data_buffer_get_length_in_pcm_frames((ma_resource_manager_data_buffer*)pDataSource, pLength);
}
static ma_result ma_resource_manager_data_buffer_cb__set_looping(ma_data_source* pDataSource, ma_bool32 isLooping)
{
ma_resource_manager_data_buffer* pDataBuffer = (ma_resource_manager_data_buffer*)pDataSource;
MA_ASSERT(pDataBuffer != NULL);
c89atomic_exchange_32(&pDataBuffer->isLooping, isLooping);
/* The looping state needs to be set on the connector as well or else looping won't work when we read audio data. */
ma_data_source_set_looping(ma_resource_manager_data_buffer_get_connector(pDataBuffer), isLooping);
return MA_SUCCESS;
}
static ma_data_source_vtable g_ma_resource_manager_data_buffer_vtable =
{
ma_resource_manager_data_buffer_cb__read_pcm_frames,
ma_resource_manager_data_buffer_cb__seek_to_pcm_frame,
ma_resource_manager_data_buffer_cb__get_data_format,
ma_resource_manager_data_buffer_cb__get_cursor_in_pcm_frames,
ma_resource_manager_data_buffer_cb__get_length_in_pcm_frames,
ma_resource_manager_data_buffer_cb__set_looping,
0
};
static ma_result ma_resource_manager_data_buffer_init_ex_internal(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_uint32 hashedName32, ma_resource_manager_data_buffer* pDataBuffer)
{
ma_result result = MA_SUCCESS;
ma_resource_manager_data_buffer_node* pDataBufferNode;
ma_data_source_config dataSourceConfig;
ma_bool32 async;
ma_uint32 flags;
ma_resource_manager_pipeline_notifications notifications;
if (pDataBuffer == NULL) {
if (pConfig != NULL && pConfig->pNotifications != NULL) {
ma_resource_manager_pipeline_notifications_signal_all_notifications(pConfig->pNotifications);
}
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pDataBuffer);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->pNotifications != NULL) {
notifications = *pConfig->pNotifications; /* From here on out we should be referencing `notifications` instead of `pNotifications`. Set this to NULL to catch errors at testing time. */
} else {
MA_ZERO_OBJECT(¬ifications);
}
/* For safety, always remove the ASYNC flag if threading is disabled on the resource manager. */
flags = pConfig->flags;
if (ma_resource_manager_is_threading_enabled(pResourceManager) == MA_FALSE) {
flags &= ~MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC;
}
async = (flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC) != 0;
/*
Fences need to be acquired before doing anything. These must be aquired and released outside of
the node to ensure there's no holes where ma_fence_wait() could prematurely return before the
data buffer has completed initialization.
When loading asynchronously, the node acquisition routine below will acquire the fences on this
thread and then release them on the async thread when the operation is complete.
These fences are always released at the "done" tag at the end of this function. They'll be
acquired a second if loading asynchronously. This double acquisition system is just done to
simplify code maintanence.
*/
ma_resource_manager_pipeline_notifications_acquire_all_fences(¬ifications);
{
/* We first need to acquire a node. If ASYNC is not set, this will not return until the entire sound has been loaded. */
result = ma_resource_manager_data_buffer_node_acquire(pResourceManager, pConfig->pFilePath, pConfig->pFilePathW, hashedName32, flags, NULL, notifications.init.pFence, notifications.done.pFence, &pDataBufferNode);
if (result != MA_SUCCESS) {
ma_resource_manager_pipeline_notifications_signal_all_notifications(¬ifications);
goto done;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
worker thread is aware of it's busy state. If the LOAD_DATA_BUFFER job sees a status other
than MA_BUSY, it'll assume an error and fall through to an early exit.
*/
c89atomic_exchange_i32(&pDataBuffer->result, MA_BUSY);
/* Acquire fences a second time. These will be released by the async thread. */
ma_resource_manager_pipeline_notifications_acquire_all_fences(¬ifications);
job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER);
job.order = ma_resource_manager_data_buffer_next_execution_order(pDataBuffer);
job.data.resourceManager.loadDataBuffer.pDataBuffer = pDataBuffer;
job.data.resourceManager.loadDataBuffer.pInitNotification = ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) ? &initNotification : notifications.init.pNotification;
job.data.resourceManager.loadDataBuffer.pDoneNotification = notifications.done.pNotification;
job.data.resourceManager.loadDataBuffer.pInitFence = notifications.init.pFence;
job.data.resourceManager.loadDataBuffer.pDoneFence = notifications.done.pFence;
job.data.resourceManager.loadDataBuffer.rangeBegInPCMFrames = pConfig->rangeBegInPCMFrames;
job.data.resourceManager.loadDataBuffer.rangeEndInPCMFrames = pConfig->rangeEndInPCMFrames;
job.data.resourceManager.loadDataBuffer.loopPointBegInPCMFrames = pConfig->loopPointBegInPCMFrames;
job.data.resourceManager.loadDataBuffer.loopPointEndInPCMFrames = pConfig->loopPointEndInPCMFrames;
job.data.resourceManager.loadDataBuffer.isLooping = pConfig->isLooping;
result = ma_resource_manager_post_job(pResourceManager, &job);
if (result != MA_SUCCESS) {
/* We failed to post the job. Most likely there isn't enough room in the queue's buffer. */
ma_log_postf(ma_resource_manager_get_log(pResourceManager), MA_LOG_LEVEL_ERROR, "Failed to post MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_BUFFER job. %s.\n", ma_result_description(result));
c89atomic_exchange_i32(&pDataBuffer->result, result);
/* Release the fences after the result has been set on the data buffer. */
ma_resource_manager_pipeline_notifications_release_all_fences(¬ifications);
} else {
if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) {
ma_resource_manager_inline_notification_wait(&initNotification);
if (notifications.init.pNotification != NULL) {
ma_async_notification_signal(notifications.init.pNotification);
}
/* NOTE: Do not release the init fence here. It will have been done by the job. */
/* Make sure we return an error if initialization failed on the async thread. */
result = ma_resource_manager_data_buffer_result(pDataBuffer);
if (result == MA_BUSY) {
result = MA_SUCCESS;
}
}
}
if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) {
ma_resource_manager_inline_notification_uninit(&initNotification);
}
}
if (result != MA_SUCCESS) {
ma_resource_manager_data_buffer_node_unacquire(pResourceManager, pDataBufferNode, NULL, NULL);
goto done;
}
}
done:
if (result == MA_SUCCESS) {
if (pConfig->initialSeekPointInPCMFrames > 0) {
ma_resource_manager_data_buffer_seek_to_pcm_frame(pDataBuffer, pConfig->initialSeekPointInPCMFrames);
}
}
ma_resource_manager_pipeline_notifications_release_all_fences(¬ifications);
return result;
}
MA_API ma_result ma_resource_manager_data_buffer_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_buffer* pDataBuffer)
{
return ma_resource_manager_data_buffer_init_ex_internal(pResourceManager, pConfig, 0, pDataBuffer);
}
MA_API ma_result ma_resource_manager_data_buffer_init(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_buffer* pDataBuffer)
{
ma_resource_manager_data_source_config config;
config = ma_resource_manager_data_source_config_init();
config.pFilePath = pFilePath;
config.flags = flags;
config.pNotifications = pNotifications;
return ma_resource_manager_data_buffer_init_ex(pResourceManager, &config, pDataBuffer);
}
MA_API ma_result ma_resource_manager_data_buffer_init_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_buffer* pDataBuffer)
{
ma_resource_manager_data_source_config config;
config = ma_resource_manager_data_source_config_init();
config.pFilePathW = pFilePath;
config.flags = flags;
config.pNotifications = pNotifications;
return ma_resource_manager_data_buffer_init_ex(pResourceManager, &config, pDataBuffer);
}
MA_API ma_result ma_resource_manager_data_buffer_init_copy(ma_resource_manager* pResourceManager, const ma_resource_manager_data_buffer* pExistingDataBuffer, ma_resource_manager_data_buffer* pDataBuffer)
{
ma_resource_manager_data_source_config config;
if (pExistingDataBuffer == NULL) {
return MA_INVALID_ARGS;
}
MA_ASSERT(pExistingDataBuffer->pNode != NULL); /* <-- If you've triggered this, you've passed in an invalid existing data buffer. */
config = ma_resource_manager_data_source_config_init();
config.flags = pExistingDataBuffer->flags;
return ma_resource_manager_data_buffer_init_ex_internal(pResourceManager, &config, pExistingDataBuffer->pNode->hashedName32, pDataBuffer);
}
static ma_result ma_resource_manager_data_buffer_uninit_internal(ma_resource_manager_data_buffer* pDataBuffer)
{
MA_ASSERT(pDataBuffer != NULL);
/* The connector should be uninitialized first. */
ma_resource_manager_data_buffer_uninit_connector(pDataBuffer->pResourceManager, pDataBuffer);
/* With the connector uninitialized we can unacquire the node. */
ma_resource_manager_data_buffer_node_unacquire(pDataBuffer->pResourceManager, pDataBuffer->pNode, NULL, NULL);
/* The base data source needs to be uninitialized as well. */
ma_data_source_uninit(&pDataBuffer->ds);
return MA_SUCCESS;
}
MA_API ma_result ma_resource_manager_data_buffer_uninit(ma_resource_manager_data_buffer* pDataBuffer)
{
ma_result result;
if (pDataBuffer == NULL) {
return MA_INVALID_ARGS;
}
if (ma_resource_manager_data_buffer_result(pDataBuffer) == MA_SUCCESS) {
/* The data buffer can be deleted synchronously. */
return ma_resource_manager_data_buffer_uninit_internal(pDataBuffer);
} else {
/*
The data buffer needs to be deleted asynchronously because it's still loading. With the status set to MA_UNAVAILABLE, no more pages will
be loaded and the uninitialization should happen fairly quickly. Since the caller owns the data buffer, we need to wait for this event
to get processed before returning.
*/
ma_resource_manager_inline_notification notification;
ma_job job;
/*
We need to mark the node as unavailable so we don't try reading from it anymore, but also to
let the loading thread know that it needs to abort it's loading procedure.
*/
c89atomic_exchange_i32(&pDataBuffer->result, MA_UNAVAILABLE);
result = ma_resource_manager_inline_notification_init(pDataBuffer->pResourceManager, ¬ification);
if (result != MA_SUCCESS) {
return result; /* Failed to create the notification. This should rarely, if ever, happen. */
}
job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER);
job.order = ma_resource_manager_data_buffer_next_execution_order(pDataBuffer);
job.data.resourceManager.freeDataBuffer.pDataBuffer = pDataBuffer;
job.data.resourceManager.freeDataBuffer.pDoneNotification = ¬ification;
job.data.resourceManager.freeDataBuffer.pDoneFence = NULL;
result = ma_resource_manager_post_job(pDataBuffer->pResourceManager, &job);
if (result != MA_SUCCESS) {
ma_resource_manager_inline_notification_uninit(¬ification);
return result;
}
ma_resource_manager_inline_notification_wait_and_uninit(¬ification);
}
return result;
}
MA_API ma_result ma_resource_manager_data_buffer_read_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
ma_result result = MA_SUCCESS;
ma_uint64 framesRead = 0;
ma_bool32 isDecodedBufferBusy = MA_FALSE;
/* Safety. */
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
if (frameCount == 0) {
return MA_INVALID_ARGS;
}
/*
We cannot be using the data buffer after it's been uninitialized. If you trigger this assert it means you're trying to read from the data buffer after
it's been uninitialized or is in the process of uninitializing.
*/
MA_ASSERT(ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) != MA_UNAVAILABLE);
/* If the node is not initialized we need to abort with a busy code. */
if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode) == ma_resource_manager_data_supply_type_unknown) {
return MA_BUSY; /* Still loading. */
}
if (pDataBuffer->seekToCursorOnNextRead) {
pDataBuffer->seekToCursorOnNextRead = MA_FALSE;
result = ma_data_source_seek_to_pcm_frame(ma_resource_manager_data_buffer_get_connector(pDataBuffer), pDataBuffer->seekTargetInPCMFrames);
if (result != MA_SUCCESS) {
return result;
}
}
/*
For decoded buffers (not paged) we need to check beforehand how many frames we have available. We cannot
exceed this amount. We'll read as much as we can, and then return MA_BUSY.
*/
if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode) == ma_resource_manager_data_supply_type_decoded) {
ma_uint64 availableFrames;
isDecodedBufferBusy = (ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) == MA_BUSY);
if (ma_resource_manager_data_buffer_get_available_frames(pDataBuffer, &availableFrames) == MA_SUCCESS) {
/* Don't try reading more than the available frame count. */
if (frameCount > availableFrames) {
frameCount = availableFrames;
/*
If there's no frames available we want to set the status to MA_AT_END. The logic below
will check if the node is busy, and if so, change it to MA_BUSY. The reason we do this
is because we don't want to call `ma_data_source_read_pcm_frames()` if the frame count
is 0 because that'll result in a situation where it's possible MA_AT_END won't get
returned.
*/
if (frameCount == 0) {
result = MA_AT_END;
}
} else {
isDecodedBufferBusy = MA_FALSE; /* We have enough frames available in the buffer to avoid a MA_BUSY status. */
}
}
}
/* Don't attempt to read anything if we've got no frames available. */
if (frameCount > 0) {
result = ma_data_source_read_pcm_frames(ma_resource_manager_data_buffer_get_connector(pDataBuffer), pFramesOut, frameCount, &framesRead);
}
/*
If we returned MA_AT_END, but the node is still loading, we don't want to return that code or else the caller will interpret the sound
as at the end and terminate decoding.
*/
if (result == MA_AT_END) {
if (ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) == MA_BUSY) {
result = MA_BUSY;
}
}
if (isDecodedBufferBusy) {
result = MA_BUSY;
}
if (pFramesRead != NULL) {
*pFramesRead = framesRead;
}
if (result == MA_SUCCESS && framesRead == 0) {
result = MA_AT_END;
}
return result;
}
MA_API ma_result ma_resource_manager_data_buffer_seek_to_pcm_frame(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64 frameIndex)
{
ma_result result;
/* We cannot be using the data source after it's been uninitialized. */
MA_ASSERT(ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) != MA_UNAVAILABLE);
/* If we haven't yet got a connector we need to abort. */
if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode) == ma_resource_manager_data_supply_type_unknown) {
pDataBuffer->seekTargetInPCMFrames = frameIndex;
pDataBuffer->seekToCursorOnNextRead = MA_TRUE;
return MA_BUSY; /* Still loading. */
}
result = ma_data_source_seek_to_pcm_frame(ma_resource_manager_data_buffer_get_connector(pDataBuffer), frameIndex);
if (result != MA_SUCCESS) {
return result;
}
pDataBuffer->seekTargetInPCMFrames = ~(ma_uint64)0; /* <-- For identification purposes. */
pDataBuffer->seekToCursorOnNextRead = MA_FALSE;
return MA_SUCCESS;
}
MA_API ma_result ma_resource_manager_data_buffer_get_data_format(ma_resource_manager_data_buffer* pDataBuffer, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
/* We cannot be using the data source after it's been uninitialized. */
MA_ASSERT(ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) != MA_UNAVAILABLE);
switch (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode))
{
case ma_resource_manager_data_supply_type_encoded:
{
return ma_data_source_get_data_format(&pDataBuffer->connector.decoder, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
};
case ma_resource_manager_data_supply_type_decoded:
{
*pFormat = pDataBuffer->pNode->data.backend.decoded.format;
*pChannels = pDataBuffer->pNode->data.backend.decoded.channels;
*pSampleRate = pDataBuffer->pNode->data.backend.decoded.sampleRate;
ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pDataBuffer->pNode->data.backend.decoded.channels);
return MA_SUCCESS;
};
case ma_resource_manager_data_supply_type_decoded_paged:
{
*pFormat = pDataBuffer->pNode->data.backend.decodedPaged.data.format;
*pChannels = pDataBuffer->pNode->data.backend.decodedPaged.data.channels;
*pSampleRate = pDataBuffer->pNode->data.backend.decodedPaged.sampleRate;
ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, pDataBuffer->pNode->data.backend.decoded.channels);
return MA_SUCCESS;
};
case ma_resource_manager_data_supply_type_unknown:
{
return MA_BUSY; /* Still loading. */
};
default:
{
/* Unknown supply type. Should never hit this. */
return MA_INVALID_ARGS;
}
}
}
MA_API ma_result ma_resource_manager_data_buffer_get_cursor_in_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pCursor)
{
/* We cannot be using the data source after it's been uninitialized. */
MA_ASSERT(ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) != MA_UNAVAILABLE);
if (pDataBuffer == NULL || pCursor == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = 0;
switch (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode))
{
case ma_resource_manager_data_supply_type_encoded:
{
return ma_decoder_get_cursor_in_pcm_frames(&pDataBuffer->connector.decoder, pCursor);
};
case ma_resource_manager_data_supply_type_decoded:
{
return ma_audio_buffer_get_cursor_in_pcm_frames(&pDataBuffer->connector.buffer, pCursor);
};
case ma_resource_manager_data_supply_type_decoded_paged:
{
return ma_paged_audio_buffer_get_cursor_in_pcm_frames(&pDataBuffer->connector.pagedBuffer, pCursor);
};
case ma_resource_manager_data_supply_type_unknown:
{
return MA_BUSY;
};
default:
{
return MA_INVALID_ARGS;
}
}
}
MA_API ma_result ma_resource_manager_data_buffer_get_length_in_pcm_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pLength)
{
/* We cannot be using the data source after it's been uninitialized. */
MA_ASSERT(ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) != MA_UNAVAILABLE);
if (pDataBuffer == NULL || pLength == NULL) {
return MA_INVALID_ARGS;
}
if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode) == ma_resource_manager_data_supply_type_unknown) {
return MA_BUSY; /* Still loading. */
}
return ma_data_source_get_length_in_pcm_frames(ma_resource_manager_data_buffer_get_connector(pDataBuffer), pLength);
}
MA_API ma_result ma_resource_manager_data_buffer_result(const ma_resource_manager_data_buffer* pDataBuffer)
{
if (pDataBuffer == NULL) {
return MA_INVALID_ARGS;
}
return (ma_result)c89atomic_load_i32((ma_result*)&pDataBuffer->result); /* Need a naughty const-cast here. */
}
MA_API ma_result ma_resource_manager_data_buffer_set_looping(ma_resource_manager_data_buffer* pDataBuffer, ma_bool32 isLooping)
{
return ma_data_source_set_looping(pDataBuffer, isLooping);
}
MA_API ma_bool32 ma_resource_manager_data_buffer_is_looping(const ma_resource_manager_data_buffer* pDataBuffer)
{
return ma_data_source_is_looping(pDataBuffer);
}
MA_API ma_result ma_resource_manager_data_buffer_get_available_frames(ma_resource_manager_data_buffer* pDataBuffer, ma_uint64* pAvailableFrames)
{
if (pAvailableFrames == NULL) {
return MA_INVALID_ARGS;
}
*pAvailableFrames = 0;
if (pDataBuffer == NULL) {
return MA_INVALID_ARGS;
}
if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode) == ma_resource_manager_data_supply_type_unknown) {
if (ma_resource_manager_data_buffer_node_result(pDataBuffer->pNode) == MA_BUSY) {
return MA_BUSY;
} else {
return MA_INVALID_OPERATION; /* No connector. */
}
}
switch (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBuffer->pNode))
{
case ma_resource_manager_data_supply_type_encoded:
{
return ma_decoder_get_available_frames(&pDataBuffer->connector.decoder, pAvailableFrames);
};
case ma_resource_manager_data_supply_type_decoded:
{
return ma_audio_buffer_get_available_frames(&pDataBuffer->connector.buffer, pAvailableFrames);
};
case ma_resource_manager_data_supply_type_decoded_paged:
{
ma_uint64 cursor;
ma_paged_audio_buffer_get_cursor_in_pcm_frames(&pDataBuffer->connector.pagedBuffer, &cursor);
if (pDataBuffer->pNode->data.backend.decodedPaged.decodedFrameCount > cursor) {
*pAvailableFrames = pDataBuffer->pNode->data.backend.decodedPaged.decodedFrameCount - cursor;
} else {
*pAvailableFrames = 0;
}
return MA_SUCCESS;
};
case ma_resource_manager_data_supply_type_unknown:
default:
{
/* Unknown supply type. Should never hit this. */
return MA_INVALID_ARGS;
}
}
}
MA_API ma_result ma_resource_manager_register_file(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags)
{
return ma_resource_manager_data_buffer_node_acquire(pResourceManager, pFilePath, NULL, 0, flags, NULL, NULL, NULL, NULL);
}
MA_API ma_result ma_resource_manager_register_file_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags)
{
return ma_resource_manager_data_buffer_node_acquire(pResourceManager, NULL, pFilePath, 0, flags, NULL, NULL, NULL, NULL);
}
static ma_result ma_resource_manager_register_data(ma_resource_manager* pResourceManager, const char* pName, const wchar_t* pNameW, ma_resource_manager_data_supply* pExistingData)
{
return ma_resource_manager_data_buffer_node_acquire(pResourceManager, pName, pNameW, 0, 0, pExistingData, NULL, NULL, NULL);
}
static ma_result ma_resource_manager_register_decoded_data_internal(ma_resource_manager* pResourceManager, const char* pName, const wchar_t* pNameW, const void* pData, ma_uint64 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
{
ma_resource_manager_data_supply data;
data.type = ma_resource_manager_data_supply_type_decoded;
data.backend.decoded.pData = pData;
data.backend.decoded.totalFrameCount = frameCount;
data.backend.decoded.format = format;
data.backend.decoded.channels = channels;
data.backend.decoded.sampleRate = sampleRate;
return ma_resource_manager_register_data(pResourceManager, pName, pNameW, &data);
}
MA_API ma_result ma_resource_manager_register_decoded_data(ma_resource_manager* pResourceManager, const char* pName, const void* pData, ma_uint64 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
{
return ma_resource_manager_register_decoded_data_internal(pResourceManager, pName, NULL, pData, frameCount, format, channels, sampleRate);
}
MA_API ma_result ma_resource_manager_register_decoded_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName, const void* pData, ma_uint64 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
{
return ma_resource_manager_register_decoded_data_internal(pResourceManager, NULL, pName, pData, frameCount, format, channels, sampleRate);
}
static ma_result ma_resource_manager_register_encoded_data_internal(ma_resource_manager* pResourceManager, const char* pName, const wchar_t* pNameW, const void* pData, size_t sizeInBytes)
{
ma_resource_manager_data_supply data;
data.type = ma_resource_manager_data_supply_type_encoded;
data.backend.encoded.pData = pData;
data.backend.encoded.sizeInBytes = sizeInBytes;
return ma_resource_manager_register_data(pResourceManager, pName, pNameW, &data);
}
MA_API ma_result ma_resource_manager_register_encoded_data(ma_resource_manager* pResourceManager, const char* pName, const void* pData, size_t sizeInBytes)
{
return ma_resource_manager_register_encoded_data_internal(pResourceManager, pName, NULL, pData, sizeInBytes);
}
MA_API ma_result ma_resource_manager_register_encoded_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName, const void* pData, size_t sizeInBytes)
{
return ma_resource_manager_register_encoded_data_internal(pResourceManager, NULL, pName, pData, sizeInBytes);
}
MA_API ma_result ma_resource_manager_unregister_file(ma_resource_manager* pResourceManager, const char* pFilePath)
{
return ma_resource_manager_unregister_data(pResourceManager, pFilePath);
}
MA_API ma_result ma_resource_manager_unregister_file_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath)
{
return ma_resource_manager_unregister_data_w(pResourceManager, pFilePath);
}
MA_API ma_result ma_resource_manager_unregister_data(ma_resource_manager* pResourceManager, const char* pName)
{
return ma_resource_manager_data_buffer_node_unacquire(pResourceManager, NULL, pName, NULL);
}
MA_API ma_result ma_resource_manager_unregister_data_w(ma_resource_manager* pResourceManager, const wchar_t* pName)
{
return ma_resource_manager_data_buffer_node_unacquire(pResourceManager, NULL, NULL, pName);
}
static ma_uint32 ma_resource_manager_data_stream_next_execution_order(ma_resource_manager_data_stream* pDataStream)
{
MA_ASSERT(pDataStream != NULL);
return c89atomic_fetch_add_32(&pDataStream->executionCounter, 1);
}
static ma_bool32 ma_resource_manager_data_stream_is_decoder_at_end(const ma_resource_manager_data_stream* pDataStream)
{
MA_ASSERT(pDataStream != NULL);
return c89atomic_load_32((ma_bool32*)&pDataStream->isDecoderAtEnd);
}
static ma_uint32 ma_resource_manager_data_stream_seek_counter(const ma_resource_manager_data_stream* pDataStream)
{
MA_ASSERT(pDataStream != NULL);
return c89atomic_load_32((ma_uint32*)&pDataStream->seekCounter);
}
static ma_result ma_resource_manager_data_stream_cb__read_pcm_frames(ma_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
return ma_resource_manager_data_stream_read_pcm_frames((ma_resource_manager_data_stream*)pDataSource, pFramesOut, frameCount, pFramesRead);
}
static ma_result ma_resource_manager_data_stream_cb__seek_to_pcm_frame(ma_data_source* pDataSource, ma_uint64 frameIndex)
{
return ma_resource_manager_data_stream_seek_to_pcm_frame((ma_resource_manager_data_stream*)pDataSource, frameIndex);
}
static ma_result ma_resource_manager_data_stream_cb__get_data_format(ma_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
return ma_resource_manager_data_stream_get_data_format((ma_resource_manager_data_stream*)pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
static ma_result ma_resource_manager_data_stream_cb__get_cursor_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pCursor)
{
return ma_resource_manager_data_stream_get_cursor_in_pcm_frames((ma_resource_manager_data_stream*)pDataSource, pCursor);
}
static ma_result ma_resource_manager_data_stream_cb__get_length_in_pcm_frames(ma_data_source* pDataSource, ma_uint64* pLength)
{
return ma_resource_manager_data_stream_get_length_in_pcm_frames((ma_resource_manager_data_stream*)pDataSource, pLength);
}
static ma_result ma_resource_manager_data_stream_cb__set_looping(ma_data_source* pDataSource, ma_bool32 isLooping)
{
ma_resource_manager_data_stream* pDataStream = (ma_resource_manager_data_stream*)pDataSource;
MA_ASSERT(pDataStream != NULL);
c89atomic_exchange_32(&pDataStream->isLooping, isLooping);
return MA_SUCCESS;
}
static ma_data_source_vtable g_ma_resource_manager_data_stream_vtable =
{
ma_resource_manager_data_stream_cb__read_pcm_frames,
ma_resource_manager_data_stream_cb__seek_to_pcm_frame,
ma_resource_manager_data_stream_cb__get_data_format,
ma_resource_manager_data_stream_cb__get_cursor_in_pcm_frames,
ma_resource_manager_data_stream_cb__get_length_in_pcm_frames,
ma_resource_manager_data_stream_cb__set_looping,
MA_DATA_SOURCE_SELF_MANAGED_RANGE_AND_LOOP_POINT
};
static void ma_resource_manager_data_stream_set_absolute_cursor(ma_resource_manager_data_stream* pDataStream, ma_uint64 absoluteCursor)
{
/* Loop if possible. */
if (absoluteCursor > pDataStream->totalLengthInPCMFrames && pDataStream->totalLengthInPCMFrames > 0) {
absoluteCursor = absoluteCursor % pDataStream->totalLengthInPCMFrames;
}
c89atomic_exchange_64(&pDataStream->absoluteCursor, absoluteCursor);
}
MA_API ma_result ma_resource_manager_data_stream_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_stream* pDataStream)
{
ma_result result;
ma_data_source_config dataSourceConfig;
char* pFilePathCopy = NULL;
wchar_t* pFilePathWCopy = NULL;
ma_job job;
ma_bool32 waitBeforeReturning = MA_FALSE;
ma_resource_manager_inline_notification waitNotification;
ma_resource_manager_pipeline_notifications notifications;
if (pDataStream == NULL) {
if (pConfig != NULL && pConfig->pNotifications != NULL) {
ma_resource_manager_pipeline_notifications_signal_all_notifications(pConfig->pNotifications);
}
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pDataStream);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->pNotifications != NULL) {
notifications = *pConfig->pNotifications; /* From here on out, `notifications` should be used instead of `pNotifications`. Setting this to NULL to catch any errors at testing time. */
} else {
MA_ZERO_OBJECT(¬ifications);
}
dataSourceConfig = ma_data_source_config_init();
dataSourceConfig.vtable = &g_ma_resource_manager_data_stream_vtable;
result = ma_data_source_init(&dataSourceConfig, &pDataStream->ds);
if (result != MA_SUCCESS) {
ma_resource_manager_pipeline_notifications_signal_all_notifications(¬ifications);
return result;
}
pDataStream->pResourceManager = pResourceManager;
pDataStream->flags = pConfig->flags;
pDataStream->result = MA_BUSY;
ma_data_source_set_range_in_pcm_frames(pDataStream, pConfig->rangeBegInPCMFrames, pConfig->rangeEndInPCMFrames);
ma_data_source_set_loop_point_in_pcm_frames(pDataStream, pConfig->loopPointBegInPCMFrames, pConfig->loopPointEndInPCMFrames);
ma_data_source_set_looping(pDataStream, pConfig->isLooping);
if (pResourceManager == NULL || (pConfig->pFilePath == NULL && pConfig->pFilePathW == NULL)) {
ma_resource_manager_pipeline_notifications_signal_all_notifications(¬ifications);
return MA_INVALID_ARGS;
}
/* We want all access to the VFS and the internal decoder to happen on the job thread just to keep things easier to manage for the VFS. */
/* We need a copy of the file path. We should probably make this more efficient, but for now we'll do a transient memory allocation. */
if (pConfig->pFilePath != NULL) {
pFilePathCopy = ma_copy_string(pConfig->pFilePath, &pResourceManager->config.allocationCallbacks);
} else {
pFilePathWCopy = ma_copy_string_w(pConfig->pFilePathW, &pResourceManager->config.allocationCallbacks);
}
if (pFilePathCopy == NULL && pFilePathWCopy == NULL) {
ma_resource_manager_pipeline_notifications_signal_all_notifications(¬ifications);
return MA_OUT_OF_MEMORY;
}
/*
We need to check for the presence of MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC. If it's not set, we need to wait before returning. Otherwise we
can return immediately. Likewise, we'll also check for MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT and do the same.
*/
if ((pConfig->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_ASYNC) == 0 || (pConfig->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) {
waitBeforeReturning = MA_TRUE;
ma_resource_manager_inline_notification_init(pResourceManager, &waitNotification);
}
ma_resource_manager_pipeline_notifications_acquire_all_fences(¬ifications);
/* Set the absolute cursor to our initial seek position so retrieval of the cursor returns a good value. */
ma_resource_manager_data_stream_set_absolute_cursor(pDataStream, pConfig->initialSeekPointInPCMFrames);
/* We now have everything we need to post the job. This is the last thing we need to do from here. The rest will be done by the job thread. */
job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_LOAD_DATA_STREAM);
job.order = ma_resource_manager_data_stream_next_execution_order(pDataStream);
job.data.resourceManager.loadDataStream.pDataStream = pDataStream;
job.data.resourceManager.loadDataStream.pFilePath = pFilePathCopy;
job.data.resourceManager.loadDataStream.pFilePathW = pFilePathWCopy;
job.data.resourceManager.loadDataStream.initialSeekPoint = pConfig->initialSeekPointInPCMFrames;
job.data.resourceManager.loadDataStream.pInitNotification = (waitBeforeReturning == MA_TRUE) ? &waitNotification : notifications.init.pNotification;
job.data.resourceManager.loadDataStream.pInitFence = notifications.init.pFence;
result = ma_resource_manager_post_job(pResourceManager, &job);
if (result != MA_SUCCESS) {
ma_resource_manager_pipeline_notifications_signal_all_notifications(¬ifications);
ma_resource_manager_pipeline_notifications_release_all_fences(¬ifications);
if (waitBeforeReturning) {
ma_resource_manager_inline_notification_uninit(&waitNotification);
}
ma_free(pFilePathCopy, &pResourceManager->config.allocationCallbacks);
ma_free(pFilePathWCopy, &pResourceManager->config.allocationCallbacks);
return result;
}
/* Wait if needed. */
if (waitBeforeReturning) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
return MA_SUCCESS;
}
MA_API ma_result ma_resource_manager_data_stream_init(ma_resource_manager* pResourceManager, const char* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_stream* pDataStream)
{
ma_resource_manager_data_source_config config;
config = ma_resource_manager_data_source_config_init();
config.pFilePath = pFilePath;
config.flags = flags;
config.pNotifications = pNotifications;
return ma_resource_manager_data_stream_init_ex(pResourceManager, &config, pDataStream);
}
MA_API ma_result ma_resource_manager_data_stream_init_w(ma_resource_manager* pResourceManager, const wchar_t* pFilePath, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_stream* pDataStream)
{
ma_resource_manager_data_source_config config;
config = ma_resource_manager_data_source_config_init();
config.pFilePathW = pFilePath;
config.flags = flags;
config.pNotifications = pNotifications;
return ma_resource_manager_data_stream_init_ex(pResourceManager, &config, pDataStream);
}
MA_API ma_result ma_resource_manager_data_stream_uninit(ma_resource_manager_data_stream* pDataStream)
{
ma_resource_manager_inline_notification freeEvent;
ma_job job;
if (pDataStream == NULL) {
return MA_INVALID_ARGS;
}
/* The first thing to do is set the result to unavailable. This will prevent future page decoding. */
c89atomic_exchange_i32(&pDataStream->result, MA_UNAVAILABLE);
/*
We need to post a job to ensure we're not in the middle or decoding or anything. Because the object is owned by the caller, we'll need
to wait for it to complete before returning which means we need an event.
*/
ma_resource_manager_inline_notification_init(pDataStream->pResourceManager, &freeEvent);
job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_STREAM);
job.order = ma_resource_manager_data_stream_next_execution_order(pDataStream);
job.data.resourceManager.freeDataStream.pDataStream = pDataStream;
job.data.resourceManager.freeDataStream.pDoneNotification = &freeEvent;
job.data.resourceManager.freeDataStream.pDoneFence = NULL;
ma_resource_manager_post_job(pDataStream->pResourceManager, &job);
/* We need to wait for the job to finish processing before we return. */
ma_resource_manager_inline_notification_wait_and_uninit(&freeEvent);
return MA_SUCCESS;
}
static ma_uint32 ma_resource_manager_data_stream_get_page_size_in_frames(ma_resource_manager_data_stream* pDataStream)
{
MA_ASSERT(pDataStream != NULL);
MA_ASSERT(pDataStream->isDecoderInitialized == MA_TRUE);
return MA_RESOURCE_MANAGER_PAGE_SIZE_IN_MILLISECONDS * (pDataStream->decoder.outputSampleRate/1000);
}
static void* ma_resource_manager_data_stream_get_page_data_pointer(ma_resource_manager_data_stream* pDataStream, ma_uint32 pageIndex, ma_uint32 relativeCursor)
{
MA_ASSERT(pDataStream != NULL);
MA_ASSERT(pDataStream->isDecoderInitialized == MA_TRUE);
MA_ASSERT(pageIndex == 0 || pageIndex == 1);
return ma_offset_ptr(pDataStream->pPageData, ((ma_resource_manager_data_stream_get_page_size_in_frames(pDataStream) * pageIndex) + relativeCursor) * ma_get_bytes_per_frame(pDataStream->decoder.outputFormat, pDataStream->decoder.outputChannels));
}
static void ma_resource_manager_data_stream_fill_page(ma_resource_manager_data_stream* pDataStream, ma_uint32 pageIndex)
{
ma_result result = MA_SUCCESS;
ma_uint64 pageSizeInFrames;
ma_uint64 totalFramesReadForThisPage = 0;
void* pPageData = ma_resource_manager_data_stream_get_page_data_pointer(pDataStream, pageIndex, 0);
pageSizeInFrames = ma_resource_manager_data_stream_get_page_size_in_frames(pDataStream);
/* The decoder needs to inherit the stream's looping and range state. */
{
ma_uint64 rangeBeg;
ma_uint64 rangeEnd;
ma_uint64 loopPointBeg;
ma_uint64 loopPointEnd;
ma_data_source_set_looping(&pDataStream->decoder, ma_resource_manager_data_stream_is_looping(pDataStream));
ma_data_source_get_range_in_pcm_frames(pDataStream, &rangeBeg, &rangeEnd);
ma_data_source_set_range_in_pcm_frames(&pDataStream->decoder, rangeBeg, rangeEnd);
ma_data_source_get_loop_point_in_pcm_frames(pDataStream, &loopPointBeg, &loopPointEnd);
ma_data_source_set_loop_point_in_pcm_frames(&pDataStream->decoder, loopPointBeg, loopPointEnd);
}
/* Just read straight from the decoder. It will deal with ranges and looping for us. */
result = ma_data_source_read_pcm_frames(&pDataStream->decoder, pPageData, pageSizeInFrames, &totalFramesReadForThisPage);
if (result == MA_AT_END || totalFramesReadForThisPage < pageSizeInFrames) {
c89atomic_exchange_32(&pDataStream->isDecoderAtEnd, MA_TRUE);
}
c89atomic_exchange_32(&pDataStream->pageFrameCount[pageIndex], (ma_uint32)totalFramesReadForThisPage);
c89atomic_exchange_32(&pDataStream->isPageValid[pageIndex], MA_TRUE);
}
static void ma_resource_manager_data_stream_fill_pages(ma_resource_manager_data_stream* pDataStream)
{
ma_uint32 iPage;
MA_ASSERT(pDataStream != NULL);
for (iPage = 0; iPage < 2; iPage += 1) {
ma_resource_manager_data_stream_fill_page(pDataStream, iPage);
}
}
static ma_result ma_resource_manager_data_stream_map(ma_resource_manager_data_stream* pDataStream, void** ppFramesOut, ma_uint64* pFrameCount)
{
ma_uint64 framesAvailable;
ma_uint64 frameCount = 0;
/* We cannot be using the data source after it's been uninitialized. */
MA_ASSERT(ma_resource_manager_data_stream_result(pDataStream) != MA_UNAVAILABLE);
if (pFrameCount != NULL) {
frameCount = *pFrameCount;
*pFrameCount = 0;
}
if (ppFramesOut != NULL) {
*ppFramesOut = NULL;
}
if (pDataStream == NULL || ppFramesOut == NULL || pFrameCount == NULL) {
return MA_INVALID_ARGS;
}
if (ma_resource_manager_data_stream_result(pDataStream) != MA_SUCCESS) {
return MA_INVALID_OPERATION;
}
/* Don't attempt to read while we're in the middle of seeking. Tell the caller that we're busy. */
if (ma_resource_manager_data_stream_seek_counter(pDataStream) > 0) {
return MA_BUSY;
}
/* If the page we're on is invalid it means we've caught up to the job thread. */
if (c89atomic_load_32(&pDataStream->isPageValid[pDataStream->currentPageIndex]) == MA_FALSE) {
framesAvailable = 0;
} else {
/*
The page we're on is valid so we must have some frames available. We need to make sure that we don't overflow into the next page, even if it's valid. The reason is
that the unmap process will only post an update for one page at a time. Keeping mapping tied to page boundaries makes this simpler.
*/
ma_uint32 currentPageFrameCount = c89atomic_load_32(&pDataStream->pageFrameCount[pDataStream->currentPageIndex]);
MA_ASSERT(currentPageFrameCount >= pDataStream->relativeCursor);
framesAvailable = currentPageFrameCount - pDataStream->relativeCursor;
}
/* If there's no frames available and the result is set to MA_AT_END we need to return MA_AT_END. */
if (framesAvailable == 0) {
if (ma_resource_manager_data_stream_is_decoder_at_end(pDataStream)) {
return MA_AT_END;
} else {
return MA_BUSY; /* There are no frames available, but we're not marked as EOF so we might have caught up to the job thread. Need to return MA_BUSY and wait for more data. */
}
}
MA_ASSERT(framesAvailable > 0);
if (frameCount > framesAvailable) {
frameCount = framesAvailable;
}
*ppFramesOut = ma_resource_manager_data_stream_get_page_data_pointer(pDataStream, pDataStream->currentPageIndex, pDataStream->relativeCursor);
*pFrameCount = frameCount;
return MA_SUCCESS;
}
static ma_result ma_resource_manager_data_stream_unmap(ma_resource_manager_data_stream* pDataStream, ma_uint64 frameCount)
{
ma_uint32 newRelativeCursor;
ma_uint32 pageSizeInFrames;
ma_job job;
/* We cannot be using the data source after it's been uninitialized. */
MA_ASSERT(ma_resource_manager_data_stream_result(pDataStream) != MA_UNAVAILABLE);
if (pDataStream == NULL) {
return MA_INVALID_ARGS;
}
if (ma_resource_manager_data_stream_result(pDataStream) != MA_SUCCESS) {
return MA_INVALID_OPERATION;
}
/* The frame count should always fit inside a 32-bit integer. */
if (frameCount > 0xFFFFFFFF) {
return MA_INVALID_ARGS;
}
pageSizeInFrames = ma_resource_manager_data_stream_get_page_size_in_frames(pDataStream);
/* The absolute cursor needs to be updated for ma_resource_manager_data_stream_get_cursor_in_pcm_frames(). */
ma_resource_manager_data_stream_set_absolute_cursor(pDataStream, c89atomic_load_64(&pDataStream->absoluteCursor) + frameCount);
/* Here is where we need to check if we need to load a new page, and if so, post a job to load it. */
newRelativeCursor = pDataStream->relativeCursor + (ma_uint32)frameCount;
/* If the new cursor has flowed over to the next page we need to mark the old one as invalid and post an event for it. */
if (newRelativeCursor >= pageSizeInFrames) {
newRelativeCursor -= pageSizeInFrames;
/* Here is where we post the job start decoding. */
job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_PAGE_DATA_STREAM);
job.order = ma_resource_manager_data_stream_next_execution_order(pDataStream);
job.data.resourceManager.pageDataStream.pDataStream = pDataStream;
job.data.resourceManager.pageDataStream.pageIndex = pDataStream->currentPageIndex;
/* The page needs to be marked as invalid so that the public API doesn't try reading from it. */
c89atomic_exchange_32(&pDataStream->isPageValid[pDataStream->currentPageIndex], MA_FALSE);
/* Before posting the job we need to make sure we set some state. */
pDataStream->relativeCursor = newRelativeCursor;
pDataStream->currentPageIndex = (pDataStream->currentPageIndex + 1) & 0x01;
return ma_resource_manager_post_job(pDataStream->pResourceManager, &job);
} else {
/* We haven't moved into a new page so we can just move the cursor forward. */
pDataStream->relativeCursor = newRelativeCursor;
return MA_SUCCESS;
}
}
MA_API ma_result ma_resource_manager_data_stream_read_pcm_frames(ma_resource_manager_data_stream* pDataStream, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
ma_result result = MA_SUCCESS;
ma_uint64 totalFramesProcessed;
ma_format format;
ma_uint32 channels;
/* Safety. */
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
if (frameCount == 0) {
return MA_INVALID_ARGS;
}
/* We cannot be using the data source after it's been uninitialized. */
MA_ASSERT(ma_resource_manager_data_stream_result(pDataStream) != MA_UNAVAILABLE);
if (pDataStream == NULL) {
return MA_INVALID_ARGS;
}
if (ma_resource_manager_data_stream_result(pDataStream) != MA_SUCCESS) {
return MA_INVALID_OPERATION;
}
/* Don't attempt to read while we're in the middle of seeking. Tell the caller that we're busy. */
if (ma_resource_manager_data_stream_seek_counter(pDataStream) > 0) {
return MA_BUSY;
}
ma_resource_manager_data_stream_get_data_format(pDataStream, &format, &channels, NULL, NULL, 0);
/* Reading is implemented in terms of map/unmap. We need to run this in a loop because mapping is clamped against page boundaries. */
totalFramesProcessed = 0;
while (totalFramesProcessed < frameCount) {
void* pMappedFrames;
ma_uint64 mappedFrameCount;
mappedFrameCount = frameCount - totalFramesProcessed;
result = ma_resource_manager_data_stream_map(pDataStream, &pMappedFrames, &mappedFrameCount);
if (result != MA_SUCCESS) {
break;
}
/* Copy the mapped data to the output buffer if we have one. It's allowed for pFramesOut to be NULL in which case a relative forward seek is performed. */
if (pFramesOut != NULL) {
ma_copy_pcm_frames(ma_offset_pcm_frames_ptr(pFramesOut, totalFramesProcessed, format, channels), pMappedFrames, mappedFrameCount, format, channels);
}
totalFramesProcessed += mappedFrameCount;
result = ma_resource_manager_data_stream_unmap(pDataStream, mappedFrameCount);
if (result != MA_SUCCESS) {
break; /* This is really bad - will only get an error here if we failed to post a job to the queue for loading the next page. */
}
}
if (pFramesRead != NULL) {
*pFramesRead = totalFramesProcessed;
}
if (result == MA_SUCCESS && totalFramesProcessed == 0) {
result = MA_AT_END;
}
return result;
}
MA_API ma_result ma_resource_manager_data_stream_seek_to_pcm_frame(ma_resource_manager_data_stream* pDataStream, ma_uint64 frameIndex)
{
ma_job job;
ma_result streamResult;
streamResult = ma_resource_manager_data_stream_result(pDataStream);
/* We cannot be using the data source after it's been uninitialized. */
MA_ASSERT(streamResult != MA_UNAVAILABLE);
if (pDataStream == NULL) {
return MA_INVALID_ARGS;
}
if (streamResult != MA_SUCCESS && streamResult != MA_BUSY) {
return MA_INVALID_OPERATION;
}
/* If we're not already seeking and we're sitting on the same frame, just make this a no-op. */
if (c89atomic_load_32(&pDataStream->seekCounter) == 0) {
if (c89atomic_load_64(&pDataStream->absoluteCursor) == frameIndex) {
return MA_SUCCESS;
}
}
/* Increment the seek counter first to indicate to read_paged_pcm_frames() and map_paged_pcm_frames() that we are in the middle of a seek and MA_BUSY should be returned. */
c89atomic_fetch_add_32(&pDataStream->seekCounter, 1);
/* Update the absolute cursor so that ma_resource_manager_data_stream_get_cursor_in_pcm_frames() returns the new position. */
ma_resource_manager_data_stream_set_absolute_cursor(pDataStream, frameIndex);
/*
We need to clear our currently loaded pages so that the stream starts playback from the new seek point as soon as possible. These are for the purpose of the public
API and will be ignored by the seek job. The seek job will operate on the assumption that both pages have been marked as invalid and the cursor is at the start of
the first page.
*/
pDataStream->relativeCursor = 0;
pDataStream->currentPageIndex = 0;
c89atomic_exchange_32(&pDataStream->isPageValid[0], MA_FALSE);
c89atomic_exchange_32(&pDataStream->isPageValid[1], MA_FALSE);
/* Make sure the data stream is not marked as at the end or else if we seek in response to hitting the end, we won't be able to read any more data. */
c89atomic_exchange_32(&pDataStream->isDecoderAtEnd, MA_FALSE);
/*
The public API is not allowed to touch the internal decoder so we need to use a job to perform the seek. When seeking, the job thread will assume both pages
are invalid and any content contained within them will be discarded and replaced with newly decoded data.
*/
job = ma_job_init(MA_JOB_TYPE_RESOURCE_MANAGER_SEEK_DATA_STREAM);
job.order = ma_resource_manager_data_stream_next_execution_order(pDataStream);
job.data.resourceManager.seekDataStream.pDataStream = pDataStream;
job.data.resourceManager.seekDataStream.frameIndex = frameIndex;
return ma_resource_manager_post_job(pDataStream->pResourceManager, &job);
}
MA_API ma_result ma_resource_manager_data_stream_get_data_format(ma_resource_manager_data_stream* pDataStream, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
/* We cannot be using the data source after it's been uninitialized. */
MA_ASSERT(ma_resource_manager_data_stream_result(pDataStream) != MA_UNAVAILABLE);
if (pFormat != NULL) {
*pFormat = ma_format_unknown;
}
if (pChannels != NULL) {
*pChannels = 0;
}
if (pSampleRate != NULL) {
*pSampleRate = 0;
}
if (pChannelMap != NULL) {
MA_ZERO_MEMORY(pChannelMap, sizeof(*pChannelMap) * channelMapCap);
}
if (pDataStream == NULL) {
return MA_INVALID_ARGS;
}
if (ma_resource_manager_data_stream_result(pDataStream) != MA_SUCCESS) {
return MA_INVALID_OPERATION;
}
/*
We're being a little bit naughty here and accessing the internal decoder from the public API. The output data format is constant, and we've defined this function
such that the application is responsible for ensuring it's not called while uninitializing so it should be safe.
*/
return ma_data_source_get_data_format(&pDataStream->decoder, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
MA_API ma_result ma_resource_manager_data_stream_get_cursor_in_pcm_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pCursor)
{
ma_result result;
if (pCursor == NULL) {
return MA_INVALID_ARGS;
}
*pCursor = 0;
/* We cannot be using the data source after it's been uninitialized. */
MA_ASSERT(ma_resource_manager_data_stream_result(pDataStream) != MA_UNAVAILABLE);
if (pDataStream == NULL) {
return MA_INVALID_ARGS;
}
/*
If the stream is in an erroneous state we need to return an invalid operation. We can allow
this to be called when the data stream is in a busy state because the caller may have asked
for an initial seek position and it's convenient to return that as the cursor position.
*/
result = ma_resource_manager_data_stream_result(pDataStream);
if (result != MA_SUCCESS && result != MA_BUSY) {
return MA_INVALID_OPERATION;
}
*pCursor = c89atomic_load_64(&pDataStream->absoluteCursor);
return MA_SUCCESS;
}
MA_API ma_result ma_resource_manager_data_stream_get_length_in_pcm_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pLength)
{
ma_result streamResult;
if (pLength == NULL) {
return MA_INVALID_ARGS;
}
*pLength = 0;
streamResult = ma_resource_manager_data_stream_result(pDataStream);
/* We cannot be using the data source after it's been uninitialized. */
MA_ASSERT(streamResult != MA_UNAVAILABLE);
if (pDataStream == NULL) {
return MA_INVALID_ARGS;
}
if (streamResult != MA_SUCCESS) {
return streamResult;
}
/*
We most definitely do not want to be calling ma_decoder_get_length_in_pcm_frames() directly. Instead we want to use a cached value that we
calculated when we initialized it on the job thread.
*/
*pLength = pDataStream->totalLengthInPCMFrames;
if (*pLength == 0) {
return MA_NOT_IMPLEMENTED; /* Some decoders may not have a known length. */
}
return MA_SUCCESS;
}
MA_API ma_result ma_resource_manager_data_stream_result(const ma_resource_manager_data_stream* pDataStream)
{
if (pDataStream == NULL) {
return MA_INVALID_ARGS;
}
return (ma_result)c89atomic_load_i32(&pDataStream->result);
}
MA_API ma_result ma_resource_manager_data_stream_set_looping(ma_resource_manager_data_stream* pDataStream, ma_bool32 isLooping)
{
return ma_data_source_set_looping(pDataStream, isLooping);
}
MA_API ma_bool32 ma_resource_manager_data_stream_is_looping(const ma_resource_manager_data_stream* pDataStream)
{
if (pDataStream == NULL) {
return MA_FALSE;
}
return c89atomic_load_32((ma_bool32*)&pDataStream->isLooping); /* Naughty const-cast. Value won't change from here in practice (maybe from another thread). */
}
MA_API ma_result ma_resource_manager_data_stream_get_available_frames(ma_resource_manager_data_stream* pDataStream, ma_uint64* pAvailableFrames)
{
ma_uint32 pageIndex0;
ma_uint32 pageIndex1;
ma_uint32 relativeCursor;
ma_uint64 availableFrames;
if (pAvailableFrames == NULL) {
return MA_INVALID_ARGS;
}
*pAvailableFrames = 0;
if (pDataStream == NULL) {
return MA_INVALID_ARGS;
}
pageIndex0 = pDataStream->currentPageIndex;
pageIndex1 = (pDataStream->currentPageIndex + 1) & 0x01;
relativeCursor = pDataStream->relativeCursor;
availableFrames = 0;
if (c89atomic_load_32(&pDataStream->isPageValid[pageIndex0])) {
availableFrames += c89atomic_load_32(&pDataStream->pageFrameCount[pageIndex0]) - relativeCursor;
if (c89atomic_load_32(&pDataStream->isPageValid[pageIndex1])) {
availableFrames += c89atomic_load_32(&pDataStream->pageFrameCount[pageIndex1]);
}
}
*pAvailableFrames = availableFrames;
return MA_SUCCESS;
}
static ma_result ma_resource_manager_data_source_preinit(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_source* pDataSource)
{
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pDataSource);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pResourceManager == NULL) {
return MA_INVALID_ARGS;
}
pDataSource->flags = pConfig->flags;
return MA_SUCCESS;
}
MA_API ma_result ma_resource_manager_data_source_init_ex(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source_config* pConfig, ma_resource_manager_data_source* pDataSource)
{
ma_result result;
result = ma_resource_manager_data_source_preinit(pResourceManager, pConfig, pDataSource);
if (result != MA_SUCCESS) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
config = ma_resource_manager_data_source_config_init();
config.pFilePath = pName;
config.flags = flags;
config.pNotifications = pNotifications;
return ma_resource_manager_data_source_init_ex(pResourceManager, &config, pDataSource);
}
MA_API ma_result ma_resource_manager_data_source_init_w(ma_resource_manager* pResourceManager, const wchar_t* pName, ma_uint32 flags, const ma_resource_manager_pipeline_notifications* pNotifications, ma_resource_manager_data_source* pDataSource)
{
ma_resource_manager_data_source_config config;
config = ma_resource_manager_data_source_config_init();
config.pFilePathW = pName;
config.flags = flags;
config.pNotifications = pNotifications;
return ma_resource_manager_data_source_init_ex(pResourceManager, &config, pDataSource);
}
MA_API ma_result ma_resource_manager_data_source_init_copy(ma_resource_manager* pResourceManager, const ma_resource_manager_data_source* pExistingDataSource, ma_resource_manager_data_source* pDataSource)
{
ma_result result;
ma_resource_manager_data_source_config config;
if (pExistingDataSource == NULL) {
return MA_INVALID_ARGS;
}
config = ma_resource_manager_data_source_config_init();
config.flags = pExistingDataSource->flags;
result = ma_resource_manager_data_source_preinit(pResourceManager, &config, pDataSource);
if (result != MA_SUCCESS) {
return result;
}
/* Copying can only be done from data buffers. Streams cannot be copied. */
if ((pExistingDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return MA_INVALID_OPERATION;
}
return ma_resource_manager_data_buffer_init_copy(pResourceManager, &pExistingDataSource->backend.buffer, &pDataSource->backend.buffer);
}
MA_API ma_result ma_resource_manager_data_source_uninit(ma_resource_manager_data_source* pDataSource)
{
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
/* All we need to is uninitialize the underlying data buffer or data stream. */
if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return ma_resource_manager_data_stream_uninit(&pDataSource->backend.stream);
} else {
return ma_resource_manager_data_buffer_uninit(&pDataSource->backend.buffer);
}
}
MA_API ma_result ma_resource_manager_data_source_read_pcm_frames(ma_resource_manager_data_source* pDataSource, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
/* Safety. */
if (pFramesRead != NULL) {
*pFramesRead = 0;
}
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return ma_resource_manager_data_stream_read_pcm_frames(&pDataSource->backend.stream, pFramesOut, frameCount, pFramesRead);
} else {
return ma_resource_manager_data_buffer_read_pcm_frames(&pDataSource->backend.buffer, pFramesOut, frameCount, pFramesRead);
}
}
MA_API ma_result ma_resource_manager_data_source_seek_to_pcm_frame(ma_resource_manager_data_source* pDataSource, ma_uint64 frameIndex)
{
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return ma_resource_manager_data_stream_seek_to_pcm_frame(&pDataSource->backend.stream, frameIndex);
} else {
return ma_resource_manager_data_buffer_seek_to_pcm_frame(&pDataSource->backend.buffer, frameIndex);
}
}
MA_API ma_result ma_resource_manager_data_source_map(ma_resource_manager_data_source* pDataSource, void** ppFramesOut, ma_uint64* pFrameCount)
{
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return ma_resource_manager_data_stream_map(&pDataSource->backend.stream, ppFramesOut, pFrameCount);
} else {
return MA_NOT_IMPLEMENTED; /* Mapping not supported with data buffers. */
}
}
MA_API ma_result ma_resource_manager_data_source_unmap(ma_resource_manager_data_source* pDataSource, ma_uint64 frameCount)
{
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return ma_resource_manager_data_stream_unmap(&pDataSource->backend.stream, frameCount);
} else {
return MA_NOT_IMPLEMENTED; /* Mapping not supported with data buffers. */
}
}
MA_API ma_result ma_resource_manager_data_source_get_data_format(ma_resource_manager_data_source* pDataSource, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return ma_resource_manager_data_stream_get_data_format(&pDataSource->backend.stream, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
} else {
return ma_resource_manager_data_buffer_get_data_format(&pDataSource->backend.buffer, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
}
MA_API ma_result ma_resource_manager_data_source_get_cursor_in_pcm_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pCursor)
{
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return ma_resource_manager_data_stream_get_cursor_in_pcm_frames(&pDataSource->backend.stream, pCursor);
} else {
return ma_resource_manager_data_buffer_get_cursor_in_pcm_frames(&pDataSource->backend.buffer, pCursor);
}
}
MA_API ma_result ma_resource_manager_data_source_get_length_in_pcm_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pLength)
{
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return ma_resource_manager_data_stream_get_length_in_pcm_frames(&pDataSource->backend.stream, pLength);
} else {
return ma_resource_manager_data_buffer_get_length_in_pcm_frames(&pDataSource->backend.buffer, pLength);
}
}
MA_API ma_result ma_resource_manager_data_source_result(const ma_resource_manager_data_source* pDataSource)
{
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return ma_resource_manager_data_stream_result(&pDataSource->backend.stream);
} else {
return ma_resource_manager_data_buffer_result(&pDataSource->backend.buffer);
}
}
MA_API ma_result ma_resource_manager_data_source_set_looping(ma_resource_manager_data_source* pDataSource, ma_bool32 isLooping)
{
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return ma_resource_manager_data_stream_set_looping(&pDataSource->backend.stream, isLooping);
} else {
return ma_resource_manager_data_buffer_set_looping(&pDataSource->backend.buffer, isLooping);
}
}
MA_API ma_bool32 ma_resource_manager_data_source_is_looping(const ma_resource_manager_data_source* pDataSource)
{
if (pDataSource == NULL) {
return MA_FALSE;
}
if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return ma_resource_manager_data_stream_is_looping(&pDataSource->backend.stream);
} else {
return ma_resource_manager_data_buffer_is_looping(&pDataSource->backend.buffer);
}
}
MA_API ma_result ma_resource_manager_data_source_get_available_frames(ma_resource_manager_data_source* pDataSource, ma_uint64* pAvailableFrames)
{
if (pAvailableFrames == NULL) {
return MA_INVALID_ARGS;
}
*pAvailableFrames = 0;
if (pDataSource == NULL) {
return MA_INVALID_ARGS;
}
if ((pDataSource->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_STREAM) != 0) {
return ma_resource_manager_data_stream_get_available_frames(&pDataSource->backend.stream, pAvailableFrames);
} else {
return ma_resource_manager_data_buffer_get_available_frames(&pDataSource->backend.buffer, pAvailableFrames);
}
}
MA_API ma_result ma_resource_manager_post_job(ma_resource_manager* pResourceManager, const ma_job* pJob)
{
if (pResourceManager == NULL) {
return MA_INVALID_ARGS;
}
return ma_job_queue_post(&pResourceManager->jobQueue, pJob);
}
MA_API ma_result ma_resource_manager_post_job_quit(ma_resource_manager* pResourceManager)
{
ma_job job = ma_job_init(MA_JOB_TYPE_QUIT);
return ma_resource_manager_post_job(pResourceManager, &job);
}
MA_API ma_result ma_resource_manager_next_job(ma_resource_manager* pResourceManager, ma_job* pJob)
{
if (pResourceManager == NULL) {
return MA_INVALID_ARGS;
}
return ma_job_queue_next(&pResourceManager->jobQueue, pJob);
}
static ma_result ma_job_process__resource_manager__load_data_buffer_node(ma_job* pJob)
{
ma_result result = MA_SUCCESS;
ma_resource_manager* pResourceManager;
ma_resource_manager_data_buffer_node* pDataBufferNode;
MA_ASSERT(pJob != NULL);
pResourceManager = (ma_resource_manager*)pJob->data.resourceManager.loadDataBufferNode.pResourceManager;
MA_ASSERT(pResourceManager != NULL);
pDataBufferNode = (ma_resource_manager_data_buffer_node*)pJob->data.resourceManager.loadDataBufferNode.pDataBufferNode;
MA_ASSERT(pDataBufferNode != NULL);
MA_ASSERT(pDataBufferNode->isDataOwnedByResourceManager == MA_TRUE); /* The data should always be owned by the resource manager. */
/* The data buffer is not getting deleted, but we may be getting executed out of order. If so, we need to push the job back onto the queue and return. */
if (pJob->order != c89atomic_load_32(&pDataBufferNode->executionPointer)) {
return ma_resource_manager_post_job(pResourceManager, pJob); /* Attempting to execute out of order. Probably interleaved with a MA_JOB_TYPE_RESOURCE_MANAGER_FREE_DATA_BUFFER job. */
}
/* First thing we need to do is check whether or not the data buffer is getting deleted. If so we just abort. */
if (ma_resource_manager_data_buffer_node_result(pDataBufferNode) != MA_BUSY) {
result = ma_resource_manager_data_buffer_node_result(pDataBufferNode); /* The data buffer may be getting deleted before it's even been loaded. */
goto done;
}
/*
We're ready to start loading. Essentially what we're doing here is initializing the data supply
of the node. Once this is complete, data buffers can have their connectors initialized which
will allow then to have audio data read from them.
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
pResourceManager = pDataBuffer->pResourceManager;
if (pJob->order != c89atomic_load_32(&pDataBuffer->executionPointer)) {
return ma_resource_manager_post_job(pResourceManager, pJob); /* Out of order. */
}
ma_resource_manager_data_buffer_uninit_internal(pDataBuffer);
/* The event needs to be signalled last. */
if (pJob->data.resourceManager.freeDataBuffer.pDoneNotification != NULL) {
ma_async_notification_signal(pJob->data.resourceManager.freeDataBuffer.pDoneNotification);
}
if (pJob->data.resourceManager.freeDataBuffer.pDoneFence != NULL) {
ma_fence_release(pJob->data.resourceManager.freeDataBuffer.pDoneFence);
}
c89atomic_fetch_add_32(&pDataBuffer->executionPointer, 1);
return MA_SUCCESS;
}
static ma_result ma_job_process__resource_manager__load_data_stream(ma_job* pJob)
{
ma_result result = MA_SUCCESS;
ma_decoder_config decoderConfig;
ma_uint32 pageBufferSizeInBytes;
ma_resource_manager* pResourceManager;
ma_resource_manager_data_stream* pDataStream;
MA_ASSERT(pJob != NULL);
pDataStream = (ma_resource_manager_data_stream*)pJob->data.resourceManager.loadDataStream.pDataStream;
MA_ASSERT(pDataStream != NULL);
pResourceManager = pDataStream->pResourceManager;
if (pJob->order != c89atomic_load_32(&pDataStream->executionPointer)) {
return ma_resource_manager_post_job(pResourceManager, pJob); /* Out of order. */
}
if (ma_resource_manager_data_stream_result(pDataStream) != MA_BUSY) {
result = MA_INVALID_OPERATION; /* Most likely the data stream is being uninitialized. */
goto done;
}
/* We need to initialize the decoder first so we can determine the size of the pages. */
decoderConfig = ma_resource_manager__init_decoder_config(pResourceManager);
if (pJob->data.resourceManager.loadDataStream.pFilePath != NULL) {
result = ma_decoder_init_vfs(pResourceManager->config.pVFS, pJob->data.resourceManager.loadDataStream.pFilePath, &decoderConfig, &pDataStream->decoder);
} else {
result = ma_decoder_init_vfs_w(pResourceManager->config.pVFS, pJob->data.resourceManager.loadDataStream.pFilePathW, &decoderConfig, &pDataStream->decoder);
}
if (result != MA_SUCCESS) {
goto done;
}
/* Retrieve the total length of the file before marking the decoder are loaded. */
if ((pDataStream->flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_UNKNOWN_LENGTH) == 0) {
result = ma_decoder_get_length_in_pcm_frames(&pDataStream->decoder, &pDataStream->totalLengthInPCMFrames);
if (result != MA_SUCCESS) {
goto done; /* Failed to retrieve the length. */
}
} else {
pDataStream->totalLengthInPCMFrames = 0;
}
/*
Only mark the decoder as initialized when the length of the decoder has been retrieved because that can possibly require a scan over the whole file
and we don't want to have another thread trying to access the decoder while it's scanning.
*/
pDataStream->isDecoderInitialized = MA_TRUE;
/* We have the decoder so we can now initialize our page buffer. */
pageBufferSizeInBytes = ma_resource_manager_data_stream_get_page_size_in_frames(pDataStream) * 2 * ma_get_bytes_per_frame(pDataStream->decoder.outputFormat, pDataStream->decoder.outputChannels);
pDataStream->pPageData = ma_malloc(pageBufferSizeInBytes, &pResourceManager->config.allocationCallbacks);
if (pDataStream->pPageData == NULL) {
ma_decoder_uninit(&pDataStream->decoder);
result = MA_OUT_OF_MEMORY;
goto done;
}
/* Seek to our initial seek point before filling the initial pages. */
ma_decoder_seek_to_pcm_frame(&pDataStream->decoder, pJob->data.resourceManager.loadDataStream.initialSeekPoint);
/* We have our decoder and our page buffer, so now we need to fill our pages. */
ma_resource_manager_data_stream_fill_pages(pDataStream);
/* And now we're done. We want to make sure the result is MA_SUCCESS. */
result = MA_SUCCESS;
done:
ma_free(pJob->data.resourceManager.loadDataStream.pFilePath, &pResourceManager->config.allocationCallbacks);
ma_free(pJob->data.resourceManager.loadDataStream.pFilePathW, &pResourceManager->config.allocationCallbacks);
/* We can only change the status away from MA_BUSY. If it's set to anything else it means an error has occurred somewhere or the uninitialization process has started (most likely). */
c89atomic_compare_and_swap_i32(&pDataStream->result, MA_BUSY, result);
/* Only signal the other threads after the result has been set just for cleanliness sake. */
if (pJob->data.resourceManager.loadDataStream.pInitNotification != NULL) {
ma_async_notification_signal(pJob->data.resourceManager.loadDataStream.pInitNotification);
}
if (pJob->data.resourceManager.loadDataStream.pInitFence != NULL) {
ma_fence_release(pJob->data.resourceManager.loadDataStream.pInitFence);
}
c89atomic_fetch_add_32(&pDataStream->executionPointer, 1);
return result;
}
static ma_result ma_job_process__resource_manager__free_data_stream(ma_job* pJob)
{
ma_resource_manager* pResourceManager;
ma_resource_manager_data_stream* pDataStream;
MA_ASSERT(pJob != NULL);
pDataStream = (ma_resource_manager_data_stream*)pJob->data.resourceManager.freeDataStream.pDataStream;
MA_ASSERT(pDataStream != NULL);
pResourceManager = pDataStream->pResourceManager;
if (pJob->order != c89atomic_load_32(&pDataStream->executionPointer)) {
return ma_resource_manager_post_job(pResourceManager, pJob); /* Out of order. */
}
/* If our status is not MA_UNAVAILABLE we have a bug somewhere. */
MA_ASSERT(ma_resource_manager_data_stream_result(pDataStream) == MA_UNAVAILABLE);
if (pDataStream->isDecoderInitialized) {
ma_decoder_uninit(&pDataStream->decoder);
}
if (pDataStream->pPageData != NULL) {
ma_free(pDataStream->pPageData, &pResourceManager->config.allocationCallbacks);
pDataStream->pPageData = NULL; /* Just in case... */
}
ma_data_source_uninit(&pDataStream->ds);
/* The event needs to be signalled last. */
if (pJob->data.resourceManager.freeDataStream.pDoneNotification != NULL) {
ma_async_notification_signal(pJob->data.resourceManager.freeDataStream.pDoneNotification);
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
/*c89atomic_fetch_add_32(&pDataStream->executionPointer, 1);*/
return MA_SUCCESS;
}
static ma_result ma_job_process__resource_manager__page_data_stream(ma_job* pJob)
{
ma_result result = MA_SUCCESS;
ma_resource_manager* pResourceManager;
ma_resource_manager_data_stream* pDataStream;
MA_ASSERT(pJob != NULL);
pDataStream = (ma_resource_manager_data_stream*)pJob->data.resourceManager.pageDataStream.pDataStream;
MA_ASSERT(pDataStream != NULL);
pResourceManager = pDataStream->pResourceManager;
if (pJob->order != c89atomic_load_32(&pDataStream->executionPointer)) {
return ma_resource_manager_post_job(pResourceManager, pJob); /* Out of order. */
}
/* For streams, the status should be MA_SUCCESS. */
if (ma_resource_manager_data_stream_result(pDataStream) != MA_SUCCESS) {
result = MA_INVALID_OPERATION;
goto done;
}
ma_resource_manager_data_stream_fill_page(pDataStream, pJob->data.resourceManager.pageDataStream.pageIndex);
done:
c89atomic_fetch_add_32(&pDataStream->executionPointer, 1);
return result;
}
static ma_result ma_job_process__resource_manager__seek_data_stream(ma_job* pJob)
{
ma_result result = MA_SUCCESS;
ma_resource_manager* pResourceManager;
ma_resource_manager_data_stream* pDataStream;
MA_ASSERT(pJob != NULL);
pDataStream = (ma_resource_manager_data_stream*)pJob->data.resourceManager.seekDataStream.pDataStream;
MA_ASSERT(pDataStream != NULL);
pResourceManager = pDataStream->pResourceManager;
if (pJob->order != c89atomic_load_32(&pDataStream->executionPointer)) {
return ma_resource_manager_post_job(pResourceManager, pJob); /* Out of order. */
}
/* For streams the status should be MA_SUCCESS for this to do anything. */
if (ma_resource_manager_data_stream_result(pDataStream) != MA_SUCCESS || pDataStream->isDecoderInitialized == MA_FALSE) {
result = MA_INVALID_OPERATION;
goto done;
}
/*
With seeking we just assume both pages are invalid and the relative frame cursor at position 0. This is basically exactly the same as loading, except
instead of initializing the decoder, we seek to a frame.
*/
ma_decoder_seek_to_pcm_frame(&pDataStream->decoder, pJob->data.resourceManager.seekDataStream.frameIndex);
/* After seeking we'll need to reload the pages. */
ma_resource_manager_data_stream_fill_pages(pDataStream);
/* We need to let the public API know that we're done seeking. */
c89atomic_fetch_sub_32(&pDataStream->seekCounter, 1);
done:
c89atomic_fetch_add_32(&pDataStream->executionPointer, 1);
return result;
}
MA_API ma_result ma_resource_manager_process_job(ma_resource_manager* pResourceManager, ma_job* pJob)
{
if (pResourceManager == NULL || pJob == NULL) {
return MA_INVALID_ARGS;
}
return ma_job_process(pJob);
}
MA_API ma_result ma_resource_manager_process_next_job(ma_resource_manager* pResourceManager)
{
ma_result result;
ma_job job;
if (pResourceManager == NULL) {
return MA_INVALID_ARGS;
}
/* This will return MA_CANCELLED if the next job is a quit job. */
result = ma_resource_manager_next_job(pResourceManager, &job);
if (result != MA_SUCCESS) {
return result;
}
return ma_job_process(&job);
}
#else
/* We'll get here if the resource manager is being excluded from the build. We need to define the job processing callbacks as no-ops. */
static ma_result ma_job_process__resource_manager__load_data_buffer_node(ma_job* pJob) { return ma_job_process__noop(pJob); }
static ma_result ma_job_process__resource_manager__free_data_buffer_node(ma_job* pJob) { return ma_job_process__noop(pJob); }
static ma_result ma_job_process__resource_manager__page_data_buffer_node(ma_job* pJob) { return ma_job_process__noop(pJob); }
static ma_result ma_job_process__resource_manager__load_data_buffer(ma_job* pJob) { return ma_job_process__noop(pJob); }
static ma_result ma_job_process__resource_manager__free_data_buffer(ma_job* pJob) { return ma_job_process__noop(pJob); }
static ma_result ma_job_process__resource_manager__load_data_stream(ma_job* pJob) { return ma_job_process__noop(pJob); }
static ma_result ma_job_process__resource_manager__free_data_stream(ma_job* pJob) { return ma_job_process__noop(pJob); }
static ma_result ma_job_process__resource_manager__page_data_stream(ma_job* pJob) { return ma_job_process__noop(pJob); }
static ma_result ma_job_process__resource_manager__seek_data_stream(ma_job* pJob) { return ma_job_process__noop(pJob); }
#endif /* MA_NO_RESOURCE_MANAGER */
#ifndef MA_NO_NODE_GRAPH
/* 10ms @ 48K = 480. Must never exceed 65535. */
#ifndef MA_DEFAULT_NODE_CACHE_CAP_IN_FRAMES_PER_BUS
#define MA_DEFAULT_NODE_CACHE_CAP_IN_FRAMES_PER_BUS 480
#endif
static ma_result ma_node_read_pcm_frames(ma_node* pNode, ma_uint32 outputBusIndex, float* pFramesOut, ma_uint32 frameCount, ma_uint32* pFramesRead, ma_uint64 globalTime);
MA_API void ma_debug_fill_pcm_frames_with_sine_wave(float* pFramesOut, ma_uint32 frameCount, ma_format format, ma_uint32 channels, ma_uint32 sampleRate)
{
#ifndef MA_NO_GENERATION
{
ma_waveform_config waveformConfig;
ma_waveform waveform;
waveformConfig = ma_waveform_config_init(format, channels, sampleRate, ma_waveform_type_sine, 1.0, 400);
ma_waveform_init(&waveformConfig, &waveform);
ma_waveform_read_pcm_frames(&waveform, pFramesOut, frameCount, NULL);
}
#else
{
(void)pFramesOut;
(void)frameCount;
(void)format;
(void)channels;
(void)sampleRate;
#if defined(MA_DEBUG_OUTPUT)
{
#if _MSC_VER
#pragma message ("ma_debug_fill_pcm_frames_with_sine_wave() will do nothing because MA_NO_GENERATION is enabled.")
#endif
}
#endif
}
#endif
}
static ma_result ma_mix_pcm_frames_f32(float* pDst, const float* pSrc, ma_uint64 frameCount, ma_uint32 channels, float volume)
{
ma_uint64 iSample;
ma_uint64 sampleCount;
if (pDst == NULL || pSrc == NULL || channels == 0) {
return MA_INVALID_ARGS;
}
if (volume == 0) {
return MA_SUCCESS; /* No changes if the volume is 0. */
}
sampleCount = frameCount * channels;
if (volume == 1) {
for (iSample = 0; iSample < sampleCount; iSample += 1) {
pDst[iSample] += pSrc[iSample];
}
} else {
for (iSample = 0; iSample < sampleCount; iSample += 1) {
pDst[iSample] += ma_apply_volume_unclipped_f32(pSrc[iSample], volume);
}
}
return MA_SUCCESS;
}
MA_API ma_node_graph_config ma_node_graph_config_init(ma_uint32 channels)
{
ma_node_graph_config config;
MA_ZERO_OBJECT(&config);
config.channels = channels;
config.nodeCacheCapInFrames = MA_DEFAULT_NODE_CACHE_CAP_IN_FRAMES_PER_BUS;
return config;
}
static void ma_node_graph_set_is_reading(ma_node_graph* pNodeGraph, ma_bool32 isReading)
{
MA_ASSERT(pNodeGraph != NULL);
c89atomic_exchange_32(&pNodeGraph->isReading, isReading);
}
#if 0
static ma_bool32 ma_node_graph_is_reading(ma_node_graph* pNodeGraph)
{
MA_ASSERT(pNodeGraph != NULL);
return c89atomic_load_32(&pNodeGraph->isReading);
}
#endif
static void ma_node_graph_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_node_graph* pNodeGraph = (ma_node_graph*)pNode;
ma_uint64 framesRead;
ma_node_graph_read_pcm_frames(pNodeGraph, ppFramesOut[0], *pFrameCountOut, &framesRead);
*pFrameCountOut = (ma_uint32)framesRead; /* Safe cast. */
(void)ppFramesIn;
(void)pFrameCountIn;
}
static ma_node_vtable g_node_graph_node_vtable =
{
ma_node_graph_node_process_pcm_frames,
NULL, /* onGetRequiredInputFrameCount */
0, /* 0 input buses. */
1, /* 1 output bus. */
0 /* Flags. */
};
static void ma_node_graph_endpoint_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
MA_ASSERT(pNode != NULL);
MA_ASSERT(ma_node_get_input_bus_count(pNode) == 1);
MA_ASSERT(ma_node_get_output_bus_count(pNode) == 1);
/* Input channel count needs to be the same as the output channel count. */
MA_ASSERT(ma_node_get_input_channels(pNode, 0) == ma_node_get_output_channels(pNode, 0));
/* We don't need to do anything here because it's a passthrough. */
(void)pNode;
(void)ppFramesIn;
(void)pFrameCountIn;
(void)ppFramesOut;
(void)pFrameCountOut;
#if 0
/* The data has already been mixed. We just need to move it to the output buffer. */
if (ppFramesIn != NULL) {
ma_copy_pcm_frames(ppFramesOut[0], ppFramesIn[0], *pFrameCountOut, ma_format_f32, ma_node_get_output_channels(pNode, 0));
}
#endif
}
static ma_node_vtable g_node_graph_endpoint_vtable =
{
ma_node_graph_endpoint_process_pcm_frames,
NULL, /* onGetRequiredInputFrameCount */
1, /* 1 input bus. */
1, /* 1 output bus. */
MA_NODE_FLAG_PASSTHROUGH /* Flags. The endpoint is a passthrough. */
};
MA_API ma_result ma_node_graph_init(const ma_node_graph_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_node_graph* pNodeGraph)
{
ma_result result;
ma_node_config baseConfig;
ma_node_config endpointConfig;
if (pNodeGraph == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pNodeGraph);
pNodeGraph->nodeCacheCapInFrames = pConfig->nodeCacheCapInFrames;
if (pNodeGraph->nodeCacheCapInFrames == 0) {
pNodeGraph->nodeCacheCapInFrames = MA_DEFAULT_NODE_CACHE_CAP_IN_FRAMES_PER_BUS;
}
/* Base node so we can use the node graph as a node into another graph. */
baseConfig = ma_node_config_init();
baseConfig.vtable = &g_node_graph_node_vtable;
baseConfig.pOutputChannels = &pConfig->channels;
result = ma_node_init(pNodeGraph, &baseConfig, pAllocationCallbacks, &pNodeGraph->base);
if (result != MA_SUCCESS) {
return result;
}
/* Endpoint. */
endpointConfig = ma_node_config_init();
endpointConfig.vtable = &g_node_graph_endpoint_vtable;
endpointConfig.pInputChannels = &pConfig->channels;
endpointConfig.pOutputChannels = &pConfig->channels;
result = ma_node_init(pNodeGraph, &endpointConfig, pAllocationCallbacks, &pNodeGraph->endpoint);
if (result != MA_SUCCESS) {
ma_node_uninit(&pNodeGraph->base, pAllocationCallbacks);
return result;
}
return MA_SUCCESS;
}
MA_API void ma_node_graph_uninit(ma_node_graph* pNodeGraph, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pNodeGraph == NULL) {
return;
}
ma_node_uninit(&pNodeGraph->endpoint, pAllocationCallbacks);
}
MA_API ma_node* ma_node_graph_get_endpoint(ma_node_graph* pNodeGraph)
{
if (pNodeGraph == NULL) {
return NULL;
}
return &pNodeGraph->endpoint;
}
MA_API ma_result ma_node_graph_read_pcm_frames(ma_node_graph* pNodeGraph, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
ma_result result = MA_SUCCESS;
ma_uint64 totalFramesRead;
ma_uint32 channels;
if (pFramesRead != NULL) {
*pFramesRead = 0; /* Safety. */
}
if (pNodeGraph == NULL) {
return MA_INVALID_ARGS;
}
channels = ma_node_get_output_channels(&pNodeGraph->endpoint, 0);
/* We'll be nice and try to do a full read of all frameCount frames. */
totalFramesRead = 0;
while (totalFramesRead < frameCount) {
ma_uint32 framesJustRead;
ma_uint64 framesToRead = frameCount - totalFramesRead;
if (framesToRead > 0xFFFFFFFF) {
framesToRead = 0xFFFFFFFF;
}
ma_node_graph_set_is_reading(pNodeGraph, MA_TRUE);
{
result = ma_node_read_pcm_frames(&pNodeGraph->endpoint, 0, (float*)ma_offset_pcm_frames_ptr(pFramesOut, totalFramesRead, ma_format_f32, channels), (ma_uint32)framesToRead, &framesJustRead, ma_node_get_time(&pNodeGraph->endpoint));
}
ma_node_graph_set_is_reading(pNodeGraph, MA_FALSE);
totalFramesRead += framesJustRead;
if (result != MA_SUCCESS) {
break;
}
/* Abort if we weren't able to read any frames or else we risk getting stuck in a loop. */
if (framesJustRead == 0) {
break;
}
}
/* Let's go ahead and silence any leftover frames just for some added safety to ensure the caller doesn't try emitting garbage out of the speakers. */
if (totalFramesRead < frameCount) {
ma_silence_pcm_frames(ma_offset_pcm_frames_ptr(pFramesOut, totalFramesRead, ma_format_f32, channels), (frameCount - totalFramesRead), ma_format_f32, channels);
}
if (pFramesRead != NULL) {
*pFramesRead = totalFramesRead;
}
return result;
}
MA_API ma_uint32 ma_node_graph_get_channels(const ma_node_graph* pNodeGraph)
{
if (pNodeGraph == NULL) {
return 0;
}
return ma_node_get_output_channels(&pNodeGraph->endpoint, 0);
}
MA_API ma_uint64 ma_node_graph_get_time(const ma_node_graph* pNodeGraph)
{
if (pNodeGraph == NULL) {
return 0;
}
return ma_node_get_time(&pNodeGraph->endpoint); /* Global time is just the local time of the endpoint. */
}
MA_API ma_result ma_node_graph_set_time(ma_node_graph* pNodeGraph, ma_uint64 globalTime)
{
if (pNodeGraph == NULL) {
return MA_INVALID_ARGS;
}
return ma_node_set_time(&pNodeGraph->endpoint, globalTime); /* Global time is just the local time of the endpoint. */
}
#define MA_NODE_OUTPUT_BUS_FLAG_HAS_READ 0x01 /* Whether or not this bus ready to read more data. Only used on nodes with multiple output buses. */
static ma_result ma_node_output_bus_init(ma_node* pNode, ma_uint32 outputBusIndex, ma_uint32 channels, ma_node_output_bus* pOutputBus)
{
MA_ASSERT(pOutputBus != NULL);
MA_ASSERT(outputBusIndex < MA_MAX_NODE_BUS_COUNT);
MA_ASSERT(outputBusIndex < ma_node_get_output_bus_count(pNode));
MA_ASSERT(channels < 256);
MA_ZERO_OBJECT(pOutputBus);
if (channels == 0) {
return MA_INVALID_ARGS;
}
pOutputBus->pNode = pNode;
pOutputBus->outputBusIndex = (ma_uint8)outputBusIndex;
pOutputBus->channels = (ma_uint8)channels;
pOutputBus->flags = MA_NODE_OUTPUT_BUS_FLAG_HAS_READ; /* <-- Important that this flag is set by default. */
pOutputBus->volume = 1;
return MA_SUCCESS;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
}
ma_node_input_bus_unlock(pInputBus);
/*
Mark the node as attached last. This is used to controlling whether or the output bus will be
iterated on the audio thread. Mainly required for detachment purposes.
*/
ma_node_output_bus_set_is_attached(pOutputBus, MA_TRUE);
}
ma_node_output_bus_unlock(pOutputBus);
}
static ma_node_output_bus* ma_node_input_bus_next(ma_node_input_bus* pInputBus, ma_node_output_bus* pOutputBus)
{
ma_node_output_bus* pNext;
MA_ASSERT(pInputBus != NULL);
if (pOutputBus == NULL) {
return NULL;
}
ma_node_input_bus_next_begin(pInputBus);
{
pNext = pOutputBus;
for (;;) {
pNext = (ma_node_output_bus*)c89atomic_load_ptr(&pNext->pNext);
if (pNext == NULL) {
break; /* Reached the end. */
}
if (ma_node_output_bus_is_attached(pNext) == MA_FALSE) {
continue; /* The node is not attached. Keep checking. */
}
/* The next node has been selected. */
break;
}
/* We need to increment the reference count of the selected node. */
if (pNext != NULL) {
c89atomic_fetch_add_32(&pNext->refCount, 1);
}
/* The previous node is no longer being referenced. */
c89atomic_fetch_sub_32(&pOutputBus->refCount, 1);
}
ma_node_input_bus_next_end(pInputBus);
return pNext;
}
static ma_node_output_bus* ma_node_input_bus_first(ma_node_input_bus* pInputBus)
{
return ma_node_input_bus_next(pInputBus, &pInputBus->head);
}
static ma_result ma_node_input_bus_read_pcm_frames(ma_node* pInputNode, ma_node_input_bus* pInputBus, float* pFramesOut, ma_uint32 frameCount, ma_uint32* pFramesRead, ma_uint64 globalTime)
{
ma_result result = MA_SUCCESS;
ma_node_output_bus* pOutputBus;
ma_node_output_bus* pFirst;
ma_uint32 inputChannels;
ma_bool32 doesOutputBufferHaveContent = MA_FALSE;
/*
This will be called from the audio thread which means we can't be doing any locking. Basically,
this function will not perfom any locking, whereas attaching and detaching will, but crafted in
such a way that we don't need to perform any locking here. The important thing to remember is
to always iterate in a forward direction.
In order to process any data we need to first read from all input buses. That's where this
function comes in. This iterates over each of the attachments and accumulates/mixes them. We
also convert the channels to the nodes output channel count before mixing. We want to do this
channel conversion so that the caller of this function can invoke the processing callback
without having to do it themselves.
When we iterate over each of the attachments on the input bus, we need to read as much data as
we can from each of them so that we don't end up with holes between each of the attachments. To
do this, we need to read from each attachment in a loop and read as many frames as we can, up
to `frameCount`.
*/
MA_ASSERT(pInputNode != NULL);
MA_ASSERT(pFramesRead != NULL); /* pFramesRead is critical and must always be specified. On input it's undefined and on output it'll be set to the number of frames actually read. */
*pFramesRead = 0; /* Safety. */
inputChannels = ma_node_input_bus_get_channels(pInputBus);
/*
We need to be careful with how we call ma_node_input_bus_first() and ma_node_input_bus_next(). They
are both critical to our lock-free thread-safety system. We can only call ma_node_input_bus_first()
once per iteration, however we have an optimization to checks whether or not it's the first item in
the list. We therefore need to store a pointer to the first item rather than repeatedly calling
ma_node_input_bus_first(). It's safe to keep hold of this pointer, so long as we don't dereference it
after calling ma_node_input_bus_next(), which we won't be.
*/
pFirst = ma_node_input_bus_first(pInputBus);
if (pFirst == NULL) {
return MA_SUCCESS; /* No attachments. Read nothing. */
}
for (pOutputBus = pFirst; pOutputBus != NULL; pOutputBus = ma_node_input_bus_next(pInputBus, pOutputBus)) {
ma_uint32 framesProcessed = 0;
ma_bool32 isSilentOutput = MA_FALSE;
MA_ASSERT(pOutputBus->pNode != NULL);
isSilentOutput = (((ma_node_base*)pOutputBus->pNode)->vtable->flags & MA_NODE_FLAG_SILENT_OUTPUT) != 0;
if (pFramesOut != NULL) {
/* Read. */
float temp[MA_DATA_CONVERTER_STACK_BUFFER_SIZE / sizeof(float)];
ma_uint32 tempCapInFrames = ma_countof(temp) / inputChannels;
while (framesProcessed < frameCount) {
float* pRunningFramesOut;
ma_uint32 framesToRead;
ma_uint32 framesJustRead;
framesToRead = frameCount - framesProcessed;
if (framesToRead > tempCapInFrames) {
framesToRead = tempCapInFrames;
}
pRunningFramesOut = ma_offset_pcm_frames_ptr_f32(pFramesOut, framesProcessed, inputChannels);
if (doesOutputBufferHaveContent == MA_FALSE) {
/* Fast path. First attachment. We just read straight into the output buffer (no mixing required). */
result = ma_node_read_pcm_frames(pOutputBus->pNode, pOutputBus->outputBusIndex, pRunningFramesOut, framesToRead, &framesJustRead, globalTime + framesProcessed);
} else {
/* Slow path. Not the first attachment. Mixing required. */
result = ma_node_read_pcm_frames(pOutputBus->pNode, pOutputBus->outputBusIndex, temp, framesToRead, &framesJustRead, globalTime + framesProcessed);
if (result == MA_SUCCESS || result == MA_AT_END) {
if (isSilentOutput == MA_FALSE) { /* Don't mix if the node outputs silence. */
ma_mix_pcm_frames_f32(pRunningFramesOut, temp, framesJustRead, inputChannels, /*volume*/1);
}
}
}
framesProcessed += framesJustRead;
/* If we reached the end or otherwise failed to read any data we need to finish up with this output node. */
if (result != MA_SUCCESS) {
break;
}
/* If we didn't read anything, abort so we don't get stuck in a loop. */
if (framesJustRead == 0) {
break;
}
}
/* If it's the first attachment we didn't do any mixing. Any leftover samples need to be silenced. */
if (pOutputBus == pFirst && framesProcessed < frameCount) {
ma_silence_pcm_frames(ma_offset_pcm_frames_ptr(pFramesOut, framesProcessed, ma_format_f32, inputChannels), (frameCount - framesProcessed), ma_format_f32, inputChannels);
}
if (isSilentOutput == MA_FALSE) {
doesOutputBufferHaveContent = MA_TRUE;
}
} else {
/* Seek. */
ma_node_read_pcm_frames(pOutputBus->pNode, pOutputBus->outputBusIndex, NULL, frameCount, &framesProcessed, globalTime);
}
}
/* If we didn't output anything, output silence. */
if (doesOutputBufferHaveContent == MA_FALSE && pFramesOut != NULL) {
ma_silence_pcm_frames(pFramesOut, frameCount, ma_format_f32, inputChannels);
}
/* In this path we always "process" the entire amount. */
*pFramesRead = frameCount;
return result;
}
MA_API ma_node_config ma_node_config_init(void)
{
ma_node_config config;
MA_ZERO_OBJECT(&config);
config.initialState = ma_node_state_started; /* Nodes are started by default. */
config.inputBusCount = MA_NODE_BUS_COUNT_UNKNOWN;
config.outputBusCount = MA_NODE_BUS_COUNT_UNKNOWN;
return config;
}
static ma_result ma_node_detach_full(ma_node* pNode);
static float* ma_node_get_cached_input_ptr(ma_node* pNode, ma_uint32 inputBusIndex)
{
ma_node_base* pNodeBase = (ma_node_base*)pNode;
ma_uint32 iInputBus;
float* pBasePtr;
MA_ASSERT(pNodeBase != NULL);
/* Input data is stored at the front of the buffer. */
pBasePtr = pNodeBase->pCachedData;
for (iInputBus = 0; iInputBus < inputBusIndex; iInputBus += 1) {
pBasePtr += pNodeBase->cachedDataCapInFramesPerBus * ma_node_input_bus_get_channels(&pNodeBase->pInputBuses[iInputBus]);
}
return pBasePtr;
}
static float* ma_node_get_cached_output_ptr(ma_node* pNode, ma_uint32 outputBusIndex)
{
ma_node_base* pNodeBase = (ma_node_base*)pNode;
ma_uint32 iInputBus;
ma_uint32 iOutputBus;
float* pBasePtr;
MA_ASSERT(pNodeBase != NULL);
/* Cached output data starts after the input data. */
pBasePtr = pNodeBase->pCachedData;
for (iInputBus = 0; iInputBus < ma_node_get_input_bus_count(pNodeBase); iInputBus += 1) {
pBasePtr += pNodeBase->cachedDataCapInFramesPerBus * ma_node_input_bus_get_channels(&pNodeBase->pInputBuses[iInputBus]);
}
for (iOutputBus = 0; iOutputBus < outputBusIndex; iOutputBus += 1) {
pBasePtr += pNodeBase->cachedDataCapInFramesPerBus * ma_node_output_bus_get_channels(&pNodeBase->pOutputBuses[iOutputBus]);
}
return pBasePtr;
}
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if (pConfig->pInputChannels[0] != pConfig->pOutputChannels[0]) {
return MA_INVALID_ARGS; /* Passthrough nodes must have the same number of channels between input and output nodes. */
}
}
*pInputBusCount = inputBusCount;
*pOutputBusCount = outputBusCount;
return MA_SUCCESS;
}
static ma_result ma_node_get_heap_layout(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, ma_node_heap_layout* pHeapLayout)
{
ma_result result;
ma_uint32 inputBusCount;
ma_uint32 outputBusCount;
MA_ASSERT(pHeapLayout != NULL);
MA_ZERO_OBJECT(pHeapLayout);
if (pConfig == NULL || pConfig->vtable == NULL || pConfig->vtable->onProcess == NULL) {
return MA_INVALID_ARGS;
}
result = ma_node_translate_bus_counts(pConfig, &inputBusCount, &outputBusCount);
if (result != MA_SUCCESS) {
return result;
}
pHeapLayout->sizeInBytes = 0;
/* Input buses. */
if (inputBusCount > MA_MAX_NODE_LOCAL_BUS_COUNT) {
pHeapLayout->inputBusOffset = pHeapLayout->sizeInBytes;
pHeapLayout->sizeInBytes += ma_align_64(sizeof(ma_node_input_bus) * inputBusCount);
} else {
pHeapLayout->inputBusOffset = MA_SIZE_MAX; /* MA_SIZE_MAX indicates that no heap allocation is required for the input bus. */
}
/* Output buses. */
if (outputBusCount > MA_MAX_NODE_LOCAL_BUS_COUNT) {
pHeapLayout->outputBusOffset = pHeapLayout->sizeInBytes;
pHeapLayout->sizeInBytes += ma_align_64(sizeof(ma_node_output_bus) * outputBusCount);
} else {
pHeapLayout->outputBusOffset = MA_SIZE_MAX;
}
/*
Cached audio data.
We need to allocate memory for a caching both input and output data. We have an optimization
where no caching is necessary for specific conditions:
- The node has 0 inputs and 1 output.
When a node meets the above conditions, no cache is allocated.
The size choice for this buffer is a little bit finicky. We don't want to be too wasteful by
allocating too much, but at the same time we want it be large enough so that enough frames can
be processed for each call to ma_node_read_pcm_frames() so that it keeps things efficient. For
now I'm going with 10ms @ 48K which is 480 frames per bus. This is configurable at compile
time. It might also be worth investigating whether or not this can be configured at run time.
*/
if (inputBusCount == 0 && outputBusCount == 1) {
/* Fast path. No cache needed. */
pHeapLayout->cachedDataOffset = MA_SIZE_MAX;
} else {
/* Slow path. Cache needed. */
size_t cachedDataSizeInBytes = 0;
ma_uint32 iBus;
for (iBus = 0; iBus < inputBusCount; iBus += 1) {
cachedDataSizeInBytes += pNodeGraph->nodeCacheCapInFrames * ma_get_bytes_per_frame(ma_format_f32, pConfig->pInputChannels[iBus]);
}
for (iBus = 0; iBus < outputBusCount; iBus += 1) {
cachedDataSizeInBytes += pNodeGraph->nodeCacheCapInFrames * ma_get_bytes_per_frame(ma_format_f32, pConfig->pOutputChannels[iBus]);
}
pHeapLayout->cachedDataOffset = pHeapLayout->sizeInBytes;
pHeapLayout->sizeInBytes += ma_align_64(cachedDataSizeInBytes);
}
/*
Not technically part of the heap, but we can output the input and output bus counts so we can
avoid a redundant call to ma_node_translate_bus_counts().
*/
pHeapLayout->inputBusCount = inputBusCount;
pHeapLayout->outputBusCount = outputBusCount;
/* Make sure allocation size is aligned. */
pHeapLayout->sizeInBytes = ma_align_64(pHeapLayout->sizeInBytes);
return MA_SUCCESS;
}
MA_API ma_result ma_node_get_heap_size(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, size_t* pHeapSizeInBytes)
{
ma_result result;
ma_node_heap_layout heapLayout;
if (pHeapSizeInBytes == NULL) {
return MA_INVALID_ARGS;
}
*pHeapSizeInBytes = 0;
result = ma_node_get_heap_layout(pNodeGraph, pConfig, &heapLayout);
if (result != MA_SUCCESS) {
return result;
}
*pHeapSizeInBytes = heapLayout.sizeInBytes;
return MA_SUCCESS;
}
MA_API ma_result ma_node_init_preallocated(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, void* pHeap, ma_node* pNode)
{
ma_node_base* pNodeBase = (ma_node_base*)pNode;
ma_result result;
ma_node_heap_layout heapLayout;
ma_uint32 iInputBus;
ma_uint32 iOutputBus;
if (pNodeBase == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pNodeBase);
result = ma_node_get_heap_layout(pNodeGraph, pConfig, &heapLayout);
if (result != MA_SUCCESS) {
return result;
}
pNodeBase->_pHeap = pHeap;
MA_ZERO_MEMORY(pHeap, heapLayout.sizeInBytes);
pNodeBase->pNodeGraph = pNodeGraph;
pNodeBase->vtable = pConfig->vtable;
pNodeBase->state = pConfig->initialState;
pNodeBase->stateTimes[ma_node_state_started] = 0;
pNodeBase->stateTimes[ma_node_state_stopped] = (ma_uint64)(ma_int64)-1; /* Weird casting for VC6 compatibility. */
pNodeBase->inputBusCount = heapLayout.inputBusCount;
pNodeBase->outputBusCount = heapLayout.outputBusCount;
if (heapLayout.inputBusOffset != MA_SIZE_MAX) {
pNodeBase->pInputBuses = (ma_node_input_bus*)ma_offset_ptr(pHeap, heapLayout.inputBusOffset);
} else {
pNodeBase->pInputBuses = pNodeBase->_inputBuses;
}
if (heapLayout.outputBusOffset != MA_SIZE_MAX) {
pNodeBase->pOutputBuses = (ma_node_output_bus*)ma_offset_ptr(pHeap, heapLayout.inputBusOffset);
} else {
pNodeBase->pOutputBuses = pNodeBase->_outputBuses;
}
if (heapLayout.cachedDataOffset != MA_SIZE_MAX) {
pNodeBase->pCachedData = (float*)ma_offset_ptr(pHeap, heapLayout.cachedDataOffset);
pNodeBase->cachedDataCapInFramesPerBus = pNodeGraph->nodeCacheCapInFrames;
} else {
pNodeBase->pCachedData = NULL;
}
/* We need to run an initialization step for each input and output bus. */
for (iInputBus = 0; iInputBus < ma_node_get_input_bus_count(pNodeBase); iInputBus += 1) {
result = ma_node_input_bus_init(pConfig->pInputChannels[iInputBus], &pNodeBase->pInputBuses[iInputBus]);
if (result != MA_SUCCESS) {
return result;
}
}
for (iOutputBus = 0; iOutputBus < ma_node_get_output_bus_count(pNodeBase); iOutputBus += 1) {
result = ma_node_output_bus_init(pNodeBase, iOutputBus, pConfig->pOutputChannels[iOutputBus], &pNodeBase->pOutputBuses[iOutputBus]);
if (result != MA_SUCCESS) {
return result;
}
}
/* The cached data needs to be initialized to silence (or a sine wave tone if we're debugging). */
if (pNodeBase->pCachedData != NULL) {
ma_uint32 iBus;
#if 1 /* Toggle this between 0 and 1 to turn debugging on or off. 1 = fill with a sine wave for debugging; 0 = fill with silence. */
/* For safety we'll go ahead and default the buffer to silence. */
for (iBus = 0; iBus < ma_node_get_input_bus_count(pNodeBase); iBus += 1) {
ma_silence_pcm_frames(ma_node_get_cached_input_ptr(pNode, iBus), pNodeBase->cachedDataCapInFramesPerBus, ma_format_f32, ma_node_input_bus_get_channels(&pNodeBase->pInputBuses[iBus]));
}
for (iBus = 0; iBus < ma_node_get_output_bus_count(pNodeBase); iBus += 1) {
ma_silence_pcm_frames(ma_node_get_cached_output_ptr(pNode, iBus), pNodeBase->cachedDataCapInFramesPerBus, ma_format_f32, ma_node_output_bus_get_channels(&pNodeBase->pOutputBuses[iBus]));
}
#else
/* For debugging. Default to a sine wave. */
for (iBus = 0; iBus < ma_node_get_input_bus_count(pNodeBase); iBus += 1) {
ma_debug_fill_pcm_frames_with_sine_wave(ma_node_get_cached_input_ptr(pNode, iBus), pNodeBase->cachedDataCapInFramesPerBus, ma_format_f32, ma_node_input_bus_get_channels(&pNodeBase->pInputBuses[iBus]), 48000);
}
for (iBus = 0; iBus < ma_node_get_output_bus_count(pNodeBase); iBus += 1) {
ma_debug_fill_pcm_frames_with_sine_wave(ma_node_get_cached_output_ptr(pNode, iBus), pNodeBase->cachedDataCapInFramesPerBus, ma_format_f32, ma_node_output_bus_get_channels(&pNodeBase->pOutputBuses[iBus]), 48000);
}
#endif
}
return MA_SUCCESS;
}
MA_API ma_result ma_node_init(ma_node_graph* pNodeGraph, const ma_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_node* pNode)
{
ma_result result;
size_t heapSizeInBytes;
void* pHeap;
result = ma_node_get_heap_size(pNodeGraph, pConfig, &heapSizeInBytes);
if (result != MA_SUCCESS) {
return result;
}
if (heapSizeInBytes > 0) {
pHeap = ma_malloc(heapSizeInBytes, pAllocationCallbacks);
if (pHeap == NULL) {
return MA_OUT_OF_MEMORY;
}
} else {
pHeap = NULL;
}
result = ma_node_init_preallocated(pNodeGraph, pConfig, pHeap, pNode);
if (result != MA_SUCCESS) {
ma_free(pHeap, pAllocationCallbacks);
return result;
}
((ma_node_base*)pNode)->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
MA_API void ma_node_uninit(ma_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_node_base* pNodeBase = (ma_node_base*)pNode;
if (pNodeBase == NULL) {
return;
}
/*
The first thing we need to do is fully detach the node. This will detach all inputs and
outputs. We need to do this first because it will sever the connection with the node graph and
allow us to complete uninitialization without needing to worry about thread-safety with the
audio thread. The detachment process will wait for any local processing of the node to finish.
*/
ma_node_detach_full(pNode);
/*
At this point the node should be completely unreferenced by the node graph and we can finish up
the uninitialization process without needing to worry about thread-safety.
*/
if (pNodeBase->_ownsHeap) {
ma_free(pNodeBase->_pHeap, pAllocationCallbacks);
}
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return ma_node_state_stopped;
}
return ma_node_get_state_by_time_range(pNode, globalTime, globalTime);
}
MA_API ma_node_state ma_node_get_state_by_time_range(const ma_node* pNode, ma_uint64 globalTimeBeg, ma_uint64 globalTimeEnd)
{
ma_node_state state;
if (pNode == NULL) {
return ma_node_state_stopped;
}
state = ma_node_get_state(pNode);
/* An explicitly stopped node is always stopped. */
if (state == ma_node_state_stopped) {
return ma_node_state_stopped;
}
/*
Getting here means the node is marked as started, but it may still not be truly started due to
it's start time not having been reached yet. Also, the stop time may have also been reached in
which case it'll be considered stopped.
*/
if (ma_node_get_state_time(pNode, ma_node_state_started) > globalTimeBeg) {
return ma_node_state_stopped; /* Start time has not yet been reached. */
}
if (ma_node_get_state_time(pNode, ma_node_state_stopped) <= globalTimeEnd) {
return ma_node_state_stopped; /* Stop time has been reached. */
}
/* Getting here means the node is marked as started and is within it's start/stop times. */
return ma_node_state_started;
}
MA_API ma_uint64 ma_node_get_time(const ma_node* pNode)
{
if (pNode == NULL) {
return 0;
}
return c89atomic_load_64(&((ma_node_base*)pNode)->localTime);
}
MA_API ma_result ma_node_set_time(ma_node* pNode, ma_uint64 localTime)
{
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
c89atomic_exchange_64(&((ma_node_base*)pNode)->localTime, localTime);
return MA_SUCCESS;
}
static void ma_node_process_pcm_frames_internal(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_node_base* pNodeBase = (ma_node_base*)pNode;
MA_ASSERT(pNode != NULL);
if (pNodeBase->vtable->onProcess) {
pNodeBase->vtable->onProcess(pNode, ppFramesIn, pFrameCountIn, ppFramesOut, pFrameCountOut);
}
}
static ma_result ma_node_read_pcm_frames(ma_node* pNode, ma_uint32 outputBusIndex, float* pFramesOut, ma_uint32 frameCount, ma_uint32* pFramesRead, ma_uint64 globalTime)
{
ma_node_base* pNodeBase = (ma_node_base*)pNode;
ma_result result = MA_SUCCESS;
ma_uint32 iInputBus;
ma_uint32 iOutputBus;
ma_uint32 inputBusCount;
ma_uint32 outputBusCount;
ma_uint32 totalFramesRead = 0;
float* ppFramesIn[MA_MAX_NODE_BUS_COUNT];
float* ppFramesOut[MA_MAX_NODE_BUS_COUNT];
ma_uint64 globalTimeBeg;
ma_uint64 globalTimeEnd;
ma_uint64 startTime;
ma_uint64 stopTime;
ma_uint32 timeOffsetBeg;
ma_uint32 timeOffsetEnd;
ma_uint32 frameCountIn;
ma_uint32 frameCountOut;
/*
pFramesRead is mandatory. It must be used to determine how many frames were read. It's normal and
expected that the number of frames read may be different to that requested. Therefore, the caller
must look at this value to correctly determine how many frames were read.
*/
MA_ASSERT(pFramesRead != NULL); /* <-- If you've triggered this assert, you're using this function wrong. You *must* use this variable and inspect it after the call returns. */
if (pFramesRead == NULL) {
return MA_INVALID_ARGS;
}
*pFramesRead = 0; /* Safety. */
if (pNodeBase == NULL) {
return MA_INVALID_ARGS;
}
if (outputBusIndex >= ma_node_get_output_bus_count(pNodeBase)) {
return MA_INVALID_ARGS; /* Invalid output bus index. */
}
/* Don't do anything if we're in a stopped state. */
if (ma_node_get_state_by_time_range(pNode, globalTime, globalTime + frameCount) != ma_node_state_started) {
return MA_SUCCESS; /* We're in a stopped state. This is not an error - we just need to not read anything. */
}
globalTimeBeg = globalTime;
globalTimeEnd = globalTime + frameCount;
startTime = ma_node_get_state_time(pNode, ma_node_state_started);
stopTime = ma_node_get_state_time(pNode, ma_node_state_stopped);
/*
At this point we know that we are inside our start/stop times. However, we may need to adjust
our frame count and output pointer to accomodate since we could be straddling the time period
that this function is getting called for.
It's possible (and likely) that the start time does not line up with the output buffer. We
therefore need to offset it by a number of frames to accomodate. The same thing applies for
the stop time.
*/
timeOffsetBeg = (globalTimeBeg < startTime) ? (ma_uint32)(globalTimeEnd - startTime) : 0;
timeOffsetEnd = (globalTimeEnd > stopTime) ? (ma_uint32)(globalTimeEnd - stopTime) : 0;
/* Trim based on the start offset. We need to silence the start of the buffer. */
if (timeOffsetBeg > 0) {
ma_silence_pcm_frames(pFramesOut, timeOffsetBeg, ma_format_f32, ma_node_get_output_channels(pNode, outputBusIndex));
pFramesOut += timeOffsetBeg * ma_node_get_output_channels(pNode, outputBusIndex);
frameCount -= timeOffsetBeg;
}
/* Trim based on the end offset. We don't need to silence the tail section because we'll just have a reduced value written to pFramesRead. */
if (timeOffsetEnd > 0) {
frameCount -= timeOffsetEnd;
}
/* We run on different paths depending on the bus counts. */
inputBusCount = ma_node_get_input_bus_count(pNode);
outputBusCount = ma_node_get_output_bus_count(pNode);
/*
Run a simplified path when there are no inputs and one output. In this case there's nothing to
actually read and we can go straight to output. This is a very common scenario because the vast
majority of data source nodes will use this setup so this optimization I think is worthwhile.
*/
if (inputBusCount == 0 && outputBusCount == 1) {
/* Fast path. No need to read from input and no need for any caching. */
frameCountIn = 0;
frameCountOut = frameCount; /* Just read as much as we can. The callback will return what was actually read. */
ppFramesOut[0] = pFramesOut;
ma_node_process_pcm_frames_internal(pNode, NULL, &frameCountIn, ppFramesOut, &frameCountOut);
totalFramesRead = frameCountOut;
} else {
/* Slow path. Need to read input data. */
if ((pNodeBase->vtable->flags & MA_NODE_FLAG_PASSTHROUGH) != 0) {
/*
Fast path. We're running a passthrough. We need to read directly into the output buffer, but
still fire the callback so that event handling and trigger nodes can do their thing. Since
it's a passthrough there's no need for any kind of caching logic.
*/
MA_ASSERT(outputBusCount == inputBusCount);
MA_ASSERT(outputBusCount == 1);
MA_ASSERT(outputBusIndex == 0);
/* We just read directly from input bus to output buffer, and then afterwards fire the callback. */
ppFramesOut[0] = pFramesOut;
ppFramesIn[0] = ppFramesOut[0];
result = ma_node_input_bus_read_pcm_frames(pNodeBase, &pNodeBase->pInputBuses[0], ppFramesIn[0], frameCount, &totalFramesRead, globalTime);
if (result == MA_SUCCESS) {
/* Even though it's a passthrough, we still need to fire the callback. */
frameCountIn = totalFramesRead;
frameCountOut = totalFramesRead;
if (totalFramesRead > 0) {
ma_node_process_pcm_frames_internal(pNode, (const float**)ppFramesIn, &frameCountIn, ppFramesOut, &frameCountOut); /* From GCC: expected 'const float **' but argument is of type 'float **'. Shouldn't this be implicit? Excplicit c...
}
/*
A passthrough should never have modified the input and output frame counts. If you're
triggering these assers you need to fix your processing callback.
*/
MA_ASSERT(frameCountIn == totalFramesRead);
MA_ASSERT(frameCountOut == totalFramesRead);
}
} else {
/* Slow path. Need to do caching. */
ma_uint32 framesToProcessIn;
ma_uint32 framesToProcessOut;
ma_bool32 consumeNullInput = MA_FALSE;
/*
We use frameCount as a basis for the number of frames to read since that's what's being
requested, however we still need to clamp it to whatever can fit in the cache.
This will also be used as the basis for determining how many input frames to read. This is
not ideal because it can result in too many input frames being read which introduces latency.
To solve this, nodes can implement an optional callback called onGetRequiredInputFrameCount
which is used as hint to miniaudio as to how many input frames it needs to read at a time. This
callback is completely optional, and if it's not set, miniaudio will assume `frameCount`.
This function will be called multiple times for each period of time, once for each output node.
We cannot read from each input node each time this function is called. Instead we need to check
whether or not this is first output bus to be read from for this time period, and if so, read
from our input data.
To determine whether or not we're ready to read data, we check a flag. There will be one flag
for each output. When the flag is set, it means data has been read previously and that we're
ready to advance time forward for our input nodes by reading fresh data.
*/
framesToProcessOut = frameCount;
if (framesToProcessOut > pNodeBase->cachedDataCapInFramesPerBus) {
framesToProcessOut = pNodeBase->cachedDataCapInFramesPerBus;
}
framesToProcessIn = frameCount;
if (pNodeBase->vtable->onGetRequiredInputFrameCount) {
pNodeBase->vtable->onGetRequiredInputFrameCount(pNode, framesToProcessOut, &framesToProcessIn); /* <-- It does not matter if this fails. */
}
if (framesToProcessIn > pNodeBase->cachedDataCapInFramesPerBus) {
framesToProcessIn = pNodeBase->cachedDataCapInFramesPerBus;
}
MA_ASSERT(framesToProcessIn <= 0xFFFF);
MA_ASSERT(framesToProcessOut <= 0xFFFF);
if (ma_node_output_bus_has_read(&pNodeBase->pOutputBuses[outputBusIndex])) {
/* Getting here means we need to do another round of processing. */
pNodeBase->cachedFrameCountOut = 0;
for (;;) {
frameCountOut = 0;
/*
We need to prepare our output frame pointers for processing. In the same iteration we need
to mark every output bus as unread so that future calls to this function for different buses
for the current time period don't pull in data when they should instead be reading from cache.
*/
for (iOutputBus = 0; iOutputBus < outputBusCount; iOutputBus += 1) {
ma_node_output_bus_set_has_read(&pNodeBase->pOutputBuses[iOutputBus], MA_FALSE); /* <-- This is what tells the next calls to this function for other output buses for this time period to read from cache instead of pulling in mo...
ppFramesOut[iOutputBus] = ma_node_get_cached_output_ptr(pNode, iOutputBus);
}
/* We only need to read from input buses if there isn't already some data in the cache. */
if (pNodeBase->cachedFrameCountIn == 0) {
ma_uint32 maxFramesReadIn = 0;
/* Here is where we pull in data from the input buses. This is what will trigger an advance in time. */
for (iInputBus = 0; iInputBus < inputBusCount; iInputBus += 1) {
ma_uint32 framesRead;
/* The first thing to do is get the offset within our bulk allocation to store this input data. */
ppFramesIn[iInputBus] = ma_node_get_cached_input_ptr(pNode, iInputBus);
/* Once we've determined our destination pointer we can read. Note that we must inspect the number of frames read and fill any leftovers with silence for safety. */
result = ma_node_input_bus_read_pcm_frames(pNodeBase, &pNodeBase->pInputBuses[iInputBus], ppFramesIn[iInputBus], framesToProcessIn, &framesRead, globalTime);
if (result != MA_SUCCESS) {
/* It doesn't really matter if we fail because we'll just fill with silence. */
framesRead = 0; /* Just for safety, but I don't think it's really needed. */
}
/* TODO: Minor optimization opportunity here. If no frames were read and the buffer is already filled with silence, no need to re-silence it. */
/* Any leftover frames need to silenced for safety. */
if (framesRead < framesToProcessIn) {
ma_silence_pcm_frames(ppFramesIn[iInputBus] + (framesRead * ma_node_get_input_channels(pNodeBase, iInputBus)), (framesToProcessIn - framesRead), ma_format_f32, ma_node_get_input_channels(pNodeBase, iInputBus));
}
maxFramesReadIn = ma_max(maxFramesReadIn, framesRead);
}
/* This was a fresh load of input data so reset our consumption counter. */
pNodeBase->consumedFrameCountIn = 0;
/*
We don't want to keep processing if there's nothing to process, so set the number of cached
input frames to the maximum number we read from each attachment (the lesser will be padded
with silence). If we didn't read anything, this will be set to 0 and the entire buffer will
have been assigned to silence. This being equal to 0 is an important property for us because
it allows us to detect when NULL can be passed into the processing callback for the input
buffer for the purpose of continuous processing.
*/
pNodeBase->cachedFrameCountIn = (ma_uint16)maxFramesReadIn;
} else {
/* We don't need to read anything, but we do need to prepare our input frame pointers. */
for (iInputBus = 0; iInputBus < inputBusCount; iInputBus += 1) {
ppFramesIn[iInputBus] = ma_node_get_cached_input_ptr(pNode, iInputBus) + (pNodeBase->consumedFrameCountIn * ma_node_get_input_channels(pNodeBase, iInputBus));
}
}
/*
At this point we have our input data so now we need to do some processing. Sneaky little
optimization here - we can set the pointer to the output buffer for this output bus so
that the final copy into the output buffer is done directly by onProcess().
*/
if (pFramesOut != NULL) {
ppFramesOut[outputBusIndex] = ma_offset_pcm_frames_ptr_f32(pFramesOut, pNodeBase->cachedFrameCountOut, ma_node_get_output_channels(pNode, outputBusIndex));
}
/* Give the processing function the entire capacity of the output buffer. */
frameCountOut = (framesToProcessOut - pNodeBase->cachedFrameCountOut);
/*
We need to treat nodes with continuous processing a little differently. For these ones,
we always want to fire the callback with the requested number of frames, regardless of
pNodeBase->cachedFrameCountIn, which could be 0. Also, we want to check if we can pass
in NULL for the input buffer to the callback.
*/
if ((pNodeBase->vtable->flags & MA_NODE_FLAG_CONTINUOUS_PROCESSING) != 0) {
/* We're using continuous processing. Make sure we specify the whole frame count at all times. */
frameCountIn = framesToProcessIn; /* Give the processing function as much input data as we've got in the buffer, including any silenced padding from short reads. */
if ((pNodeBase->vtable->flags & MA_NODE_FLAG_ALLOW_NULL_INPUT) != 0 && pNodeBase->consumedFrameCountIn == 0 && pNodeBase->cachedFrameCountIn == 0) {
consumeNullInput = MA_TRUE;
} else {
consumeNullInput = MA_FALSE;
}
/*
Since we're using continuous processing we're always passing in a full frame count
regardless of how much input data was read. If this is greater than what we read as
input, we'll end up with an underflow. We instead need to make sure our cached frame
count is set to the number of frames we'll be passing to the data callback. Not
doing this will result in an underflow when we "consume" the cached data later on.
Note that this check needs to be done after the "consumeNullInput" check above because
we use the property of cachedFrameCountIn being 0 to determine whether or not we
should be passing in a null pointer to the processing callback for when the node is
configured with MA_NODE_FLAG_ALLOW_NULL_INPUT.
*/
if (pNodeBase->cachedFrameCountIn < (ma_uint16)frameCountIn) {
pNodeBase->cachedFrameCountIn = (ma_uint16)frameCountIn;
}
} else {
frameCountIn = pNodeBase->cachedFrameCountIn; /* Give the processing function as much valid input data as we've got. */
consumeNullInput = MA_FALSE;
}
/*
Process data slightly differently depending on whether or not we're consuming NULL
input (checked just above).
*/
if (consumeNullInput) {
ma_node_process_pcm_frames_internal(pNode, NULL, &frameCountIn, ppFramesOut, &frameCountOut);
} else {
/*
We want to skip processing if there's no input data, but we can only do that safely if
we know that there is no chance of any output frames being produced. If continuous
processing is being used, this won't be a problem because the input frame count will
always be non-0. However, if continuous processing is *not* enabled and input and output
data is processed at different rates, we still need to process that last input frame
because there could be a few excess output frames needing to be produced from cached
data. The `MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES` flag is used as the indicator for
determining whether or not we need to process the node even when there are no input
frames available right now.
*/
if (frameCountIn > 0 || (pNodeBase->vtable->flags & MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES) != 0) {
ma_node_process_pcm_frames_internal(pNode, (const float**)ppFramesIn, &frameCountIn, ppFramesOut, &frameCountOut); /* From GCC: expected 'const float **' but argument is of type 'float **'. Shouldn't this be implicit? E...
} else {
frameCountOut = 0; /* No data was processed. */
}
}
/*
Thanks to our sneaky optimization above we don't need to do any data copying directly into
the output buffer - the onProcess() callback just did that for us. We do, however, need to
apply the number of input and output frames that were processed. Note that due to continuous
processing above, we need to do explicit checks here. If we just consumed a NULL input
buffer it means that no actual input data was processed from the internal buffers and we
don't want to be modifying any counters.
*/
if (consumeNullInput == MA_FALSE) {
pNodeBase->consumedFrameCountIn += (ma_uint16)frameCountIn;
pNodeBase->cachedFrameCountIn -= (ma_uint16)frameCountIn;
}
/* The cached output frame count is always equal to what we just read. */
pNodeBase->cachedFrameCountOut += (ma_uint16)frameCountOut;
/* If we couldn't process any data, we're done. The loop needs to be terminated here or else we'll get stuck in a loop. */
if (pNodeBase->cachedFrameCountOut == framesToProcessOut || (frameCountOut == 0 && frameCountIn == 0)) {
break;
}
}
} else {
/*
We're not needing to read anything from the input buffer so just read directly from our
already-processed data.
*/
if (pFramesOut != NULL) {
ma_copy_pcm_frames(pFramesOut, ma_node_get_cached_output_ptr(pNodeBase, outputBusIndex), pNodeBase->cachedFrameCountOut, ma_format_f32, ma_node_get_output_channels(pNodeBase, outputBusIndex));
}
}
/* The number of frames read is always equal to the number of cached output frames. */
totalFramesRead = pNodeBase->cachedFrameCountOut;
/* Now that we've read the data, make sure our read flag is set. */
ma_node_output_bus_set_has_read(&pNodeBase->pOutputBuses[outputBusIndex], MA_TRUE);
}
}
/* Apply volume, if necessary. */
ma_apply_volume_factor_f32(pFramesOut, totalFramesRead * ma_node_get_output_channels(pNodeBase, outputBusIndex), ma_node_output_bus_get_volume(&pNodeBase->pOutputBuses[outputBusIndex]));
/* Advance our local time forward. */
c89atomic_fetch_add_64(&pNodeBase->localTime, (ma_uint64)totalFramesRead);
*pFramesRead = totalFramesRead + timeOffsetBeg; /* Must include the silenced section at the start of the buffer. */
return result;
}
/* Data source node. */
MA_API ma_data_source_node_config ma_data_source_node_config_init(ma_data_source* pDataSource)
{
ma_data_source_node_config config;
MA_ZERO_OBJECT(&config);
config.nodeConfig = ma_node_config_init();
config.pDataSource = pDataSource;
return config;
}
static void ma_data_source_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_data_source_node* pDataSourceNode = (ma_data_source_node*)pNode;
ma_format format;
ma_uint32 channels;
ma_uint32 frameCount;
ma_uint64 framesRead = 0;
MA_ASSERT(pDataSourceNode != NULL);
MA_ASSERT(pDataSourceNode->pDataSource != NULL);
MA_ASSERT(ma_node_get_input_bus_count(pDataSourceNode) == 0);
MA_ASSERT(ma_node_get_output_bus_count(pDataSourceNode) == 1);
/* We don't want to read from ppFramesIn at all. Instead we read from the data source. */
(void)ppFramesIn;
(void)pFrameCountIn;
frameCount = *pFrameCountOut;
/* miniaudio should never be calling this with a frame count of zero. */
MA_ASSERT(frameCount > 0);
if (ma_data_source_get_data_format(pDataSourceNode->pDataSource, &format, &channels, NULL, NULL, 0) == MA_SUCCESS) { /* <-- Don't care about sample rate here. */
/* The node graph system requires samples be in floating point format. This is checked in ma_data_source_node_init(). */
MA_ASSERT(format == ma_format_f32);
(void)format; /* Just to silence some static analysis tools. */
ma_data_source_read_pcm_frames(pDataSourceNode->pDataSource, ppFramesOut[0], frameCount, &framesRead);
}
*pFrameCountOut = (ma_uint32)framesRead;
}
static ma_node_vtable g_ma_data_source_node_vtable =
{
ma_data_source_node_process_pcm_frames,
NULL, /* onGetRequiredInputFrameCount */
0, /* 0 input buses. */
1, /* 1 output bus. */
0
};
MA_API ma_result ma_data_source_node_init(ma_node_graph* pNodeGraph, const ma_data_source_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_data_source_node* pDataSourceNode)
{
ma_result result;
ma_format format; /* For validating the format, which must be ma_format_f32. */
ma_uint32 channels; /* For specifying the channel count of the output bus. */
ma_node_config baseConfig;
if (pDataSourceNode == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pDataSourceNode);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
result = ma_data_source_get_data_format(pConfig->pDataSource, &format, &channels, NULL, NULL, 0); /* Don't care about sample rate. This will check pDataSource for NULL. */
if (result != MA_SUCCESS) {
return result;
}
MA_ASSERT(format == ma_format_f32); /* <-- If you've triggered this it means your data source is not outputting floating-point samples. You must configure your data source to use ma_format_f32. */
if (format != ma_format_f32) {
return MA_INVALID_ARGS; /* Invalid format. */
}
/* The channel count is defined by the data source. If the caller has manually changed the channels we just ignore it. */
baseConfig = pConfig->nodeConfig;
baseConfig.vtable = &g_ma_data_source_node_vtable; /* Explicitly set the vtable here to prevent callers from setting it incorrectly. */
/*
The channel count is defined by the data source. It is invalid for the caller to manually set
the channel counts in the config. `ma_data_source_node_config_init()` will have defaulted the
channel count pointer to NULL which is how it must remain. If you trigger any of these asserts
it means you're explicitly setting the channel count. Instead, configure the output channel
count of your data source to be the necessary channel count.
*/
if (baseConfig.pOutputChannels != NULL) {
return MA_INVALID_ARGS;
}
baseConfig.pOutputChannels = &channels;
result = ma_node_init(pNodeGraph, &baseConfig, pAllocationCallbacks, &pDataSourceNode->base);
if (result != MA_SUCCESS) {
return result;
}
pDataSourceNode->pDataSource = pConfig->pDataSource;
return MA_SUCCESS;
}
MA_API void ma_data_source_node_uninit(ma_data_source_node* pDataSourceNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_node_uninit(&pDataSourceNode->base, pAllocationCallbacks);
}
MA_API ma_result ma_data_source_node_set_looping(ma_data_source_node* pDataSourceNode, ma_bool32 isLooping)
{
if (pDataSourceNode == NULL) {
return MA_INVALID_ARGS;
}
return ma_data_source_set_looping(pDataSourceNode->pDataSource, isLooping);
}
MA_API ma_bool32 ma_data_source_node_is_looping(ma_data_source_node* pDataSourceNode)
{
if (pDataSourceNode == NULL) {
return MA_FALSE;
}
return ma_data_source_is_looping(pDataSourceNode->pDataSource);
}
/* Splitter Node. */
MA_API ma_splitter_node_config ma_splitter_node_config_init(ma_uint32 channels)
{
ma_splitter_node_config config;
MA_ZERO_OBJECT(&config);
config.nodeConfig = ma_node_config_init();
config.channels = channels;
return config;
}
static void ma_splitter_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_node_base* pNodeBase = (ma_node_base*)pNode;
ma_uint32 iOutputBus;
ma_uint32 channels;
MA_ASSERT(pNodeBase != NULL);
MA_ASSERT(ma_node_get_input_bus_count(pNodeBase) == 1);
MA_ASSERT(ma_node_get_output_bus_count(pNodeBase) >= 2);
/* We don't need to consider the input frame count - it'll be the same as the output frame count and we process everything. */
(void)pFrameCountIn;
/* NOTE: This assumes the same number of channels for all inputs and outputs. This was checked in ma_splitter_node_init(). */
channels = ma_node_get_input_channels(pNodeBase, 0);
/* Splitting is just copying the first input bus and copying it over to each output bus. */
for (iOutputBus = 0; iOutputBus < ma_node_get_output_bus_count(pNodeBase); iOutputBus += 1) {
ma_copy_pcm_frames(ppFramesOut[iOutputBus], ppFramesIn[0], *pFrameCountOut, ma_format_f32, channels);
}
}
static ma_node_vtable g_ma_splitter_node_vtable =
{
ma_splitter_node_process_pcm_frames,
NULL, /* onGetRequiredInputFrameCount */
1, /* 1 input bus. */
2, /* 2 output buses. */
0
};
MA_API ma_result ma_splitter_node_init(ma_node_graph* pNodeGraph, const ma_splitter_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_splitter_node* pSplitterNode)
{
ma_result result;
ma_node_config baseConfig;
ma_uint32 pInputChannels[1];
ma_uint32 pOutputChannels[2];
if (pSplitterNode == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pSplitterNode);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
/* Splitters require the same number of channels between inputs and outputs. */
pInputChannels[0] = pConfig->channels;
pOutputChannels[0] = pConfig->channels;
pOutputChannels[1] = pConfig->channels;
baseConfig = pConfig->nodeConfig;
baseConfig.vtable = &g_ma_splitter_node_vtable;
baseConfig.pInputChannels = pInputChannels;
baseConfig.pOutputChannels = pOutputChannels;
result = ma_node_init(pNodeGraph, &baseConfig, pAllocationCallbacks, &pSplitterNode->base);
if (result != MA_SUCCESS) {
return result; /* Failed to initialize the base node. */
}
return MA_SUCCESS;
}
MA_API void ma_splitter_node_uninit(ma_splitter_node* pSplitterNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_node_uninit(pSplitterNode, pAllocationCallbacks);
}
/*
Biquad Node
*/
MA_API ma_biquad_node_config ma_biquad_node_config_init(ma_uint32 channels, float b0, float b1, float b2, float a0, float a1, float a2)
{
ma_biquad_node_config config;
config.nodeConfig = ma_node_config_init();
config.biquad = ma_biquad_config_init(ma_format_f32, channels, b0, b1, b2, a0, a1, a2);
return config;
}
static void ma_biquad_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_biquad_node* pLPFNode = (ma_biquad_node*)pNode;
MA_ASSERT(pNode != NULL);
(void)pFrameCountIn;
ma_biquad_process_pcm_frames(&pLPFNode->biquad, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
}
static ma_node_vtable g_ma_biquad_node_vtable =
{
ma_biquad_node_process_pcm_frames,
NULL, /* onGetRequiredInputFrameCount */
1, /* One input. */
1, /* One output. */
0 /* Default flags. */
};
MA_API ma_result ma_biquad_node_init(ma_node_graph* pNodeGraph, const ma_biquad_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_biquad_node* pNode)
{
ma_result result;
ma_node_config baseNodeConfig;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pNode);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->biquad.format != ma_format_f32) {
return MA_INVALID_ARGS; /* The format must be f32. */
}
result = ma_biquad_init(&pConfig->biquad, pAllocationCallbacks, &pNode->biquad);
if (result != MA_SUCCESS) {
return result;
}
baseNodeConfig = ma_node_config_init();
baseNodeConfig.vtable = &g_ma_biquad_node_vtable;
baseNodeConfig.pInputChannels = &pConfig->biquad.channels;
baseNodeConfig.pOutputChannels = &pConfig->biquad.channels;
result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
if (result != MA_SUCCESS) {
return result;
}
return result;
}
MA_API ma_result ma_biquad_node_reinit(const ma_biquad_config* pConfig, ma_biquad_node* pNode)
{
ma_biquad_node* pLPFNode = (ma_biquad_node*)pNode;
MA_ASSERT(pNode != NULL);
return ma_biquad_reinit(pConfig, &pLPFNode->biquad);
}
MA_API void ma_biquad_node_uninit(ma_biquad_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_biquad_node* pLPFNode = (ma_biquad_node*)pNode;
if (pNode == NULL) {
return;
}
ma_node_uninit(pNode, pAllocationCallbacks);
ma_biquad_uninit(&pLPFNode->biquad, pAllocationCallbacks);
}
/*
Low Pass Filter Node
*/
MA_API ma_lpf_node_config ma_lpf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
{
ma_lpf_node_config config;
config.nodeConfig = ma_node_config_init();
config.lpf = ma_lpf_config_init(ma_format_f32, channels, sampleRate, cutoffFrequency, order);
return config;
}
static void ma_lpf_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_lpf_node* pLPFNode = (ma_lpf_node*)pNode;
MA_ASSERT(pNode != NULL);
(void)pFrameCountIn;
ma_lpf_process_pcm_frames(&pLPFNode->lpf, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
}
static ma_node_vtable g_ma_lpf_node_vtable =
{
ma_lpf_node_process_pcm_frames,
NULL, /* onGetRequiredInputFrameCount */
1, /* One input. */
1, /* One output. */
0 /* Default flags. */
};
MA_API ma_result ma_lpf_node_init(ma_node_graph* pNodeGraph, const ma_lpf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_lpf_node* pNode)
{
ma_result result;
ma_node_config baseNodeConfig;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pNode);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->lpf.format != ma_format_f32) {
return MA_INVALID_ARGS; /* The format must be f32. */
}
result = ma_lpf_init(&pConfig->lpf, pAllocationCallbacks, &pNode->lpf);
if (result != MA_SUCCESS) {
return result;
}
baseNodeConfig = ma_node_config_init();
baseNodeConfig.vtable = &g_ma_lpf_node_vtable;
baseNodeConfig.pInputChannels = &pConfig->lpf.channels;
baseNodeConfig.pOutputChannels = &pConfig->lpf.channels;
result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
if (result != MA_SUCCESS) {
return result;
}
return result;
}
MA_API ma_result ma_lpf_node_reinit(const ma_lpf_config* pConfig, ma_lpf_node* pNode)
{
ma_lpf_node* pLPFNode = (ma_lpf_node*)pNode;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
return ma_lpf_reinit(pConfig, &pLPFNode->lpf);
}
MA_API void ma_lpf_node_uninit(ma_lpf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_lpf_node* pLPFNode = (ma_lpf_node*)pNode;
if (pNode == NULL) {
return;
}
ma_node_uninit(pNode, pAllocationCallbacks);
ma_lpf_uninit(&pLPFNode->lpf, pAllocationCallbacks);
}
/*
High Pass Filter Node
*/
MA_API ma_hpf_node_config ma_hpf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
{
ma_hpf_node_config config;
config.nodeConfig = ma_node_config_init();
config.hpf = ma_hpf_config_init(ma_format_f32, channels, sampleRate, cutoffFrequency, order);
return config;
}
static void ma_hpf_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_hpf_node* pHPFNode = (ma_hpf_node*)pNode;
MA_ASSERT(pNode != NULL);
(void)pFrameCountIn;
ma_hpf_process_pcm_frames(&pHPFNode->hpf, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
}
static ma_node_vtable g_ma_hpf_node_vtable =
{
ma_hpf_node_process_pcm_frames,
NULL, /* onGetRequiredInputFrameCount */
1, /* One input. */
1, /* One output. */
0 /* Default flags. */
};
MA_API ma_result ma_hpf_node_init(ma_node_graph* pNodeGraph, const ma_hpf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hpf_node* pNode)
{
ma_result result;
ma_node_config baseNodeConfig;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pNode);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->hpf.format != ma_format_f32) {
return MA_INVALID_ARGS; /* The format must be f32. */
}
result = ma_hpf_init(&pConfig->hpf, pAllocationCallbacks, &pNode->hpf);
if (result != MA_SUCCESS) {
return result;
}
baseNodeConfig = ma_node_config_init();
baseNodeConfig.vtable = &g_ma_hpf_node_vtable;
baseNodeConfig.pInputChannels = &pConfig->hpf.channels;
baseNodeConfig.pOutputChannels = &pConfig->hpf.channels;
result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
if (result != MA_SUCCESS) {
return result;
}
return result;
}
MA_API ma_result ma_hpf_node_reinit(const ma_hpf_config* pConfig, ma_hpf_node* pNode)
{
ma_hpf_node* pHPFNode = (ma_hpf_node*)pNode;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
return ma_hpf_reinit(pConfig, &pHPFNode->hpf);
}
MA_API void ma_hpf_node_uninit(ma_hpf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_hpf_node* pHPFNode = (ma_hpf_node*)pNode;
if (pNode == NULL) {
return;
}
ma_node_uninit(pNode, pAllocationCallbacks);
ma_hpf_uninit(&pHPFNode->hpf, pAllocationCallbacks);
}
/*
Band Pass Filter Node
*/
MA_API ma_bpf_node_config ma_bpf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double cutoffFrequency, ma_uint32 order)
{
ma_bpf_node_config config;
config.nodeConfig = ma_node_config_init();
config.bpf = ma_bpf_config_init(ma_format_f32, channels, sampleRate, cutoffFrequency, order);
return config;
}
static void ma_bpf_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_bpf_node* pBPFNode = (ma_bpf_node*)pNode;
MA_ASSERT(pNode != NULL);
(void)pFrameCountIn;
ma_bpf_process_pcm_frames(&pBPFNode->bpf, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
}
static ma_node_vtable g_ma_bpf_node_vtable =
{
ma_bpf_node_process_pcm_frames,
NULL, /* onGetRequiredInputFrameCount */
1, /* One input. */
1, /* One output. */
0 /* Default flags. */
};
MA_API ma_result ma_bpf_node_init(ma_node_graph* pNodeGraph, const ma_bpf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_bpf_node* pNode)
{
ma_result result;
ma_node_config baseNodeConfig;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pNode);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->bpf.format != ma_format_f32) {
return MA_INVALID_ARGS; /* The format must be f32. */
}
result = ma_bpf_init(&pConfig->bpf, pAllocationCallbacks, &pNode->bpf);
if (result != MA_SUCCESS) {
return result;
}
baseNodeConfig = ma_node_config_init();
baseNodeConfig.vtable = &g_ma_bpf_node_vtable;
baseNodeConfig.pInputChannels = &pConfig->bpf.channels;
baseNodeConfig.pOutputChannels = &pConfig->bpf.channels;
result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
if (result != MA_SUCCESS) {
return result;
}
return result;
}
MA_API ma_result ma_bpf_node_reinit(const ma_bpf_config* pConfig, ma_bpf_node* pNode)
{
ma_bpf_node* pBPFNode = (ma_bpf_node*)pNode;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
return ma_bpf_reinit(pConfig, &pBPFNode->bpf);
}
MA_API void ma_bpf_node_uninit(ma_bpf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_bpf_node* pBPFNode = (ma_bpf_node*)pNode;
if (pNode == NULL) {
return;
}
ma_node_uninit(pNode, pAllocationCallbacks);
ma_bpf_uninit(&pBPFNode->bpf, pAllocationCallbacks);
}
/*
Notching Filter Node
*/
MA_API ma_notch_node_config ma_notch_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double q, double frequency)
{
ma_notch_node_config config;
config.nodeConfig = ma_node_config_init();
config.notch = ma_notch2_config_init(ma_format_f32, channels, sampleRate, q, frequency);
return config;
}
static void ma_notch_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_notch_node* pBPFNode = (ma_notch_node*)pNode;
MA_ASSERT(pNode != NULL);
(void)pFrameCountIn;
ma_notch2_process_pcm_frames(&pBPFNode->notch, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
}
static ma_node_vtable g_ma_notch_node_vtable =
{
ma_notch_node_process_pcm_frames,
NULL, /* onGetRequiredInputFrameCount */
1, /* One input. */
1, /* One output. */
0 /* Default flags. */
};
MA_API ma_result ma_notch_node_init(ma_node_graph* pNodeGraph, const ma_notch_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_notch_node* pNode)
{
ma_result result;
ma_node_config baseNodeConfig;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pNode);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->notch.format != ma_format_f32) {
return MA_INVALID_ARGS; /* The format must be f32. */
}
result = ma_notch2_init(&pConfig->notch, pAllocationCallbacks, &pNode->notch);
if (result != MA_SUCCESS) {
return result;
}
baseNodeConfig = ma_node_config_init();
baseNodeConfig.vtable = &g_ma_notch_node_vtable;
baseNodeConfig.pInputChannels = &pConfig->notch.channels;
baseNodeConfig.pOutputChannels = &pConfig->notch.channels;
result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
if (result != MA_SUCCESS) {
return result;
}
return result;
}
MA_API ma_result ma_notch_node_reinit(const ma_notch_config* pConfig, ma_notch_node* pNode)
{
ma_notch_node* pNotchNode = (ma_notch_node*)pNode;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
return ma_notch2_reinit(pConfig, &pNotchNode->notch);
}
MA_API void ma_notch_node_uninit(ma_notch_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_notch_node* pNotchNode = (ma_notch_node*)pNode;
if (pNode == NULL) {
return;
}
ma_node_uninit(pNode, pAllocationCallbacks);
ma_notch2_uninit(&pNotchNode->notch, pAllocationCallbacks);
}
/*
Peaking Filter Node
*/
MA_API ma_peak_node_config ma_peak_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency)
{
ma_peak_node_config config;
config.nodeConfig = ma_node_config_init();
config.peak = ma_peak2_config_init(ma_format_f32, channels, sampleRate, gainDB, q, frequency);
return config;
}
static void ma_peak_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_peak_node* pBPFNode = (ma_peak_node*)pNode;
MA_ASSERT(pNode != NULL);
(void)pFrameCountIn;
ma_peak2_process_pcm_frames(&pBPFNode->peak, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
}
static ma_node_vtable g_ma_peak_node_vtable =
{
ma_peak_node_process_pcm_frames,
NULL, /* onGetRequiredInputFrameCount */
1, /* One input. */
1, /* One output. */
0 /* Default flags. */
};
MA_API ma_result ma_peak_node_init(ma_node_graph* pNodeGraph, const ma_peak_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_peak_node* pNode)
{
ma_result result;
ma_node_config baseNodeConfig;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pNode);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->peak.format != ma_format_f32) {
return MA_INVALID_ARGS; /* The format must be f32. */
}
result = ma_peak2_init(&pConfig->peak, pAllocationCallbacks, &pNode->peak);
if (result != MA_SUCCESS) {
ma_node_uninit(pNode, pAllocationCallbacks);
return result;
}
baseNodeConfig = ma_node_config_init();
baseNodeConfig.vtable = &g_ma_peak_node_vtable;
baseNodeConfig.pInputChannels = &pConfig->peak.channels;
baseNodeConfig.pOutputChannels = &pConfig->peak.channels;
result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
if (result != MA_SUCCESS) {
return result;
}
return result;
}
MA_API ma_result ma_peak_node_reinit(const ma_peak_config* pConfig, ma_peak_node* pNode)
{
ma_peak_node* pPeakNode = (ma_peak_node*)pNode;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
return ma_peak2_reinit(pConfig, &pPeakNode->peak);
}
MA_API void ma_peak_node_uninit(ma_peak_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_peak_node* pPeakNode = (ma_peak_node*)pNode;
if (pNode == NULL) {
return;
}
ma_node_uninit(pNode, pAllocationCallbacks);
ma_peak2_uninit(&pPeakNode->peak, pAllocationCallbacks);
}
/*
Low Shelf Filter Node
*/
MA_API ma_loshelf_node_config ma_loshelf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency)
{
ma_loshelf_node_config config;
config.nodeConfig = ma_node_config_init();
config.loshelf = ma_loshelf2_config_init(ma_format_f32, channels, sampleRate, gainDB, q, frequency);
return config;
}
static void ma_loshelf_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_loshelf_node* pBPFNode = (ma_loshelf_node*)pNode;
MA_ASSERT(pNode != NULL);
(void)pFrameCountIn;
ma_loshelf2_process_pcm_frames(&pBPFNode->loshelf, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
}
static ma_node_vtable g_ma_loshelf_node_vtable =
{
ma_loshelf_node_process_pcm_frames,
NULL, /* onGetRequiredInputFrameCount */
1, /* One input. */
1, /* One output. */
0 /* Default flags. */
};
MA_API ma_result ma_loshelf_node_init(ma_node_graph* pNodeGraph, const ma_loshelf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_loshelf_node* pNode)
{
ma_result result;
ma_node_config baseNodeConfig;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pNode);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->loshelf.format != ma_format_f32) {
return MA_INVALID_ARGS; /* The format must be f32. */
}
result = ma_loshelf2_init(&pConfig->loshelf, pAllocationCallbacks, &pNode->loshelf);
if (result != MA_SUCCESS) {
return result;
}
baseNodeConfig = ma_node_config_init();
baseNodeConfig.vtable = &g_ma_loshelf_node_vtable;
baseNodeConfig.pInputChannels = &pConfig->loshelf.channels;
baseNodeConfig.pOutputChannels = &pConfig->loshelf.channels;
result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
if (result != MA_SUCCESS) {
return result;
}
return result;
}
MA_API ma_result ma_loshelf_node_reinit(const ma_loshelf_config* pConfig, ma_loshelf_node* pNode)
{
ma_loshelf_node* pLoshelfNode = (ma_loshelf_node*)pNode;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
return ma_loshelf2_reinit(pConfig, &pLoshelfNode->loshelf);
}
MA_API void ma_loshelf_node_uninit(ma_loshelf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_loshelf_node* pLoshelfNode = (ma_loshelf_node*)pNode;
if (pNode == NULL) {
return;
}
ma_node_uninit(pNode, pAllocationCallbacks);
ma_loshelf2_uninit(&pLoshelfNode->loshelf, pAllocationCallbacks);
}
/*
High Shelf Filter Node
*/
MA_API ma_hishelf_node_config ma_hishelf_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, double gainDB, double q, double frequency)
{
ma_hishelf_node_config config;
config.nodeConfig = ma_node_config_init();
config.hishelf = ma_hishelf2_config_init(ma_format_f32, channels, sampleRate, gainDB, q, frequency);
return config;
}
static void ma_hishelf_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_hishelf_node* pBPFNode = (ma_hishelf_node*)pNode;
MA_ASSERT(pNode != NULL);
(void)pFrameCountIn;
ma_hishelf2_process_pcm_frames(&pBPFNode->hishelf, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
}
static ma_node_vtable g_ma_hishelf_node_vtable =
{
ma_hishelf_node_process_pcm_frames,
NULL, /* onGetRequiredInputFrameCount */
1, /* One input. */
1, /* One output. */
0 /* Default flags. */
};
MA_API ma_result ma_hishelf_node_init(ma_node_graph* pNodeGraph, const ma_hishelf_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_hishelf_node* pNode)
{
ma_result result;
ma_node_config baseNodeConfig;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pNode);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->hishelf.format != ma_format_f32) {
return MA_INVALID_ARGS; /* The format must be f32. */
}
result = ma_hishelf2_init(&pConfig->hishelf, pAllocationCallbacks, &pNode->hishelf);
if (result != MA_SUCCESS) {
return result;
}
baseNodeConfig = ma_node_config_init();
baseNodeConfig.vtable = &g_ma_hishelf_node_vtable;
baseNodeConfig.pInputChannels = &pConfig->hishelf.channels;
baseNodeConfig.pOutputChannels = &pConfig->hishelf.channels;
result = ma_node_init(pNodeGraph, &baseNodeConfig, pAllocationCallbacks, pNode);
if (result != MA_SUCCESS) {
return result;
}
return result;
}
MA_API ma_result ma_hishelf_node_reinit(const ma_hishelf_config* pConfig, ma_hishelf_node* pNode)
{
ma_hishelf_node* pHishelfNode = (ma_hishelf_node*)pNode;
if (pNode == NULL) {
return MA_INVALID_ARGS;
}
return ma_hishelf2_reinit(pConfig, &pHishelfNode->hishelf);
}
MA_API void ma_hishelf_node_uninit(ma_hishelf_node* pNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
ma_hishelf_node* pHishelfNode = (ma_hishelf_node*)pNode;
if (pNode == NULL) {
return;
}
ma_node_uninit(pNode, pAllocationCallbacks);
ma_hishelf2_uninit(&pHishelfNode->hishelf, pAllocationCallbacks);
}
MA_API ma_delay_node_config ma_delay_node_config_init(ma_uint32 channels, ma_uint32 sampleRate, ma_uint32 delayInFrames, float decay)
{
ma_delay_node_config config;
config.nodeConfig = ma_node_config_init();
config.delay = ma_delay_config_init(channels, sampleRate, delayInFrames, decay);
return config;
}
static void ma_delay_node_process_pcm_frames(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_delay_node* pDelayNode = (ma_delay_node*)pNode;
(void)pFrameCountIn;
ma_delay_process_pcm_frames(&pDelayNode->delay, ppFramesOut[0], ppFramesIn[0], *pFrameCountOut);
}
static ma_node_vtable g_ma_delay_node_vtable =
{
ma_delay_node_process_pcm_frames,
NULL,
1, /* 1 input channels. */
1, /* 1 output channel. */
MA_NODE_FLAG_CONTINUOUS_PROCESSING /* Delay requires continuous processing to ensure the tail get's processed. */
};
MA_API ma_result ma_delay_node_init(ma_node_graph* pNodeGraph, const ma_delay_node_config* pConfig, const ma_allocation_callbacks* pAllocationCallbacks, ma_delay_node* pDelayNode)
{
ma_result result;
ma_node_config baseConfig;
if (pDelayNode == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pDelayNode);
result = ma_delay_init(&pConfig->delay, pAllocationCallbacks, &pDelayNode->delay);
if (result != MA_SUCCESS) {
return result;
}
baseConfig = pConfig->nodeConfig;
baseConfig.vtable = &g_ma_delay_node_vtable;
baseConfig.pInputChannels = &pConfig->delay.channels;
baseConfig.pOutputChannels = &pConfig->delay.channels;
result = ma_node_init(pNodeGraph, &baseConfig, pAllocationCallbacks, &pDelayNode->baseNode);
if (result != MA_SUCCESS) {
ma_delay_uninit(&pDelayNode->delay, pAllocationCallbacks);
return result;
}
return result;
}
MA_API void ma_delay_node_uninit(ma_delay_node* pDelayNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
if (pDelayNode == NULL) {
return;
}
/* The base node is always uninitialized first. */
ma_node_uninit(pDelayNode, pAllocationCallbacks);
ma_delay_uninit(&pDelayNode->delay, pAllocationCallbacks);
}
MA_API void ma_delay_node_set_wet(ma_delay_node* pDelayNode, float value)
{
if (pDelayNode == NULL) {
return;
}
ma_delay_set_wet(&pDelayNode->delay, value);
}
MA_API float ma_delay_node_get_wet(const ma_delay_node* pDelayNode)
{
if (pDelayNode == NULL) {
return 0;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
Engine
**************************************************************************************************************************************************************/
#define MA_SEEK_TARGET_NONE (~(ma_uint64)0)
MA_API ma_engine_node_config ma_engine_node_config_init(ma_engine* pEngine, ma_engine_node_type type, ma_uint32 flags)
{
ma_engine_node_config config;
MA_ZERO_OBJECT(&config);
config.pEngine = pEngine;
config.type = type;
config.isPitchDisabled = (flags & MA_SOUND_FLAG_NO_PITCH) != 0;
config.isSpatializationDisabled = (flags & MA_SOUND_FLAG_NO_SPATIALIZATION) != 0;
return config;
}
static void ma_engine_node_update_pitch_if_required(ma_engine_node* pEngineNode)
{
ma_bool32 isUpdateRequired = MA_FALSE;
float newPitch;
MA_ASSERT(pEngineNode != NULL);
newPitch = c89atomic_load_explicit_f32(&pEngineNode->pitch, c89atomic_memory_order_acquire);
if (pEngineNode->oldPitch != newPitch) {
pEngineNode->oldPitch = newPitch;
isUpdateRequired = MA_TRUE;
}
if (pEngineNode->oldDopplerPitch != pEngineNode->spatializer.dopplerPitch) {
pEngineNode->oldDopplerPitch = pEngineNode->spatializer.dopplerPitch;
isUpdateRequired = MA_TRUE;
}
if (isUpdateRequired) {
float basePitch = (float)pEngineNode->sampleRate / ma_engine_get_sample_rate(pEngineNode->pEngine);
ma_linear_resampler_set_rate_ratio(&pEngineNode->resampler, basePitch * pEngineNode->oldPitch * pEngineNode->oldDopplerPitch);
}
}
static ma_bool32 ma_engine_node_is_pitching_enabled(const ma_engine_node* pEngineNode)
{
MA_ASSERT(pEngineNode != NULL);
/* Don't try to be clever by skiping resampling in the pitch=1 case or else you'll glitch when moving away from 1. */
return !c89atomic_load_explicit_32(&pEngineNode->isPitchDisabled, c89atomic_memory_order_acquire);
}
static ma_bool32 ma_engine_node_is_spatialization_enabled(const ma_engine_node* pEngineNode)
{
MA_ASSERT(pEngineNode != NULL);
return !c89atomic_load_explicit_32(&pEngineNode->isSpatializationDisabled, c89atomic_memory_order_acquire);
}
static ma_uint64 ma_engine_node_get_required_input_frame_count(const ma_engine_node* pEngineNode, ma_uint64 outputFrameCount)
{
ma_uint64 inputFrameCount = 0;
if (ma_engine_node_is_pitching_enabled(pEngineNode)) {
ma_result result = ma_linear_resampler_get_required_input_frame_count(&pEngineNode->resampler, outputFrameCount, &inputFrameCount);
if (result != MA_SUCCESS) {
inputFrameCount = 0;
}
} else {
inputFrameCount = outputFrameCount; /* No resampling, so 1:1. */
}
return inputFrameCount;
}
static void ma_engine_node_process_pcm_frames__general(ma_engine_node* pEngineNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
ma_uint32 frameCountIn;
ma_uint32 frameCountOut;
ma_uint32 totalFramesProcessedIn;
ma_uint32 totalFramesProcessedOut;
ma_uint32 channelsIn;
ma_uint32 channelsOut;
ma_bool32 isPitchingEnabled;
ma_bool32 isFadingEnabled;
ma_bool32 isSpatializationEnabled;
ma_bool32 isPanningEnabled;
frameCountIn = *pFrameCountIn;
frameCountOut = *pFrameCountOut;
channelsIn = ma_spatializer_get_input_channels(&pEngineNode->spatializer);
channelsOut = ma_spatializer_get_output_channels(&pEngineNode->spatializer);
totalFramesProcessedIn = 0;
totalFramesProcessedOut = 0;
isPitchingEnabled = ma_engine_node_is_pitching_enabled(pEngineNode);
isFadingEnabled = pEngineNode->fader.volumeBeg != 1 || pEngineNode->fader.volumeEnd != 1;
isSpatializationEnabled = ma_engine_node_is_spatialization_enabled(pEngineNode);
isPanningEnabled = pEngineNode->panner.pan != 0 && channelsOut != 1;
/* Keep going while we've still got data available for processing. */
while (totalFramesProcessedOut < frameCountOut) {
/*
We need to process in a specific order. We always do resampling first because it's likely
we're going to be increasing the channel count after spatialization. Also, I want to do
fading based on the output sample rate.
We'll first read into a buffer from the resampler. Then we'll do all processing that
operates on the on the input channel count. We'll then get the spatializer to output to
the output buffer and then do all effects from that point directly in the output buffer
in-place.
Note that we're always running the resampler. If we try to be clever and skip resampling
when the pitch is 1, we'll get a glitch when we move away from 1, back to 1, and then
away from 1 again. We'll want to implement any pitch=1 optimizations in the resampler
itself.
There's a small optimization here that we'll utilize since it might be a fairly common
case. When the input and output channel counts are the same, we'll read straight into the
output buffer from the resampler and do everything in-place.
*/
const float* pRunningFramesIn;
float* pRunningFramesOut;
float* pWorkingBuffer; /* This is the buffer that we'll be processing frames in. This is in input channels. */
float temp[MA_DATA_CONVERTER_STACK_BUFFER_SIZE / sizeof(float)];
ma_uint32 tempCapInFrames = ma_countof(temp) / channelsIn;
ma_uint32 framesAvailableIn;
ma_uint32 framesAvailableOut;
ma_uint32 framesJustProcessedIn;
ma_uint32 framesJustProcessedOut;
ma_bool32 isWorkingBufferValid = MA_FALSE;
framesAvailableIn = frameCountIn - totalFramesProcessedIn;
framesAvailableOut = frameCountOut - totalFramesProcessedOut;
pRunningFramesIn = ma_offset_pcm_frames_const_ptr_f32(ppFramesIn[0], totalFramesProcessedIn, channelsIn);
pRunningFramesOut = ma_offset_pcm_frames_ptr_f32(ppFramesOut[0], totalFramesProcessedOut, channelsOut);
if (channelsIn == channelsOut) {
/* Fast path. Channel counts are the same. No need for an intermediary input buffer. */
pWorkingBuffer = pRunningFramesOut;
} else {
/* Slow path. Channel counts are different. Need to use an intermediary input buffer. */
pWorkingBuffer = temp;
if (framesAvailableOut > tempCapInFrames) {
framesAvailableOut = tempCapInFrames;
}
}
/* First is resampler. */
if (isPitchingEnabled) {
ma_uint64 resampleFrameCountIn = framesAvailableIn;
ma_uint64 resampleFrameCountOut = framesAvailableOut;
ma_linear_resampler_process_pcm_frames(&pEngineNode->resampler, pRunningFramesIn, &resampleFrameCountIn, pWorkingBuffer, &resampleFrameCountOut);
isWorkingBufferValid = MA_TRUE;
framesJustProcessedIn = (ma_uint32)resampleFrameCountIn;
framesJustProcessedOut = (ma_uint32)resampleFrameCountOut;
} else {
framesJustProcessedIn = ma_min(framesAvailableIn, framesAvailableOut);
framesJustProcessedOut = framesJustProcessedIn; /* When no resampling is being performed, the number of output frames is the same as input frames. */
}
/* Fading. */
if (isFadingEnabled) {
if (isWorkingBufferValid) {
ma_fader_process_pcm_frames(&pEngineNode->fader, pWorkingBuffer, pWorkingBuffer, framesJustProcessedOut); /* In-place processing. */
} else {
ma_fader_process_pcm_frames(&pEngineNode->fader, pWorkingBuffer, pRunningFramesIn, framesJustProcessedOut);
isWorkingBufferValid = MA_TRUE;
}
}
/*
If at this point we still haven't actually done anything with the working buffer we need
to just read straight from the input buffer.
*/
if (isWorkingBufferValid == MA_FALSE) {
pWorkingBuffer = (float*)pRunningFramesIn; /* Naughty const cast, but it's safe at this point because we won't ever be writing to it from this point out. */
}
/* Spatialization. */
if (isSpatializationEnabled) {
ma_uint32 iListener;
/*
When determining the listener to use, we first check to see if the sound is pinned to a
specific listener. If so, we use that. Otherwise we just use the closest listener.
*/
if (pEngineNode->pinnedListenerIndex != MA_LISTENER_INDEX_CLOSEST && pEngineNode->pinnedListenerIndex < ma_engine_get_listener_count(pEngineNode->pEngine)) {
iListener = pEngineNode->pinnedListenerIndex;
} else {
iListener = ma_engine_find_closest_listener(pEngineNode->pEngine, pEngineNode->spatializer.position.x, pEngineNode->spatializer.position.y, pEngineNode->spatializer.position.z);
}
ma_spatializer_process_pcm_frames(&pEngineNode->spatializer, &pEngineNode->pEngine->listeners[iListener], pRunningFramesOut, pWorkingBuffer, framesJustProcessedOut);
} else {
/* No spatialization, but we still need to do channel conversion. */
if (channelsIn == channelsOut) {
/* No channel conversion required. Just copy straight to the output buffer. */
ma_copy_pcm_frames(pRunningFramesOut, pWorkingBuffer, framesJustProcessedOut, ma_format_f32, channelsOut);
} else {
/* Channel conversion required. TODO: Add support for channel maps here. */
ma_channel_map_apply_f32(pRunningFramesOut, NULL, channelsOut, pWorkingBuffer, NULL, channelsIn, framesJustProcessedOut, ma_channel_mix_mode_simple, pEngineNode->pEngine->monoExpansionMode);
}
}
/* At this point we can guarantee that the output buffer contains valid data. We can process everything in place now. */
/* Panning. */
if (isPanningEnabled) {
ma_panner_process_pcm_frames(&pEngineNode->panner, pRunningFramesOut, pRunningFramesOut, framesJustProcessedOut); /* In-place processing. */
}
/* We're done for this chunk. */
totalFramesProcessedIn += framesJustProcessedIn;
totalFramesProcessedOut += framesJustProcessedOut;
/* If we didn't process any output frames this iteration it means we've either run out of input data, or run out of room in the output buffer. */
if (framesJustProcessedOut == 0) {
break;
}
}
/* At this point we're done processing. */
*pFrameCountIn = totalFramesProcessedIn;
*pFrameCountOut = totalFramesProcessedOut;
}
static void ma_engine_node_process_pcm_frames__sound(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
/* For sounds, we need to first read from the data source. Then we need to apply the engine effects (pan, pitch, fades, etc.). */
ma_result result = MA_SUCCESS;
ma_sound* pSound = (ma_sound*)pNode;
ma_uint32 frameCount = *pFrameCountOut;
ma_uint32 totalFramesRead = 0;
ma_format dataSourceFormat;
ma_uint32 dataSourceChannels;
ma_uint8 temp[MA_DATA_CONVERTER_STACK_BUFFER_SIZE];
ma_uint32 tempCapInFrames;
ma_uint64 seekTarget;
/* This is a data source node which means no input buses. */
(void)ppFramesIn;
(void)pFrameCountIn;
/* If we're marked at the end we need to stop the sound and do nothing. */
if (ma_sound_at_end(pSound)) {
ma_sound_stop(pSound);
*pFrameCountOut = 0;
return;
}
/* If we're seeking, do so now before reading. */
seekTarget = c89atomic_load_64(&pSound->seekTarget);
if (seekTarget != MA_SEEK_TARGET_NONE) {
ma_data_source_seek_to_pcm_frame(pSound->pDataSource, seekTarget);
/* Any time-dependant effects need to have their times updated. */
ma_node_set_time(pSound, seekTarget);
c89atomic_exchange_64(&pSound->seekTarget, MA_SEEK_TARGET_NONE);
}
/*
We want to update the pitch once. For sounds, this can be either at the start or at the end. If
we don't force this to only ever be updating once, we could end up in a situation where
retrieving the required input frame count ends up being different to what we actually retrieve.
What could happen is that the required input frame count is calculated, the pitch is update,
and then this processing function is called resulting in a different number of input frames
being processed. Do not call this in ma_engine_node_process_pcm_frames__general() or else
you'll hit the aforementioned bug.
*/
ma_engine_node_update_pitch_if_required(&pSound->engineNode);
/*
For the convenience of the caller, we're doing to allow data sources to use non-floating-point formats and channel counts that differ
from the main engine.
*/
result = ma_data_source_get_data_format(pSound->pDataSource, &dataSourceFormat, &dataSourceChannels, NULL, NULL, 0);
if (result == MA_SUCCESS) {
tempCapInFrames = sizeof(temp) / ma_get_bytes_per_frame(dataSourceFormat, dataSourceChannels);
/* Keep reading until we've read as much as was requested or we reach the end of the data source. */
while (totalFramesRead < frameCount) {
ma_uint32 framesRemaining = frameCount - totalFramesRead;
ma_uint32 framesToRead;
ma_uint64 framesJustRead;
ma_uint32 frameCountIn;
ma_uint32 frameCountOut;
const float* pRunningFramesIn;
float* pRunningFramesOut;
/*
The first thing we need to do is read into the temporary buffer. We can calculate exactly
how many input frames we'll need after resampling.
*/
framesToRead = (ma_uint32)ma_engine_node_get_required_input_frame_count(&pSound->engineNode, framesRemaining);
if (framesToRead > tempCapInFrames) {
framesToRead = tempCapInFrames;
}
result = ma_data_source_read_pcm_frames(pSound->pDataSource, temp, framesToRead, &framesJustRead);
/* If we reached the end of the sound we'll want to mark it as at the end and stop it. This should never be returned for looping sounds. */
if (result == MA_AT_END) {
c89atomic_exchange_32(&pSound->atEnd, MA_TRUE); /* This will be set to false in ma_sound_start(). */
}
pRunningFramesOut = ma_offset_pcm_frames_ptr_f32(ppFramesOut[0], totalFramesRead, ma_engine_get_channels(ma_sound_get_engine(pSound)));
frameCountIn = (ma_uint32)framesJustRead;
frameCountOut = framesRemaining;
/* Convert if necessary. */
if (dataSourceFormat == ma_format_f32) {
/* Fast path. No data conversion necessary. */
pRunningFramesIn = (float*)temp;
ma_engine_node_process_pcm_frames__general(&pSound->engineNode, &pRunningFramesIn, &frameCountIn, &pRunningFramesOut, &frameCountOut);
} else {
/* Slow path. Need to do sample format conversion to f32. If we give the f32 buffer the same count as the first temp buffer, we're guaranteed it'll be large enough. */
float tempf32[MA_DATA_CONVERTER_STACK_BUFFER_SIZE]; /* Do not do `MA_DATA_CONVERTER_STACK_BUFFER_SIZE/sizeof(float)` here like we've done in other places. */
ma_convert_pcm_frames_format(tempf32, ma_format_f32, temp, dataSourceFormat, framesJustRead, dataSourceChannels, ma_dither_mode_none);
/* Now that we have our samples in f32 format we can process like normal. */
pRunningFramesIn = tempf32;
ma_engine_node_process_pcm_frames__general(&pSound->engineNode, &pRunningFramesIn, &frameCountIn, &pRunningFramesOut, &frameCountOut);
}
/* We should have processed all of our input frames since we calculated the required number of input frames at the top. */
MA_ASSERT(frameCountIn == framesJustRead);
totalFramesRead += (ma_uint32)frameCountOut; /* Safe cast. */
if (result != MA_SUCCESS || ma_sound_at_end(pSound)) {
break; /* Might have reached the end. */
}
}
}
*pFrameCountOut = totalFramesRead;
}
static void ma_engine_node_process_pcm_frames__group(ma_node* pNode, const float** ppFramesIn, ma_uint32* pFrameCountIn, float** ppFramesOut, ma_uint32* pFrameCountOut)
{
/*
Make sure the pitch is updated before trying to read anything. It's important that this is done
only once and not in ma_engine_node_process_pcm_frames__general(). The reason for this is that
ma_engine_node_process_pcm_frames__general() will call ma_engine_node_get_required_input_frame_count(),
and if another thread modifies the pitch just after that call it can result in a glitch due to
the input rate changing.
*/
ma_engine_node_update_pitch_if_required((ma_engine_node*)pNode);
/* For groups, the input data has already been read and we just need to apply the effect. */
ma_engine_node_process_pcm_frames__general((ma_engine_node*)pNode, ppFramesIn, pFrameCountIn, ppFramesOut, pFrameCountOut);
}
static ma_result ma_engine_node_get_required_input_frame_count__group(ma_node* pNode, ma_uint32 outputFrameCount, ma_uint32* pInputFrameCount)
{
ma_uint64 inputFrameCount;
MA_ASSERT(pInputFrameCount != NULL);
/* Our pitch will affect this calculation. We need to update it. */
ma_engine_node_update_pitch_if_required((ma_engine_node*)pNode);
inputFrameCount = ma_engine_node_get_required_input_frame_count((ma_engine_node*)pNode, outputFrameCount);
if (inputFrameCount > 0xFFFFFFFF) {
inputFrameCount = 0xFFFFFFFF; /* Will never happen because miniaudio will only ever process in relatively small chunks. */
}
*pInputFrameCount = (ma_uint32)inputFrameCount;
return MA_SUCCESS;
}
static ma_node_vtable g_ma_engine_node_vtable__sound =
{
ma_engine_node_process_pcm_frames__sound,
NULL, /* onGetRequiredInputFrameCount */
0, /* Sounds are data source nodes which means they have zero inputs (their input is drawn from the data source itself). */
1, /* Sounds have one output bus. */
0 /* Default flags. */
};
static ma_node_vtable g_ma_engine_node_vtable__group =
{
ma_engine_node_process_pcm_frames__group,
ma_engine_node_get_required_input_frame_count__group,
1, /* Groups have one input bus. */
1, /* Groups have one output bus. */
MA_NODE_FLAG_DIFFERENT_PROCESSING_RATES /* The engine node does resampling so should let miniaudio know about it. */
};
static ma_node_config ma_engine_node_base_node_config_init(const ma_engine_node_config* pConfig)
{
ma_node_config baseNodeConfig;
if (pConfig->type == ma_engine_node_type_sound) {
/* Sound. */
baseNodeConfig = ma_node_config_init();
baseNodeConfig.vtable = &g_ma_engine_node_vtable__sound;
baseNodeConfig.initialState = ma_node_state_stopped; /* Sounds are stopped by default. */
} else {
/* Group. */
baseNodeConfig = ma_node_config_init();
baseNodeConfig.vtable = &g_ma_engine_node_vtable__group;
baseNodeConfig.initialState = ma_node_state_started; /* Groups are started by default. */
}
return baseNodeConfig;
}
static ma_spatializer_config ma_engine_node_spatializer_config_init(const ma_node_config* pBaseNodeConfig)
{
return ma_spatializer_config_init(pBaseNodeConfig->pInputChannels[0], pBaseNodeConfig->pOutputChannels[0]);
}
typedef struct
{
size_t sizeInBytes;
size_t baseNodeOffset;
size_t resamplerOffset;
size_t spatializerOffset;
} ma_engine_node_heap_layout;
static ma_result ma_engine_node_get_heap_layout(const ma_engine_node_config* pConfig, ma_engine_node_heap_layout* pHeapLayout)
{
ma_result result;
size_t tempHeapSize;
ma_node_config baseNodeConfig;
ma_linear_resampler_config resamplerConfig;
ma_spatializer_config spatializerConfig;
ma_uint32 channelsIn;
ma_uint32 channelsOut;
MA_ASSERT(pHeapLayout);
MA_ZERO_OBJECT(pHeapLayout);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
if (pConfig->pEngine == NULL) {
return MA_INVALID_ARGS; /* An engine must be specified. */
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
return result;
}
pEngineNode->_ownsHeap = MA_TRUE;
return MA_SUCCESS;
}
MA_API void ma_engine_node_uninit(ma_engine_node* pEngineNode, const ma_allocation_callbacks* pAllocationCallbacks)
{
/*
The base node always needs to be uninitialized first to ensure it's detached from the graph completely before we
destroy anything that might be in the middle of being used by the processing function.
*/
ma_node_uninit(&pEngineNode->baseNode, pAllocationCallbacks);
/* Now that the node has been uninitialized we can safely uninitialize the rest. */
ma_spatializer_uninit(&pEngineNode->spatializer, pAllocationCallbacks);
ma_linear_resampler_uninit(&pEngineNode->resampler, pAllocationCallbacks);
/* Free the heap last. */
if (pEngineNode->_ownsHeap) {
ma_free(pEngineNode->_pHeap, pAllocationCallbacks);
}
}
MA_API ma_sound_config ma_sound_config_init(void)
{
ma_sound_config config;
MA_ZERO_OBJECT(&config);
config.rangeEndInPCMFrames = ~((ma_uint64)0);
config.loopPointEndInPCMFrames = ~((ma_uint64)0);
return config;
}
MA_API ma_sound_group_config ma_sound_group_config_init(void)
{
ma_sound_group_config config;
MA_ZERO_OBJECT(&config);
return config;
}
MA_API ma_engine_config ma_engine_config_init(void)
{
ma_engine_config config;
MA_ZERO_OBJECT(&config);
config.listenerCount = 1; /* Always want at least one listener. */
config.monoExpansionMode = ma_mono_expansion_mode_default;
return config;
}
#if !defined(MA_NO_DEVICE_IO)
static void ma_engine_data_callback_internal(ma_device* pDevice, void* pFramesOut, const void* pFramesIn, ma_uint32 frameCount)
{
ma_engine* pEngine = (ma_engine*)pDevice->pUserData;
(void)pFramesIn;
/*
Experiment: Try processing a resource manager job if we're on the Emscripten build.
This serves two purposes:
1) It ensures jobs are actually processed at some point since we cannot guarantee that the
caller is doing the right thing and calling ma_resource_manager_process_next_job(); and
2) It's an attempt at working around an issue where processing jobs on the Emscripten main
loop doesn't work as well as it should. When trying to load sounds without the `DECODE`
flag or with the `ASYNC` flag, the sound data is just not able to be loaded in time
before the callback is processed. I think it's got something to do with the single-
threaded nature of Web, but I'm not entirely sure.
*/
#if !defined(MA_NO_RESOURCE_MANAGER) && defined(MA_EMSCRIPTEN)
{
if (pEngine->pResourceManager != NULL) {
if ((pEngine->pResourceManager->config.flags & MA_RESOURCE_MANAGER_FLAG_NO_THREADING) != 0) {
ma_resource_manager_process_next_job(pEngine->pResourceManager);
}
}
}
#endif
ma_engine_read_pcm_frames(pEngine, pFramesOut, frameCount, NULL);
}
#endif
MA_API ma_result ma_engine_init(const ma_engine_config* pConfig, ma_engine* pEngine)
{
ma_result result;
ma_node_graph_config nodeGraphConfig;
ma_engine_config engineConfig;
ma_spatializer_listener_config listenerConfig;
ma_uint32 iListener;
if (pEngine == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pEngine);
/* The config is allowed to be NULL in which case we use defaults for everything. */
if (pConfig != NULL) {
engineConfig = *pConfig;
} else {
engineConfig = ma_engine_config_init();
}
pEngine->monoExpansionMode = engineConfig.monoExpansionMode;
ma_allocation_callbacks_init_copy(&pEngine->allocationCallbacks, &engineConfig.allocationCallbacks);
#if !defined(MA_NO_RESOURCE_MANAGER)
{
pEngine->pResourceManager = engineConfig.pResourceManager;
}
#endif
#if !defined(MA_NO_DEVICE_IO)
{
pEngine->pDevice = engineConfig.pDevice;
/* If we don't have a device, we need one. */
if (pEngine->pDevice == NULL && engineConfig.noDevice == MA_FALSE) {
ma_device_config deviceConfig;
pEngine->pDevice = (ma_device*)ma_malloc(sizeof(*pEngine->pDevice), &pEngine->allocationCallbacks);
if (pEngine->pDevice == NULL) {
return MA_OUT_OF_MEMORY;
}
deviceConfig = ma_device_config_init(ma_device_type_playback);
deviceConfig.playback.pDeviceID = engineConfig.pPlaybackDeviceID;
deviceConfig.playback.format = ma_format_f32;
deviceConfig.playback.channels = engineConfig.channels;
deviceConfig.sampleRate = engineConfig.sampleRate;
deviceConfig.dataCallback = ma_engine_data_callback_internal;
deviceConfig.pUserData = pEngine;
deviceConfig.periodSizeInFrames = engineConfig.periodSizeInFrames;
deviceConfig.periodSizeInMilliseconds = engineConfig.periodSizeInMilliseconds;
deviceConfig.noPreSilencedOutputBuffer = MA_TRUE; /* We'll always be outputting to every frame in the callback so there's no need for a pre-silenced buffer. */
deviceConfig.noClip = MA_TRUE; /* The engine will do clipping itself. */
if (engineConfig.pContext == NULL) {
ma_context_config contextConfig = ma_context_config_init();
contextConfig.allocationCallbacks = pEngine->allocationCallbacks;
contextConfig.pLog = engineConfig.pLog;
/* If the engine config does not specify a log, use the resource manager's if we have one. */
#ifndef MA_NO_RESOURCE_MANAGER
{
if (contextConfig.pLog == NULL && engineConfig.pResourceManager != NULL) {
contextConfig.pLog = ma_resource_manager_get_log(engineConfig.pResourceManager);
}
}
#endif
result = ma_device_init_ex(NULL, 0, &contextConfig, &deviceConfig, pEngine->pDevice);
} else {
result = ma_device_init(engineConfig.pContext, &deviceConfig, pEngine->pDevice);
}
if (result != MA_SUCCESS) {
ma_free(pEngine->pDevice, &pEngine->allocationCallbacks);
pEngine->pDevice = NULL;
return result;
}
pEngine->ownsDevice = MA_TRUE;
}
/* Update the channel count and sample rate of the engine config so we can reference it below. */
if (pEngine->pDevice != NULL) {
engineConfig.channels = pEngine->pDevice->playback.channels;
engineConfig.sampleRate = pEngine->pDevice->sampleRate;
}
}
#endif
if (engineConfig.channels == 0 || engineConfig.sampleRate == 0) {
return MA_INVALID_ARGS;
}
pEngine->sampleRate = engineConfig.sampleRate;
/* The engine always uses either the log that was passed into the config, or the context's log is available. */
if (engineConfig.pLog != NULL) {
pEngine->pLog = engineConfig.pLog;
} else {
#if !defined(MA_NO_DEVICE_IO)
{
pEngine->pLog = ma_device_get_log(pEngine->pDevice);
}
#else
{
pEngine->pLog = NULL;
}
#endif
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
MA_API void ma_engine_uninit(ma_engine* pEngine)
{
ma_uint32 iListener;
if (pEngine == NULL) {
return;
}
/* The device must be uninitialized before the node graph to ensure the audio thread doesn't try accessing it. */
#if !defined(MA_NO_DEVICE_IO)
{
if (pEngine->ownsDevice) {
ma_device_uninit(pEngine->pDevice);
ma_free(pEngine->pDevice, &pEngine->allocationCallbacks);
} else {
if (pEngine->pDevice != NULL) {
ma_device_stop(pEngine->pDevice);
}
}
}
#endif
/*
All inlined sounds need to be deleted. I'm going to use a lock here just to future proof in case
I want to do some kind of garbage collection later on.
*/
ma_spinlock_lock(&pEngine->inlinedSoundLock);
{
for (;;) {
ma_sound_inlined* pSoundToDelete = pEngine->pInlinedSoundHead;
if (pSoundToDelete == NULL) {
break; /* Done. */
}
pEngine->pInlinedSoundHead = pSoundToDelete->pNext;
ma_sound_uninit(&pSoundToDelete->sound);
ma_free(pSoundToDelete, &pEngine->allocationCallbacks);
}
}
ma_spinlock_unlock(&pEngine->inlinedSoundLock);
for (iListener = 0; iListener < pEngine->listenerCount; iListener += 1) {
ma_spatializer_listener_uninit(&pEngine->listeners[iListener], &pEngine->allocationCallbacks);
}
/* Make sure the node graph is uninitialized after the audio thread has been shutdown to prevent accessing of the node graph after being uninitialized. */
ma_node_graph_uninit(&pEngine->nodeGraph, &pEngine->allocationCallbacks);
/* Uninitialize the resource manager last to ensure we don't have a thread still trying to access it. */
#ifndef MA_NO_RESOURCE_MANAGER
if (pEngine->ownsResourceManager) {
ma_resource_manager_uninit(pEngine->pResourceManager);
ma_free(pEngine->pResourceManager, &pEngine->allocationCallbacks);
}
#endif
}
MA_API ma_result ma_engine_read_pcm_frames(ma_engine* pEngine, void* pFramesOut, ma_uint64 frameCount, ma_uint64* pFramesRead)
{
return ma_node_graph_read_pcm_frames(&pEngine->nodeGraph, pFramesOut, frameCount, pFramesRead);
}
MA_API ma_node_graph* ma_engine_get_node_graph(ma_engine* pEngine)
{
if (pEngine == NULL) {
return NULL;
}
return &pEngine->nodeGraph;
}
#if !defined(MA_NO_RESOURCE_MANAGER)
MA_API ma_resource_manager* ma_engine_get_resource_manager(ma_engine* pEngine)
{
if (pEngine == NULL) {
return NULL;
}
#if !defined(MA_NO_RESOURCE_MANAGER)
{
return pEngine->pResourceManager;
}
#else
{
return NULL;
}
#endif
}
#endif
MA_API ma_device* ma_engine_get_device(ma_engine* pEngine)
{
if (pEngine == NULL) {
return NULL;
}
#if !defined(MA_NO_DEVICE_IO)
{
return pEngine->pDevice;
}
#else
{
return NULL;
}
#endif
}
MA_API ma_log* ma_engine_get_log(ma_engine* pEngine)
{
if (pEngine == NULL) {
return NULL;
}
if (pEngine->pLog != NULL) {
return pEngine->pLog;
} else {
#if !defined(MA_NO_DEVICE_IO)
{
return ma_device_get_log(ma_engine_get_device(pEngine));
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
pSound->pDataSource = pConfig->pDataSource;
if (pConfig->pDataSource != NULL) {
type = ma_engine_node_type_sound;
} else {
type = ma_engine_node_type_group;
}
/*
Sounds are engine nodes. Before we can initialize this we need to determine the channel count.
If we can't do this we need to abort. It's up to the caller to ensure they're using a data
source that provides this information upfront.
*/
engineNodeConfig = ma_engine_node_config_init(pEngine, type, pConfig->flags);
engineNodeConfig.channelsIn = pConfig->channelsIn;
engineNodeConfig.channelsOut = pConfig->channelsOut;
/* If we're loading from a data source the input channel count needs to be the data source's native channel count. */
if (pConfig->pDataSource != NULL) {
result = ma_data_source_get_data_format(pConfig->pDataSource, NULL, &engineNodeConfig.channelsIn, &engineNodeConfig.sampleRate, NULL, 0);
if (result != MA_SUCCESS) {
return result; /* Failed to retrieve the channel count. */
}
if (engineNodeConfig.channelsIn == 0) {
return MA_INVALID_OPERATION; /* Invalid channel count. */
}
if (engineNodeConfig.channelsOut == MA_SOUND_SOURCE_CHANNEL_COUNT) {
engineNodeConfig.channelsOut = engineNodeConfig.channelsIn;
}
}
/* Getting here means we should have a valid channel count and we can initialize the engine node. */
result = ma_engine_node_init(&engineNodeConfig, &pEngine->allocationCallbacks, &pSound->engineNode);
if (result != MA_SUCCESS) {
return result;
}
/* If no attachment is specified, attach the sound straight to the endpoint. */
if (pConfig->pInitialAttachment == NULL) {
/* No group. Attach straight to the endpoint by default, unless the caller has requested that do not. */
if ((pConfig->flags & MA_SOUND_FLAG_NO_DEFAULT_ATTACHMENT) == 0) {
result = ma_node_attach_output_bus(pSound, 0, ma_node_graph_get_endpoint(&pEngine->nodeGraph), 0);
}
} else {
/* An attachment is specified. Attach to it by default. The sound has only a single output bus, and the config will specify which input bus to attach to. */
result = ma_node_attach_output_bus(pSound, 0, pConfig->pInitialAttachment, pConfig->initialAttachmentInputBusIndex);
}
if (result != MA_SUCCESS) {
ma_engine_node_uninit(&pSound->engineNode, &pEngine->allocationCallbacks);
return result;
}
/* Apply initial range and looping state to the data source if applicable. */
if (pConfig->rangeBegInPCMFrames != 0 || pConfig->rangeEndInPCMFrames != ~((ma_uint64)0)) {
ma_data_source_set_range_in_pcm_frames(ma_sound_get_data_source(pSound), pConfig->rangeBegInPCMFrames, pConfig->rangeEndInPCMFrames);
}
if (pConfig->loopPointBegInPCMFrames != 0 || pConfig->loopPointEndInPCMFrames != ~((ma_uint64)0)) {
ma_data_source_set_range_in_pcm_frames(ma_sound_get_data_source(pSound), pConfig->loopPointBegInPCMFrames, pConfig->loopPointEndInPCMFrames);
}
ma_sound_set_looping(pSound, pConfig->isLooping);
return MA_SUCCESS;
}
#ifndef MA_NO_RESOURCE_MANAGER
MA_API ma_result ma_sound_init_from_file_internal(ma_engine* pEngine, const ma_sound_config* pConfig, ma_sound* pSound)
{
ma_result result = MA_SUCCESS;
ma_uint32 flags;
ma_sound_config config;
ma_resource_manager_pipeline_notifications notifications;
/*
The engine requires knowledge of the channel count of the underlying data source before it can
initialize the sound. Therefore, we need to make the resource manager wait until initialization
of the underlying data source to be initialized so we can get access to the channel count. To
do this, the MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT is forced.
Because we're initializing the data source before the sound, there's a chance the notification
will get triggered before this function returns. This is OK, so long as the caller is aware of
it and can avoid accessing the sound from within the notification.
*/
flags = pConfig->flags | MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT;
pSound->pResourceManagerDataSource = (ma_resource_manager_data_source*)ma_malloc(sizeof(*pSound->pResourceManagerDataSource), &pEngine->allocationCallbacks);
if (pSound->pResourceManagerDataSource == NULL) {
return MA_OUT_OF_MEMORY;
}
notifications = ma_resource_manager_pipeline_notifications_init();
notifications.done.pFence = pConfig->pDoneFence;
/*
We must wrap everything around the fence if one was specified. This ensures ma_fence_wait() does
not return prematurely before the sound has finished initializing.
*/
if (notifications.done.pFence) { ma_fence_acquire(notifications.done.pFence); }
{
ma_resource_manager_data_source_config resourceManagerDataSourceConfig = ma_resource_manager_data_source_config_init();
resourceManagerDataSourceConfig.pFilePath = pConfig->pFilePath;
resourceManagerDataSourceConfig.pFilePathW = pConfig->pFilePathW;
resourceManagerDataSourceConfig.flags = flags;
resourceManagerDataSourceConfig.pNotifications = ¬ifications;
resourceManagerDataSourceConfig.initialSeekPointInPCMFrames = pConfig->initialSeekPointInPCMFrames;
resourceManagerDataSourceConfig.rangeBegInPCMFrames = pConfig->rangeBegInPCMFrames;
resourceManagerDataSourceConfig.rangeEndInPCMFrames = pConfig->rangeEndInPCMFrames;
resourceManagerDataSourceConfig.loopPointBegInPCMFrames = pConfig->loopPointBegInPCMFrames;
resourceManagerDataSourceConfig.loopPointEndInPCMFrames = pConfig->loopPointEndInPCMFrames;
resourceManagerDataSourceConfig.isLooping = pConfig->isLooping;
result = ma_resource_manager_data_source_init_ex(pEngine->pResourceManager, &resourceManagerDataSourceConfig, pSound->pResourceManagerDataSource);
if (result != MA_SUCCESS) {
goto done;
}
pSound->ownsDataSource = MA_TRUE; /* <-- Important. Not setting this will result in the resource manager data source never getting uninitialized. */
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
*/
return ma_sound_init_from_data_source_internal(pEngine, pConfig, pSound);
}
}
MA_API void ma_sound_uninit(ma_sound* pSound)
{
if (pSound == NULL) {
return;
}
/*
Always uninitialize the node first. This ensures it's detached from the graph and does not return until it has done
so which makes thread safety beyond this point trivial.
*/
ma_engine_node_uninit(&pSound->engineNode, &pSound->engineNode.pEngine->allocationCallbacks);
/* Once the sound is detached from the group we can guarantee that it won't be referenced by the mixer thread which means it's safe for us to destroy the data source. */
#ifndef MA_NO_RESOURCE_MANAGER
if (pSound->ownsDataSource) {
ma_resource_manager_data_source_uninit(pSound->pResourceManagerDataSource);
ma_free(pSound->pResourceManagerDataSource, &pSound->engineNode.pEngine->allocationCallbacks);
pSound->pDataSource = NULL;
}
#else
MA_ASSERT(pSound->ownsDataSource == MA_FALSE);
#endif
}
MA_API ma_engine* ma_sound_get_engine(const ma_sound* pSound)
{
if (pSound == NULL) {
return NULL;
}
return pSound->engineNode.pEngine;
}
MA_API ma_data_source* ma_sound_get_data_source(const ma_sound* pSound)
{
if (pSound == NULL) {
return NULL;
}
return pSound->pDataSource;
}
MA_API ma_result ma_sound_start(ma_sound* pSound)
{
if (pSound == NULL) {
return MA_INVALID_ARGS;
}
/* If the sound is already playing, do nothing. */
if (ma_sound_is_playing(pSound)) {
return MA_SUCCESS;
}
/* If the sound is at the end it means we want to start from the start again. */
if (ma_sound_at_end(pSound)) {
ma_result result = ma_data_source_seek_to_pcm_frame(pSound->pDataSource, 0);
if (result != MA_SUCCESS && result != MA_NOT_IMPLEMENTED) {
return result; /* Failed to seek back to the start. */
}
/* Make sure we clear the end indicator. */
c89atomic_exchange_32(&pSound->atEnd, MA_FALSE);
}
/* Make sure the sound is started. If there's a start delay, the sound won't actually start until the start time is reached. */
ma_node_set_state(pSound, ma_node_state_started);
return MA_SUCCESS;
}
MA_API ma_result ma_sound_stop(ma_sound* pSound)
{
if (pSound == NULL) {
return MA_INVALID_ARGS;
}
/* This will stop the sound immediately. Use ma_sound_set_stop_time() to stop the sound at a specific time. */
ma_node_set_state(pSound, ma_node_state_stopped);
return MA_SUCCESS;
}
MA_API void ma_sound_set_volume(ma_sound* pSound, float volume)
{
if (pSound == NULL) {
return;
}
/* The volume is controlled via the output bus. */
ma_node_set_output_bus_volume(pSound, 0, volume);
}
MA_API float ma_sound_get_volume(const ma_sound* pSound)
{
if (pSound == NULL) {
return 0;
}
return ma_node_get_output_bus_volume(pSound, 0);
}
MA_API void ma_sound_set_pan(ma_sound* pSound, float pan)
{
if (pSound == NULL) {
return;
}
ma_panner_set_pan(&pSound->engineNode.panner, pan);
}
MA_API float ma_sound_get_pan(const ma_sound* pSound)
{
if (pSound == NULL) {
return 0;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
return;
}
ma_spatializer_set_cone(&pSound->engineNode.spatializer, innerAngleInRadians, outerAngleInRadians, outerGain);
}
MA_API void ma_sound_get_cone(const ma_sound* pSound, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain)
{
if (pInnerAngleInRadians != NULL) {
*pInnerAngleInRadians = 0;
}
if (pOuterAngleInRadians != NULL) {
*pOuterAngleInRadians = 0;
}
if (pOuterGain != NULL) {
*pOuterGain = 0;
}
ma_spatializer_get_cone(&pSound->engineNode.spatializer, pInnerAngleInRadians, pOuterAngleInRadians, pOuterGain);
}
MA_API void ma_sound_set_doppler_factor(ma_sound* pSound, float dopplerFactor)
{
if (pSound == NULL) {
return;
}
ma_spatializer_set_doppler_factor(&pSound->engineNode.spatializer, dopplerFactor);
}
MA_API float ma_sound_get_doppler_factor(const ma_sound* pSound)
{
if (pSound == NULL) {
return 0;
}
return ma_spatializer_get_doppler_factor(&pSound->engineNode.spatializer);
}
MA_API void ma_sound_set_directional_attenuation_factor(ma_sound* pSound, float directionalAttenuationFactor)
{
if (pSound == NULL) {
return;
}
ma_spatializer_set_directional_attenuation_factor(&pSound->engineNode.spatializer, directionalAttenuationFactor);
}
MA_API float ma_sound_get_directional_attenuation_factor(const ma_sound* pSound)
{
if (pSound == NULL) {
return 1;
}
return ma_spatializer_get_directional_attenuation_factor(&pSound->engineNode.spatializer);
}
MA_API void ma_sound_set_fade_in_pcm_frames(ma_sound* pSound, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInFrames)
{
if (pSound == NULL) {
return;
}
ma_fader_set_fade(&pSound->engineNode.fader, volumeBeg, volumeEnd, fadeLengthInFrames);
}
MA_API void ma_sound_set_fade_in_milliseconds(ma_sound* pSound, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInMilliseconds)
{
if (pSound == NULL) {
return;
}
ma_sound_set_fade_in_pcm_frames(pSound, volumeBeg, volumeEnd, (fadeLengthInMilliseconds * pSound->engineNode.fader.config.sampleRate) / 1000);
}
MA_API float ma_sound_get_current_fade_volume(ma_sound* pSound)
{
if (pSound == NULL) {
return MA_INVALID_ARGS;
}
return ma_fader_get_current_volume(&pSound->engineNode.fader);
}
MA_API void ma_sound_set_start_time_in_pcm_frames(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInFrames)
{
if (pSound == NULL) {
return;
}
ma_node_set_state_time(pSound, ma_node_state_started, absoluteGlobalTimeInFrames);
}
MA_API void ma_sound_set_start_time_in_milliseconds(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInMilliseconds)
{
if (pSound == NULL) {
return;
}
ma_sound_set_start_time_in_pcm_frames(pSound, absoluteGlobalTimeInMilliseconds * ma_engine_get_sample_rate(ma_sound_get_engine(pSound)) / 1000);
}
MA_API void ma_sound_set_stop_time_in_pcm_frames(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInFrames)
{
if (pSound == NULL) {
return;
}
ma_node_set_state_time(pSound, ma_node_state_stopped, absoluteGlobalTimeInFrames);
}
MA_API void ma_sound_set_stop_time_in_milliseconds(ma_sound* pSound, ma_uint64 absoluteGlobalTimeInMilliseconds)
{
if (pSound == NULL) {
return;
}
ma_sound_set_stop_time_in_pcm_frames(pSound, absoluteGlobalTimeInMilliseconds * ma_engine_get_sample_rate(ma_sound_get_engine(pSound)) / 1000);
}
MA_API ma_bool32 ma_sound_is_playing(const ma_sound* pSound)
{
if (pSound == NULL) {
return MA_FALSE;
}
return ma_node_get_state_by_time(pSound, ma_engine_get_time(ma_sound_get_engine(pSound))) == ma_node_state_started;
}
MA_API ma_uint64 ma_sound_get_time_in_pcm_frames(const ma_sound* pSound)
{
if (pSound == NULL) {
return 0;
}
return ma_node_get_time(pSound);
}
MA_API void ma_sound_set_looping(ma_sound* pSound, ma_bool32 isLooping)
{
if (pSound == NULL) {
return;
}
/* Looping is only a valid concept if the sound is backed by a data source. */
if (pSound->pDataSource == NULL) {
return;
}
/* The looping state needs to be applied to the data source in order for any looping to actually happen. */
ma_data_source_set_looping(pSound->pDataSource, isLooping);
}
MA_API ma_bool32 ma_sound_is_looping(const ma_sound* pSound)
{
if (pSound == NULL) {
return MA_FALSE;
}
/* There is no notion of looping for sounds that are not backed by a data source. */
if (pSound->pDataSource == NULL) {
return MA_FALSE;
}
return ma_data_source_is_looping(pSound->pDataSource);
}
MA_API ma_bool32 ma_sound_at_end(const ma_sound* pSound)
{
if (pSound == NULL) {
return MA_FALSE;
}
/* There is no notion of an end of a sound if it's not backed by a data source. */
if (pSound->pDataSource == NULL) {
return MA_FALSE;
}
return c89atomic_load_32(&pSound->atEnd);
}
MA_API ma_result ma_sound_seek_to_pcm_frame(ma_sound* pSound, ma_uint64 frameIndex)
{
if (pSound == NULL) {
return MA_INVALID_ARGS;
}
/* Seeking is only valid for sounds that are backed by a data source. */
if (pSound->pDataSource == NULL) {
return MA_INVALID_OPERATION;
}
/* We can't be seeking while reading at the same time. We just set the seek target and get the mixing thread to do the actual seek. */
c89atomic_exchange_64(&pSound->seekTarget, frameIndex);
return MA_SUCCESS;
}
MA_API ma_result ma_sound_get_data_format(ma_sound* pSound, ma_format* pFormat, ma_uint32* pChannels, ma_uint32* pSampleRate, ma_channel* pChannelMap, size_t channelMapCap)
{
if (pSound == NULL) {
return MA_INVALID_ARGS;
}
/* The data format is retrieved directly from the data source if the sound is backed by one. Otherwise we pull it from the node. */
if (pSound->pDataSource == NULL) {
ma_uint32 channels;
if (pFormat != NULL) {
*pFormat = ma_format_f32;
}
channels = ma_node_get_input_channels(&pSound->engineNode, 0);
if (pChannels != NULL) {
*pChannels = channels;
}
if (pSampleRate != NULL) {
*pSampleRate = pSound->engineNode.resampler.config.sampleRateIn;
}
if (pChannelMap != NULL) {
ma_channel_map_init_standard(ma_standard_channel_map_default, pChannelMap, channelMapCap, channels);
}
return MA_SUCCESS;
} else {
return ma_data_source_get_data_format(pSound->pDataSource, pFormat, pChannels, pSampleRate, pChannelMap, channelMapCap);
}
}
MA_API ma_result ma_sound_get_cursor_in_pcm_frames(ma_sound* pSound, ma_uint64* pCursor)
{
if (pSound == NULL) {
return MA_INVALID_ARGS;
}
/* The notion of a cursor is only valid for sounds that are backed by a data source. */
if (pSound->pDataSource == NULL) {
return MA_INVALID_OPERATION;
}
return ma_data_source_get_cursor_in_pcm_frames(pSound->pDataSource, pCursor);
}
MA_API ma_result ma_sound_get_length_in_pcm_frames(ma_sound* pSound, ma_uint64* pLength)
{
if (pSound == NULL) {
return MA_INVALID_ARGS;
}
/* The notion of a sound length is only valid for sounds that are backed by a data source. */
if (pSound->pDataSource == NULL) {
return MA_INVALID_OPERATION;
}
return ma_data_source_get_length_in_pcm_frames(pSound->pDataSource, pLength);
}
MA_API ma_result ma_sound_get_cursor_in_seconds(ma_sound* pSound, float* pCursor)
{
if (pSound == NULL) {
return MA_INVALID_ARGS;
}
/* The notion of a cursor is only valid for sounds that are backed by a data source. */
if (pSound->pDataSource == NULL) {
return MA_INVALID_OPERATION;
}
return ma_data_source_get_cursor_in_seconds(pSound->pDataSource, pCursor);
}
MA_API ma_result ma_sound_get_length_in_seconds(ma_sound* pSound, float* pLength)
{
if (pSound == NULL) {
return MA_INVALID_ARGS;
}
/* The notion of a sound length is only valid for sounds that are backed by a data source. */
if (pSound->pDataSource == NULL) {
return MA_INVALID_OPERATION;
}
return ma_data_source_get_length_in_seconds(pSound->pDataSource, pLength);
}
MA_API ma_result ma_sound_group_init(ma_engine* pEngine, ma_uint32 flags, ma_sound_group* pParentGroup, ma_sound_group* pGroup)
{
ma_sound_group_config config = ma_sound_group_config_init();
config.flags = flags;
config.pInitialAttachment = pParentGroup;
return ma_sound_group_init_ex(pEngine, &config, pGroup);
}
MA_API ma_result ma_sound_group_init_ex(ma_engine* pEngine, const ma_sound_group_config* pConfig, ma_sound_group* pGroup)
{
ma_sound_config soundConfig;
if (pGroup == NULL) {
return MA_INVALID_ARGS;
}
MA_ZERO_OBJECT(pGroup);
if (pConfig == NULL) {
return MA_INVALID_ARGS;
}
/* A sound group is just a sound without a data source. */
soundConfig = *pConfig;
soundConfig.pFilePath = NULL;
soundConfig.pFilePathW = NULL;
soundConfig.pDataSource = NULL;
/*
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
MA_API void ma_sound_group_set_max_gain(ma_sound_group* pGroup, float maxGain)
{
ma_sound_set_max_gain(pGroup, maxGain);
}
MA_API float ma_sound_group_get_max_gain(const ma_sound_group* pGroup)
{
return ma_sound_get_max_gain(pGroup);
}
MA_API void ma_sound_group_set_min_distance(ma_sound_group* pGroup, float minDistance)
{
ma_sound_set_min_distance(pGroup, minDistance);
}
MA_API float ma_sound_group_get_min_distance(const ma_sound_group* pGroup)
{
return ma_sound_get_min_distance(pGroup);
}
MA_API void ma_sound_group_set_max_distance(ma_sound_group* pGroup, float maxDistance)
{
ma_sound_set_max_distance(pGroup, maxDistance);
}
MA_API float ma_sound_group_get_max_distance(const ma_sound_group* pGroup)
{
return ma_sound_get_max_distance(pGroup);
}
MA_API void ma_sound_group_set_cone(ma_sound_group* pGroup, float innerAngleInRadians, float outerAngleInRadians, float outerGain)
{
ma_sound_set_cone(pGroup, innerAngleInRadians, outerAngleInRadians, outerGain);
}
MA_API void ma_sound_group_get_cone(const ma_sound_group* pGroup, float* pInnerAngleInRadians, float* pOuterAngleInRadians, float* pOuterGain)
{
ma_sound_get_cone(pGroup, pInnerAngleInRadians, pOuterAngleInRadians, pOuterGain);
}
MA_API void ma_sound_group_set_doppler_factor(ma_sound_group* pGroup, float dopplerFactor)
{
ma_sound_set_doppler_factor(pGroup, dopplerFactor);
}
MA_API float ma_sound_group_get_doppler_factor(const ma_sound_group* pGroup)
{
return ma_sound_get_doppler_factor(pGroup);
}
MA_API void ma_sound_group_set_directional_attenuation_factor(ma_sound_group* pGroup, float directionalAttenuationFactor)
{
ma_sound_set_directional_attenuation_factor(pGroup, directionalAttenuationFactor);
}
MA_API float ma_sound_group_get_directional_attenuation_factor(const ma_sound_group* pGroup)
{
return ma_sound_get_directional_attenuation_factor(pGroup);
}
MA_API void ma_sound_group_set_fade_in_pcm_frames(ma_sound_group* pGroup, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInFrames)
{
ma_sound_set_fade_in_pcm_frames(pGroup, volumeBeg, volumeEnd, fadeLengthInFrames);
}
MA_API void ma_sound_group_set_fade_in_milliseconds(ma_sound_group* pGroup, float volumeBeg, float volumeEnd, ma_uint64 fadeLengthInMilliseconds)
{
ma_sound_set_fade_in_milliseconds(pGroup, volumeBeg, volumeEnd, fadeLengthInMilliseconds);
}
MA_API float ma_sound_group_get_current_fade_volume(ma_sound_group* pGroup)
{
return ma_sound_get_current_fade_volume(pGroup);
}
MA_API void ma_sound_group_set_start_time_in_pcm_frames(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInFrames)
{
ma_sound_set_start_time_in_pcm_frames(pGroup, absoluteGlobalTimeInFrames);
}
MA_API void ma_sound_group_set_start_time_in_milliseconds(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInMilliseconds)
{
ma_sound_set_start_time_in_milliseconds(pGroup, absoluteGlobalTimeInMilliseconds);
}
MA_API void ma_sound_group_set_stop_time_in_pcm_frames(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInFrames)
{
ma_sound_set_stop_time_in_pcm_frames(pGroup, absoluteGlobalTimeInFrames);
}
MA_API void ma_sound_group_set_stop_time_in_milliseconds(ma_sound_group* pGroup, ma_uint64 absoluteGlobalTimeInMilliseconds)
{
ma_sound_set_stop_time_in_milliseconds(pGroup, absoluteGlobalTimeInMilliseconds);
}
MA_API ma_bool32 ma_sound_group_is_playing(const ma_sound_group* pGroup)
{
return ma_sound_is_playing(pGroup);
}
MA_API ma_uint64 ma_sound_group_get_time_in_pcm_frames(const ma_sound_group* pGroup)
{
return ma_sound_get_time_in_pcm_frames(pGroup);
}
#endif /* MA_NO_ENGINE */
/**************************************************************************************************************************************************************
***************************************************************************************************************************************************************
Auto Generated
==============
All code below is auto-generated from a tool. This mostly consists of decoding backend implementations such as dr_wav, dr_flac, etc. If you find a bug in the
code below please report the bug to the respective repository for the relevant project (probably dr_libs).
***************************************************************************************************************************************************************
**************************************************************************************************************************************************************/
#if !defined(MA_NO_WAV) && (!defined(MA_NO_DECODING) || !defined(MA_NO_ENCODING))
#if !defined(DR_WAV_IMPLEMENTATION) && !defined(DRWAV_IMPLEMENTATION) /* For backwards compatibility. Will be removed in version 0.11 for cleanliness. */
/* dr_wav_c begin */
#ifndef dr_wav_c
#define dr_wav_c
#include <stdlib.h>
#include <string.h>
#include <limits.h>
#ifndef DR_WAV_NO_STDIO
#include <stdio.h>
#include <wchar.h>
#endif
#ifndef DRWAV_ASSERT
#include <assert.h>
#define DRWAV_ASSERT(expression) assert(expression)
#endif
#ifndef DRWAV_MALLOC
#define DRWAV_MALLOC(sz) malloc((sz))
#endif
#ifndef DRWAV_REALLOC
#define DRWAV_REALLOC(p, sz) realloc((p), (sz))
#endif
#ifndef DRWAV_FREE
#define DRWAV_FREE(p) free((p))
#endif
#ifndef DRWAV_COPY_MEMORY
#define DRWAV_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz))
#endif
#ifndef DRWAV_ZERO_MEMORY
#define DRWAV_ZERO_MEMORY(p, sz) memset((p), 0, (sz))
#endif
#ifndef DRWAV_ZERO_OBJECT
#define DRWAV_ZERO_OBJECT(p) DRWAV_ZERO_MEMORY((p), sizeof(*p))
#endif
#define drwav_countof(x) (sizeof(x) / sizeof(x[0]))
#define drwav_align(x, a) ((((x) + (a) - 1) / (a)) * (a))
#define drwav_min(a, b) (((a) < (b)) ? (a) : (b))
#define drwav_max(a, b) (((a) > (b)) ? (a) : (b))
#define drwav_clamp(x, lo, hi) (drwav_max((lo), drwav_min((hi), (x))))
#define drwav_offset_ptr(p, offset) (((drwav_uint8*)(p)) + (offset))
#define DRWAV_MAX_SIMD_VECTOR_SIZE 64
#if defined(__x86_64__) || defined(_M_X64)
#define DRWAV_X64
#elif defined(__i386) || defined(_M_IX86)
#define DRWAV_X86
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
return NULL;
}
DRWAV_PRIVATE void* drwav__realloc_from_callbacks(void* p, size_t szNew, size_t szOld, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (pAllocationCallbacks == NULL) {
return NULL;
}
if (pAllocationCallbacks->onRealloc != NULL) {
return pAllocationCallbacks->onRealloc(p, szNew, pAllocationCallbacks->pUserData);
}
if (pAllocationCallbacks->onMalloc != NULL && pAllocationCallbacks->onFree != NULL) {
void* p2;
p2 = pAllocationCallbacks->onMalloc(szNew, pAllocationCallbacks->pUserData);
if (p2 == NULL) {
return NULL;
}
if (p != NULL) {
DRWAV_COPY_MEMORY(p2, p, szOld);
pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData);
}
return p2;
}
return NULL;
}
DRWAV_PRIVATE void drwav__free_from_callbacks(void* p, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (p == NULL || pAllocationCallbacks == NULL) {
return;
}
if (pAllocationCallbacks->onFree != NULL) {
pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData);
}
}
DRWAV_PRIVATE drwav_allocation_callbacks drwav_copy_allocation_callbacks_or_defaults(const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (pAllocationCallbacks != NULL) {
return *pAllocationCallbacks;
} else {
drwav_allocation_callbacks allocationCallbacks;
allocationCallbacks.pUserData = NULL;
allocationCallbacks.onMalloc = drwav__malloc_default;
allocationCallbacks.onRealloc = drwav__realloc_default;
allocationCallbacks.onFree = drwav__free_default;
return allocationCallbacks;
}
}
static DRWAV_INLINE drwav_bool32 drwav__is_compressed_format_tag(drwav_uint16 formatTag)
{
return
formatTag == DR_WAVE_FORMAT_ADPCM ||
formatTag == DR_WAVE_FORMAT_DVI_ADPCM;
}
DRWAV_PRIVATE unsigned int drwav__chunk_padding_size_riff(drwav_uint64 chunkSize)
{
return (unsigned int)(chunkSize % 2);
}
DRWAV_PRIVATE unsigned int drwav__chunk_padding_size_w64(drwav_uint64 chunkSize)
{
return (unsigned int)(chunkSize % 8);
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut);
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ima(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut);
DRWAV_PRIVATE drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount);
DRWAV_PRIVATE drwav_result drwav__read_chunk_header(drwav_read_proc onRead, void* pUserData, drwav_container container, drwav_uint64* pRunningBytesReadOut, drwav_chunk_header* pHeaderOut)
{
if (container == drwav_container_riff || container == drwav_container_rf64) {
drwav_uint8 sizeInBytes[4];
if (onRead(pUserData, pHeaderOut->id.fourcc, 4) != 4) {
return DRWAV_AT_END;
}
if (onRead(pUserData, sizeInBytes, 4) != 4) {
return DRWAV_INVALID_FILE;
}
pHeaderOut->sizeInBytes = drwav_bytes_to_u32(sizeInBytes);
pHeaderOut->paddingSize = drwav__chunk_padding_size_riff(pHeaderOut->sizeInBytes);
*pRunningBytesReadOut += 8;
} else {
drwav_uint8 sizeInBytes[8];
if (onRead(pUserData, pHeaderOut->id.guid, 16) != 16) {
return DRWAV_AT_END;
}
if (onRead(pUserData, sizeInBytes, 8) != 8) {
return DRWAV_INVALID_FILE;
}
pHeaderOut->sizeInBytes = drwav_bytes_to_u64(sizeInBytes) - 24;
pHeaderOut->paddingSize = drwav__chunk_padding_size_w64(pHeaderOut->sizeInBytes);
*pRunningBytesReadOut += 24;
}
return DRWAV_SUCCESS;
}
DRWAV_PRIVATE drwav_bool32 drwav__seek_forward(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData)
{
drwav_uint64 bytesRemainingToSeek = offset;
while (bytesRemainingToSeek > 0) {
if (bytesRemainingToSeek > 0x7FFFFFFF) {
if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_current)) {
return DRWAV_FALSE;
}
bytesRemainingToSeek -= 0x7FFFFFFF;
} else {
if (!onSeek(pUserData, (int)bytesRemainingToSeek, drwav_seek_origin_current)) {
return DRWAV_FALSE;
}
bytesRemainingToSeek = 0;
}
}
return DRWAV_TRUE;
}
DRWAV_PRIVATE drwav_bool32 drwav__seek_from_start(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData)
{
if (offset <= 0x7FFFFFFF) {
return onSeek(pUserData, (int)offset, drwav_seek_origin_start);
}
if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_start)) {
return DRWAV_FALSE;
}
offset -= 0x7FFFFFFF;
for (;;) {
if (offset <= 0x7FFFFFFF) {
return onSeek(pUserData, (int)offset, drwav_seek_origin_current);
}
if (!onSeek(pUserData, 0x7FFFFFFF, drwav_seek_origin_current)) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
} else {
}
} else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_labelled_cue_region, "ltxt")) {
if (subchunkDataSize >= DRWAV_LIST_LABELLED_TEXT_BYTES) {
drwav_uint64 stringSizeWithNullTerminator = subchunkDataSize - DRWAV_LIST_LABELLED_TEXT_BYTES;
if (pParser->stage == drwav__metadata_parser_stage_count) {
pParser->metadataCount += 1;
drwav__metadata_request_extra_memory_for_stage_2(pParser, (size_t)stringSizeWithNullTerminator, 1);
} else {
subchunkBytesRead = drwav__read_list_labelled_cue_region_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor], subchunkDataSize);
if (subchunkBytesRead == subchunkDataSize) {
pParser->metadataCursor += 1;
} else {
}
}
} else {
}
} else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_software, "ISFT")) {
subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_software);
} else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_copyright, "ICOP")) {
subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_copyright);
} else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_title, "INAM")) {
subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_title);
} else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_artist, "IART")) {
subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_artist);
} else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_comment, "ICMT")) {
subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_comment);
} else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_date, "ICRD")) {
subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_date);
} else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_genre, "IGNR")) {
subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_genre);
} else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_album, "IPRD")) {
subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_album);
} else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_tracknumber, "ITRK")) {
subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_tracknumber);
} else if ((allowedMetadataTypes & drwav_metadata_type_unknown) != 0) {
subchunkBytesRead = drwav__metadata_process_unknown_chunk(pParser, subchunkId, subchunkDataSize, listType);
}
bytesRead += subchunkBytesRead;
DRWAV_ASSERT(subchunkBytesRead <= subchunkDataSize);
if (subchunkBytesRead < subchunkDataSize) {
drwav_uint64 bytesToSeek = subchunkDataSize - subchunkBytesRead;
if (!pParser->onSeek(pParser->pReadSeekUserData, (int)bytesToSeek, drwav_seek_origin_current)) {
break;
}
bytesRead += bytesToSeek;
}
if ((subchunkDataSize % 2) == 1) {
if (!pParser->onSeek(pParser->pReadSeekUserData, 1, drwav_seek_origin_current)) {
break;
}
bytesRead += 1;
}
}
} else if ((allowedMetadataTypes & drwav_metadata_type_unknown) != 0) {
bytesRead = drwav__metadata_process_unknown_chunk(pParser, pChunkID, pChunkHeader->sizeInBytes, drwav_metadata_location_top_level);
}
return bytesRead;
}
DRWAV_PRIVATE drwav_uint32 drwav_get_bytes_per_pcm_frame(drwav* pWav)
{
drwav_uint32 bytesPerFrame;
if ((pWav->bitsPerSample & 0x7) == 0) {
bytesPerFrame = (pWav->bitsPerSample * pWav->fmt.channels) >> 3;
} else {
bytesPerFrame = pWav->fmt.blockAlign;
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW || pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) {
if (bytesPerFrame != pWav->fmt.channels) {
return 0;
}
}
return bytesPerFrame;
}
DRWAV_API drwav_uint16 drwav_fmt_get_format(const drwav_fmt* pFMT)
{
if (pFMT == NULL) {
return 0;
}
if (pFMT->formatTag != DR_WAVE_FORMAT_EXTENSIBLE) {
return pFMT->formatTag;
} else {
return drwav_bytes_to_u16(pFMT->subFormat);
}
}
DRWAV_PRIVATE drwav_bool32 drwav_preinit(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pReadSeekUserData, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (pWav == NULL || onRead == NULL || onSeek == NULL) {
return DRWAV_FALSE;
}
DRWAV_ZERO_MEMORY(pWav, sizeof(*pWav));
pWav->onRead = onRead;
pWav->onSeek = onSeek;
pWav->pUserData = pReadSeekUserData;
pWav->allocationCallbacks = drwav_copy_allocation_callbacks_or_defaults(pAllocationCallbacks);
if (pWav->allocationCallbacks.onFree == NULL || (pWav->allocationCallbacks.onMalloc == NULL && pWav->allocationCallbacks.onRealloc == NULL)) {
return DRWAV_FALSE;
}
return DRWAV_TRUE;
}
DRWAV_PRIVATE drwav_bool32 drwav_init__internal(drwav* pWav, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags)
{
drwav_uint64 cursor;
drwav_bool32 sequential;
drwav_uint8 riff[4];
drwav_fmt fmt;
unsigned short translatedFormatTag;
drwav_bool32 foundDataChunk;
drwav_uint64 dataChunkSize = 0;
drwav_uint64 sampleCountFromFactChunk = 0;
drwav_uint64 chunkSize;
drwav__metadata_parser metadataParser;
cursor = 0;
sequential = (flags & DRWAV_SEQUENTIAL) != 0;
if (drwav__on_read(pWav->onRead, pWav->pUserData, riff, sizeof(riff), &cursor) != sizeof(riff)) {
return DRWAV_FALSE;
}
if (drwav_fourcc_equal(riff, "RIFF")) {
pWav->container = drwav_container_riff;
} else if (drwav_fourcc_equal(riff, "riff")) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
break;
}
if (pWav->container == drwav_container_riff) {
if (drwav_fourcc_equal(header.id.fourcc, "fact")) {
drwav_uint32 sampleCount;
if (drwav__on_read(pWav->onRead, pWav->pUserData, &sampleCount, 4, &cursor) != 4) {
return DRWAV_FALSE;
}
chunkSize -= 4;
if (!foundDataChunk) {
pWav->dataChunkDataPos = cursor;
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
sampleCountFromFactChunk = sampleCount;
} else {
sampleCountFromFactChunk = 0;
}
}
} else if (pWav->container == drwav_container_w64) {
if (drwav_guid_equal(header.id.guid, drwavGUID_W64_FACT)) {
if (drwav__on_read(pWav->onRead, pWav->pUserData, &sampleCountFromFactChunk, 8, &cursor) != 8) {
return DRWAV_FALSE;
}
chunkSize -= 8;
if (!foundDataChunk) {
pWav->dataChunkDataPos = cursor;
}
}
} else if (pWav->container == drwav_container_rf64) {
}
chunkSize += header.paddingSize;
if (!drwav__seek_forward(pWav->onSeek, chunkSize, pWav->pUserData)) {
break;
}
cursor += chunkSize;
if (!foundDataChunk) {
pWav->dataChunkDataPos = cursor;
}
}
pWav->pMetadata = metadataParser.pMetadata;
pWav->metadataCount = metadataParser.metadataCount;
if (!foundDataChunk) {
return DRWAV_FALSE;
}
if (!sequential) {
if (!drwav__seek_from_start(pWav->onSeek, pWav->dataChunkDataPos, pWav->pUserData)) {
return DRWAV_FALSE;
}
cursor = pWav->dataChunkDataPos;
}
pWav->fmt = fmt;
pWav->sampleRate = fmt.sampleRate;
pWav->channels = fmt.channels;
pWav->bitsPerSample = fmt.bitsPerSample;
pWav->bytesRemaining = dataChunkSize;
pWav->translatedFormatTag = translatedFormatTag;
pWav->dataChunkDataSize = dataChunkSize;
if (sampleCountFromFactChunk != 0) {
pWav->totalPCMFrameCount = sampleCountFromFactChunk;
} else {
drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return DRWAV_FALSE;
}
pWav->totalPCMFrameCount = dataChunkSize / bytesPerFrame;
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
drwav_uint64 totalBlockHeaderSizeInBytes;
drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign;
if ((blockCount * fmt.blockAlign) < dataChunkSize) {
blockCount += 1;
}
totalBlockHeaderSizeInBytes = blockCount * (6*fmt.channels);
pWav->totalPCMFrameCount = ((dataChunkSize - totalBlockHeaderSizeInBytes) * 2) / fmt.channels;
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
drwav_uint64 totalBlockHeaderSizeInBytes;
drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign;
if ((blockCount * fmt.blockAlign) < dataChunkSize) {
blockCount += 1;
}
totalBlockHeaderSizeInBytes = blockCount * (4*fmt.channels);
pWav->totalPCMFrameCount = ((dataChunkSize - totalBlockHeaderSizeInBytes) * 2) / fmt.channels;
pWav->totalPCMFrameCount += blockCount;
}
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
if (pWav->channels > 2) {
return DRWAV_FALSE;
}
}
if (drwav_get_bytes_per_pcm_frame(pWav) == 0) {
return DRWAV_FALSE;
}
#ifdef DR_WAV_LIBSNDFILE_COMPAT
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign;
pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (6*pWav->channels))) * 2)) / fmt.channels;
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign;
pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (4*pWav->channels))) * 2) + (blockCount * pWav->channels)) / fmt.channels;
}
#endif
return DRWAV_TRUE;
}
DRWAV_API drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks)
{
return drwav_init_ex(pWav, onRead, onSeek, NULL, pUserData, NULL, 0, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_chunk_proc onChunk, void* pReadSeekUserData, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (!drwav_preinit(pWav, onRead, onSeek, pReadSeekUserData, pAllocationCallbacks)) {
return DRWAV_FALSE;
}
return drwav_init__internal(pWav, onChunk, pChunkUserData, flags);
}
DRWAV_API drwav_bool32 drwav_init_with_metadata(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (!drwav_preinit(pWav, onRead, onSeek, pUserData, pAllocationCallbacks)) {
return DRWAV_FALSE;
}
pWav->allowedMetadataTypes = drwav_metadata_type_all_including_unknown;
return drwav_init__internal(pWav, NULL, NULL, flags);
}
DRWAV_API drwav_metadata* drwav_take_ownership_of_metadata(drwav* pWav)
{
drwav_metadata *result = pWav->pMetadata;
pWav->pMetadata = NULL;
pWav->metadataCount = 0;
return result;
}
DRWAV_PRIVATE size_t drwav__write(drwav* pWav, const void* pData, size_t dataSize)
{
DRWAV_ASSERT(pWav != NULL);
DRWAV_ASSERT(pWav->onWrite != NULL);
return pWav->onWrite(pWav->pUserData, pData, dataSize);
}
DRWAV_PRIVATE size_t drwav__write_byte(drwav* pWav, drwav_uint8 byte)
{
DRWAV_ASSERT(pWav != NULL);
DRWAV_ASSERT(pWav->onWrite != NULL);
return pWav->onWrite(pWav->pUserData, &byte, 1);
}
DRWAV_PRIVATE size_t drwav__write_u16ne_to_le(drwav* pWav, drwav_uint16 value)
{
DRWAV_ASSERT(pWav != NULL);
DRWAV_ASSERT(pWav->onWrite != NULL);
if (!drwav__is_little_endian()) {
value = drwav__bswap16(value);
}
return drwav__write(pWav, &value, 2);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
runningPos += drwav__write(pWav, "ds64", 4);
runningPos += drwav__write_u32ne_to_le(pWav, initialds64ChunkSize);
runningPos += drwav__write_u64ne_to_le(pWav, initialRiffChunkSize);
runningPos += drwav__write_u64ne_to_le(pWav, initialDataChunkSize);
runningPos += drwav__write_u64ne_to_le(pWav, totalSampleCount);
runningPos += drwav__write_u32ne_to_le(pWav, 0);
}
if (pFormat->container == drwav_container_riff || pFormat->container == drwav_container_rf64) {
chunkSizeFMT = 16;
runningPos += drwav__write(pWav, "fmt ", 4);
runningPos += drwav__write_u32ne_to_le(pWav, (drwav_uint32)chunkSizeFMT);
} else if (pFormat->container == drwav_container_w64) {
chunkSizeFMT = 40;
runningPos += drwav__write(pWav, drwavGUID_W64_FMT, 16);
runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeFMT);
}
runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.formatTag);
runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.channels);
runningPos += drwav__write_u32ne_to_le(pWav, pWav->fmt.sampleRate);
runningPos += drwav__write_u32ne_to_le(pWav, pWav->fmt.avgBytesPerSec);
runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.blockAlign);
runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.bitsPerSample);
if (!pWav->isSequentialWrite && pWav->pMetadata != NULL && pWav->metadataCount > 0 && (pFormat->container == drwav_container_riff || pFormat->container == drwav_container_rf64)) {
runningPos += drwav__write_or_count_metadata(pWav, pWav->pMetadata, pWav->metadataCount);
}
pWav->dataChunkDataPos = runningPos;
if (pFormat->container == drwav_container_riff) {
drwav_uint32 chunkSizeDATA = (drwav_uint32)initialDataChunkSize;
runningPos += drwav__write(pWav, "data", 4);
runningPos += drwav__write_u32ne_to_le(pWav, chunkSizeDATA);
} else if (pFormat->container == drwav_container_w64) {
drwav_uint64 chunkSizeDATA = 24 + initialDataChunkSize;
runningPos += drwav__write(pWav, drwavGUID_W64_DATA, 16);
runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeDATA);
} else if (pFormat->container == drwav_container_rf64) {
runningPos += drwav__write(pWav, "data", 4);
runningPos += drwav__write_u32ne_to_le(pWav, 0xFFFFFFFF);
}
pWav->container = pFormat->container;
pWav->channels = (drwav_uint16)pFormat->channels;
pWav->sampleRate = pFormat->sampleRate;
pWav->bitsPerSample = (drwav_uint16)pFormat->bitsPerSample;
pWav->translatedFormatTag = (drwav_uint16)pFormat->format;
pWav->dataChunkDataPos = runningPos;
return DRWAV_TRUE;
}
DRWAV_API drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (!drwav_preinit_write(pWav, pFormat, DRWAV_FALSE, onWrite, onSeek, pUserData, pAllocationCallbacks)) {
return DRWAV_FALSE;
}
return drwav_init_write__internal(pWav, pFormat, 0);
}
DRWAV_API drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (!drwav_preinit_write(pWav, pFormat, DRWAV_TRUE, onWrite, NULL, pUserData, pAllocationCallbacks)) {
return DRWAV_FALSE;
}
return drwav_init_write__internal(pWav, pFormat, totalSampleCount);
}
DRWAV_API drwav_bool32 drwav_init_write_sequential_pcm_frames(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (pFormat == NULL) {
return DRWAV_FALSE;
}
return drwav_init_write_sequential(pWav, pFormat, totalPCMFrameCount*pFormat->channels, onWrite, pUserData, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_write_with_metadata(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks, drwav_metadata* pMetadata,...
{
if (!drwav_preinit_write(pWav, pFormat, DRWAV_FALSE, onWrite, onSeek, pUserData, pAllocationCallbacks)) {
return DRWAV_FALSE;
}
pWav->pMetadata = pMetadata;
pWav->metadataCount = metadataCount;
return drwav_init_write__internal(pWav, pFormat, 0);
}
DRWAV_API drwav_uint64 drwav_target_write_size_bytes(const drwav_data_format* pFormat, drwav_uint64 totalFrameCount, drwav_metadata* pMetadata, drwav_uint32 metadataCount)
{
drwav_uint64 targetDataSizeBytes = (drwav_uint64)((drwav_int64)totalFrameCount * pFormat->channels * pFormat->bitsPerSample/8.0);
drwav_uint64 riffChunkSizeBytes;
drwav_uint64 fileSizeBytes = 0;
if (pFormat->container == drwav_container_riff) {
riffChunkSizeBytes = drwav__riff_chunk_size_riff(targetDataSizeBytes, pMetadata, metadataCount);
fileSizeBytes = (8 + riffChunkSizeBytes);
} else if (pFormat->container == drwav_container_w64) {
riffChunkSizeBytes = drwav__riff_chunk_size_w64(targetDataSizeBytes);
fileSizeBytes = riffChunkSizeBytes;
} else if (pFormat->container == drwav_container_rf64) {
riffChunkSizeBytes = drwav__riff_chunk_size_rf64(targetDataSizeBytes, pMetadata, metadataCount);
fileSizeBytes = (8 + riffChunkSizeBytes);
}
return fileSizeBytes;
}
#ifndef DR_WAV_NO_STDIO
#include <errno.h>
DRWAV_PRIVATE drwav_result drwav_result_from_errno(int e)
{
switch (e)
{
case 0: return DRWAV_SUCCESS;
#ifdef EPERM
case EPERM: return DRWAV_INVALID_OPERATION;
#endif
#ifdef ENOENT
case ENOENT: return DRWAV_DOES_NOT_EXIST;
#endif
#ifdef ESRCH
case ESRCH: return DRWAV_DOES_NOT_EXIST;
#endif
#ifdef EINTR
case EINTR: return DRWAV_INTERRUPT;
#endif
#ifdef EIO
case EIO: return DRWAV_IO_ERROR;
#endif
#ifdef ENXIO
case ENXIO: return DRWAV_DOES_NOT_EXIST;
#endif
#ifdef E2BIG
case E2BIG: return DRWAV_INVALID_ARGS;
#endif
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
if (drwav_wfopen(&pFile, filename, L"rb", pAllocationCallbacks) != DRWAV_SUCCESS) {
return DRWAV_FALSE;
}
return drwav_init_file__internal_FILE(pWav, pFile, onChunk, pChunkUserData, flags, drwav_metadata_type_none, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_file_with_metadata(drwav* pWav, const char* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
{
FILE* pFile;
if (drwav_fopen(&pFile, filename, "rb") != DRWAV_SUCCESS) {
return DRWAV_FALSE;
}
return drwav_init_file__internal_FILE(pWav, pFile, NULL, NULL, flags, drwav_metadata_type_all_including_unknown, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_file_with_metadata_w(drwav* pWav, const wchar_t* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
{
FILE* pFile;
if (drwav_wfopen(&pFile, filename, L"rb", pAllocationCallbacks) != DRWAV_SUCCESS) {
return DRWAV_FALSE;
}
return drwav_init_file__internal_FILE(pWav, pFile, NULL, NULL, flags, drwav_metadata_type_all_including_unknown, pAllocationCallbacks);
}
DRWAV_PRIVATE drwav_bool32 drwav_init_file_write__internal_FILE(drwav* pWav, FILE* pFile, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks)
{
drwav_bool32 result;
result = drwav_preinit_write(pWav, pFormat, isSequential, drwav__on_write_stdio, drwav__on_seek_stdio, (void*)pFile, pAllocationCallbacks);
if (result != DRWAV_TRUE) {
fclose(pFile);
return result;
}
result = drwav_init_write__internal(pWav, pFormat, totalSampleCount);
if (result != DRWAV_TRUE) {
fclose(pFile);
return result;
}
return DRWAV_TRUE;
}
DRWAV_PRIVATE drwav_bool32 drwav_init_file_write__internal(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks)
{
FILE* pFile;
if (drwav_fopen(&pFile, filename, "wb") != DRWAV_SUCCESS) {
return DRWAV_FALSE;
}
return drwav_init_file_write__internal_FILE(pWav, pFile, pFormat, totalSampleCount, isSequential, pAllocationCallbacks);
}
DRWAV_PRIVATE drwav_bool32 drwav_init_file_write_w__internal(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks)
{
FILE* pFile;
if (drwav_wfopen(&pFile, filename, L"wb", pAllocationCallbacks) != DRWAV_SUCCESS) {
return DRWAV_FALSE;
}
return drwav_init_file_write__internal_FILE(pWav, pFile, pFormat, totalSampleCount, isSequential, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks)
{
return drwav_init_file_write__internal(pWav, filename, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks)
{
return drwav_init_file_write__internal(pWav, filename, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (pFormat == NULL) {
return DRWAV_FALSE;
}
return drwav_init_file_write_sequential(pWav, filename, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_file_write_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks)
{
return drwav_init_file_write_w__internal(pWav, filename, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_file_write_sequential_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks)
{
return drwav_init_file_write_w__internal(pWav, filename, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (pFormat == NULL) {
return DRWAV_FALSE;
}
return drwav_init_file_write_sequential_w(pWav, filename, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks);
}
#endif
DRWAV_PRIVATE size_t drwav__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead)
{
drwav* pWav = (drwav*)pUserData;
size_t bytesRemaining;
DRWAV_ASSERT(pWav != NULL);
DRWAV_ASSERT(pWav->memoryStream.dataSize >= pWav->memoryStream.currentReadPos);
bytesRemaining = pWav->memoryStream.dataSize - pWav->memoryStream.currentReadPos;
if (bytesToRead > bytesRemaining) {
bytesToRead = bytesRemaining;
}
if (bytesToRead > 0) {
DRWAV_COPY_MEMORY(pBufferOut, pWav->memoryStream.data + pWav->memoryStream.currentReadPos, bytesToRead);
pWav->memoryStream.currentReadPos += bytesToRead;
}
return bytesToRead;
}
DRWAV_PRIVATE drwav_bool32 drwav__on_seek_memory(void* pUserData, int offset, drwav_seek_origin origin)
{
drwav* pWav = (drwav*)pUserData;
DRWAV_ASSERT(pWav != NULL);
if (origin == drwav_seek_origin_current) {
if (offset > 0) {
if (pWav->memoryStream.currentReadPos + offset > pWav->memoryStream.dataSize) {
return DRWAV_FALSE;
}
} else {
if (pWav->memoryStream.currentReadPos < (size_t)-offset) {
return DRWAV_FALSE;
}
}
pWav->memoryStream.currentReadPos += offset;
} else {
if ((drwav_uint32)offset <= pWav->memoryStream.dataSize) {
pWav->memoryStream.currentReadPos = offset;
} else {
return DRWAV_FALSE;
}
}
return DRWAV_TRUE;
}
DRWAV_PRIVATE size_t drwav__on_write_memory(void* pUserData, const void* pDataIn, size_t bytesToWrite)
{
drwav* pWav = (drwav*)pUserData;
size_t bytesRemaining;
DRWAV_ASSERT(pWav != NULL);
DRWAV_ASSERT(pWav->memoryStreamWrite.dataCapacity >= pWav->memoryStreamWrite.currentWritePos);
bytesRemaining = pWav->memoryStreamWrite.dataCapacity - pWav->memoryStreamWrite.currentWritePos;
if (bytesRemaining < bytesToWrite) {
void* pNewData;
size_t newDataCapacity = (pWav->memoryStreamWrite.dataCapacity == 0) ? 256 : pWav->memoryStreamWrite.dataCapacity * 2;
if ((newDataCapacity - pWav->memoryStreamWrite.currentWritePos) < bytesToWrite) {
newDataCapacity = pWav->memoryStreamWrite.currentWritePos + bytesToWrite;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
}
return DRWAV_TRUE;
}
DRWAV_API drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize, const drwav_allocation_callbacks* pAllocationCallbacks)
{
return drwav_init_memory_ex(pWav, data, dataSize, NULL, NULL, 0, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (data == NULL || dataSize == 0) {
return DRWAV_FALSE;
}
if (!drwav_preinit(pWav, drwav__on_read_memory, drwav__on_seek_memory, pWav, pAllocationCallbacks)) {
return DRWAV_FALSE;
}
pWav->memoryStream.data = (const drwav_uint8*)data;
pWav->memoryStream.dataSize = dataSize;
pWav->memoryStream.currentReadPos = 0;
return drwav_init__internal(pWav, onChunk, pChunkUserData, flags);
}
DRWAV_API drwav_bool32 drwav_init_memory_with_metadata(drwav* pWav, const void* data, size_t dataSize, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (data == NULL || dataSize == 0) {
return DRWAV_FALSE;
}
if (!drwav_preinit(pWav, drwav__on_read_memory, drwav__on_seek_memory, pWav, pAllocationCallbacks)) {
return DRWAV_FALSE;
}
pWav->memoryStream.data = (const drwav_uint8*)data;
pWav->memoryStream.dataSize = dataSize;
pWav->memoryStream.currentReadPos = 0;
pWav->allowedMetadataTypes = drwav_metadata_type_all_including_unknown;
return drwav_init__internal(pWav, NULL, NULL, flags);
}
DRWAV_PRIVATE drwav_bool32 drwav_init_memory_write__internal(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallb...
{
if (ppData == NULL || pDataSize == NULL) {
return DRWAV_FALSE;
}
*ppData = NULL;
*pDataSize = 0;
if (!drwav_preinit_write(pWav, pFormat, isSequential, drwav__on_write_memory, drwav__on_seek_memory_write, pWav, pAllocationCallbacks)) {
return DRWAV_FALSE;
}
pWav->memoryStreamWrite.ppData = ppData;
pWav->memoryStreamWrite.pDataSize = pDataSize;
pWav->memoryStreamWrite.dataSize = 0;
pWav->memoryStreamWrite.dataCapacity = 0;
pWav->memoryStreamWrite.currentWritePos = 0;
return drwav_init_write__internal(pWav, pFormat, totalSampleCount);
}
DRWAV_API drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks)
{
return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks)
{
return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks);
}
DRWAV_API drwav_bool32 drwav_init_memory_write_sequential_pcm_frames(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (pFormat == NULL) {
return DRWAV_FALSE;
}
return drwav_init_memory_write_sequential(pWav, ppData, pDataSize, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks);
}
DRWAV_API drwav_result drwav_uninit(drwav* pWav)
{
drwav_result result = DRWAV_SUCCESS;
if (pWav == NULL) {
return DRWAV_INVALID_ARGS;
}
if (pWav->onWrite != NULL) {
drwav_uint32 paddingSize = 0;
if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rf64) {
paddingSize = drwav__chunk_padding_size_riff(pWav->dataChunkDataSize);
} else {
paddingSize = drwav__chunk_padding_size_w64(pWav->dataChunkDataSize);
}
if (paddingSize > 0) {
drwav_uint64 paddingData = 0;
drwav__write(pWav, &paddingData, paddingSize);
}
if (pWav->onSeek && !pWav->isSequentialWrite) {
if (pWav->container == drwav_container_riff) {
if (pWav->onSeek(pWav->pUserData, 4, drwav_seek_origin_start)) {
drwav_uint32 riffChunkSize = drwav__riff_chunk_size_riff(pWav->dataChunkDataSize, pWav->pMetadata, pWav->metadataCount);
drwav__write_u32ne_to_le(pWav, riffChunkSize);
}
if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos - 4, drwav_seek_origin_start)) {
drwav_uint32 dataChunkSize = drwav__data_chunk_size_riff(pWav->dataChunkDataSize);
drwav__write_u32ne_to_le(pWav, dataChunkSize);
}
} else if (pWav->container == drwav_container_w64) {
if (pWav->onSeek(pWav->pUserData, 16, drwav_seek_origin_start)) {
drwav_uint64 riffChunkSize = drwav__riff_chunk_size_w64(pWav->dataChunkDataSize);
drwav__write_u64ne_to_le(pWav, riffChunkSize);
}
if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos - 8, drwav_seek_origin_start)) {
drwav_uint64 dataChunkSize = drwav__data_chunk_size_w64(pWav->dataChunkDataSize);
drwav__write_u64ne_to_le(pWav, dataChunkSize);
}
} else if (pWav->container == drwav_container_rf64) {
int ds64BodyPos = 12 + 8;
if (pWav->onSeek(pWav->pUserData, ds64BodyPos + 0, drwav_seek_origin_start)) {
drwav_uint64 riffChunkSize = drwav__riff_chunk_size_rf64(pWav->dataChunkDataSize, pWav->pMetadata, pWav->metadataCount);
drwav__write_u64ne_to_le(pWav, riffChunkSize);
}
if (pWav->onSeek(pWav->pUserData, ds64BodyPos + 8, drwav_seek_origin_start)) {
drwav_uint64 dataChunkSize = drwav__data_chunk_size_rf64(pWav->dataChunkDataSize);
drwav__write_u64ne_to_le(pWav, dataChunkSize);
}
}
}
if (pWav->isSequentialWrite) {
if (pWav->dataChunkDataSize != pWav->dataChunkDataSizeTargetWrite) {
result = DRWAV_INVALID_FILE;
}
}
} else {
if (pWav->pMetadata != NULL) {
pWav->allocationCallbacks.onFree(pWav->pMetadata, pWav->allocationCallbacks.pUserData);
}
}
#ifndef DR_WAV_NO_STDIO
if (pWav->onRead == drwav__on_read_stdio || pWav->onWrite == drwav__on_write_stdio) {
fclose((FILE*)pWav->pUserData);
}
#endif
return result;
}
DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut)
{
size_t bytesRead;
drwav_uint32 bytesPerFrame;
if (pWav == NULL || bytesToRead == 0) {
return 0;
}
if (bytesToRead > pWav->bytesRemaining) {
bytesToRead = (size_t)pWav->bytesRemaining;
}
if (bytesToRead == 0) {
return 0;
}
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
if (pBufferOut != NULL) {
bytesRead = pWav->onRead(pWav->pUserData, pBufferOut, bytesToRead);
} else {
bytesRead = 0;
while (bytesRead < bytesToRead) {
size_t bytesToSeek = (bytesToRead - bytesRead);
if (bytesToSeek > 0x7FFFFFFF) {
bytesToSeek = 0x7FFFFFFF;
}
if (pWav->onSeek(pWav->pUserData, (int)bytesToSeek, drwav_seek_origin_current) == DRWAV_FALSE) {
break;
}
bytesRead += bytesToSeek;
}
while (bytesRead < bytesToRead) {
drwav_uint8 buffer[4096];
size_t bytesSeeked;
size_t bytesToSeek = (bytesToRead - bytesRead);
if (bytesToSeek > sizeof(buffer)) {
bytesToSeek = sizeof(buffer);
}
bytesSeeked = pWav->onRead(pWav->pUserData, buffer, bytesToSeek);
bytesRead += bytesSeeked;
if (bytesSeeked < bytesToSeek) {
break;
}
}
}
pWav->readCursorInPCMFrames += bytesRead / bytesPerFrame;
pWav->bytesRemaining -= bytesRead;
return bytesRead;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_le(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut)
{
drwav_uint32 bytesPerFrame;
drwav_uint64 bytesToRead;
if (pWav == NULL || framesToRead == 0) {
return 0;
}
if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) {
return 0;
}
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesToRead = framesToRead * bytesPerFrame;
if (bytesToRead > DRWAV_SIZE_MAX) {
bytesToRead = (DRWAV_SIZE_MAX / bytesPerFrame) * bytesPerFrame;
}
if (bytesToRead == 0) {
return 0;
}
return drwav_read_raw(pWav, (size_t)bytesToRead, pBufferOut) / bytesPerFrame;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut)
{
drwav_uint64 framesRead = drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut);
if (pBufferOut != NULL) {
drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
drwav__bswap_samples(pBufferOut, framesRead*pWav->channels, bytesPerFrame/pWav->channels, pWav->translatedFormatTag);
}
return framesRead;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut)
{
if (drwav__is_little_endian()) {
return drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut);
} else {
return drwav_read_pcm_frames_be(pWav, framesToRead, pBufferOut);
}
}
DRWAV_PRIVATE drwav_bool32 drwav_seek_to_first_pcm_frame(drwav* pWav)
{
if (pWav->onWrite != NULL) {
return DRWAV_FALSE;
}
if (!pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos, drwav_seek_origin_start)) {
return DRWAV_FALSE;
}
if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) {
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
DRWAV_ZERO_OBJECT(&pWav->msadpcm);
} else if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
DRWAV_ZERO_OBJECT(&pWav->ima);
} else {
DRWAV_ASSERT(DRWAV_FALSE);
}
}
pWav->readCursorInPCMFrames = 0;
pWav->bytesRemaining = pWav->dataChunkDataSize;
return DRWAV_TRUE;
}
DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex)
{
if (pWav == NULL || pWav->onSeek == NULL) {
return DRWAV_FALSE;
}
if (pWav->onWrite != NULL) {
return DRWAV_FALSE;
}
if (pWav->totalPCMFrameCount == 0) {
return DRWAV_TRUE;
}
if (targetFrameIndex > pWav->totalPCMFrameCount) {
targetFrameIndex = pWav->totalPCMFrameCount;
}
if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) {
if (targetFrameIndex < pWav->readCursorInPCMFrames) {
if (!drwav_seek_to_first_pcm_frame(pWav)) {
return DRWAV_FALSE;
}
}
if (targetFrameIndex > pWav->readCursorInPCMFrames) {
drwav_uint64 offsetInFrames = targetFrameIndex - pWav->readCursorInPCMFrames;
drwav_int16 devnull[2048];
while (offsetInFrames > 0) {
drwav_uint64 framesRead = 0;
drwav_uint64 framesToRead = offsetInFrames;
if (framesToRead > drwav_countof(devnull)/pWav->channels) {
framesToRead = drwav_countof(devnull)/pWav->channels;
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
framesRead = drwav_read_pcm_frames_s16__msadpcm(pWav, framesToRead, devnull);
} else if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
framesRead = drwav_read_pcm_frames_s16__ima(pWav, framesToRead, devnull);
} else {
DRWAV_ASSERT(DRWAV_FALSE);
}
if (framesRead != framesToRead) {
return DRWAV_FALSE;
}
offsetInFrames -= framesRead;
}
}
} else {
drwav_uint64 totalSizeInBytes;
drwav_uint64 currentBytePos;
drwav_uint64 targetBytePos;
drwav_uint64 offset;
drwav_uint32 bytesPerFrame;
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return DRWAV_FALSE;
}
totalSizeInBytes = pWav->totalPCMFrameCount * bytesPerFrame;
DRWAV_ASSERT(totalSizeInBytes >= pWav->bytesRemaining);
currentBytePos = totalSizeInBytes - pWav->bytesRemaining;
targetBytePos = targetFrameIndex * bytesPerFrame;
if (currentBytePos < targetBytePos) {
offset = (targetBytePos - currentBytePos);
} else {
if (!drwav_seek_to_first_pcm_frame(pWav)) {
return DRWAV_FALSE;
}
offset = targetBytePos;
}
while (offset > 0) {
int offset32 = ((offset > INT_MAX) ? INT_MAX : (int)offset);
if (!pWav->onSeek(pWav->pUserData, offset32, drwav_seek_origin_current)) {
return DRWAV_FALSE;
}
pWav->readCursorInPCMFrames += offset32 / bytesPerFrame;
pWav->bytesRemaining -= offset32;
offset -= offset32;
}
}
return DRWAV_TRUE;
}
DRWAV_API drwav_result drwav_get_cursor_in_pcm_frames(drwav* pWav, drwav_uint64* pCursor)
{
if (pCursor == NULL) {
return DRWAV_INVALID_ARGS;
}
*pCursor = 0;
if (pWav == NULL) {
return DRWAV_INVALID_ARGS;
}
*pCursor = pWav->readCursorInPCMFrames;
return DRWAV_SUCCESS;
}
DRWAV_API drwav_result drwav_get_length_in_pcm_frames(drwav* pWav, drwav_uint64* pLength)
{
if (pLength == NULL) {
return DRWAV_INVALID_ARGS;
}
*pLength = 0;
if (pWav == NULL) {
return DRWAV_INVALID_ARGS;
}
*pLength = pWav->totalPCMFrameCount;
return DRWAV_SUCCESS;
}
DRWAV_API size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData)
{
size_t bytesWritten;
if (pWav == NULL || bytesToWrite == 0 || pData == NULL) {
return 0;
}
bytesWritten = pWav->onWrite(pWav->pUserData, pData, bytesToWrite);
pWav->dataChunkDataSize += bytesWritten;
return bytesWritten;
}
DRWAV_API drwav_uint64 drwav_write_pcm_frames_le(drwav* pWav, drwav_uint64 framesToWrite, const void* pData)
{
drwav_uint64 bytesToWrite;
drwav_uint64 bytesWritten;
const drwav_uint8* pRunningData;
if (pWav == NULL || framesToWrite == 0 || pData == NULL) {
return 0;
}
bytesToWrite = ((framesToWrite * pWav->channels * pWav->bitsPerSample) / 8);
if (bytesToWrite > DRWAV_SIZE_MAX) {
return 0;
}
bytesWritten = 0;
pRunningData = (const drwav_uint8*)pData;
while (bytesToWrite > 0) {
size_t bytesJustWritten;
drwav_uint64 bytesToWriteThisIteration;
bytesToWriteThisIteration = bytesToWrite;
DRWAV_ASSERT(bytesToWriteThisIteration <= DRWAV_SIZE_MAX);
bytesJustWritten = drwav_write_raw(pWav, (size_t)bytesToWriteThisIteration, pRunningData);
if (bytesJustWritten == 0) {
break;
}
bytesToWrite -= bytesJustWritten;
bytesWritten += bytesJustWritten;
pRunningData += bytesJustWritten;
}
return (bytesWritten * 8) / pWav->bitsPerSample / pWav->channels;
}
DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 framesToWrite, const void* pData)
{
drwav_uint64 bytesToWrite;
drwav_uint64 bytesWritten;
drwav_uint32 bytesPerSample;
const drwav_uint8* pRunningData;
if (pWav == NULL || framesToWrite == 0 || pData == NULL) {
return 0;
}
bytesToWrite = ((framesToWrite * pWav->channels * pWav->bitsPerSample) / 8);
if (bytesToWrite > DRWAV_SIZE_MAX) {
return 0;
}
bytesWritten = 0;
pRunningData = (const drwav_uint8*)pData;
bytesPerSample = drwav_get_bytes_per_pcm_frame(pWav) / pWav->channels;
if (bytesPerSample == 0) {
return 0;
}
while (bytesToWrite > 0) {
drwav_uint8 temp[4096];
drwav_uint32 sampleCount;
size_t bytesJustWritten;
drwav_uint64 bytesToWriteThisIteration;
bytesToWriteThisIteration = bytesToWrite;
DRWAV_ASSERT(bytesToWriteThisIteration <= DRWAV_SIZE_MAX);
sampleCount = sizeof(temp)/bytesPerSample;
if (bytesToWriteThisIteration > ((drwav_uint64)sampleCount)*bytesPerSample) {
bytesToWriteThisIteration = ((drwav_uint64)sampleCount)*bytesPerSample;
}
DRWAV_COPY_MEMORY(temp, pRunningData, (size_t)bytesToWriteThisIteration);
drwav__bswap_samples(temp, sampleCount, bytesPerSample, pWav->translatedFormatTag);
bytesJustWritten = drwav_write_raw(pWav, (size_t)bytesToWriteThisIteration, temp);
if (bytesJustWritten == 0) {
break;
}
bytesToWrite -= bytesJustWritten;
bytesWritten += bytesJustWritten;
pRunningData += bytesJustWritten;
}
return (bytesWritten * 8) / pWav->bitsPerSample / pWav->channels;
}
DRWAV_API drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData)
{
if (drwav__is_little_endian()) {
return drwav_write_pcm_frames_le(pWav, framesToWrite, pData);
} else {
return drwav_write_pcm_frames_be(pWav, framesToWrite, pData);
}
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__msadpcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
drwav_uint64 totalFramesRead = 0;
DRWAV_ASSERT(pWav != NULL);
DRWAV_ASSERT(framesToRead > 0);
while (pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) {
DRWAV_ASSERT(framesToRead > 0);
if (pWav->msadpcm.cachedFrameCount == 0 && pWav->msadpcm.bytesRemainingInBlock == 0) {
if (pWav->channels == 1) {
drwav_uint8 header[7];
if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) {
return totalFramesRead;
}
pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header);
pWav->msadpcm.predictor[0] = header[0];
pWav->msadpcm.delta[0] = drwav_bytes_to_s16(header + 1);
pWav->msadpcm.prevFrames[0][1] = (drwav_int32)drwav_bytes_to_s16(header + 3);
pWav->msadpcm.prevFrames[0][0] = (drwav_int32)drwav_bytes_to_s16(header + 5);
pWav->msadpcm.cachedFrames[2] = pWav->msadpcm.prevFrames[0][0];
pWav->msadpcm.cachedFrames[3] = pWav->msadpcm.prevFrames[0][1];
pWav->msadpcm.cachedFrameCount = 2;
} else {
drwav_uint8 header[14];
if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) {
return totalFramesRead;
}
pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header);
pWav->msadpcm.predictor[0] = header[0];
pWav->msadpcm.predictor[1] = header[1];
pWav->msadpcm.delta[0] = drwav_bytes_to_s16(header + 2);
pWav->msadpcm.delta[1] = drwav_bytes_to_s16(header + 4);
pWav->msadpcm.prevFrames[0][1] = (drwav_int32)drwav_bytes_to_s16(header + 6);
pWav->msadpcm.prevFrames[1][1] = (drwav_int32)drwav_bytes_to_s16(header + 8);
pWav->msadpcm.prevFrames[0][0] = (drwav_int32)drwav_bytes_to_s16(header + 10);
pWav->msadpcm.prevFrames[1][0] = (drwav_int32)drwav_bytes_to_s16(header + 12);
pWav->msadpcm.cachedFrames[0] = pWav->msadpcm.prevFrames[0][0];
pWav->msadpcm.cachedFrames[1] = pWav->msadpcm.prevFrames[1][0];
pWav->msadpcm.cachedFrames[2] = pWav->msadpcm.prevFrames[0][1];
pWav->msadpcm.cachedFrames[3] = pWav->msadpcm.prevFrames[1][1];
pWav->msadpcm.cachedFrameCount = 2;
}
}
while (framesToRead > 0 && pWav->msadpcm.cachedFrameCount > 0 && pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) {
if (pBufferOut != NULL) {
drwav_uint32 iSample = 0;
for (iSample = 0; iSample < pWav->channels; iSample += 1) {
pBufferOut[iSample] = (drwav_int16)pWav->msadpcm.cachedFrames[(drwav_countof(pWav->msadpcm.cachedFrames) - (pWav->msadpcm.cachedFrameCount*pWav->channels)) + iSample];
}
pBufferOut += pWav->channels;
}
framesToRead -= 1;
totalFramesRead += 1;
pWav->readCursorInPCMFrames += 1;
pWav->msadpcm.cachedFrameCount -= 1;
}
if (framesToRead == 0) {
break;
}
if (pWav->msadpcm.cachedFrameCount == 0) {
if (pWav->msadpcm.bytesRemainingInBlock == 0) {
continue;
} else {
static drwav_int32 adaptationTable[] = {
230, 230, 230, 230, 307, 409, 512, 614,
768, 614, 512, 409, 307, 230, 230, 230
};
static drwav_int32 coeff1Table[] = { 256, 512, 0, 192, 240, 460, 392 };
static drwav_int32 coeff2Table[] = { 0, -256, 0, 64, 0, -208, -232 };
drwav_uint8 nibbles;
drwav_int32 nibble0;
drwav_int32 nibble1;
if (pWav->onRead(pWav->pUserData, &nibbles, 1) != 1) {
return totalFramesRead;
}
pWav->msadpcm.bytesRemainingInBlock -= 1;
nibble0 = ((nibbles & 0xF0) >> 4); if ((nibbles & 0x80)) { nibble0 |= 0xFFFFFFF0UL; }
nibble1 = ((nibbles & 0x0F) >> 0); if ((nibbles & 0x08)) { nibble1 |= 0xFFFFFFF0UL; }
if (pWav->channels == 1) {
drwav_int32 newSample0;
drwav_int32 newSample1;
newSample0 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8;
newSample0 += nibble0 * pWav->msadpcm.delta[0];
newSample0 = drwav_clamp(newSample0, -32768, 32767);
pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8;
if (pWav->msadpcm.delta[0] < 16) {
pWav->msadpcm.delta[0] = 16;
}
pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1];
pWav->msadpcm.prevFrames[0][1] = newSample0;
newSample1 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8;
newSample1 += nibble1 * pWav->msadpcm.delta[0];
newSample1 = drwav_clamp(newSample1, -32768, 32767);
pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[0]) >> 8;
if (pWav->msadpcm.delta[0] < 16) {
pWav->msadpcm.delta[0] = 16;
}
pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1];
pWav->msadpcm.prevFrames[0][1] = newSample1;
pWav->msadpcm.cachedFrames[2] = newSample0;
pWav->msadpcm.cachedFrames[3] = newSample1;
pWav->msadpcm.cachedFrameCount = 2;
} else {
drwav_int32 newSample0;
drwav_int32 newSample1;
newSample0 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8;
newSample0 += nibble0 * pWav->msadpcm.delta[0];
newSample0 = drwav_clamp(newSample0, -32768, 32767);
pWav->msadpcm.delta[0] = (adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8;
if (pWav->msadpcm.delta[0] < 16) {
pWav->msadpcm.delta[0] = 16;
}
pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1];
pWav->msadpcm.prevFrames[0][1] = newSample0;
newSample1 = ((pWav->msadpcm.prevFrames[1][1] * coeff1Table[pWav->msadpcm.predictor[1]]) + (pWav->msadpcm.prevFrames[1][0] * coeff2Table[pWav->msadpcm.predictor[1]])) >> 8;
newSample1 += nibble1 * pWav->msadpcm.delta[1];
newSample1 = drwav_clamp(newSample1, -32768, 32767);
pWav->msadpcm.delta[1] = (adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[1]) >> 8;
if (pWav->msadpcm.delta[1] < 16) {
pWav->msadpcm.delta[1] = 16;
}
pWav->msadpcm.prevFrames[1][0] = pWav->msadpcm.prevFrames[1][1];
pWav->msadpcm.prevFrames[1][1] = newSample1;
pWav->msadpcm.cachedFrames[2] = newSample0;
pWav->msadpcm.cachedFrames[3] = newSample1;
pWav->msadpcm.cachedFrameCount = 1;
}
}
}
}
return totalFramesRead;
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ima(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
drwav_uint64 totalFramesRead = 0;
drwav_uint32 iChannel;
static drwav_int32 indexTable[16] = {
-1, -1, -1, -1, 2, 4, 6, 8,
-1, -1, -1, -1, 2, 4, 6, 8
};
static drwav_int32 stepTable[89] = {
7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
};
DRWAV_ASSERT(pWav != NULL);
DRWAV_ASSERT(framesToRead > 0);
while (pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) {
DRWAV_ASSERT(framesToRead > 0);
if (pWav->ima.cachedFrameCount == 0 && pWav->ima.bytesRemainingInBlock == 0) {
if (pWav->channels == 1) {
drwav_uint8 header[4];
if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) {
return totalFramesRead;
}
pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header);
if (header[2] >= drwav_countof(stepTable)) {
pWav->onSeek(pWav->pUserData, pWav->ima.bytesRemainingInBlock, drwav_seek_origin_current);
pWav->ima.bytesRemainingInBlock = 0;
return totalFramesRead;
}
pWav->ima.predictor[0] = drwav_bytes_to_s16(header + 0);
pWav->ima.stepIndex[0] = drwav_clamp(header[2], 0, (drwav_int32)drwav_countof(stepTable)-1);
pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 1] = pWav->ima.predictor[0];
pWav->ima.cachedFrameCount = 1;
} else {
drwav_uint8 header[8];
if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) {
return totalFramesRead;
}
pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header);
if (header[2] >= drwav_countof(stepTable) || header[6] >= drwav_countof(stepTable)) {
pWav->onSeek(pWav->pUserData, pWav->ima.bytesRemainingInBlock, drwav_seek_origin_current);
pWav->ima.bytesRemainingInBlock = 0;
return totalFramesRead;
}
pWav->ima.predictor[0] = drwav_bytes_to_s16(header + 0);
pWav->ima.stepIndex[0] = drwav_clamp(header[2], 0, (drwav_int32)drwav_countof(stepTable)-1);
pWav->ima.predictor[1] = drwav_bytes_to_s16(header + 4);
pWav->ima.stepIndex[1] = drwav_clamp(header[6], 0, (drwav_int32)drwav_countof(stepTable)-1);
pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 2] = pWav->ima.predictor[0];
pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 1] = pWav->ima.predictor[1];
pWav->ima.cachedFrameCount = 1;
}
}
while (framesToRead > 0 && pWav->ima.cachedFrameCount > 0 && pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) {
if (pBufferOut != NULL) {
drwav_uint32 iSample;
for (iSample = 0; iSample < pWav->channels; iSample += 1) {
pBufferOut[iSample] = (drwav_int16)pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + iSample];
}
pBufferOut += pWav->channels;
}
framesToRead -= 1;
totalFramesRead += 1;
pWav->readCursorInPCMFrames += 1;
pWav->ima.cachedFrameCount -= 1;
}
if (framesToRead == 0) {
break;
}
if (pWav->ima.cachedFrameCount == 0) {
if (pWav->ima.bytesRemainingInBlock == 0) {
continue;
} else {
pWav->ima.cachedFrameCount = 8;
for (iChannel = 0; iChannel < pWav->channels; ++iChannel) {
drwav_uint32 iByte;
drwav_uint8 nibbles[4];
if (pWav->onRead(pWav->pUserData, &nibbles, 4) != 4) {
pWav->ima.cachedFrameCount = 0;
return totalFramesRead;
}
pWav->ima.bytesRemainingInBlock -= 4;
for (iByte = 0; iByte < 4; ++iByte) {
drwav_uint8 nibble0 = ((nibbles[iByte] & 0x0F) >> 0);
drwav_uint8 nibble1 = ((nibbles[iByte] & 0xF0) >> 4);
drwav_int32 step = stepTable[pWav->ima.stepIndex[iChannel]];
drwav_int32 predictor = pWav->ima.predictor[iChannel];
drwav_int32 diff = step >> 3;
if (nibble0 & 1) diff += step >> 2;
if (nibble0 & 2) diff += step >> 1;
if (nibble0 & 4) diff += step;
if (nibble0 & 8) diff = -diff;
predictor = drwav_clamp(predictor + diff, -32768, 32767);
pWav->ima.predictor[iChannel] = predictor;
pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble0], 0, (drwav_int32)drwav_countof(stepTable)-1);
pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + (iByte*2+0)*pWav->channels + iChannel] = predictor;
step = stepTable[pWav->ima.stepIndex[iChannel]];
predictor = pWav->ima.predictor[iChannel];
diff = step >> 3;
if (nibble1 & 1) diff += step >> 2;
if (nibble1 & 2) diff += step >> 1;
if (nibble1 & 4) diff += step;
if (nibble1 & 8) diff = -diff;
predictor = drwav_clamp(predictor + diff, -32768, 32767);
pWav->ima.predictor[iChannel] = predictor;
pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble1], 0, (drwav_int32)drwav_countof(stepTable)-1);
pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + (iByte*2+1)*pWav->channels + iChannel] = predictor;
}
}
}
}
}
return totalFramesRead;
}
#ifndef DR_WAV_NO_CONVERSION_API
static unsigned short g_drwavAlawTable[256] = {
0xEA80, 0xEB80, 0xE880, 0xE980, 0xEE80, 0xEF80, 0xEC80, 0xED80, 0xE280, 0xE380, 0xE080, 0xE180, 0xE680, 0xE780, 0xE480, 0xE580,
0xF540, 0xF5C0, 0xF440, 0xF4C0, 0xF740, 0xF7C0, 0xF640, 0xF6C0, 0xF140, 0xF1C0, 0xF040, 0xF0C0, 0xF340, 0xF3C0, 0xF240, 0xF2C0,
0xAA00, 0xAE00, 0xA200, 0xA600, 0xBA00, 0xBE00, 0xB200, 0xB600, 0x8A00, 0x8E00, 0x8200, 0x8600, 0x9A00, 0x9E00, 0x9200, 0x9600,
0xD500, 0xD700, 0xD100, 0xD300, 0xDD00, 0xDF00, 0xD900, 0xDB00, 0xC500, 0xC700, 0xC100, 0xC300, 0xCD00, 0xCF00, 0xC900, 0xCB00,
0xFEA8, 0xFEB8, 0xFE88, 0xFE98, 0xFEE8, 0xFEF8, 0xFEC8, 0xFED8, 0xFE28, 0xFE38, 0xFE08, 0xFE18, 0xFE68, 0xFE78, 0xFE48, 0xFE58,
0xFFA8, 0xFFB8, 0xFF88, 0xFF98, 0xFFE8, 0xFFF8, 0xFFC8, 0xFFD8, 0xFF28, 0xFF38, 0xFF08, 0xFF18, 0xFF68, 0xFF78, 0xFF48, 0xFF58,
0xFAA0, 0xFAE0, 0xFA20, 0xFA60, 0xFBA0, 0xFBE0, 0xFB20, 0xFB60, 0xF8A0, 0xF8E0, 0xF820, 0xF860, 0xF9A0, 0xF9E0, 0xF920, 0xF960,
0xFD50, 0xFD70, 0xFD10, 0xFD30, 0xFDD0, 0xFDF0, 0xFD90, 0xFDB0, 0xFC50, 0xFC70, 0xFC10, 0xFC30, 0xFCD0, 0xFCF0, 0xFC90, 0xFCB0,
0x1580, 0x1480, 0x1780, 0x1680, 0x1180, 0x1080, 0x1380, 0x1280, 0x1D80, 0x1C80, 0x1F80, 0x1E80, 0x1980, 0x1880, 0x1B80, 0x1A80,
0x0AC0, 0x0A40, 0x0BC0, 0x0B40, 0x08C0, 0x0840, 0x09C0, 0x0940, 0x0EC0, 0x0E40, 0x0FC0, 0x0F40, 0x0CC0, 0x0C40, 0x0DC0, 0x0D40,
0x5600, 0x5200, 0x5E00, 0x5A00, 0x4600, 0x4200, 0x4E00, 0x4A00, 0x7600, 0x7200, 0x7E00, 0x7A00, 0x6600, 0x6200, 0x6E00, 0x6A00,
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
};
static DRWAV_INLINE drwav_int16 drwav__alaw_to_s16(drwav_uint8 sampleIn)
{
return (short)g_drwavAlawTable[sampleIn];
}
static DRWAV_INLINE drwav_int16 drwav__mulaw_to_s16(drwav_uint8 sampleIn)
{
return (short)g_drwavMulawTable[sampleIn];
}
DRWAV_PRIVATE void drwav__pcm_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample)
{
size_t i;
if (bytesPerSample == 1) {
drwav_u8_to_s16(pOut, pIn, totalSampleCount);
return;
}
if (bytesPerSample == 2) {
for (i = 0; i < totalSampleCount; ++i) {
*pOut++ = ((const drwav_int16*)pIn)[i];
}
return;
}
if (bytesPerSample == 3) {
drwav_s24_to_s16(pOut, pIn, totalSampleCount);
return;
}
if (bytesPerSample == 4) {
drwav_s32_to_s16(pOut, (const drwav_int32*)pIn, totalSampleCount);
return;
}
if (bytesPerSample > 8) {
DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut));
return;
}
for (i = 0; i < totalSampleCount; ++i) {
drwav_uint64 sample = 0;
unsigned int shift = (8 - bytesPerSample) * 8;
unsigned int j;
for (j = 0; j < bytesPerSample; j += 1) {
DRWAV_ASSERT(j < 8);
sample |= (drwav_uint64)(pIn[j]) << shift;
shift += 8;
}
pIn += j;
*pOut++ = (drwav_int16)((drwav_int64)sample >> 48);
}
}
DRWAV_PRIVATE void drwav__ieee_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample)
{
if (bytesPerSample == 4) {
drwav_f32_to_s16(pOut, (const float*)pIn, totalSampleCount);
return;
} else if (bytesPerSample == 8) {
drwav_f64_to_s16(pOut, (const double*)pIn, totalSampleCount);
return;
} else {
DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut));
return;
}
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__pcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
drwav_uint32 bytesPerSample;
drwav_uint64 samplesRead;
if ((pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 16) || pBufferOut == NULL) {
return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut);
}
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesPerSample = bytesPerFrame / pWav->channels;
if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
return 0;
}
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
samplesRead = framesRead * pWav->channels;
if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
DRWAV_ASSERT(DRWAV_FALSE);
break;
}
drwav__pcm_to_s16(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ieee(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
drwav_uint32 bytesPerSample;
drwav_uint64 samplesRead;
if (pBufferOut == NULL) {
return drwav_read_pcm_frames(pWav, framesToRead, NULL);
}
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesPerSample = bytesPerFrame / pWav->channels;
if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
return 0;
}
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
samplesRead = framesRead * pWav->channels;
if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
DRWAV_ASSERT(DRWAV_FALSE);
break;
}
drwav__ieee_to_s16(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__alaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
drwav_uint32 bytesPerSample;
drwav_uint64 samplesRead;
if (pBufferOut == NULL) {
return drwav_read_pcm_frames(pWav, framesToRead, NULL);
}
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesPerSample = bytesPerFrame / pWav->channels;
if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
return 0;
}
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
samplesRead = framesRead * pWav->channels;
if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
DRWAV_ASSERT(DRWAV_FALSE);
break;
}
drwav_alaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead);
pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__mulaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
drwav_uint32 bytesPerSample;
drwav_uint64 samplesRead;
if (pBufferOut == NULL) {
return drwav_read_pcm_frames(pWav, framesToRead, NULL);
}
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesPerSample = bytesPerFrame / pWav->channels;
if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
return 0;
}
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
samplesRead = framesRead * pWav->channels;
if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
DRWAV_ASSERT(DRWAV_FALSE);
break;
}
drwav_mulaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead);
pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
if (pWav == NULL || framesToRead == 0) {
return 0;
}
if (pBufferOut == NULL) {
return drwav_read_pcm_frames(pWav, framesToRead, NULL);
}
if (framesToRead * pWav->channels * sizeof(drwav_int16) > DRWAV_SIZE_MAX) {
framesToRead = DRWAV_SIZE_MAX / sizeof(drwav_int16) / pWav->channels;
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) {
return drwav_read_pcm_frames_s16__pcm(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) {
return drwav_read_pcm_frames_s16__ieee(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) {
return drwav_read_pcm_frames_s16__alaw(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) {
return drwav_read_pcm_frames_s16__mulaw(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) {
return drwav_read_pcm_frames_s16__msadpcm(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
return drwav_read_pcm_frames_s16__ima(pWav, framesToRead, pBufferOut);
}
return 0;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16le(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToRead, pBufferOut);
if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) {
drwav__bswap_samples_s16(pBufferOut, framesRead*pWav->channels);
}
return framesRead;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16be(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut)
{
drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToRead, pBufferOut);
if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) {
drwav__bswap_samples_s16(pBufferOut, framesRead*pWav->channels);
}
return framesRead;
}
DRWAV_API void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount)
{
int r;
size_t i;
for (i = 0; i < sampleCount; ++i) {
int x = pIn[i];
r = x << 8;
r = r - 32768;
pOut[i] = (short)r;
}
}
DRWAV_API void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount)
{
int r;
size_t i;
for (i = 0; i < sampleCount; ++i) {
int x = ((int)(((unsigned int)(((const drwav_uint8*)pIn)[i*3+0]) << 8) | ((unsigned int)(((const drwav_uint8*)pIn)[i*3+1]) << 16) | ((unsigned int)(((const drwav_uint8*)pIn)[i*3+2])) << 24)) >> 8;
r = x >> 8;
pOut[i] = (short)r;
}
}
DRWAV_API void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount)
{
int r;
size_t i;
for (i = 0; i < sampleCount; ++i) {
int x = pIn[i];
r = x >> 16;
pOut[i] = (short)r;
}
}
DRWAV_API void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount)
{
int r;
size_t i;
for (i = 0; i < sampleCount; ++i) {
float x = pIn[i];
float c;
c = ((x < -1) ? -1 : ((x > 1) ? 1 : x));
c = c + 1;
r = (int)(c * 32767.5f);
r = r - 32768;
pOut[i] = (short)r;
}
}
DRWAV_API void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount)
{
int r;
size_t i;
for (i = 0; i < sampleCount; ++i) {
double x = pIn[i];
double c;
c = ((x < -1) ? -1 : ((x > 1) ? 1 : x));
c = c + 1;
r = (int)(c * 32767.5);
r = r - 32768;
pOut[i] = (short)r;
}
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
DRWAV_API void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount)
{
size_t i;
for (i = 0; i < sampleCount; ++i) {
pOut[i] = drwav__mulaw_to_s16(pIn[i]);
}
}
DRWAV_PRIVATE void drwav__pcm_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount, unsigned int bytesPerSample)
{
unsigned int i;
if (bytesPerSample == 1) {
drwav_u8_to_f32(pOut, pIn, sampleCount);
return;
}
if (bytesPerSample == 2) {
drwav_s16_to_f32(pOut, (const drwav_int16*)pIn, sampleCount);
return;
}
if (bytesPerSample == 3) {
drwav_s24_to_f32(pOut, pIn, sampleCount);
return;
}
if (bytesPerSample == 4) {
drwav_s32_to_f32(pOut, (const drwav_int32*)pIn, sampleCount);
return;
}
if (bytesPerSample > 8) {
DRWAV_ZERO_MEMORY(pOut, sampleCount * sizeof(*pOut));
return;
}
for (i = 0; i < sampleCount; ++i) {
drwav_uint64 sample = 0;
unsigned int shift = (8 - bytesPerSample) * 8;
unsigned int j;
for (j = 0; j < bytesPerSample; j += 1) {
DRWAV_ASSERT(j < 8);
sample |= (drwav_uint64)(pIn[j]) << shift;
shift += 8;
}
pIn += j;
*pOut++ = (float)((drwav_int64)sample / 9223372036854775807.0);
}
}
DRWAV_PRIVATE void drwav__ieee_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount, unsigned int bytesPerSample)
{
if (bytesPerSample == 4) {
unsigned int i;
for (i = 0; i < sampleCount; ++i) {
*pOut++ = ((const float*)pIn)[i];
}
return;
} else if (bytesPerSample == 8) {
drwav_f64_to_f32(pOut, (const double*)pIn, sampleCount);
return;
} else {
DRWAV_ZERO_MEMORY(pOut, sampleCount * sizeof(*pOut));
return;
}
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__pcm(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
drwav_uint32 bytesPerSample;
drwav_uint64 samplesRead;
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesPerSample = bytesPerFrame / pWav->channels;
if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
return 0;
}
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
samplesRead = framesRead * pWav->channels;
if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
DRWAV_ASSERT(DRWAV_FALSE);
break;
}
drwav__pcm_to_f32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__msadpcm_ima(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_int16 samples16[2048];
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels);
drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToReadThisIteration, samples16);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
drwav_s16_to_f32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels));
pBufferOut += framesRead*pWav->channels;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__ieee(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
drwav_uint32 bytesPerSample;
drwav_uint64 samplesRead;
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT && pWav->bitsPerSample == 32) {
return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut);
}
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesPerSample = bytesPerFrame / pWav->channels;
if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
return 0;
}
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
samplesRead = framesRead * pWav->channels;
if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
DRWAV_ASSERT(DRWAV_FALSE);
break;
}
drwav__ieee_to_f32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__alaw(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
drwav_uint32 bytesPerSample;
drwav_uint64 samplesRead;
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesPerSample = bytesPerFrame / pWav->channels;
if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
return 0;
}
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
samplesRead = framesRead * pWav->channels;
if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
DRWAV_ASSERT(DRWAV_FALSE);
break;
}
drwav_alaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead);
pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__mulaw(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
drwav_uint32 bytesPerSample;
drwav_uint64 samplesRead;
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesPerSample = bytesPerFrame / pWav->channels;
if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
return 0;
}
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
samplesRead = framesRead * pWav->channels;
if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
DRWAV_ASSERT(DRWAV_FALSE);
break;
}
drwav_mulaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead);
pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
if (pWav == NULL || framesToRead == 0) {
return 0;
}
if (pBufferOut == NULL) {
return drwav_read_pcm_frames(pWav, framesToRead, NULL);
}
if (framesToRead * pWav->channels * sizeof(float) > DRWAV_SIZE_MAX) {
framesToRead = DRWAV_SIZE_MAX / sizeof(float) / pWav->channels;
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) {
return drwav_read_pcm_frames_f32__pcm(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
return drwav_read_pcm_frames_f32__msadpcm_ima(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) {
return drwav_read_pcm_frames_f32__ieee(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) {
return drwav_read_pcm_frames_f32__alaw(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) {
return drwav_read_pcm_frames_f32__mulaw(pWav, framesToRead, pBufferOut);
}
return 0;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
drwav_uint64 framesRead = drwav_read_pcm_frames_f32(pWav, framesToRead, pBufferOut);
if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) {
drwav__bswap_samples_f32(pBufferOut, framesRead*pWav->channels);
}
return framesRead;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32be(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut)
{
drwav_uint64 framesRead = drwav_read_pcm_frames_f32(pWav, framesToRead, pBufferOut);
if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) {
drwav__bswap_samples_f32(pBufferOut, framesRead*pWav->channels);
}
return framesRead;
}
DRWAV_API void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount)
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
}
#ifdef DR_WAV_LIBSNDFILE_COMPAT
for (i = 0; i < sampleCount; ++i) {
*pOut++ = (pIn[i] / 256.0f) * 2 - 1;
}
#else
for (i = 0; i < sampleCount; ++i) {
float x = pIn[i];
x = x * 0.00784313725490196078f;
x = x - 1;
*pOut++ = x;
}
#endif
}
DRWAV_API void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount)
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
}
for (i = 0; i < sampleCount; ++i) {
*pOut++ = pIn[i] * 0.000030517578125f;
}
}
DRWAV_API void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount)
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
}
for (i = 0; i < sampleCount; ++i) {
double x;
drwav_uint32 a = ((drwav_uint32)(pIn[i*3+0]) << 8);
drwav_uint32 b = ((drwav_uint32)(pIn[i*3+1]) << 16);
drwav_uint32 c = ((drwav_uint32)(pIn[i*3+2]) << 24);
x = (double)((drwav_int32)(a | b | c) >> 8);
*pOut++ = (float)(x * 0.00000011920928955078125);
}
}
DRWAV_API void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount)
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
}
for (i = 0; i < sampleCount; ++i) {
*pOut++ = (float)(pIn[i] / 2147483648.0);
}
}
DRWAV_API void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount)
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
}
for (i = 0; i < sampleCount; ++i) {
*pOut++ = drwav__mulaw_to_s16(pIn[i]) / 32768.0f;
}
}
DRWAV_PRIVATE void drwav__pcm_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample)
{
unsigned int i;
if (bytesPerSample == 1) {
drwav_u8_to_s32(pOut, pIn, totalSampleCount);
return;
}
if (bytesPerSample == 2) {
drwav_s16_to_s32(pOut, (const drwav_int16*)pIn, totalSampleCount);
return;
}
if (bytesPerSample == 3) {
drwav_s24_to_s32(pOut, pIn, totalSampleCount);
return;
}
if (bytesPerSample == 4) {
for (i = 0; i < totalSampleCount; ++i) {
*pOut++ = ((const drwav_int32*)pIn)[i];
}
return;
}
if (bytesPerSample > 8) {
DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut));
return;
}
for (i = 0; i < totalSampleCount; ++i) {
drwav_uint64 sample = 0;
unsigned int shift = (8 - bytesPerSample) * 8;
unsigned int j;
for (j = 0; j < bytesPerSample; j += 1) {
DRWAV_ASSERT(j < 8);
sample |= (drwav_uint64)(pIn[j]) << shift;
shift += 8;
}
pIn += j;
*pOut++ = (drwav_int32)((drwav_int64)sample >> 32);
}
}
DRWAV_PRIVATE void drwav__ieee_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample)
{
if (bytesPerSample == 4) {
drwav_f32_to_s32(pOut, (const float*)pIn, totalSampleCount);
return;
} else if (bytesPerSample == 8) {
drwav_f64_to_s32(pOut, (const double*)pIn, totalSampleCount);
return;
} else {
DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut));
return;
}
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__pcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
drwav_uint32 bytesPerSample;
drwav_uint64 samplesRead;
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 32) {
return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut);
}
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesPerSample = bytesPerFrame / pWav->channels;
if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
return 0;
}
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
samplesRead = framesRead * pWav->channels;
if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
DRWAV_ASSERT(DRWAV_FALSE);
break;
}
drwav__pcm_to_s32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__msadpcm_ima(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
drwav_uint64 totalFramesRead = 0;
drwav_int16 samples16[2048];
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels);
drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToReadThisIteration, samples16);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
drwav_s16_to_s32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels));
pBufferOut += framesRead*pWav->channels;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__ieee(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
drwav_uint32 bytesPerSample;
drwav_uint64 samplesRead;
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesPerSample = bytesPerFrame / pWav->channels;
if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
return 0;
}
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
samplesRead = framesRead * pWav->channels;
if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
DRWAV_ASSERT(DRWAV_FALSE);
break;
}
drwav__ieee_to_s32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample);
pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__alaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
drwav_uint32 bytesPerSample;
drwav_uint64 samplesRead;
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesPerSample = bytesPerFrame / pWav->channels;
if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
return 0;
}
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
samplesRead = framesRead * pWav->channels;
if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
DRWAV_ASSERT(DRWAV_FALSE);
break;
}
drwav_alaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead);
pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__mulaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
drwav_uint64 totalFramesRead;
drwav_uint8 sampleData[4096] = {0};
drwav_uint32 bytesPerFrame;
drwav_uint32 bytesPerSample;
drwav_uint64 samplesRead;
bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav);
if (bytesPerFrame == 0) {
return 0;
}
bytesPerSample = bytesPerFrame / pWav->channels;
if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) {
return 0;
}
totalFramesRead = 0;
while (framesToRead > 0) {
drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame);
drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData);
if (framesRead == 0) {
break;
}
DRWAV_ASSERT(framesRead <= framesToReadThisIteration);
samplesRead = framesRead * pWav->channels;
if ((samplesRead * bytesPerSample) > sizeof(sampleData)) {
DRWAV_ASSERT(DRWAV_FALSE);
break;
}
drwav_mulaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead);
pBufferOut += samplesRead;
framesToRead -= framesRead;
totalFramesRead += framesRead;
}
return totalFramesRead;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
if (pWav == NULL || framesToRead == 0) {
return 0;
}
if (pBufferOut == NULL) {
return drwav_read_pcm_frames(pWav, framesToRead, NULL);
}
if (framesToRead * pWav->channels * sizeof(drwav_int32) > DRWAV_SIZE_MAX) {
framesToRead = DRWAV_SIZE_MAX / sizeof(drwav_int32) / pWav->channels;
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) {
return drwav_read_pcm_frames_s32__pcm(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) {
return drwav_read_pcm_frames_s32__msadpcm_ima(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) {
return drwav_read_pcm_frames_s32__ieee(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) {
return drwav_read_pcm_frames_s32__alaw(pWav, framesToRead, pBufferOut);
}
if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) {
return drwav_read_pcm_frames_s32__mulaw(pWav, framesToRead, pBufferOut);
}
return 0;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
drwav_uint64 framesRead = drwav_read_pcm_frames_s32(pWav, framesToRead, pBufferOut);
if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) {
drwav__bswap_samples_s32(pBufferOut, framesRead*pWav->channels);
}
return framesRead;
}
DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32be(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut)
{
drwav_uint64 framesRead = drwav_read_pcm_frames_s32(pWav, framesToRead, pBufferOut);
if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) {
drwav__bswap_samples_s32(pBufferOut, framesRead*pWav->channels);
}
return framesRead;
}
DRWAV_API void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount)
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
}
for (i = 0; i < sampleCount; ++i) {
*pOut++ = ((int)pIn[i] - 128) << 24;
}
}
DRWAV_API void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount)
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
}
for (i = 0; i < sampleCount; ++i) {
*pOut++ = pIn[i] << 16;
}
}
DRWAV_API void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount)
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
}
for (i = 0; i < sampleCount; ++i) {
unsigned int s0 = pIn[i*3 + 0];
unsigned int s1 = pIn[i*3 + 1];
unsigned int s2 = pIn[i*3 + 2];
drwav_int32 sample32 = (drwav_int32)((s0 << 8) | (s1 << 16) | (s2 << 24));
*pOut++ = sample32;
}
}
DRWAV_API void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount)
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
}
for (i = 0; i < sampleCount; ++i) {
*pOut++ = (drwav_int32)(2147483648.0 * pIn[i]);
}
}
DRWAV_API void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount)
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
}
for (i = 0; i < sampleCount; ++i) {
*pOut++ = (drwav_int32)(2147483648.0 * pIn[i]);
}
}
DRWAV_API void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount)
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
}
for (i = 0; i < sampleCount; ++i) {
*pOut++ = ((drwav_int32)drwav__alaw_to_s16(pIn[i])) << 16;
}
}
DRWAV_API void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount)
{
size_t i;
if (pOut == NULL || pIn == NULL) {
return;
}
for (i= 0; i < sampleCount; ++i) {
*pOut++ = ((drwav_int32)drwav__mulaw_to_s16(pIn[i])) << 16;
}
}
DRWAV_PRIVATE drwav_int16* drwav__read_pcm_frames_and_close_s16(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount)
{
drwav_uint64 sampleDataSize;
drwav_int16* pSampleData;
drwav_uint64 framesRead;
DRWAV_ASSERT(pWav != NULL);
sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(drwav_int16);
if (sampleDataSize > DRWAV_SIZE_MAX) {
drwav_uninit(pWav);
return NULL;
}
pSampleData = (drwav_int16*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks);
if (pSampleData == NULL) {
drwav_uninit(pWav);
return NULL;
}
framesRead = drwav_read_pcm_frames_s16(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData);
if (framesRead != pWav->totalPCMFrameCount) {
drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks);
drwav_uninit(pWav);
return NULL;
}
drwav_uninit(pWav);
if (sampleRate) {
*sampleRate = pWav->sampleRate;
}
if (channels) {
*channels = pWav->channels;
}
if (totalFrameCount) {
*totalFrameCount = pWav->totalPCMFrameCount;
}
return pSampleData;
}
DRWAV_PRIVATE float* drwav__read_pcm_frames_and_close_f32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount)
{
drwav_uint64 sampleDataSize;
float* pSampleData;
drwav_uint64 framesRead;
DRWAV_ASSERT(pWav != NULL);
sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(float);
if (sampleDataSize > DRWAV_SIZE_MAX) {
drwav_uninit(pWav);
return NULL;
}
pSampleData = (float*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks);
if (pSampleData == NULL) {
drwav_uninit(pWav);
return NULL;
}
framesRead = drwav_read_pcm_frames_f32(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData);
if (framesRead != pWav->totalPCMFrameCount) {
drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks);
drwav_uninit(pWav);
return NULL;
}
drwav_uninit(pWav);
if (sampleRate) {
*sampleRate = pWav->sampleRate;
}
if (channels) {
*channels = pWav->channels;
}
if (totalFrameCount) {
*totalFrameCount = pWav->totalPCMFrameCount;
}
return pSampleData;
}
DRWAV_PRIVATE drwav_int32* drwav__read_pcm_frames_and_close_s32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount)
{
drwav_uint64 sampleDataSize;
drwav_int32* pSampleData;
drwav_uint64 framesRead;
DRWAV_ASSERT(pWav != NULL);
sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(drwav_int32);
if (sampleDataSize > DRWAV_SIZE_MAX) {
drwav_uninit(pWav);
return NULL;
}
pSampleData = (drwav_int32*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks);
if (pSampleData == NULL) {
drwav_uninit(pWav);
return NULL;
}
framesRead = drwav_read_pcm_frames_s32(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData);
if (framesRead != pWav->totalPCMFrameCount) {
drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks);
drwav_uninit(pWav);
return NULL;
}
drwav_uninit(pWav);
if (sampleRate) {
*sampleRate = pWav->sampleRate;
}
if (channels) {
*channels = pWav->channels;
}
if (totalFrameCount) {
*totalFrameCount = pWav->totalPCMFrameCount;
}
return pSampleData;
}
DRWAV_API drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAl...
{
drwav wav;
if (channelsOut) {
*channelsOut = 0;
}
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (totalFrameCountOut) {
*totalFrameCountOut = 0;
}
if (!drwav_init(&wav, onRead, onSeek, pUserData, pAllocationCallbacks)) {
return NULL;
}
return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
}
DRWAV_API float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocati...
{
drwav wav;
if (channelsOut) {
*channelsOut = 0;
}
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (totalFrameCountOut) {
*totalFrameCountOut = 0;
}
if (!drwav_init(&wav, onRead, onSeek, pUserData, pAllocationCallbacks)) {
return NULL;
}
return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
}
DRWAV_API drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAl...
{
drwav wav;
if (channelsOut) {
*channelsOut = 0;
}
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (totalFrameCountOut) {
*totalFrameCountOut = 0;
}
if (!drwav_init(&wav, onRead, onSeek, pUserData, pAllocationCallbacks)) {
return NULL;
}
return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
}
#ifndef DR_WAV_NO_STDIO
DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
{
drwav wav;
if (channelsOut) {
*channelsOut = 0;
}
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (totalFrameCountOut) {
*totalFrameCountOut = 0;
}
if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) {
return NULL;
}
return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
}
DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
{
drwav wav;
if (channelsOut) {
*channelsOut = 0;
}
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (totalFrameCountOut) {
*totalFrameCountOut = 0;
}
if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) {
return NULL;
}
return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
}
DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
{
drwav wav;
if (channelsOut) {
*channelsOut = 0;
}
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (totalFrameCountOut) {
*totalFrameCountOut = 0;
}
if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) {
return NULL;
}
return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
}
DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
{
drwav wav;
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (channelsOut) {
*channelsOut = 0;
}
if (totalFrameCountOut) {
*totalFrameCountOut = 0;
}
if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) {
return NULL;
}
return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
}
DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
{
drwav wav;
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (channelsOut) {
*channelsOut = 0;
}
if (totalFrameCountOut) {
*totalFrameCountOut = 0;
}
if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) {
return NULL;
}
return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
}
DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
{
drwav wav;
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (channelsOut) {
*channelsOut = 0;
}
if (totalFrameCountOut) {
*totalFrameCountOut = 0;
}
if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) {
return NULL;
}
return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
}
#endif
DRWAV_API drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
{
drwav wav;
if (channelsOut) {
*channelsOut = 0;
}
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (totalFrameCountOut) {
*totalFrameCountOut = 0;
}
if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) {
return NULL;
}
return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
}
DRWAV_API float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
{
drwav wav;
if (channelsOut) {
*channelsOut = 0;
}
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (totalFrameCountOut) {
*totalFrameCountOut = 0;
}
if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) {
return NULL;
}
return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
}
DRWAV_API drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks)
{
drwav wav;
if (channelsOut) {
*channelsOut = 0;
}
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (totalFrameCountOut) {
*totalFrameCountOut = 0;
}
if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) {
return NULL;
}
return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut);
}
#endif
DRWAV_API void drwav_free(void* p, const drwav_allocation_callbacks* pAllocationCallbacks)
{
if (pAllocationCallbacks != NULL) {
drwav__free_from_callbacks(p, pAllocationCallbacks);
} else {
drwav__free_default(p, NULL);
}
}
DRWAV_API drwav_uint16 drwav_bytes_to_u16(const drwav_uint8* data)
{
return ((drwav_uint16)data[0] << 0) | ((drwav_uint16)data[1] << 8);
}
DRWAV_API drwav_int16 drwav_bytes_to_s16(const drwav_uint8* data)
{
return (drwav_int16)drwav_bytes_to_u16(data);
}
DRWAV_API drwav_uint32 drwav_bytes_to_u32(const drwav_uint8* data)
{
return ((drwav_uint32)data[0] << 0) | ((drwav_uint32)data[1] << 8) | ((drwav_uint32)data[2] << 16) | ((drwav_uint32)data[3] << 24);
}
DRWAV_API float drwav_bytes_to_f32(const drwav_uint8* data)
{
union {
drwav_uint32 u32;
float f32;
} value;
value.u32 = drwav_bytes_to_u32(data);
return value.f32;
}
DRWAV_API drwav_int32 drwav_bytes_to_s32(const drwav_uint8* data)
{
return (drwav_int32)drwav_bytes_to_u32(data);
}
DRWAV_API drwav_uint64 drwav_bytes_to_u64(const drwav_uint8* data)
{
return
((drwav_uint64)data[0] << 0) | ((drwav_uint64)data[1] << 8) | ((drwav_uint64)data[2] << 16) | ((drwav_uint64)data[3] << 24) |
((drwav_uint64)data[4] << 32) | ((drwav_uint64)data[5] << 40) | ((drwav_uint64)data[6] << 48) | ((drwav_uint64)data[7] << 56);
}
DRWAV_API drwav_int64 drwav_bytes_to_s64(const drwav_uint8* data)
{
return (drwav_int64)drwav_bytes_to_u64(data);
}
DRWAV_API drwav_bool32 drwav_guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16])
{
int i;
for (i = 0; i < 16; i += 1) {
if (a[i] != b[i]) {
return DRWAV_FALSE;
}
}
return DRWAV_TRUE;
}
DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b)
{
return
a[0] == b[0] &&
a[1] == b[1] &&
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
drflac_uint32 samplesInPartition;
drflac_uint32 partitionsRemaining;
DRFLAC_ASSERT(bs != NULL);
DRFLAC_ASSERT(blockSize != 0);
if (!drflac__read_uint8(bs, 2, &residualMethod)) {
return DRFLAC_FALSE;
}
if (residualMethod != DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE && residualMethod != DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2) {
return DRFLAC_FALSE;
}
if (!drflac__read_uint8(bs, 4, &partitionOrder)) {
return DRFLAC_FALSE;
}
if (partitionOrder > 8) {
return DRFLAC_FALSE;
}
if ((blockSize / (1 << partitionOrder)) <= order) {
return DRFLAC_FALSE;
}
samplesInPartition = (blockSize / (1 << partitionOrder)) - order;
partitionsRemaining = (1 << partitionOrder);
for (;;)
{
drflac_uint8 riceParam = 0;
if (residualMethod == DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE) {
if (!drflac__read_uint8(bs, 4, &riceParam)) {
return DRFLAC_FALSE;
}
if (riceParam == 15) {
riceParam = 0xFF;
}
} else if (residualMethod == DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2) {
if (!drflac__read_uint8(bs, 5, &riceParam)) {
return DRFLAC_FALSE;
}
if (riceParam == 31) {
riceParam = 0xFF;
}
}
if (riceParam != 0xFF) {
if (!drflac__read_and_seek_residual__rice(bs, samplesInPartition, riceParam)) {
return DRFLAC_FALSE;
}
} else {
drflac_uint8 unencodedBitsPerSample = 0;
if (!drflac__read_uint8(bs, 5, &unencodedBitsPerSample)) {
return DRFLAC_FALSE;
}
if (!drflac__seek_bits(bs, unencodedBitsPerSample * samplesInPartition)) {
return DRFLAC_FALSE;
}
}
if (partitionsRemaining == 1) {
break;
}
partitionsRemaining -= 1;
samplesInPartition = blockSize / (1 << partitionOrder);
}
return DRFLAC_TRUE;
}
static drflac_bool32 drflac__decode_samples__constant(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 subframeBitsPerSample, drflac_int32* pDecodedSamples)
{
drflac_uint32 i;
drflac_int32 sample;
if (!drflac__read_int32(bs, subframeBitsPerSample, &sample)) {
return DRFLAC_FALSE;
}
for (i = 0; i < blockSize; ++i) {
pDecodedSamples[i] = sample;
}
return DRFLAC_TRUE;
}
static drflac_bool32 drflac__decode_samples__verbatim(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 subframeBitsPerSample, drflac_int32* pDecodedSamples)
{
drflac_uint32 i;
for (i = 0; i < blockSize; ++i) {
drflac_int32 sample;
if (!drflac__read_int32(bs, subframeBitsPerSample, &sample)) {
return DRFLAC_FALSE;
}
pDecodedSamples[i] = sample;
}
return DRFLAC_TRUE;
}
static drflac_bool32 drflac__decode_samples__fixed(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 subframeBitsPerSample, drflac_uint8 lpcOrder, drflac_int32* pDecodedSamples)
{
drflac_uint32 i;
static drflac_int32 lpcCoefficientsTable[5][4] = {
{0, 0, 0, 0},
{1, 0, 0, 0},
{2, -1, 0, 0},
{3, -3, 1, 0},
{4, -6, 4, -1}
};
for (i = 0; i < lpcOrder; ++i) {
drflac_int32 sample;
if (!drflac__read_int32(bs, subframeBitsPerSample, &sample)) {
return DRFLAC_FALSE;
}
pDecodedSamples[i] = sample;
}
if (!drflac__decode_samples_with_residual(bs, subframeBitsPerSample, blockSize, lpcOrder, 0, 4, lpcCoefficientsTable[lpcOrder], pDecodedSamples)) {
return DRFLAC_FALSE;
}
return DRFLAC_TRUE;
}
static drflac_bool32 drflac__decode_samples__lpc(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 bitsPerSample, drflac_uint8 lpcOrder, drflac_int32* pDecodedSamples)
{
drflac_uint8 i;
drflac_uint8 lpcPrecision;
drflac_int8 lpcShift;
drflac_int32 coefficients[32];
for (i = 0; i < lpcOrder; ++i) {
drflac_int32 sample;
if (!drflac__read_int32(bs, bitsPerSample, &sample)) {
return DRFLAC_FALSE;
}
pDecodedSamples[i] = sample;
}
if (!drflac__read_uint8(bs, 4, &lpcPrecision)) {
return DRFLAC_FALSE;
}
if (lpcPrecision == 15) {
return DRFLAC_FALSE;
}
lpcPrecision += 1;
if (!drflac__read_int8(bs, 5, &lpcShift)) {
return DRFLAC_FALSE;
}
if (lpcShift < 0) {
return DRFLAC_FALSE;
}
DRFLAC_ZERO_MEMORY(coefficients, sizeof(coefficients));
for (i = 0; i < lpcOrder; ++i) {
if (!drflac__read_int32(bs, lpcPrecision, coefficients + i)) {
return DRFLAC_FALSE;
}
}
if (!drflac__decode_samples_with_residual(bs, bitsPerSample, blockSize, lpcOrder, lpcShift, lpcPrecision, coefficients, pDecodedSamples)) {
return DRFLAC_FALSE;
}
return DRFLAC_TRUE;
}
static drflac_bool32 drflac__read_next_flac_frame_header(drflac_bs* bs, drflac_uint8 streaminfoBitsPerSample, drflac_frame_header* header)
{
const drflac_uint32 sampleRateTable[12] = {0, 88200, 176400, 192000, 8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000};
const drflac_uint8 bitsPerSampleTable[8] = {0, 8, 12, (drflac_uint8)-1, 16, 20, 24, (drflac_uint8)-1};
DRFLAC_ASSERT(bs != NULL);
DRFLAC_ASSERT(header != NULL);
for (;;) {
drflac_uint8 crc8 = 0xCE;
drflac_uint8 reserved = 0;
drflac_uint8 blockingStrategy = 0;
drflac_uint8 blockSize = 0;
drflac_uint8 sampleRate = 0;
drflac_uint8 channelAssignment = 0;
drflac_uint8 bitsPerSample = 0;
drflac_bool32 isVariableBlockSize;
if (!drflac__find_and_seek_to_next_sync_code(bs)) {
return DRFLAC_FALSE;
}
if (!drflac__read_uint8(bs, 1, &reserved)) {
return DRFLAC_FALSE;
}
if (reserved == 1) {
continue;
}
crc8 = drflac_crc8(crc8, reserved, 1);
if (!drflac__read_uint8(bs, 1, &blockingStrategy)) {
return DRFLAC_FALSE;
}
crc8 = drflac_crc8(crc8, blockingStrategy, 1);
if (!drflac__read_uint8(bs, 4, &blockSize)) {
return DRFLAC_FALSE;
}
if (blockSize == 0) {
continue;
}
crc8 = drflac_crc8(crc8, blockSize, 4);
if (!drflac__read_uint8(bs, 4, &sampleRate)) {
return DRFLAC_FALSE;
}
crc8 = drflac_crc8(crc8, sampleRate, 4);
if (!drflac__read_uint8(bs, 4, &channelAssignment)) {
return DRFLAC_FALSE;
}
if (channelAssignment > 10) {
continue;
}
crc8 = drflac_crc8(crc8, channelAssignment, 4);
if (!drflac__read_uint8(bs, 3, &bitsPerSample)) {
return DRFLAC_FALSE;
}
if (bitsPerSample == 3 || bitsPerSample == 7) {
continue;
}
crc8 = drflac_crc8(crc8, bitsPerSample, 3);
if (!drflac__read_uint8(bs, 1, &reserved)) {
return DRFLAC_FALSE;
}
if (reserved == 1) {
continue;
}
crc8 = drflac_crc8(crc8, reserved, 1);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
} else if (blockSize == 6) {
if (!drflac__read_uint16(bs, 8, &header->blockSizeInPCMFrames)) {
return DRFLAC_FALSE;
}
crc8 = drflac_crc8(crc8, header->blockSizeInPCMFrames, 8);
header->blockSizeInPCMFrames += 1;
} else if (blockSize == 7) {
if (!drflac__read_uint16(bs, 16, &header->blockSizeInPCMFrames)) {
return DRFLAC_FALSE;
}
crc8 = drflac_crc8(crc8, header->blockSizeInPCMFrames, 16);
if (header->blockSizeInPCMFrames == 0xFFFF) {
return DRFLAC_FALSE;
}
header->blockSizeInPCMFrames += 1;
} else {
DRFLAC_ASSERT(blockSize >= 8);
header->blockSizeInPCMFrames = 256 * (1 << (blockSize - 8));
}
if (sampleRate <= 11) {
header->sampleRate = sampleRateTable[sampleRate];
} else if (sampleRate == 12) {
if (!drflac__read_uint32(bs, 8, &header->sampleRate)) {
return DRFLAC_FALSE;
}
crc8 = drflac_crc8(crc8, header->sampleRate, 8);
header->sampleRate *= 1000;
} else if (sampleRate == 13) {
if (!drflac__read_uint32(bs, 16, &header->sampleRate)) {
return DRFLAC_FALSE;
}
crc8 = drflac_crc8(crc8, header->sampleRate, 16);
} else if (sampleRate == 14) {
if (!drflac__read_uint32(bs, 16, &header->sampleRate)) {
return DRFLAC_FALSE;
}
crc8 = drflac_crc8(crc8, header->sampleRate, 16);
header->sampleRate *= 10;
} else {
continue;
}
header->channelAssignment = channelAssignment;
header->bitsPerSample = bitsPerSampleTable[bitsPerSample];
if (header->bitsPerSample == 0) {
header->bitsPerSample = streaminfoBitsPerSample;
}
if (header->bitsPerSample != streaminfoBitsPerSample) {
return DRFLAC_FALSE;
}
if (!drflac__read_uint8(bs, 8, &header->crc8)) {
return DRFLAC_FALSE;
}
#ifndef DR_FLAC_NO_CRC
if (header->crc8 != crc8) {
continue;
}
#endif
return DRFLAC_TRUE;
}
}
static drflac_bool32 drflac__read_subframe_header(drflac_bs* bs, drflac_subframe* pSubframe)
{
drflac_uint8 header;
int type;
if (!drflac__read_uint8(bs, 8, &header)) {
return DRFLAC_FALSE;
}
if ((header & 0x80) != 0) {
return DRFLAC_FALSE;
}
type = (header & 0x7E) >> 1;
if (type == 0) {
pSubframe->subframeType = DRFLAC_SUBFRAME_CONSTANT;
} else if (type == 1) {
pSubframe->subframeType = DRFLAC_SUBFRAME_VERBATIM;
} else {
if ((type & 0x20) != 0) {
pSubframe->subframeType = DRFLAC_SUBFRAME_LPC;
pSubframe->lpcOrder = (drflac_uint8)(type & 0x1F) + 1;
} else if ((type & 0x08) != 0) {
pSubframe->subframeType = DRFLAC_SUBFRAME_FIXED;
pSubframe->lpcOrder = (drflac_uint8)(type & 0x07);
if (pSubframe->lpcOrder > 4) {
pSubframe->subframeType = DRFLAC_SUBFRAME_RESERVED;
pSubframe->lpcOrder = 0;
}
} else {
pSubframe->subframeType = DRFLAC_SUBFRAME_RESERVED;
}
}
if (pSubframe->subframeType == DRFLAC_SUBFRAME_RESERVED) {
return DRFLAC_FALSE;
}
pSubframe->wastedBitsPerSample = 0;
if ((header & 0x01) == 1) {
unsigned int wastedBitsPerSample;
if (!drflac__seek_past_next_set_bit(bs, &wastedBitsPerSample)) {
return DRFLAC_FALSE;
}
pSubframe->wastedBitsPerSample = (drflac_uint8)wastedBitsPerSample + 1;
}
return DRFLAC_TRUE;
}
static drflac_bool32 drflac__decode_subframe(drflac_bs* bs, drflac_frame* frame, int subframeIndex, drflac_int32* pDecodedSamplesOut)
{
drflac_subframe* pSubframe;
drflac_uint32 subframeBitsPerSample;
DRFLAC_ASSERT(bs != NULL);
DRFLAC_ASSERT(frame != NULL);
pSubframe = frame->subframes + subframeIndex;
if (!drflac__read_subframe_header(bs, pSubframe)) {
return DRFLAC_FALSE;
}
subframeBitsPerSample = frame->header.bitsPerSample;
if ((frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE || frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE) && subframeIndex == 1) {
subframeBitsPerSample += 1;
} else if (frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE && subframeIndex == 0) {
subframeBitsPerSample += 1;
}
if (subframeBitsPerSample > 32) {
return DRFLAC_FALSE;
}
if (pSubframe->wastedBitsPerSample >= subframeBitsPerSample) {
return DRFLAC_FALSE;
}
subframeBitsPerSample -= pSubframe->wastedBitsPerSample;
pSubframe->pSamplesS32 = pDecodedSamplesOut;
switch (pSubframe->subframeType)
{
case DRFLAC_SUBFRAME_CONSTANT:
{
drflac__decode_samples__constant(bs, frame->header.blockSizeInPCMFrames, subframeBitsPerSample, pSubframe->pSamplesS32);
} break;
case DRFLAC_SUBFRAME_VERBATIM:
{
drflac__decode_samples__verbatim(bs, frame->header.blockSizeInPCMFrames, subframeBitsPerSample, pSubframe->pSamplesS32);
} break;
case DRFLAC_SUBFRAME_FIXED:
{
drflac__decode_samples__fixed(bs, frame->header.blockSizeInPCMFrames, subframeBitsPerSample, pSubframe->lpcOrder, pSubframe->pSamplesS32);
} break;
case DRFLAC_SUBFRAME_LPC:
{
drflac__decode_samples__lpc(bs, frame->header.blockSizeInPCMFrames, subframeBitsPerSample, pSubframe->lpcOrder, pSubframe->pSamplesS32);
} break;
default: return DRFLAC_FALSE;
}
return DRFLAC_TRUE;
}
static drflac_bool32 drflac__seek_subframe(drflac_bs* bs, drflac_frame* frame, int subframeIndex)
{
drflac_subframe* pSubframe;
drflac_uint32 subframeBitsPerSample;
DRFLAC_ASSERT(bs != NULL);
DRFLAC_ASSERT(frame != NULL);
pSubframe = frame->subframes + subframeIndex;
if (!drflac__read_subframe_header(bs, pSubframe)) {
return DRFLAC_FALSE;
}
subframeBitsPerSample = frame->header.bitsPerSample;
if ((frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE || frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE) && subframeIndex == 1) {
subframeBitsPerSample += 1;
} else if (frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE && subframeIndex == 0) {
subframeBitsPerSample += 1;
}
if (pSubframe->wastedBitsPerSample >= subframeBitsPerSample) {
return DRFLAC_FALSE;
}
subframeBitsPerSample -= pSubframe->wastedBitsPerSample;
pSubframe->pSamplesS32 = NULL;
switch (pSubframe->subframeType)
{
case DRFLAC_SUBFRAME_CONSTANT:
{
if (!drflac__seek_bits(bs, subframeBitsPerSample)) {
return DRFLAC_FALSE;
}
} break;
case DRFLAC_SUBFRAME_VERBATIM:
{
unsigned int bitsToSeek = frame->header.blockSizeInPCMFrames * subframeBitsPerSample;
if (!drflac__seek_bits(bs, bitsToSeek)) {
return DRFLAC_FALSE;
}
} break;
case DRFLAC_SUBFRAME_FIXED:
{
unsigned int bitsToSeek = pSubframe->lpcOrder * subframeBitsPerSample;
if (!drflac__seek_bits(bs, bitsToSeek)) {
return DRFLAC_FALSE;
}
if (!drflac__read_and_seek_residual(bs, frame->header.blockSizeInPCMFrames, pSubframe->lpcOrder)) {
return DRFLAC_FALSE;
}
} break;
case DRFLAC_SUBFRAME_LPC:
{
drflac_uint8 lpcPrecision;
unsigned int bitsToSeek = pSubframe->lpcOrder * subframeBitsPerSample;
if (!drflac__seek_bits(bs, bitsToSeek)) {
return DRFLAC_FALSE;
}
if (!drflac__read_uint8(bs, 4, &lpcPrecision)) {
return DRFLAC_FALSE;
}
if (lpcPrecision == 15) {
return DRFLAC_FALSE;
}
lpcPrecision += 1;
bitsToSeek = (pSubframe->lpcOrder * lpcPrecision) + 5;
if (!drflac__seek_bits(bs, bitsToSeek)) {
return DRFLAC_FALSE;
}
if (!drflac__read_and_seek_residual(bs, frame->header.blockSizeInPCMFrames, pSubframe->lpcOrder)) {
return DRFLAC_FALSE;
}
} break;
default: return DRFLAC_FALSE;
}
return DRFLAC_TRUE;
}
static DRFLAC_INLINE drflac_uint8 drflac__get_channel_count_from_channel_assignment(drflac_int8 channelAssignment)
{
drflac_uint8 lookup[] = {1, 2, 3, 4, 5, 6, 7, 8, 2, 2, 2};
DRFLAC_ASSERT(channelAssignment <= 10);
return lookup[channelAssignment];
}
static drflac_result drflac__decode_flac_frame(drflac* pFlac)
{
int channelCount;
int i;
drflac_uint8 paddingSizeInBits;
drflac_uint16 desiredCRC16;
#ifndef DR_FLAC_NO_CRC
drflac_uint16 actualCRC16;
#endif
DRFLAC_ZERO_MEMORY(pFlac->currentFLACFrame.subframes, sizeof(pFlac->currentFLACFrame.subframes));
if (pFlac->currentFLACFrame.header.blockSizeInPCMFrames > pFlac->maxBlockSizeInPCMFrames) {
return DRFLAC_ERROR;
}
channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment);
if (channelCount != (int)pFlac->channels) {
return DRFLAC_ERROR;
}
for (i = 0; i < channelCount; ++i) {
if (!drflac__decode_subframe(&pFlac->bs, &pFlac->currentFLACFrame, i, pFlac->pDecodedSamples + (pFlac->currentFLACFrame.header.blockSizeInPCMFrames * i))) {
return DRFLAC_ERROR;
}
}
paddingSizeInBits = (drflac_uint8)(DRFLAC_CACHE_L1_BITS_REMAINING(&pFlac->bs) & 7);
if (paddingSizeInBits > 0) {
drflac_uint8 padding = 0;
if (!drflac__read_uint8(&pFlac->bs, paddingSizeInBits, &padding)) {
return DRFLAC_AT_END;
}
}
#ifndef DR_FLAC_NO_CRC
actualCRC16 = drflac__flush_crc16(&pFlac->bs);
#endif
if (!drflac__read_uint16(&pFlac->bs, 16, &desiredCRC16)) {
return DRFLAC_AT_END;
}
#ifndef DR_FLAC_NO_CRC
if (actualCRC16 != desiredCRC16) {
return DRFLAC_CRC_MISMATCH;
}
#endif
pFlac->currentFLACFrame.pcmFramesRemaining = pFlac->currentFLACFrame.header.blockSizeInPCMFrames;
return DRFLAC_SUCCESS;
}
static drflac_result drflac__seek_flac_frame(drflac* pFlac)
{
int channelCount;
int i;
drflac_uint16 desiredCRC16;
#ifndef DR_FLAC_NO_CRC
drflac_uint16 actualCRC16;
#endif
channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment);
for (i = 0; i < channelCount; ++i) {
if (!drflac__seek_subframe(&pFlac->bs, &pFlac->currentFLACFrame, i)) {
return DRFLAC_ERROR;
}
}
if (!drflac__seek_bits(&pFlac->bs, DRFLAC_CACHE_L1_BITS_REMAINING(&pFlac->bs) & 7)) {
return DRFLAC_ERROR;
}
#ifndef DR_FLAC_NO_CRC
actualCRC16 = drflac__flush_crc16(&pFlac->bs);
#endif
if (!drflac__read_uint16(&pFlac->bs, 16, &desiredCRC16)) {
return DRFLAC_AT_END;
}
#ifndef DR_FLAC_NO_CRC
if (actualCRC16 != desiredCRC16) {
return DRFLAC_CRC_MISMATCH;
}
#endif
return DRFLAC_SUCCESS;
}
static drflac_bool32 drflac__read_and_decode_next_flac_frame(drflac* pFlac)
{
DRFLAC_ASSERT(pFlac != NULL);
for (;;) {
drflac_result result;
if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) {
return DRFLAC_FALSE;
}
result = drflac__decode_flac_frame(pFlac);
if (result != DRFLAC_SUCCESS) {
if (result == DRFLAC_CRC_MISMATCH) {
continue;
} else {
return DRFLAC_FALSE;
}
}
return DRFLAC_TRUE;
}
}
static void drflac__get_pcm_frame_range_of_current_flac_frame(drflac* pFlac, drflac_uint64* pFirstPCMFrame, drflac_uint64* pLastPCMFrame)
{
drflac_uint64 firstPCMFrame;
drflac_uint64 lastPCMFrame;
DRFLAC_ASSERT(pFlac != NULL);
firstPCMFrame = pFlac->currentFLACFrame.header.pcmFrameNumber;
if (firstPCMFrame == 0) {
firstPCMFrame = ((drflac_uint64)pFlac->currentFLACFrame.header.flacFrameNumber) * pFlac->maxBlockSizeInPCMFrames;
}
lastPCMFrame = firstPCMFrame + pFlac->currentFLACFrame.header.blockSizeInPCMFrames;
if (lastPCMFrame > 0) {
lastPCMFrame -= 1;
}
if (pFirstPCMFrame) {
*pFirstPCMFrame = firstPCMFrame;
}
if (pLastPCMFrame) {
*pLastPCMFrame = lastPCMFrame;
}
}
static drflac_bool32 drflac__seek_to_first_frame(drflac* pFlac)
{
drflac_bool32 result;
DRFLAC_ASSERT(pFlac != NULL);
result = drflac__seek_to_byte(&pFlac->bs, pFlac->firstFLACFramePosInBytes);
DRFLAC_ZERO_MEMORY(&pFlac->currentFLACFrame, sizeof(pFlac->currentFLACFrame));
pFlac->currentPCMFrame = 0;
return result;
}
static DRFLAC_INLINE drflac_result drflac__seek_to_next_flac_frame(drflac* pFlac)
{
DRFLAC_ASSERT(pFlac != NULL);
return drflac__seek_flac_frame(pFlac);
}
static drflac_uint64 drflac__seek_forward_by_pcm_frames(drflac* pFlac, drflac_uint64 pcmFramesToSeek)
{
drflac_uint64 pcmFramesRead = 0;
while (pcmFramesToSeek > 0) {
if (pFlac->currentFLACFrame.pcmFramesRemaining == 0) {
if (!drflac__read_and_decode_next_flac_frame(pFlac)) {
break;
}
} else {
if (pFlac->currentFLACFrame.pcmFramesRemaining > pcmFramesToSeek) {
pcmFramesRead += pcmFramesToSeek;
pFlac->currentFLACFrame.pcmFramesRemaining -= (drflac_uint32)pcmFramesToSeek;
pcmFramesToSeek = 0;
} else {
pcmFramesRead += pFlac->currentFLACFrame.pcmFramesRemaining;
pcmFramesToSeek -= pFlac->currentFLACFrame.pcmFramesRemaining;
pFlac->currentFLACFrame.pcmFramesRemaining = 0;
}
}
}
pFlac->currentPCMFrame += pcmFramesRead;
return pcmFramesRead;
}
static drflac_bool32 drflac__seek_to_pcm_frame__brute_force(drflac* pFlac, drflac_uint64 pcmFrameIndex)
{
drflac_bool32 isMidFrame = DRFLAC_FALSE;
drflac_uint64 runningPCMFrameCount;
DRFLAC_ASSERT(pFlac != NULL);
if (pcmFrameIndex >= pFlac->currentPCMFrame) {
runningPCMFrameCount = pFlac->currentPCMFrame;
if (pFlac->currentPCMFrame == 0 && pFlac->currentFLACFrame.pcmFramesRemaining == 0) {
if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) {
return DRFLAC_FALSE;
}
} else {
isMidFrame = DRFLAC_TRUE;
}
} else {
runningPCMFrameCount = 0;
if (!drflac__seek_to_first_frame(pFlac)) {
return DRFLAC_FALSE;
}
if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) {
return DRFLAC_FALSE;
}
}
for (;;) {
drflac_uint64 pcmFrameCountInThisFLACFrame;
drflac_uint64 firstPCMFrameInFLACFrame = 0;
drflac_uint64 lastPCMFrameInFLACFrame = 0;
drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &firstPCMFrameInFLACFrame, &lastPCMFrameInFLACFrame);
pcmFrameCountInThisFLACFrame = (lastPCMFrameInFLACFrame - firstPCMFrameInFLACFrame) + 1;
if (pcmFrameIndex < (runningPCMFrameCount + pcmFrameCountInThisFLACFrame)) {
drflac_uint64 pcmFramesToDecode = pcmFrameIndex - runningPCMFrameCount;
if (!isMidFrame) {
drflac_result result = drflac__decode_flac_frame(pFlac);
if (result == DRFLAC_SUCCESS) {
return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode;
} else {
if (result == DRFLAC_CRC_MISMATCH) {
goto next_iteration;
} else {
return DRFLAC_FALSE;
}
}
} else {
return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode;
}
} else {
if (!isMidFrame) {
drflac_result result = drflac__seek_to_next_flac_frame(pFlac);
if (result == DRFLAC_SUCCESS) {
runningPCMFrameCount += pcmFrameCountInThisFLACFrame;
} else {
if (result == DRFLAC_CRC_MISMATCH) {
goto next_iteration;
} else {
return DRFLAC_FALSE;
}
}
} else {
runningPCMFrameCount += pFlac->currentFLACFrame.pcmFramesRemaining;
pFlac->currentFLACFrame.pcmFramesRemaining = 0;
isMidFrame = DRFLAC_FALSE;
}
if (pcmFrameIndex == pFlac->totalPCMFrameCount && runningPCMFrameCount == pFlac->totalPCMFrameCount) {
return DRFLAC_TRUE;
}
}
next_iteration:
if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) {
return DRFLAC_FALSE;
}
}
}
#if !defined(DR_FLAC_NO_CRC)
#define DRFLAC_BINARY_SEARCH_APPROX_COMPRESSION_RATIO 0.6f
static drflac_bool32 drflac__seek_to_approximate_flac_frame_to_byte(drflac* pFlac, drflac_uint64 targetByte, drflac_uint64 rangeLo, drflac_uint64 rangeHi, drflac_uint64* pLastSuccessfulSeekOffset)
{
DRFLAC_ASSERT(pFlac != NULL);
DRFLAC_ASSERT(pLastSuccessfulSeekOffset != NULL);
DRFLAC_ASSERT(targetByte >= rangeLo);
DRFLAC_ASSERT(targetByte <= rangeHi);
*pLastSuccessfulSeekOffset = pFlac->firstFLACFramePosInBytes;
for (;;) {
drflac_uint64 lastTargetByte = targetByte;
if (!drflac__seek_to_byte(&pFlac->bs, targetByte)) {
if (targetByte == 0) {
drflac__seek_to_first_frame(pFlac);
return DRFLAC_FALSE;
}
targetByte = rangeLo + ((rangeHi - rangeLo)/2);
rangeHi = targetByte;
} else {
DRFLAC_ZERO_MEMORY(&pFlac->currentFLACFrame, sizeof(pFlac->currentFLACFrame));
#if 1
if (!drflac__read_and_decode_next_flac_frame(pFlac)) {
targetByte = rangeLo + ((rangeHi - rangeLo)/2);
rangeHi = targetByte;
} else {
break;
}
#else
if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) {
targetByte = rangeLo + ((rangeHi - rangeLo)/2);
rangeHi = targetByte;
} else {
break;
}
#endif
}
if(targetByte == lastTargetByte) {
return DRFLAC_FALSE;
}
}
drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &pFlac->currentPCMFrame, NULL);
DRFLAC_ASSERT(targetByte <= rangeHi);
*pLastSuccessfulSeekOffset = targetByte;
return DRFLAC_TRUE;
}
static drflac_bool32 drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(drflac* pFlac, drflac_uint64 offset)
{
#if 0
if (drflac__decode_flac_frame(pFlac) != DRFLAC_SUCCESS) {
if (drflac__read_and_decode_next_flac_frame(pFlac) == DRFLAC_FALSE) {
return DRFLAC_FALSE;
}
}
#endif
return drflac__seek_forward_by_pcm_frames(pFlac, offset) == offset;
}
static drflac_bool32 drflac__seek_to_pcm_frame__binary_search_internal(drflac* pFlac, drflac_uint64 pcmFrameIndex, drflac_uint64 byteRangeLo, drflac_uint64 byteRangeHi)
{
drflac_uint64 targetByte;
drflac_uint64 pcmRangeLo = pFlac->totalPCMFrameCount;
drflac_uint64 pcmRangeHi = 0;
drflac_uint64 lastSuccessfulSeekOffset = (drflac_uint64)-1;
drflac_uint64 closestSeekOffsetBeforeTargetPCMFrame = byteRangeLo;
drflac_uint32 seekForwardThreshold = (pFlac->maxBlockSizeInPCMFrames != 0) ? pFlac->maxBlockSizeInPCMFrames*2 : 4096;
targetByte = byteRangeLo + (drflac_uint64)(((drflac_int64)((pcmFrameIndex - pFlac->currentPCMFrame) * pFlac->channels * pFlac->bitsPerSample)/8.0f) * DRFLAC_BINARY_SEARCH_APPROX_COMPRESSION_RATIO);
if (targetByte > byteRangeHi) {
targetByte = byteRangeHi;
}
for (;;) {
if (drflac__seek_to_approximate_flac_frame_to_byte(pFlac, targetByte, byteRangeLo, byteRangeHi, &lastSuccessfulSeekOffset)) {
drflac_uint64 newPCMRangeLo;
drflac_uint64 newPCMRangeHi;
drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &newPCMRangeLo, &newPCMRangeHi);
if (pcmRangeLo == newPCMRangeLo) {
if (!drflac__seek_to_approximate_flac_frame_to_byte(pFlac, closestSeekOffsetBeforeTargetPCMFrame, closestSeekOffsetBeforeTargetPCMFrame, byteRangeHi, &lastSuccessfulSeekOffset)) {
break;
}
if (drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(pFlac, pcmFrameIndex - pFlac->currentPCMFrame)) {
return DRFLAC_TRUE;
} else {
break;
}
}
pcmRangeLo = newPCMRangeLo;
pcmRangeHi = newPCMRangeHi;
if (pcmRangeLo <= pcmFrameIndex && pcmRangeHi >= pcmFrameIndex) {
if (drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(pFlac, pcmFrameIndex - pFlac->currentPCMFrame) ) {
return DRFLAC_TRUE;
} else {
break;
}
} else {
const float approxCompressionRatio = (drflac_int64)(lastSuccessfulSeekOffset - pFlac->firstFLACFramePosInBytes) / ((drflac_int64)(pcmRangeLo * pFlac->channels * pFlac->bitsPerSample)/8.0f);
if (pcmRangeLo > pcmFrameIndex) {
byteRangeHi = lastSuccessfulSeekOffset;
if (byteRangeLo > byteRangeHi) {
byteRangeLo = byteRangeHi;
}
targetByte = byteRangeLo + ((byteRangeHi - byteRangeLo) / 2);
if (targetByte < byteRangeLo) {
targetByte = byteRangeLo;
}
} else {
if ((pcmFrameIndex - pcmRangeLo) < seekForwardThreshold) {
if (drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(pFlac, pcmFrameIndex - pFlac->currentPCMFrame)) {
return DRFLAC_TRUE;
} else {
break;
}
} else {
byteRangeLo = lastSuccessfulSeekOffset;
if (byteRangeHi < byteRangeLo) {
byteRangeHi = byteRangeLo;
}
targetByte = lastSuccessfulSeekOffset + (drflac_uint64)(((drflac_int64)((pcmFrameIndex-pcmRangeLo) * pFlac->channels * pFlac->bitsPerSample)/8.0f) * approxCompressionRatio);
if (targetByte > byteRangeHi) {
targetByte = byteRangeHi;
}
if (closestSeekOffsetBeforeTargetPCMFrame < lastSuccessfulSeekOffset) {
closestSeekOffsetBeforeTargetPCMFrame = lastSuccessfulSeekOffset;
}
}
}
}
} else {
break;
}
}
drflac__seek_to_first_frame(pFlac);
return DRFLAC_FALSE;
}
static drflac_bool32 drflac__seek_to_pcm_frame__binary_search(drflac* pFlac, drflac_uint64 pcmFrameIndex)
{
drflac_uint64 byteRangeLo;
drflac_uint64 byteRangeHi;
drflac_uint32 seekForwardThreshold = (pFlac->maxBlockSizeInPCMFrames != 0) ? pFlac->maxBlockSizeInPCMFrames*2 : 4096;
if (drflac__seek_to_first_frame(pFlac) == DRFLAC_FALSE) {
return DRFLAC_FALSE;
}
if (pcmFrameIndex < seekForwardThreshold) {
return drflac__seek_forward_by_pcm_frames(pFlac, pcmFrameIndex) == pcmFrameIndex;
}
byteRangeLo = pFlac->firstFLACFramePosInBytes;
byteRangeHi = pFlac->firstFLACFramePosInBytes + (drflac_uint64)((drflac_int64)(pFlac->totalPCMFrameCount * pFlac->channels * pFlac->bitsPerSample)/8.0f);
return drflac__seek_to_pcm_frame__binary_search_internal(pFlac, pcmFrameIndex, byteRangeLo, byteRangeHi);
}
#endif
static drflac_bool32 drflac__seek_to_pcm_frame__seek_table(drflac* pFlac, drflac_uint64 pcmFrameIndex)
{
drflac_uint32 iClosestSeekpoint = 0;
drflac_bool32 isMidFrame = DRFLAC_FALSE;
drflac_uint64 runningPCMFrameCount;
drflac_uint32 iSeekpoint;
DRFLAC_ASSERT(pFlac != NULL);
if (pFlac->pSeekpoints == NULL || pFlac->seekpointCount == 0) {
return DRFLAC_FALSE;
}
if (pFlac->pSeekpoints[0].firstPCMFrame > pcmFrameIndex) {
return DRFLAC_FALSE;
}
for (iSeekpoint = 0; iSeekpoint < pFlac->seekpointCount; ++iSeekpoint) {
if (pFlac->pSeekpoints[iSeekpoint].firstPCMFrame >= pcmFrameIndex) {
break;
}
iClosestSeekpoint = iSeekpoint;
}
if (pFlac->pSeekpoints[iClosestSeekpoint].pcmFrameCount == 0 || pFlac->pSeekpoints[iClosestSeekpoint].pcmFrameCount > pFlac->maxBlockSizeInPCMFrames) {
return DRFLAC_FALSE;
}
if (pFlac->pSeekpoints[iClosestSeekpoint].firstPCMFrame > pFlac->totalPCMFrameCount && pFlac->totalPCMFrameCount > 0) {
return DRFLAC_FALSE;
}
#if !defined(DR_FLAC_NO_CRC)
if (pFlac->totalPCMFrameCount > 0) {
drflac_uint64 byteRangeLo;
drflac_uint64 byteRangeHi;
byteRangeHi = pFlac->firstFLACFramePosInBytes + (drflac_uint64)((drflac_int64)(pFlac->totalPCMFrameCount * pFlac->channels * pFlac->bitsPerSample)/8.0f);
byteRangeLo = pFlac->firstFLACFramePosInBytes + pFlac->pSeekpoints[iClosestSeekpoint].flacFrameOffset;
if (iClosestSeekpoint < pFlac->seekpointCount-1) {
drflac_uint32 iNextSeekpoint = iClosestSeekpoint + 1;
if (pFlac->pSeekpoints[iClosestSeekpoint].flacFrameOffset >= pFlac->pSeekpoints[iNextSeekpoint].flacFrameOffset || pFlac->pSeekpoints[iNextSeekpoint].pcmFrameCount == 0) {
return DRFLAC_FALSE;
}
if (pFlac->pSeekpoints[iNextSeekpoint].firstPCMFrame != (((drflac_uint64)0xFFFFFFFF << 32) | 0xFFFFFFFF)) {
byteRangeHi = pFlac->firstFLACFramePosInBytes + pFlac->pSeekpoints[iNextSeekpoint].flacFrameOffset - 1;
}
}
if (drflac__seek_to_byte(&pFlac->bs, pFlac->firstFLACFramePosInBytes + pFlac->pSeekpoints[iClosestSeekpoint].flacFrameOffset)) {
if (drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) {
drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &pFlac->currentPCMFrame, NULL);
if (drflac__seek_to_pcm_frame__binary_search_internal(pFlac, pcmFrameIndex, byteRangeLo, byteRangeHi)) {
return DRFLAC_TRUE;
}
}
}
}
#endif
if (pcmFrameIndex >= pFlac->currentPCMFrame && pFlac->pSeekpoints[iClosestSeekpoint].firstPCMFrame <= pFlac->currentPCMFrame) {
runningPCMFrameCount = pFlac->currentPCMFrame;
if (pFlac->currentPCMFrame == 0 && pFlac->currentFLACFrame.pcmFramesRemaining == 0) {
if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) {
return DRFLAC_FALSE;
}
} else {
isMidFrame = DRFLAC_TRUE;
}
} else {
runningPCMFrameCount = pFlac->pSeekpoints[iClosestSeekpoint].firstPCMFrame;
if (!drflac__seek_to_byte(&pFlac->bs, pFlac->firstFLACFramePosInBytes + pFlac->pSeekpoints[iClosestSeekpoint].flacFrameOffset)) {
return DRFLAC_FALSE;
}
if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) {
return DRFLAC_FALSE;
}
}
for (;;) {
drflac_uint64 pcmFrameCountInThisFLACFrame;
drflac_uint64 firstPCMFrameInFLACFrame = 0;
drflac_uint64 lastPCMFrameInFLACFrame = 0;
drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &firstPCMFrameInFLACFrame, &lastPCMFrameInFLACFrame);
pcmFrameCountInThisFLACFrame = (lastPCMFrameInFLACFrame - firstPCMFrameInFLACFrame) + 1;
if (pcmFrameIndex < (runningPCMFrameCount + pcmFrameCountInThisFLACFrame)) {
drflac_uint64 pcmFramesToDecode = pcmFrameIndex - runningPCMFrameCount;
if (!isMidFrame) {
drflac_result result = drflac__decode_flac_frame(pFlac);
if (result == DRFLAC_SUCCESS) {
return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode;
} else {
if (result == DRFLAC_CRC_MISMATCH) {
goto next_iteration;
} else {
return DRFLAC_FALSE;
}
}
} else {
return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode;
}
} else {
if (!isMidFrame) {
drflac_result result = drflac__seek_to_next_flac_frame(pFlac);
if (result == DRFLAC_SUCCESS) {
runningPCMFrameCount += pcmFrameCountInThisFLACFrame;
} else {
if (result == DRFLAC_CRC_MISMATCH) {
goto next_iteration;
} else {
return DRFLAC_FALSE;
}
}
} else {
runningPCMFrameCount += pFlac->currentFLACFrame.pcmFramesRemaining;
pFlac->currentFLACFrame.pcmFramesRemaining = 0;
isMidFrame = DRFLAC_FALSE;
}
if (pcmFrameIndex == pFlac->totalPCMFrameCount && runningPCMFrameCount == pFlac->totalPCMFrameCount) {
return DRFLAC_TRUE;
}
}
next_iteration:
if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) {
return DRFLAC_FALSE;
}
}
}
#ifndef DR_FLAC_NO_OGG
typedef struct
{
drflac_uint8 capturePattern[4];
drflac_uint8 structureVersion;
drflac_uint8 headerType;
drflac_uint64 granulePosition;
drflac_uint32 serialNumber;
drflac_uint32 sequenceNumber;
drflac_uint32 checksum;
drflac_uint8 segmentCount;
drflac_uint8 segmentTable[255];
} drflac_ogg_page_header;
#endif
typedef struct
{
drflac_read_proc onRead;
drflac_seek_proc onSeek;
drflac_meta_proc onMeta;
drflac_container container;
void* pUserData;
void* pUserDataMD;
drflac_uint32 sampleRate;
drflac_uint8 channels;
drflac_uint8 bitsPerSample;
drflac_uint64 totalPCMFrameCount;
drflac_uint16 maxBlockSizeInPCMFrames;
drflac_uint64 runningFilePos;
drflac_bool32 hasStreamInfoBlock;
drflac_bool32 hasMetadataBlocks;
drflac_bs bs;
drflac_frame_header firstFrameHeader;
#ifndef DR_FLAC_NO_OGG
drflac_uint32 oggSerial;
drflac_uint64 oggFirstBytePos;
drflac_ogg_page_header oggBosHeader;
#endif
} drflac_init_info;
static DRFLAC_INLINE void drflac__decode_block_header(drflac_uint32 blockHeader, drflac_uint8* isLastBlock, drflac_uint8* blockType, drflac_uint32* blockSize)
{
blockHeader = drflac__be2host_32(blockHeader);
*isLastBlock = (drflac_uint8)((blockHeader & 0x80000000UL) >> 31);
*blockType = (drflac_uint8)((blockHeader & 0x7F000000UL) >> 24);
*blockSize = (blockHeader & 0x00FFFFFFUL);
}
static DRFLAC_INLINE drflac_bool32 drflac__read_and_decode_block_header(drflac_read_proc onRead, void* pUserData, drflac_uint8* isLastBlock, drflac_uint8* blockType, drflac_uint32* blockSize)
{
drflac_uint32 blockHeader;
*blockSize = 0;
if (onRead(pUserData, &blockHeader, 4) != 4) {
return DRFLAC_FALSE;
}
drflac__decode_block_header(blockHeader, isLastBlock, blockType, blockSize);
return DRFLAC_TRUE;
}
static drflac_bool32 drflac__read_streaminfo(drflac_read_proc onRead, void* pUserData, drflac_streaminfo* pStreamInfo)
{
drflac_uint32 blockSizes;
drflac_uint64 frameSizes = 0;
drflac_uint64 importantProps;
drflac_uint8 md5[16];
if (onRead(pUserData, &blockSizes, 4) != 4) {
return DRFLAC_FALSE;
}
if (onRead(pUserData, &frameSizes, 6) != 6) {
return DRFLAC_FALSE;
}
if (onRead(pUserData, &importantProps, 8) != 8) {
return DRFLAC_FALSE;
}
if (onRead(pUserData, md5, sizeof(md5)) != sizeof(md5)) {
return DRFLAC_FALSE;
}
blockSizes = drflac__be2host_32(blockSizes);
frameSizes = drflac__be2host_64(frameSizes);
importantProps = drflac__be2host_64(importantProps);
pStreamInfo->minBlockSizeInPCMFrames = (drflac_uint16)((blockSizes & 0xFFFF0000) >> 16);
pStreamInfo->maxBlockSizeInPCMFrames = (drflac_uint16) (blockSizes & 0x0000FFFF);
pStreamInfo->minFrameSizeInPCMFrames = (drflac_uint32)((frameSizes & (((drflac_uint64)0x00FFFFFF << 16) << 24)) >> 40);
pStreamInfo->maxFrameSizeInPCMFrames = (drflac_uint32)((frameSizes & (((drflac_uint64)0x00FFFFFF << 16) << 0)) >> 16);
pStreamInfo->sampleRate = (drflac_uint32)((importantProps & (((drflac_uint64)0x000FFFFF << 16) << 28)) >> 44);
pStreamInfo->channels = (drflac_uint8 )((importantProps & (((drflac_uint64)0x0000000E << 16) << 24)) >> 41) + 1;
pStreamInfo->bitsPerSample = (drflac_uint8 )((importantProps & (((drflac_uint64)0x0000001F << 16) << 20)) >> 36) + 1;
pStreamInfo->totalPCMFrameCount = ((importantProps & ((((drflac_uint64)0x0000000F << 16) << 16) | 0xFFFFFFFF)));
DRFLAC_COPY_MEMORY(pStreamInfo->md5, md5, sizeof(md5));
return DRFLAC_TRUE;
}
static void* drflac__malloc_default(size_t sz, void* pUserData)
{
(void)pUserData;
return DRFLAC_MALLOC(sz);
}
static void* drflac__realloc_default(void* p, size_t sz, void* pUserData)
{
(void)pUserData;
return DRFLAC_REALLOC(p, sz);
}
static void drflac__free_default(void* p, void* pUserData)
{
(void)pUserData;
DRFLAC_FREE(p);
}
static void* drflac__malloc_from_callbacks(size_t sz, const drflac_allocation_callbacks* pAllocationCallbacks)
{
if (pAllocationCallbacks == NULL) {
return NULL;
}
if (pAllocationCallbacks->onMalloc != NULL) {
return pAllocationCallbacks->onMalloc(sz, pAllocationCallbacks->pUserData);
}
if (pAllocationCallbacks->onRealloc != NULL) {
return pAllocationCallbacks->onRealloc(NULL, sz, pAllocationCallbacks->pUserData);
}
return NULL;
}
static void* drflac__realloc_from_callbacks(void* p, size_t szNew, size_t szOld, const drflac_allocation_callbacks* pAllocationCallbacks)
{
if (pAllocationCallbacks == NULL) {
return NULL;
}
if (pAllocationCallbacks->onRealloc != NULL) {
return pAllocationCallbacks->onRealloc(p, szNew, pAllocationCallbacks->pUserData);
}
if (pAllocationCallbacks->onMalloc != NULL && pAllocationCallbacks->onFree != NULL) {
void* p2;
p2 = pAllocationCallbacks->onMalloc(szNew, pAllocationCallbacks->pUserData);
if (p2 == NULL) {
return NULL;
}
if (p != NULL) {
DRFLAC_COPY_MEMORY(p2, p, szOld);
pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData);
}
return p2;
}
return NULL;
}
static void drflac__free_from_callbacks(void* p, const drflac_allocation_callbacks* pAllocationCallbacks)
{
if (p == NULL || pAllocationCallbacks == NULL) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
}
} break;
case DRFLAC_METADATA_BLOCK_TYPE_INVALID:
{
if (onMeta) {
if (!onSeek(pUserData, blockSize, drflac_seek_origin_current)) {
isLastBlock = DRFLAC_TRUE;
}
}
} break;
default:
{
if (onMeta) {
void* pRawData = drflac__malloc_from_callbacks(blockSize, pAllocationCallbacks);
if (pRawData == NULL) {
return DRFLAC_FALSE;
}
if (onRead(pUserData, pRawData, blockSize) != blockSize) {
drflac__free_from_callbacks(pRawData, pAllocationCallbacks);
return DRFLAC_FALSE;
}
metadata.pRawData = pRawData;
metadata.rawDataSize = blockSize;
onMeta(pUserDataMD, &metadata);
drflac__free_from_callbacks(pRawData, pAllocationCallbacks);
}
} break;
}
if (onMeta == NULL && blockSize > 0) {
if (!onSeek(pUserData, blockSize, drflac_seek_origin_current)) {
isLastBlock = DRFLAC_TRUE;
}
}
runningFilePos += blockSize;
if (isLastBlock) {
break;
}
}
*pSeektablePos = seektablePos;
*pSeektableSize = seektableSize;
*pFirstFramePos = runningFilePos;
return DRFLAC_TRUE;
}
static drflac_bool32 drflac__init_private__native(drflac_init_info* pInit, drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, void* pUserDataMD, drflac_bool32 relaxed)
{
drflac_uint8 isLastBlock;
drflac_uint8 blockType;
drflac_uint32 blockSize;
(void)onSeek;
pInit->container = drflac_container_native;
if (!drflac__read_and_decode_block_header(onRead, pUserData, &isLastBlock, &blockType, &blockSize)) {
return DRFLAC_FALSE;
}
if (blockType != DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO || blockSize != 34) {
if (!relaxed) {
return DRFLAC_FALSE;
} else {
pInit->hasStreamInfoBlock = DRFLAC_FALSE;
pInit->hasMetadataBlocks = DRFLAC_FALSE;
if (!drflac__read_next_flac_frame_header(&pInit->bs, 0, &pInit->firstFrameHeader)) {
return DRFLAC_FALSE;
}
if (pInit->firstFrameHeader.bitsPerSample == 0) {
return DRFLAC_FALSE;
}
pInit->sampleRate = pInit->firstFrameHeader.sampleRate;
pInit->channels = drflac__get_channel_count_from_channel_assignment(pInit->firstFrameHeader.channelAssignment);
pInit->bitsPerSample = pInit->firstFrameHeader.bitsPerSample;
pInit->maxBlockSizeInPCMFrames = 65535;
return DRFLAC_TRUE;
}
} else {
drflac_streaminfo streaminfo;
if (!drflac__read_streaminfo(onRead, pUserData, &streaminfo)) {
return DRFLAC_FALSE;
}
pInit->hasStreamInfoBlock = DRFLAC_TRUE;
pInit->sampleRate = streaminfo.sampleRate;
pInit->channels = streaminfo.channels;
pInit->bitsPerSample = streaminfo.bitsPerSample;
pInit->totalPCMFrameCount = streaminfo.totalPCMFrameCount;
pInit->maxBlockSizeInPCMFrames = streaminfo.maxBlockSizeInPCMFrames;
pInit->hasMetadataBlocks = !isLastBlock;
if (onMeta) {
drflac_metadata metadata;
metadata.type = DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO;
metadata.pRawData = NULL;
metadata.rawDataSize = 0;
metadata.data.streaminfo = streaminfo;
onMeta(pUserDataMD, &metadata);
}
return DRFLAC_TRUE;
}
}
#ifndef DR_FLAC_NO_OGG
#define DRFLAC_OGG_MAX_PAGE_SIZE 65307
#define DRFLAC_OGG_CAPTURE_PATTERN_CRC32 1605413199
typedef enum
{
drflac_ogg_recover_on_crc_mismatch,
drflac_ogg_fail_on_crc_mismatch
} drflac_ogg_crc_mismatch_recovery;
#ifndef DR_FLAC_NO_CRC
static drflac_uint32 drflac__crc32_table[] = {
0x00000000L, 0x04C11DB7L, 0x09823B6EL, 0x0D4326D9L,
0x130476DCL, 0x17C56B6BL, 0x1A864DB2L, 0x1E475005L,
0x2608EDB8L, 0x22C9F00FL, 0x2F8AD6D6L, 0x2B4BCB61L,
0x350C9B64L, 0x31CD86D3L, 0x3C8EA00AL, 0x384FBDBDL,
0x4C11DB70L, 0x48D0C6C7L, 0x4593E01EL, 0x4152FDA9L,
0x5F15ADACL, 0x5BD4B01BL, 0x569796C2L, 0x52568B75L,
0x6A1936C8L, 0x6ED82B7FL, 0x639B0DA6L, 0x675A1011L,
0x791D4014L, 0x7DDC5DA3L, 0x709F7B7AL, 0x745E66CDL,
0x9823B6E0L, 0x9CE2AB57L, 0x91A18D8EL, 0x95609039L,
0x8B27C03CL, 0x8FE6DD8BL, 0x82A5FB52L, 0x8664E6E5L,
0xBE2B5B58L, 0xBAEA46EFL, 0xB7A96036L, 0xB3687D81L,
0xAD2F2D84L, 0xA9EE3033L, 0xA4AD16EAL, 0xA06C0B5DL,
0xD4326D90L, 0xD0F37027L, 0xDDB056FEL, 0xD9714B49L,
0xC7361B4CL, 0xC3F706FBL, 0xCEB42022L, 0xCA753D95L,
0xF23A8028L, 0xF6FB9D9FL, 0xFBB8BB46L, 0xFF79A6F1L,
0xE13EF6F4L, 0xE5FFEB43L, 0xE8BCCD9AL, 0xEC7DD02DL,
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
return DRFLAC_FALSE;
}
}
#else
(void)recoveryMethod;
#endif
oggbs->currentPageHeader = header;
oggbs->bytesRemainingInPage = pageBodySize;
return DRFLAC_TRUE;
}
}
#if 0
static drflac_uint8 drflac_oggbs__get_current_segment_index(drflac_oggbs* oggbs, drflac_uint8* pBytesRemainingInSeg)
{
drflac_uint32 bytesConsumedInPage = drflac_ogg__get_page_body_size(&oggbs->currentPageHeader) - oggbs->bytesRemainingInPage;
drflac_uint8 iSeg = 0;
drflac_uint32 iByte = 0;
while (iByte < bytesConsumedInPage) {
drflac_uint8 segmentSize = oggbs->currentPageHeader.segmentTable[iSeg];
if (iByte + segmentSize > bytesConsumedInPage) {
break;
} else {
iSeg += 1;
iByte += segmentSize;
}
}
*pBytesRemainingInSeg = oggbs->currentPageHeader.segmentTable[iSeg] - (drflac_uint8)(bytesConsumedInPage - iByte);
return iSeg;
}
static drflac_bool32 drflac_oggbs__seek_to_next_packet(drflac_oggbs* oggbs)
{
for (;;) {
drflac_bool32 atEndOfPage = DRFLAC_FALSE;
drflac_uint8 bytesRemainingInSeg;
drflac_uint8 iFirstSeg = drflac_oggbs__get_current_segment_index(oggbs, &bytesRemainingInSeg);
drflac_uint32 bytesToEndOfPacketOrPage = bytesRemainingInSeg;
for (drflac_uint8 iSeg = iFirstSeg; iSeg < oggbs->currentPageHeader.segmentCount; ++iSeg) {
drflac_uint8 segmentSize = oggbs->currentPageHeader.segmentTable[iSeg];
if (segmentSize < 255) {
if (iSeg == oggbs->currentPageHeader.segmentCount-1) {
atEndOfPage = DRFLAC_TRUE;
}
break;
}
bytesToEndOfPacketOrPage += segmentSize;
}
drflac_oggbs__seek_physical(oggbs, bytesToEndOfPacketOrPage, drflac_seek_origin_current);
oggbs->bytesRemainingInPage -= bytesToEndOfPacketOrPage;
if (atEndOfPage) {
if (!drflac_oggbs__goto_next_page(oggbs)) {
return DRFLAC_FALSE;
}
if ((oggbs->currentPageHeader.headerType & 0x01) == 0) {
return DRFLAC_TRUE;
}
} else {
return DRFLAC_TRUE;
}
}
}
static drflac_bool32 drflac_oggbs__seek_to_next_frame(drflac_oggbs* oggbs)
{
return drflac_oggbs__seek_to_next_packet(oggbs);
}
#endif
static size_t drflac__on_read_ogg(void* pUserData, void* bufferOut, size_t bytesToRead)
{
drflac_oggbs* oggbs = (drflac_oggbs*)pUserData;
drflac_uint8* pRunningBufferOut = (drflac_uint8*)bufferOut;
size_t bytesRead = 0;
DRFLAC_ASSERT(oggbs != NULL);
DRFLAC_ASSERT(pRunningBufferOut != NULL);
while (bytesRead < bytesToRead) {
size_t bytesRemainingToRead = bytesToRead - bytesRead;
if (oggbs->bytesRemainingInPage >= bytesRemainingToRead) {
DRFLAC_COPY_MEMORY(pRunningBufferOut, oggbs->pageData + (oggbs->pageDataSize - oggbs->bytesRemainingInPage), bytesRemainingToRead);
bytesRead += bytesRemainingToRead;
oggbs->bytesRemainingInPage -= (drflac_uint32)bytesRemainingToRead;
break;
}
if (oggbs->bytesRemainingInPage > 0) {
DRFLAC_COPY_MEMORY(pRunningBufferOut, oggbs->pageData + (oggbs->pageDataSize - oggbs->bytesRemainingInPage), oggbs->bytesRemainingInPage);
bytesRead += oggbs->bytesRemainingInPage;
pRunningBufferOut += oggbs->bytesRemainingInPage;
oggbs->bytesRemainingInPage = 0;
}
DRFLAC_ASSERT(bytesRemainingToRead > 0);
if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_recover_on_crc_mismatch)) {
break;
}
}
return bytesRead;
}
static drflac_bool32 drflac__on_seek_ogg(void* pUserData, int offset, drflac_seek_origin origin)
{
drflac_oggbs* oggbs = (drflac_oggbs*)pUserData;
int bytesSeeked = 0;
DRFLAC_ASSERT(oggbs != NULL);
DRFLAC_ASSERT(offset >= 0);
if (origin == drflac_seek_origin_start) {
if (!drflac_oggbs__seek_physical(oggbs, (int)oggbs->firstBytePos, drflac_seek_origin_start)) {
return DRFLAC_FALSE;
}
if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_fail_on_crc_mismatch)) {
return DRFLAC_FALSE;
}
return drflac__on_seek_ogg(pUserData, offset, drflac_seek_origin_current);
}
DRFLAC_ASSERT(origin == drflac_seek_origin_current);
while (bytesSeeked < offset) {
int bytesRemainingToSeek = offset - bytesSeeked;
DRFLAC_ASSERT(bytesRemainingToSeek >= 0);
if (oggbs->bytesRemainingInPage >= (size_t)bytesRemainingToSeek) {
bytesSeeked += bytesRemainingToSeek;
(void)bytesSeeked;
oggbs->bytesRemainingInPage -= bytesRemainingToSeek;
break;
}
if (oggbs->bytesRemainingInPage > 0) {
bytesSeeked += (int)oggbs->bytesRemainingInPage;
oggbs->bytesRemainingInPage = 0;
}
DRFLAC_ASSERT(bytesRemainingToSeek > 0);
if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_fail_on_crc_mismatch)) {
return DRFLAC_FALSE;
}
}
return DRFLAC_TRUE;
}
static drflac_bool32 drflac_ogg__seek_to_pcm_frame(drflac* pFlac, drflac_uint64 pcmFrameIndex)
{
drflac_oggbs* oggbs = (drflac_oggbs*)pFlac->_oggbs;
drflac_uint64 originalBytePos;
drflac_uint64 runningGranulePosition;
drflac_uint64 runningFrameBytePos;
drflac_uint64 runningPCMFrameCount;
DRFLAC_ASSERT(oggbs != NULL);
originalBytePos = oggbs->currentBytePos;
if (!drflac__seek_to_byte(&pFlac->bs, pFlac->firstFLACFramePosInBytes)) {
return DRFLAC_FALSE;
}
oggbs->bytesRemainingInPage = 0;
runningGranulePosition = 0;
for (;;) {
if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_recover_on_crc_mismatch)) {
drflac_oggbs__seek_physical(oggbs, originalBytePos, drflac_seek_origin_start);
return DRFLAC_FALSE;
}
runningFrameBytePos = oggbs->currentBytePos - drflac_ogg__get_page_header_size(&oggbs->currentPageHeader) - oggbs->pageDataSize;
if (oggbs->currentPageHeader.granulePosition >= pcmFrameIndex) {
break;
}
if ((oggbs->currentPageHeader.headerType & 0x01) == 0) {
if (oggbs->currentPageHeader.segmentTable[0] >= 2) {
drflac_uint8 firstBytesInPage[2];
firstBytesInPage[0] = oggbs->pageData[0];
firstBytesInPage[1] = oggbs->pageData[1];
if ((firstBytesInPage[0] == 0xFF) && (firstBytesInPage[1] & 0xFC) == 0xF8) {
runningGranulePosition = oggbs->currentPageHeader.granulePosition;
}
continue;
}
}
}
if (!drflac_oggbs__seek_physical(oggbs, runningFrameBytePos, drflac_seek_origin_start)) {
return DRFLAC_FALSE;
}
if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_recover_on_crc_mismatch)) {
return DRFLAC_FALSE;
}
runningPCMFrameCount = runningGranulePosition;
for (;;) {
drflac_uint64 firstPCMFrameInFLACFrame = 0;
drflac_uint64 lastPCMFrameInFLACFrame = 0;
drflac_uint64 pcmFrameCountInThisFrame;
if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) {
return DRFLAC_FALSE;
}
drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &firstPCMFrameInFLACFrame, &lastPCMFrameInFLACFrame);
pcmFrameCountInThisFrame = (lastPCMFrameInFLACFrame - firstPCMFrameInFLACFrame) + 1;
if (pcmFrameIndex == pFlac->totalPCMFrameCount && (runningPCMFrameCount + pcmFrameCountInThisFrame) == pFlac->totalPCMFrameCount) {
drflac_result result = drflac__decode_flac_frame(pFlac);
if (result == DRFLAC_SUCCESS) {
pFlac->currentPCMFrame = pcmFrameIndex;
pFlac->currentFLACFrame.pcmFramesRemaining = 0;
return DRFLAC_TRUE;
} else {
return DRFLAC_FALSE;
}
}
if (pcmFrameIndex < (runningPCMFrameCount + pcmFrameCountInThisFrame)) {
drflac_result result = drflac__decode_flac_frame(pFlac);
if (result == DRFLAC_SUCCESS) {
drflac_uint64 pcmFramesToDecode = (size_t)(pcmFrameIndex - runningPCMFrameCount);
if (pcmFramesToDecode == 0) {
return DRFLAC_TRUE;
}
pFlac->currentPCMFrame = runningPCMFrameCount;
return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode;
} else {
if (result == DRFLAC_CRC_MISMATCH) {
continue;
} else {
return DRFLAC_FALSE;
}
}
} else {
drflac_result result = drflac__seek_to_next_flac_frame(pFlac);
if (result == DRFLAC_SUCCESS) {
runningPCMFrameCount += pcmFrameCountInThisFrame;
} else {
if (result == DRFLAC_CRC_MISMATCH) {
continue;
} else {
return DRFLAC_FALSE;
}
}
}
}
}
static drflac_bool32 drflac__init_private__ogg(drflac_init_info* pInit, drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, void* pUserDataMD, drflac_bool32 relaxed)
{
drflac_ogg_page_header header;
drflac_uint32 crc32 = DRFLAC_OGG_CAPTURE_PATTERN_CRC32;
drflac_uint32 bytesRead = 0;
(void)relaxed;
pInit->container = drflac_container_ogg;
pInit->oggFirstBytePos = 0;
if (drflac_ogg__read_page_header_after_capture_pattern(onRead, pUserData, &header, &bytesRead, &crc32) != DRFLAC_SUCCESS) {
return DRFLAC_FALSE;
}
pInit->runningFilePos += bytesRead;
for (;;) {
int pageBodySize;
if ((header.headerType & 0x02) == 0) {
return DRFLAC_FALSE;
}
pageBodySize = drflac_ogg__get_page_body_size(&header);
if (pageBodySize == 51) {
drflac_uint32 bytesRemainingInPage = pageBodySize;
drflac_uint8 packetType;
if (onRead(pUserData, &packetType, 1) != 1) {
return DRFLAC_FALSE;
}
bytesRemainingInPage -= 1;
if (packetType == 0x7F) {
drflac_uint8 sig[4];
if (onRead(pUserData, sig, 4) != 4) {
return DRFLAC_FALSE;
}
bytesRemainingInPage -= 4;
if (sig[0] == 'F' && sig[1] == 'L' && sig[2] == 'A' && sig[3] == 'C') {
drflac_uint8 mappingVersion[2];
if (onRead(pUserData, mappingVersion, 2) != 2) {
return DRFLAC_FALSE;
}
if (mappingVersion[0] != 1) {
return DRFLAC_FALSE;
}
if (!onSeek(pUserData, 2, drflac_seek_origin_current)) {
return DRFLAC_FALSE;
}
if (onRead(pUserData, sig, 4) != 4) {
return DRFLAC_FALSE;
}
if (sig[0] == 'f' && sig[1] == 'L' && sig[2] == 'a' && sig[3] == 'C') {
drflac_streaminfo streaminfo;
drflac_uint8 isLastBlock;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
allocationSize += seektableSize;
}
pFlac = (drflac*)drflac__malloc_from_callbacks(allocationSize, &allocationCallbacks);
if (pFlac == NULL) {
return NULL;
}
drflac__init_from_info(pFlac, &init);
pFlac->allocationCallbacks = allocationCallbacks;
pFlac->pDecodedSamples = (drflac_int32*)drflac_align((size_t)pFlac->pExtraData, DRFLAC_MAX_SIMD_VECTOR_SIZE);
#ifndef DR_FLAC_NO_OGG
if (init.container == drflac_container_ogg) {
drflac_oggbs* pInternalOggbs = (drflac_oggbs*)((drflac_uint8*)pFlac->pDecodedSamples + decodedSamplesAllocationSize + seektableSize);
DRFLAC_COPY_MEMORY(pInternalOggbs, &oggbs, sizeof(oggbs));
pFlac->bs.onRead = drflac__on_read_ogg;
pFlac->bs.onSeek = drflac__on_seek_ogg;
pFlac->bs.pUserData = (void*)pInternalOggbs;
pFlac->_oggbs = (void*)pInternalOggbs;
}
#endif
pFlac->firstFLACFramePosInBytes = firstFramePos;
#ifndef DR_FLAC_NO_OGG
if (init.container == drflac_container_ogg)
{
pFlac->pSeekpoints = NULL;
pFlac->seekpointCount = 0;
}
else
#endif
{
if (seektablePos != 0) {
pFlac->seekpointCount = seektableSize / sizeof(*pFlac->pSeekpoints);
pFlac->pSeekpoints = (drflac_seekpoint*)((drflac_uint8*)pFlac->pDecodedSamples + decodedSamplesAllocationSize);
DRFLAC_ASSERT(pFlac->bs.onSeek != NULL);
DRFLAC_ASSERT(pFlac->bs.onRead != NULL);
if (pFlac->bs.onSeek(pFlac->bs.pUserData, (int)seektablePos, drflac_seek_origin_start)) {
if (pFlac->bs.onRead(pFlac->bs.pUserData, pFlac->pSeekpoints, seektableSize) == seektableSize) {
drflac_uint32 iSeekpoint;
for (iSeekpoint = 0; iSeekpoint < pFlac->seekpointCount; ++iSeekpoint) {
pFlac->pSeekpoints[iSeekpoint].firstPCMFrame = drflac__be2host_64(pFlac->pSeekpoints[iSeekpoint].firstPCMFrame);
pFlac->pSeekpoints[iSeekpoint].flacFrameOffset = drflac__be2host_64(pFlac->pSeekpoints[iSeekpoint].flacFrameOffset);
pFlac->pSeekpoints[iSeekpoint].pcmFrameCount = drflac__be2host_16(pFlac->pSeekpoints[iSeekpoint].pcmFrameCount);
}
} else {
pFlac->pSeekpoints = NULL;
pFlac->seekpointCount = 0;
}
if (!pFlac->bs.onSeek(pFlac->bs.pUserData, (int)pFlac->firstFLACFramePosInBytes, drflac_seek_origin_start)) {
drflac__free_from_callbacks(pFlac, &allocationCallbacks);
return NULL;
}
} else {
pFlac->pSeekpoints = NULL;
pFlac->seekpointCount = 0;
}
}
}
if (!init.hasStreamInfoBlock) {
pFlac->currentFLACFrame.header = init.firstFrameHeader;
for (;;) {
drflac_result result = drflac__decode_flac_frame(pFlac);
if (result == DRFLAC_SUCCESS) {
break;
} else {
if (result == DRFLAC_CRC_MISMATCH) {
if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) {
drflac__free_from_callbacks(pFlac, &allocationCallbacks);
return NULL;
}
continue;
} else {
drflac__free_from_callbacks(pFlac, &allocationCallbacks);
return NULL;
}
}
}
}
return pFlac;
}
#ifndef DR_FLAC_NO_STDIO
#include <stdio.h>
#include <wchar.h>
#include <errno.h>
static drflac_result drflac_result_from_errno(int e)
{
switch (e)
{
case 0: return DRFLAC_SUCCESS;
#ifdef EPERM
case EPERM: return DRFLAC_INVALID_OPERATION;
#endif
#ifdef ENOENT
case ENOENT: return DRFLAC_DOES_NOT_EXIST;
#endif
#ifdef ESRCH
case ESRCH: return DRFLAC_DOES_NOT_EXIST;
#endif
#ifdef EINTR
case EINTR: return DRFLAC_INTERRUPT;
#endif
#ifdef EIO
case EIO: return DRFLAC_IO_ERROR;
#endif
#ifdef ENXIO
case ENXIO: return DRFLAC_DOES_NOT_EXIST;
#endif
#ifdef E2BIG
case E2BIG: return DRFLAC_INVALID_ARGS;
#endif
#ifdef ENOEXEC
case ENOEXEC: return DRFLAC_INVALID_FILE;
#endif
#ifdef EBADF
case EBADF: return DRFLAC_INVALID_FILE;
#endif
#ifdef ECHILD
case ECHILD: return DRFLAC_ERROR;
#endif
#ifdef EAGAIN
case EAGAIN: return DRFLAC_UNAVAILABLE;
#endif
#ifdef ENOMEM
case ENOMEM: return DRFLAC_OUT_OF_MEMORY;
#endif
#ifdef EACCES
case EACCES: return DRFLAC_ACCESS_DENIED;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
drflac* pFlac;
memoryStream.data = (const drflac_uint8*)pData;
memoryStream.dataSize = dataSize;
memoryStream.currentReadPos = 0;
pFlac = drflac_open_with_metadata_private(drflac__on_read_memory, drflac__on_seek_memory, onMeta, drflac_container_unknown, &memoryStream, pUserData, pAllocationCallbacks);
if (pFlac == NULL) {
return NULL;
}
pFlac->memoryStream = memoryStream;
#ifndef DR_FLAC_NO_OGG
if (pFlac->container == drflac_container_ogg)
{
drflac_oggbs* oggbs = (drflac_oggbs*)pFlac->_oggbs;
oggbs->pUserData = &pFlac->memoryStream;
}
else
#endif
{
pFlac->bs.pUserData = &pFlac->memoryStream;
}
return pFlac;
}
DRFLAC_API drflac* drflac_open(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks)
{
return drflac_open_with_metadata_private(onRead, onSeek, NULL, drflac_container_unknown, pUserData, pUserData, pAllocationCallbacks);
}
DRFLAC_API drflac* drflac_open_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks)
{
return drflac_open_with_metadata_private(onRead, onSeek, NULL, container, pUserData, pUserData, pAllocationCallbacks);
}
DRFLAC_API drflac* drflac_open_with_metadata(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks)
{
return drflac_open_with_metadata_private(onRead, onSeek, onMeta, drflac_container_unknown, pUserData, pUserData, pAllocationCallbacks);
}
DRFLAC_API drflac* drflac_open_with_metadata_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks)
{
return drflac_open_with_metadata_private(onRead, onSeek, onMeta, container, pUserData, pUserData, pAllocationCallbacks);
}
DRFLAC_API void drflac_close(drflac* pFlac)
{
if (pFlac == NULL) {
return;
}
#ifndef DR_FLAC_NO_STDIO
if (pFlac->bs.onRead == drflac__on_read_stdio) {
fclose((FILE*)pFlac->bs.pUserData);
}
#ifndef DR_FLAC_NO_OGG
if (pFlac->container == drflac_container_ogg) {
drflac_oggbs* oggbs = (drflac_oggbs*)pFlac->_oggbs;
DRFLAC_ASSERT(pFlac->bs.onRead == drflac__on_read_ogg);
if (oggbs->onRead == drflac__on_read_stdio) {
fclose((FILE*)oggbs->pUserData);
}
}
#endif
#endif
drflac__free_from_callbacks(pFlac, &pFlac->allocationCallbacks);
}
#if 0
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutpu...
{
drflac_uint64 i;
for (i = 0; i < frameCount; ++i) {
drflac_uint32 left = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
drflac_uint32 right = left - side;
pOutputSamples[i*2+0] = (drflac_int32)left;
pOutputSamples[i*2+1] = (drflac_int32)right;
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSa...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 left0 = pInputSamples0U32[i*4+0] << shift0;
drflac_uint32 left1 = pInputSamples0U32[i*4+1] << shift0;
drflac_uint32 left2 = pInputSamples0U32[i*4+2] << shift0;
drflac_uint32 left3 = pInputSamples0U32[i*4+3] << shift0;
drflac_uint32 side0 = pInputSamples1U32[i*4+0] << shift1;
drflac_uint32 side1 = pInputSamples1U32[i*4+1] << shift1;
drflac_uint32 side2 = pInputSamples1U32[i*4+2] << shift1;
drflac_uint32 side3 = pInputSamples1U32[i*4+3] << shift1;
drflac_uint32 right0 = left0 - side0;
drflac_uint32 right1 = left1 - side1;
drflac_uint32 right2 = left2 - side2;
drflac_uint32 right3 = left3 - side3;
pOutputSamples[i*8+0] = (drflac_int32)left0;
pOutputSamples[i*8+1] = (drflac_int32)right0;
pOutputSamples[i*8+2] = (drflac_int32)left1;
pOutputSamples[i*8+3] = (drflac_int32)right1;
pOutputSamples[i*8+4] = (drflac_int32)left2;
pOutputSamples[i*8+5] = (drflac_int32)right2;
pOutputSamples[i*8+6] = (drflac_int32)left3;
pOutputSamples[i*8+7] = (drflac_int32)right3;
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 left = pInputSamples0U32[i] << shift0;
drflac_uint32 side = pInputSamples1U32[i] << shift1;
drflac_uint32 right = left - side;
pOutputSamples[i*2+0] = (drflac_int32)left;
pOutputSamples[i*2+1] = (drflac_int32)right;
}
}
#if defined(DRFLAC_SUPPORT_SSE2)
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamp...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
for (i = 0; i < frameCount4; ++i) {
__m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0);
__m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1);
__m128i right = _mm_sub_epi32(left, side);
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right));
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 left = pInputSamples0U32[i] << shift0;
drflac_uint32 side = pInputSamples1U32[i] << shift1;
drflac_uint32 right = left - side;
pOutputSamples[i*2+0] = (drflac_int32)left;
pOutputSamples[i*2+1] = (drflac_int32)right;
}
}
#endif
#if defined(DRFLAC_SUPPORT_NEON)
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamp...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
int32x4_t shift0_4;
int32x4_t shift1_4;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
shift0_4 = vdupq_n_s32(shift0);
shift1_4 = vdupq_n_s32(shift1);
for (i = 0; i < frameCount4; ++i) {
uint32x4_t left;
uint32x4_t side;
uint32x4_t right;
left = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4);
side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4);
right = vsubq_u32(left, side);
drflac__vst2q_u32((drflac_uint32*)pOutputSamples + i*8, vzipq_u32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 left = pInputSamples0U32[i] << shift0;
drflac_uint32 side = pInputSamples1U32[i] << shift1;
drflac_uint32 right = left - side;
pOutputSamples[i*2+0] = (drflac_int32)left;
pOutputSamples[i*2+1] = (drflac_int32)right;
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples)
{
#if defined(DRFLAC_SUPPORT_SSE2)
if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s32__decode_left_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s32__decode_left_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#endif
{
#if 0
drflac_read_pcm_frames_s32__decode_left_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#else
drflac_read_pcm_frames_s32__decode_left_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#endif
}
}
#if 0
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutp...
{
drflac_uint64 i;
for (i = 0; i < frameCount; ++i) {
drflac_uint32 side = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
drflac_uint32 right = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
drflac_uint32 left = right + side;
pOutputSamples[i*2+0] = (drflac_int32)left;
pOutputSamples[i*2+1] = (drflac_int32)right;
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputS...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 side0 = pInputSamples0U32[i*4+0] << shift0;
drflac_uint32 side1 = pInputSamples0U32[i*4+1] << shift0;
drflac_uint32 side2 = pInputSamples0U32[i*4+2] << shift0;
drflac_uint32 side3 = pInputSamples0U32[i*4+3] << shift0;
drflac_uint32 right0 = pInputSamples1U32[i*4+0] << shift1;
drflac_uint32 right1 = pInputSamples1U32[i*4+1] << shift1;
drflac_uint32 right2 = pInputSamples1U32[i*4+2] << shift1;
drflac_uint32 right3 = pInputSamples1U32[i*4+3] << shift1;
drflac_uint32 left0 = right0 + side0;
drflac_uint32 left1 = right1 + side1;
drflac_uint32 left2 = right2 + side2;
drflac_uint32 left3 = right3 + side3;
pOutputSamples[i*8+0] = (drflac_int32)left0;
pOutputSamples[i*8+1] = (drflac_int32)right0;
pOutputSamples[i*8+2] = (drflac_int32)left1;
pOutputSamples[i*8+3] = (drflac_int32)right1;
pOutputSamples[i*8+4] = (drflac_int32)left2;
pOutputSamples[i*8+5] = (drflac_int32)right2;
pOutputSamples[i*8+6] = (drflac_int32)left3;
pOutputSamples[i*8+7] = (drflac_int32)right3;
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 side = pInputSamples0U32[i] << shift0;
drflac_uint32 right = pInputSamples1U32[i] << shift1;
drflac_uint32 left = right + side;
pOutputSamples[i*2+0] = (drflac_int32)left;
pOutputSamples[i*2+1] = (drflac_int32)right;
}
}
#if defined(DRFLAC_SUPPORT_SSE2)
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSam...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
for (i = 0; i < frameCount4; ++i) {
__m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0);
__m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1);
__m128i left = _mm_add_epi32(right, side);
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right));
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 side = pInputSamples0U32[i] << shift0;
drflac_uint32 right = pInputSamples1U32[i] << shift1;
drflac_uint32 left = right + side;
pOutputSamples[i*2+0] = (drflac_int32)left;
pOutputSamples[i*2+1] = (drflac_int32)right;
}
}
#endif
#if defined(DRFLAC_SUPPORT_NEON)
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSam...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
int32x4_t shift0_4;
int32x4_t shift1_4;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
shift0_4 = vdupq_n_s32(shift0);
shift1_4 = vdupq_n_s32(shift1);
for (i = 0; i < frameCount4; ++i) {
uint32x4_t side;
uint32x4_t right;
uint32x4_t left;
side = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4);
right = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4);
left = vaddq_u32(right, side);
drflac__vst2q_u32((drflac_uint32*)pOutputSamples + i*8, vzipq_u32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 side = pInputSamples0U32[i] << shift0;
drflac_uint32 right = pInputSamples1U32[i] << shift1;
drflac_uint32 left = right + side;
pOutputSamples[i*2+0] = (drflac_int32)left;
pOutputSamples[i*2+1] = (drflac_int32)right;
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples)
{
#if defined(DRFLAC_SUPPORT_SSE2)
if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s32__decode_right_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s32__decode_right_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#endif
{
#if 0
drflac_read_pcm_frames_s32__decode_right_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#else
drflac_read_pcm_frames_s32__decode_right_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#endif
}
}
#if 0
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutput...
{
for (drflac_uint64 i = 0; i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample);
pOutputSamples[i*2+1] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample);
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSam...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_int32 shift = unusedBitsPerSample;
if (shift > 0) {
shift -= 1;
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 temp0L;
drflac_uint32 temp1L;
drflac_uint32 temp2L;
drflac_uint32 temp3L;
drflac_uint32 temp0R;
drflac_uint32 temp1R;
drflac_uint32 temp2R;
drflac_uint32 temp3R;
drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid0 = (mid0 << 1) | (side0 & 0x01);
mid1 = (mid1 << 1) | (side1 & 0x01);
mid2 = (mid2 << 1) | (side2 & 0x01);
mid3 = (mid3 << 1) | (side3 & 0x01);
temp0L = (mid0 + side0) << shift;
temp1L = (mid1 + side1) << shift;
temp2L = (mid2 + side2) << shift;
temp3L = (mid3 + side3) << shift;
temp0R = (mid0 - side0) << shift;
temp1R = (mid1 - side1) << shift;
temp2R = (mid2 - side2) << shift;
temp3R = (mid3 - side3) << shift;
pOutputSamples[i*8+0] = (drflac_int32)temp0L;
pOutputSamples[i*8+1] = (drflac_int32)temp0R;
pOutputSamples[i*8+2] = (drflac_int32)temp1L;
pOutputSamples[i*8+3] = (drflac_int32)temp1R;
pOutputSamples[i*8+4] = (drflac_int32)temp2L;
pOutputSamples[i*8+5] = (drflac_int32)temp2R;
pOutputSamples[i*8+6] = (drflac_int32)temp3L;
pOutputSamples[i*8+7] = (drflac_int32)temp3R;
}
} else {
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 temp0L;
drflac_uint32 temp1L;
drflac_uint32 temp2L;
drflac_uint32 temp3L;
drflac_uint32 temp0R;
drflac_uint32 temp1R;
drflac_uint32 temp2R;
drflac_uint32 temp3R;
drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid0 = (mid0 << 1) | (side0 & 0x01);
mid1 = (mid1 << 1) | (side1 & 0x01);
mid2 = (mid2 << 1) | (side2 & 0x01);
mid3 = (mid3 << 1) | (side3 & 0x01);
temp0L = (drflac_uint32)((drflac_int32)(mid0 + side0) >> 1);
temp1L = (drflac_uint32)((drflac_int32)(mid1 + side1) >> 1);
temp2L = (drflac_uint32)((drflac_int32)(mid2 + side2) >> 1);
temp3L = (drflac_uint32)((drflac_int32)(mid3 + side3) >> 1);
temp0R = (drflac_uint32)((drflac_int32)(mid0 - side0) >> 1);
temp1R = (drflac_uint32)((drflac_int32)(mid1 - side1) >> 1);
temp2R = (drflac_uint32)((drflac_int32)(mid2 - side2) >> 1);
temp3R = (drflac_uint32)((drflac_int32)(mid3 - side3) >> 1);
pOutputSamples[i*8+0] = (drflac_int32)temp0L;
pOutputSamples[i*8+1] = (drflac_int32)temp0R;
pOutputSamples[i*8+2] = (drflac_int32)temp1L;
pOutputSamples[i*8+3] = (drflac_int32)temp1R;
pOutputSamples[i*8+4] = (drflac_int32)temp2L;
pOutputSamples[i*8+5] = (drflac_int32)temp2R;
pOutputSamples[i*8+6] = (drflac_int32)temp3L;
pOutputSamples[i*8+7] = (drflac_int32)temp3R;
}
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample);
pOutputSamples[i*2+1] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample);
}
}
#if defined(DRFLAC_SUPPORT_SSE2)
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSampl...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_int32 shift = unusedBitsPerSample;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
if (shift == 0) {
for (i = 0; i < frameCount4; ++i) {
__m128i mid;
__m128i side;
__m128i left;
__m128i right;
mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01)));
left = _mm_srai_epi32(_mm_add_epi32(mid, side), 1);
right = _mm_srai_epi32(_mm_sub_epi32(mid, side), 1);
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right));
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int32)(mid + side) >> 1;
pOutputSamples[i*2+1] = (drflac_int32)(mid - side) >> 1;
}
} else {
shift -= 1;
for (i = 0; i < frameCount4; ++i) {
__m128i mid;
__m128i side;
__m128i left;
__m128i right;
mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01)));
left = _mm_slli_epi32(_mm_add_epi32(mid, side), shift);
right = _mm_slli_epi32(_mm_sub_epi32(mid, side), shift);
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right));
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int32)((mid + side) << shift);
pOutputSamples[i*2+1] = (drflac_int32)((mid - side) << shift);
}
}
}
#endif
#if defined(DRFLAC_SUPPORT_NEON)
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSampl...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_int32 shift = unusedBitsPerSample;
int32x4_t wbpsShift0_4;
int32x4_t wbpsShift1_4;
uint32x4_t one4;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
wbpsShift0_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
wbpsShift1_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
one4 = vdupq_n_u32(1);
if (shift == 0) {
for (i = 0; i < frameCount4; ++i) {
uint32x4_t mid;
uint32x4_t side;
int32x4_t left;
int32x4_t right;
mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbpsShift0_4);
side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbpsShift1_4);
mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, one4));
left = vshrq_n_s32(vreinterpretq_s32_u32(vaddq_u32(mid, side)), 1);
right = vshrq_n_s32(vreinterpretq_s32_u32(vsubq_u32(mid, side)), 1);
drflac__vst2q_s32(pOutputSamples + i*8, vzipq_s32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int32)(mid + side) >> 1;
pOutputSamples[i*2+1] = (drflac_int32)(mid - side) >> 1;
}
} else {
int32x4_t shift4;
shift -= 1;
shift4 = vdupq_n_s32(shift);
for (i = 0; i < frameCount4; ++i) {
uint32x4_t mid;
uint32x4_t side;
int32x4_t left;
int32x4_t right;
mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbpsShift0_4);
side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbpsShift1_4);
mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, one4));
left = vreinterpretq_s32_u32(vshlq_u32(vaddq_u32(mid, side), shift4));
right = vreinterpretq_s32_u32(vshlq_u32(vsubq_u32(mid, side), shift4));
drflac__vst2q_s32(pOutputSamples + i*8, vzipq_s32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int32)((mid + side) << shift);
pOutputSamples[i*2+1] = (drflac_int32)((mid - side) << shift);
}
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples)
{
#if defined(DRFLAC_SUPPORT_SSE2)
if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s32__decode_mid_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s32__decode_mid_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#endif
{
#if 0
drflac_read_pcm_frames_s32__decode_mid_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#else
drflac_read_pcm_frames_s32__decode_mid_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#endif
}
}
#if 0
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int3...
{
for (drflac_uint64 i = 0; i < frameCount; ++i) {
pOutputSamples[i*2+0] = (drflac_int32)((drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample));
pOutputSamples[i*2+1] = (drflac_int32)((drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample));
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* ...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 tempL0 = pInputSamples0U32[i*4+0] << shift0;
drflac_uint32 tempL1 = pInputSamples0U32[i*4+1] << shift0;
drflac_uint32 tempL2 = pInputSamples0U32[i*4+2] << shift0;
drflac_uint32 tempL3 = pInputSamples0U32[i*4+3] << shift0;
drflac_uint32 tempR0 = pInputSamples1U32[i*4+0] << shift1;
drflac_uint32 tempR1 = pInputSamples1U32[i*4+1] << shift1;
drflac_uint32 tempR2 = pInputSamples1U32[i*4+2] << shift1;
drflac_uint32 tempR3 = pInputSamples1U32[i*4+3] << shift1;
pOutputSamples[i*8+0] = (drflac_int32)tempL0;
pOutputSamples[i*8+1] = (drflac_int32)tempR0;
pOutputSamples[i*8+2] = (drflac_int32)tempL1;
pOutputSamples[i*8+3] = (drflac_int32)tempR1;
pOutputSamples[i*8+4] = (drflac_int32)tempL2;
pOutputSamples[i*8+5] = (drflac_int32)tempR2;
pOutputSamples[i*8+6] = (drflac_int32)tempL3;
pOutputSamples[i*8+7] = (drflac_int32)tempR3;
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0);
pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1);
}
}
#if defined(DRFLAC_SUPPORT_SSE2)
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pO...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
for (i = 0; i < frameCount4; ++i) {
__m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0);
__m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1);
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right));
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0);
pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1);
}
}
#endif
#if defined(DRFLAC_SUPPORT_NEON)
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pO...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
int32x4_t shift4_0 = vdupq_n_s32(shift0);
int32x4_t shift4_1 = vdupq_n_s32(shift1);
for (i = 0; i < frameCount4; ++i) {
int32x4_t left;
int32x4_t right;
left = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift4_0));
right = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift4_1));
drflac__vst2q_s32(pOutputSamples + i*8, vzipq_s32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0);
pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1);
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputS...
{
#if defined(DRFLAC_SUPPORT_SSE2)
if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s32__decode_independent_stereo__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s32__decode_independent_stereo__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#endif
{
#if 0
drflac_read_pcm_frames_s32__decode_independent_stereo__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#else
drflac_read_pcm_frames_s32__decode_independent_stereo__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#endif
}
}
DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s32(drflac* pFlac, drflac_uint64 framesToRead, drflac_int32* pBufferOut)
{
drflac_uint64 framesRead;
drflac_uint32 unusedBitsPerSample;
if (pFlac == NULL || framesToRead == 0) {
return 0;
}
if (pBufferOut == NULL) {
return drflac__seek_forward_by_pcm_frames(pFlac, framesToRead);
}
DRFLAC_ASSERT(pFlac->bitsPerSample <= 32);
unusedBitsPerSample = 32 - pFlac->bitsPerSample;
framesRead = 0;
while (framesToRead > 0) {
if (pFlac->currentFLACFrame.pcmFramesRemaining == 0) {
if (!drflac__read_and_decode_next_flac_frame(pFlac)) {
break;
}
} else {
unsigned int channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment);
drflac_uint64 iFirstPCMFrame = pFlac->currentFLACFrame.header.blockSizeInPCMFrames - pFlac->currentFLACFrame.pcmFramesRemaining;
drflac_uint64 frameCountThisIteration = framesToRead;
if (frameCountThisIteration > pFlac->currentFLACFrame.pcmFramesRemaining) {
frameCountThisIteration = pFlac->currentFLACFrame.pcmFramesRemaining;
}
if (channelCount == 2) {
const drflac_int32* pDecodedSamples0 = pFlac->currentFLACFrame.subframes[0].pSamplesS32 + iFirstPCMFrame;
const drflac_int32* pDecodedSamples1 = pFlac->currentFLACFrame.subframes[1].pSamplesS32 + iFirstPCMFrame;
switch (pFlac->currentFLACFrame.header.channelAssignment)
{
case DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE:
{
drflac_read_pcm_frames_s32__decode_left_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut);
} break;
case DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE:
{
drflac_read_pcm_frames_s32__decode_right_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut);
} break;
case DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE:
{
drflac_read_pcm_frames_s32__decode_mid_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut);
} break;
case DRFLAC_CHANNEL_ASSIGNMENT_INDEPENDENT:
default:
{
drflac_read_pcm_frames_s32__decode_independent_stereo(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut);
} break;
}
} else {
drflac_uint64 i;
for (i = 0; i < frameCountThisIteration; ++i) {
unsigned int j;
for (j = 0; j < channelCount; ++j) {
pBufferOut[(i*channelCount)+j] = (drflac_int32)((drflac_uint32)(pFlac->currentFLACFrame.subframes[j].pSamplesS32[iFirstPCMFrame + i]) << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[j].wastedBitsPerSample));
}
}
}
framesRead += frameCountThisIteration;
pBufferOut += frameCountThisIteration * channelCount;
framesToRead -= frameCountThisIteration;
pFlac->currentPCMFrame += frameCountThisIteration;
pFlac->currentFLACFrame.pcmFramesRemaining -= (drflac_uint32)frameCountThisIteration;
}
}
return framesRead;
}
#if 0
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutpu...
{
drflac_uint64 i;
for (i = 0; i < frameCount; ++i) {
drflac_uint32 left = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
drflac_uint32 right = left - side;
left >>= 16;
right >>= 16;
pOutputSamples[i*2+0] = (drflac_int16)left;
pOutputSamples[i*2+1] = (drflac_int16)right;
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSa...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 left0 = pInputSamples0U32[i*4+0] << shift0;
drflac_uint32 left1 = pInputSamples0U32[i*4+1] << shift0;
drflac_uint32 left2 = pInputSamples0U32[i*4+2] << shift0;
drflac_uint32 left3 = pInputSamples0U32[i*4+3] << shift0;
drflac_uint32 side0 = pInputSamples1U32[i*4+0] << shift1;
drflac_uint32 side1 = pInputSamples1U32[i*4+1] << shift1;
drflac_uint32 side2 = pInputSamples1U32[i*4+2] << shift1;
drflac_uint32 side3 = pInputSamples1U32[i*4+3] << shift1;
drflac_uint32 right0 = left0 - side0;
drflac_uint32 right1 = left1 - side1;
drflac_uint32 right2 = left2 - side2;
drflac_uint32 right3 = left3 - side3;
left0 >>= 16;
left1 >>= 16;
left2 >>= 16;
left3 >>= 16;
right0 >>= 16;
right1 >>= 16;
right2 >>= 16;
right3 >>= 16;
pOutputSamples[i*8+0] = (drflac_int16)left0;
pOutputSamples[i*8+1] = (drflac_int16)right0;
pOutputSamples[i*8+2] = (drflac_int16)left1;
pOutputSamples[i*8+3] = (drflac_int16)right1;
pOutputSamples[i*8+4] = (drflac_int16)left2;
pOutputSamples[i*8+5] = (drflac_int16)right2;
pOutputSamples[i*8+6] = (drflac_int16)left3;
pOutputSamples[i*8+7] = (drflac_int16)right3;
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 left = pInputSamples0U32[i] << shift0;
drflac_uint32 side = pInputSamples1U32[i] << shift1;
drflac_uint32 right = left - side;
left >>= 16;
right >>= 16;
pOutputSamples[i*2+0] = (drflac_int16)left;
pOutputSamples[i*2+1] = (drflac_int16)right;
}
}
#if defined(DRFLAC_SUPPORT_SSE2)
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamp...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
for (i = 0; i < frameCount4; ++i) {
__m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0);
__m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1);
__m128i right = _mm_sub_epi32(left, side);
left = _mm_srai_epi32(left, 16);
right = _mm_srai_epi32(right, 16);
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 left = pInputSamples0U32[i] << shift0;
drflac_uint32 side = pInputSamples1U32[i] << shift1;
drflac_uint32 right = left - side;
left >>= 16;
right >>= 16;
pOutputSamples[i*2+0] = (drflac_int16)left;
pOutputSamples[i*2+1] = (drflac_int16)right;
}
}
#endif
#if defined(DRFLAC_SUPPORT_NEON)
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamp...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
int32x4_t shift0_4;
int32x4_t shift1_4;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
shift0_4 = vdupq_n_s32(shift0);
shift1_4 = vdupq_n_s32(shift1);
for (i = 0; i < frameCount4; ++i) {
uint32x4_t left;
uint32x4_t side;
uint32x4_t right;
left = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4);
side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4);
right = vsubq_u32(left, side);
left = vshrq_n_u32(left, 16);
right = vshrq_n_u32(right, 16);
drflac__vst2q_u16((drflac_uint16*)pOutputSamples + i*8, vzip_u16(vmovn_u32(left), vmovn_u32(right)));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 left = pInputSamples0U32[i] << shift0;
drflac_uint32 side = pInputSamples1U32[i] << shift1;
drflac_uint32 right = left - side;
left >>= 16;
right >>= 16;
pOutputSamples[i*2+0] = (drflac_int16)left;
pOutputSamples[i*2+1] = (drflac_int16)right;
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples)
{
#if defined(DRFLAC_SUPPORT_SSE2)
if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s16__decode_left_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s16__decode_left_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#endif
{
#if 0
drflac_read_pcm_frames_s16__decode_left_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#else
drflac_read_pcm_frames_s16__decode_left_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#endif
}
}
#if 0
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutp...
{
drflac_uint64 i;
for (i = 0; i < frameCount; ++i) {
drflac_uint32 side = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
drflac_uint32 right = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
drflac_uint32 left = right + side;
left >>= 16;
right >>= 16;
pOutputSamples[i*2+0] = (drflac_int16)left;
pOutputSamples[i*2+1] = (drflac_int16)right;
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputS...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 side0 = pInputSamples0U32[i*4+0] << shift0;
drflac_uint32 side1 = pInputSamples0U32[i*4+1] << shift0;
drflac_uint32 side2 = pInputSamples0U32[i*4+2] << shift0;
drflac_uint32 side3 = pInputSamples0U32[i*4+3] << shift0;
drflac_uint32 right0 = pInputSamples1U32[i*4+0] << shift1;
drflac_uint32 right1 = pInputSamples1U32[i*4+1] << shift1;
drflac_uint32 right2 = pInputSamples1U32[i*4+2] << shift1;
drflac_uint32 right3 = pInputSamples1U32[i*4+3] << shift1;
drflac_uint32 left0 = right0 + side0;
drflac_uint32 left1 = right1 + side1;
drflac_uint32 left2 = right2 + side2;
drflac_uint32 left3 = right3 + side3;
left0 >>= 16;
left1 >>= 16;
left2 >>= 16;
left3 >>= 16;
right0 >>= 16;
right1 >>= 16;
right2 >>= 16;
right3 >>= 16;
pOutputSamples[i*8+0] = (drflac_int16)left0;
pOutputSamples[i*8+1] = (drflac_int16)right0;
pOutputSamples[i*8+2] = (drflac_int16)left1;
pOutputSamples[i*8+3] = (drflac_int16)right1;
pOutputSamples[i*8+4] = (drflac_int16)left2;
pOutputSamples[i*8+5] = (drflac_int16)right2;
pOutputSamples[i*8+6] = (drflac_int16)left3;
pOutputSamples[i*8+7] = (drflac_int16)right3;
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 side = pInputSamples0U32[i] << shift0;
drflac_uint32 right = pInputSamples1U32[i] << shift1;
drflac_uint32 left = right + side;
left >>= 16;
right >>= 16;
pOutputSamples[i*2+0] = (drflac_int16)left;
pOutputSamples[i*2+1] = (drflac_int16)right;
}
}
#if defined(DRFLAC_SUPPORT_SSE2)
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSam...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
for (i = 0; i < frameCount4; ++i) {
__m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0);
__m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1);
__m128i left = _mm_add_epi32(right, side);
left = _mm_srai_epi32(left, 16);
right = _mm_srai_epi32(right, 16);
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 side = pInputSamples0U32[i] << shift0;
drflac_uint32 right = pInputSamples1U32[i] << shift1;
drflac_uint32 left = right + side;
left >>= 16;
right >>= 16;
pOutputSamples[i*2+0] = (drflac_int16)left;
pOutputSamples[i*2+1] = (drflac_int16)right;
}
}
#endif
#if defined(DRFLAC_SUPPORT_NEON)
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSam...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
int32x4_t shift0_4;
int32x4_t shift1_4;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
shift0_4 = vdupq_n_s32(shift0);
shift1_4 = vdupq_n_s32(shift1);
for (i = 0; i < frameCount4; ++i) {
uint32x4_t side;
uint32x4_t right;
uint32x4_t left;
side = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4);
right = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4);
left = vaddq_u32(right, side);
left = vshrq_n_u32(left, 16);
right = vshrq_n_u32(right, 16);
drflac__vst2q_u16((drflac_uint16*)pOutputSamples + i*8, vzip_u16(vmovn_u32(left), vmovn_u32(right)));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 side = pInputSamples0U32[i] << shift0;
drflac_uint32 right = pInputSamples1U32[i] << shift1;
drflac_uint32 left = right + side;
left >>= 16;
right >>= 16;
pOutputSamples[i*2+0] = (drflac_int16)left;
pOutputSamples[i*2+1] = (drflac_int16)right;
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples)
{
#if defined(DRFLAC_SUPPORT_SSE2)
if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s16__decode_right_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s16__decode_right_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#endif
{
#if 0
drflac_read_pcm_frames_s16__decode_right_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#else
drflac_read_pcm_frames_s16__decode_right_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#endif
}
}
#if 0
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutput...
{
for (drflac_uint64 i = 0; i < frameCount; ++i) {
drflac_uint32 mid = (drflac_uint32)pInputSamples0[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int16)(((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample) >> 16);
pOutputSamples[i*2+1] = (drflac_int16)(((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample) >> 16);
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSam...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift = unusedBitsPerSample;
if (shift > 0) {
shift -= 1;
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 temp0L;
drflac_uint32 temp1L;
drflac_uint32 temp2L;
drflac_uint32 temp3L;
drflac_uint32 temp0R;
drflac_uint32 temp1R;
drflac_uint32 temp2R;
drflac_uint32 temp3R;
drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid0 = (mid0 << 1) | (side0 & 0x01);
mid1 = (mid1 << 1) | (side1 & 0x01);
mid2 = (mid2 << 1) | (side2 & 0x01);
mid3 = (mid3 << 1) | (side3 & 0x01);
temp0L = (mid0 + side0) << shift;
temp1L = (mid1 + side1) << shift;
temp2L = (mid2 + side2) << shift;
temp3L = (mid3 + side3) << shift;
temp0R = (mid0 - side0) << shift;
temp1R = (mid1 - side1) << shift;
temp2R = (mid2 - side2) << shift;
temp3R = (mid3 - side3) << shift;
temp0L >>= 16;
temp1L >>= 16;
temp2L >>= 16;
temp3L >>= 16;
temp0R >>= 16;
temp1R >>= 16;
temp2R >>= 16;
temp3R >>= 16;
pOutputSamples[i*8+0] = (drflac_int16)temp0L;
pOutputSamples[i*8+1] = (drflac_int16)temp0R;
pOutputSamples[i*8+2] = (drflac_int16)temp1L;
pOutputSamples[i*8+3] = (drflac_int16)temp1R;
pOutputSamples[i*8+4] = (drflac_int16)temp2L;
pOutputSamples[i*8+5] = (drflac_int16)temp2R;
pOutputSamples[i*8+6] = (drflac_int16)temp3L;
pOutputSamples[i*8+7] = (drflac_int16)temp3R;
}
} else {
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 temp0L;
drflac_uint32 temp1L;
drflac_uint32 temp2L;
drflac_uint32 temp3L;
drflac_uint32 temp0R;
drflac_uint32 temp1R;
drflac_uint32 temp2R;
drflac_uint32 temp3R;
drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid0 = (mid0 << 1) | (side0 & 0x01);
mid1 = (mid1 << 1) | (side1 & 0x01);
mid2 = (mid2 << 1) | (side2 & 0x01);
mid3 = (mid3 << 1) | (side3 & 0x01);
temp0L = ((drflac_int32)(mid0 + side0) >> 1);
temp1L = ((drflac_int32)(mid1 + side1) >> 1);
temp2L = ((drflac_int32)(mid2 + side2) >> 1);
temp3L = ((drflac_int32)(mid3 + side3) >> 1);
temp0R = ((drflac_int32)(mid0 - side0) >> 1);
temp1R = ((drflac_int32)(mid1 - side1) >> 1);
temp2R = ((drflac_int32)(mid2 - side2) >> 1);
temp3R = ((drflac_int32)(mid3 - side3) >> 1);
temp0L >>= 16;
temp1L >>= 16;
temp2L >>= 16;
temp3L >>= 16;
temp0R >>= 16;
temp1R >>= 16;
temp2R >>= 16;
temp3R >>= 16;
pOutputSamples[i*8+0] = (drflac_int16)temp0L;
pOutputSamples[i*8+1] = (drflac_int16)temp0R;
pOutputSamples[i*8+2] = (drflac_int16)temp1L;
pOutputSamples[i*8+3] = (drflac_int16)temp1R;
pOutputSamples[i*8+4] = (drflac_int16)temp2L;
pOutputSamples[i*8+5] = (drflac_int16)temp2R;
pOutputSamples[i*8+6] = (drflac_int16)temp3L;
pOutputSamples[i*8+7] = (drflac_int16)temp3R;
}
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int16)(((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample) >> 16);
pOutputSamples[i*2+1] = (drflac_int16)(((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample) >> 16);
}
}
#if defined(DRFLAC_SUPPORT_SSE2)
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSampl...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift = unusedBitsPerSample;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
if (shift == 0) {
for (i = 0; i < frameCount4; ++i) {
__m128i mid;
__m128i side;
__m128i left;
__m128i right;
mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01)));
left = _mm_srai_epi32(_mm_add_epi32(mid, side), 1);
right = _mm_srai_epi32(_mm_sub_epi32(mid, side), 1);
left = _mm_srai_epi32(left, 16);
right = _mm_srai_epi32(right, 16);
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int16)(((drflac_int32)(mid + side) >> 1) >> 16);
pOutputSamples[i*2+1] = (drflac_int16)(((drflac_int32)(mid - side) >> 1) >> 16);
}
} else {
shift -= 1;
for (i = 0; i < frameCount4; ++i) {
__m128i mid;
__m128i side;
__m128i left;
__m128i right;
mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01)));
left = _mm_slli_epi32(_mm_add_epi32(mid, side), shift);
right = _mm_slli_epi32(_mm_sub_epi32(mid, side), shift);
left = _mm_srai_epi32(left, 16);
right = _mm_srai_epi32(right, 16);
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int16)(((mid + side) << shift) >> 16);
pOutputSamples[i*2+1] = (drflac_int16)(((mid - side) << shift) >> 16);
}
}
}
#endif
#if defined(DRFLAC_SUPPORT_NEON)
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSampl...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift = unusedBitsPerSample;
int32x4_t wbpsShift0_4;
int32x4_t wbpsShift1_4;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
wbpsShift0_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
wbpsShift1_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
if (shift == 0) {
for (i = 0; i < frameCount4; ++i) {
uint32x4_t mid;
uint32x4_t side;
int32x4_t left;
int32x4_t right;
mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbpsShift0_4);
side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbpsShift1_4);
mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, vdupq_n_u32(1)));
left = vshrq_n_s32(vreinterpretq_s32_u32(vaddq_u32(mid, side)), 1);
right = vshrq_n_s32(vreinterpretq_s32_u32(vsubq_u32(mid, side)), 1);
left = vshrq_n_s32(left, 16);
right = vshrq_n_s32(right, 16);
drflac__vst2q_s16(pOutputSamples + i*8, vzip_s16(vmovn_s32(left), vmovn_s32(right)));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int16)(((drflac_int32)(mid + side) >> 1) >> 16);
pOutputSamples[i*2+1] = (drflac_int16)(((drflac_int32)(mid - side) >> 1) >> 16);
}
} else {
int32x4_t shift4;
shift -= 1;
shift4 = vdupq_n_s32(shift);
for (i = 0; i < frameCount4; ++i) {
uint32x4_t mid;
uint32x4_t side;
int32x4_t left;
int32x4_t right;
mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbpsShift0_4);
side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbpsShift1_4);
mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, vdupq_n_u32(1)));
left = vreinterpretq_s32_u32(vshlq_u32(vaddq_u32(mid, side), shift4));
right = vreinterpretq_s32_u32(vshlq_u32(vsubq_u32(mid, side), shift4));
left = vshrq_n_s32(left, 16);
right = vshrq_n_s32(right, 16);
drflac__vst2q_s16(pOutputSamples + i*8, vzip_s16(vmovn_s32(left), vmovn_s32(right)));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int16)(((mid + side) << shift) >> 16);
pOutputSamples[i*2+1] = (drflac_int16)(((mid - side) << shift) >> 16);
}
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples)
{
#if defined(DRFLAC_SUPPORT_SSE2)
if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s16__decode_mid_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s16__decode_mid_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#endif
{
#if 0
drflac_read_pcm_frames_s16__decode_mid_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#else
drflac_read_pcm_frames_s16__decode_mid_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#endif
}
}
#if 0
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int1...
{
for (drflac_uint64 i = 0; i < frameCount; ++i) {
pOutputSamples[i*2+0] = (drflac_int16)((drflac_int32)((drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample)) >> 16);
pOutputSamples[i*2+1] = (drflac_int16)((drflac_int32)((drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample)) >> 16);
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* ...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 tempL0 = pInputSamples0U32[i*4+0] << shift0;
drflac_uint32 tempL1 = pInputSamples0U32[i*4+1] << shift0;
drflac_uint32 tempL2 = pInputSamples0U32[i*4+2] << shift0;
drflac_uint32 tempL3 = pInputSamples0U32[i*4+3] << shift0;
drflac_uint32 tempR0 = pInputSamples1U32[i*4+0] << shift1;
drflac_uint32 tempR1 = pInputSamples1U32[i*4+1] << shift1;
drflac_uint32 tempR2 = pInputSamples1U32[i*4+2] << shift1;
drflac_uint32 tempR3 = pInputSamples1U32[i*4+3] << shift1;
tempL0 >>= 16;
tempL1 >>= 16;
tempL2 >>= 16;
tempL3 >>= 16;
tempR0 >>= 16;
tempR1 >>= 16;
tempR2 >>= 16;
tempR3 >>= 16;
pOutputSamples[i*8+0] = (drflac_int16)tempL0;
pOutputSamples[i*8+1] = (drflac_int16)tempR0;
pOutputSamples[i*8+2] = (drflac_int16)tempL1;
pOutputSamples[i*8+3] = (drflac_int16)tempR1;
pOutputSamples[i*8+4] = (drflac_int16)tempL2;
pOutputSamples[i*8+5] = (drflac_int16)tempR2;
pOutputSamples[i*8+6] = (drflac_int16)tempL3;
pOutputSamples[i*8+7] = (drflac_int16)tempR3;
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
pOutputSamples[i*2+0] = (drflac_int16)((pInputSamples0U32[i] << shift0) >> 16);
pOutputSamples[i*2+1] = (drflac_int16)((pInputSamples1U32[i] << shift1) >> 16);
}
}
#if defined(DRFLAC_SUPPORT_SSE2)
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pO...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
for (i = 0; i < frameCount4; ++i) {
__m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0);
__m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1);
left = _mm_srai_epi32(left, 16);
right = _mm_srai_epi32(right, 16);
_mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
pOutputSamples[i*2+0] = (drflac_int16)((pInputSamples0U32[i] << shift0) >> 16);
pOutputSamples[i*2+1] = (drflac_int16)((pInputSamples1U32[i] << shift1) >> 16);
}
}
#endif
#if defined(DRFLAC_SUPPORT_NEON)
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pO...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
int32x4_t shift0_4 = vdupq_n_s32(shift0);
int32x4_t shift1_4 = vdupq_n_s32(shift1);
for (i = 0; i < frameCount4; ++i) {
int32x4_t left;
int32x4_t right;
left = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4));
right = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4));
left = vshrq_n_s32(left, 16);
right = vshrq_n_s32(right, 16);
drflac__vst2q_s16(pOutputSamples + i*8, vzip_s16(vmovn_s32(left), vmovn_s32(right)));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
pOutputSamples[i*2+0] = (drflac_int16)((pInputSamples0U32[i] << shift0) >> 16);
pOutputSamples[i*2+1] = (drflac_int16)((pInputSamples1U32[i] << shift1) >> 16);
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputS...
{
#if defined(DRFLAC_SUPPORT_SSE2)
if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s16__decode_independent_stereo__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_s16__decode_independent_stereo__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#endif
{
#if 0
drflac_read_pcm_frames_s16__decode_independent_stereo__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#else
drflac_read_pcm_frames_s16__decode_independent_stereo__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#endif
}
}
DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s16(drflac* pFlac, drflac_uint64 framesToRead, drflac_int16* pBufferOut)
{
drflac_uint64 framesRead;
drflac_uint32 unusedBitsPerSample;
if (pFlac == NULL || framesToRead == 0) {
return 0;
}
if (pBufferOut == NULL) {
return drflac__seek_forward_by_pcm_frames(pFlac, framesToRead);
}
DRFLAC_ASSERT(pFlac->bitsPerSample <= 32);
unusedBitsPerSample = 32 - pFlac->bitsPerSample;
framesRead = 0;
while (framesToRead > 0) {
if (pFlac->currentFLACFrame.pcmFramesRemaining == 0) {
if (!drflac__read_and_decode_next_flac_frame(pFlac)) {
break;
}
} else {
unsigned int channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment);
drflac_uint64 iFirstPCMFrame = pFlac->currentFLACFrame.header.blockSizeInPCMFrames - pFlac->currentFLACFrame.pcmFramesRemaining;
drflac_uint64 frameCountThisIteration = framesToRead;
if (frameCountThisIteration > pFlac->currentFLACFrame.pcmFramesRemaining) {
frameCountThisIteration = pFlac->currentFLACFrame.pcmFramesRemaining;
}
if (channelCount == 2) {
const drflac_int32* pDecodedSamples0 = pFlac->currentFLACFrame.subframes[0].pSamplesS32 + iFirstPCMFrame;
const drflac_int32* pDecodedSamples1 = pFlac->currentFLACFrame.subframes[1].pSamplesS32 + iFirstPCMFrame;
switch (pFlac->currentFLACFrame.header.channelAssignment)
{
case DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE:
{
drflac_read_pcm_frames_s16__decode_left_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut);
} break;
case DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE:
{
drflac_read_pcm_frames_s16__decode_right_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut);
} break;
case DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE:
{
drflac_read_pcm_frames_s16__decode_mid_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut);
} break;
case DRFLAC_CHANNEL_ASSIGNMENT_INDEPENDENT:
default:
{
drflac_read_pcm_frames_s16__decode_independent_stereo(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut);
} break;
}
} else {
drflac_uint64 i;
for (i = 0; i < frameCountThisIteration; ++i) {
unsigned int j;
for (j = 0; j < channelCount; ++j) {
drflac_int32 sampleS32 = (drflac_int32)((drflac_uint32)(pFlac->currentFLACFrame.subframes[j].pSamplesS32[iFirstPCMFrame + i]) << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[j].wastedBitsPerSample));
pBufferOut[(i*channelCount)+j] = (drflac_int16)(sampleS32 >> 16);
}
}
}
framesRead += frameCountThisIteration;
pBufferOut += frameCountThisIteration * channelCount;
framesToRead -= frameCountThisIteration;
pFlac->currentPCMFrame += frameCountThisIteration;
pFlac->currentFLACFrame.pcmFramesRemaining -= (drflac_uint32)frameCountThisIteration;
}
}
return framesRead;
}
#if 0
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSample...
{
drflac_uint64 i;
for (i = 0; i < frameCount; ++i) {
drflac_uint32 left = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
drflac_uint32 right = left - side;
pOutputSamples[i*2+0] = (float)((drflac_int32)left / 2147483648.0);
pOutputSamples[i*2+1] = (float)((drflac_int32)right / 2147483648.0);
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
float factor = 1 / 2147483648.0;
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 left0 = pInputSamples0U32[i*4+0] << shift0;
drflac_uint32 left1 = pInputSamples0U32[i*4+1] << shift0;
drflac_uint32 left2 = pInputSamples0U32[i*4+2] << shift0;
drflac_uint32 left3 = pInputSamples0U32[i*4+3] << shift0;
drflac_uint32 side0 = pInputSamples1U32[i*4+0] << shift1;
drflac_uint32 side1 = pInputSamples1U32[i*4+1] << shift1;
drflac_uint32 side2 = pInputSamples1U32[i*4+2] << shift1;
drflac_uint32 side3 = pInputSamples1U32[i*4+3] << shift1;
drflac_uint32 right0 = left0 - side0;
drflac_uint32 right1 = left1 - side1;
drflac_uint32 right2 = left2 - side2;
drflac_uint32 right3 = left3 - side3;
pOutputSamples[i*8+0] = (drflac_int32)left0 * factor;
pOutputSamples[i*8+1] = (drflac_int32)right0 * factor;
pOutputSamples[i*8+2] = (drflac_int32)left1 * factor;
pOutputSamples[i*8+3] = (drflac_int32)right1 * factor;
pOutputSamples[i*8+4] = (drflac_int32)left2 * factor;
pOutputSamples[i*8+5] = (drflac_int32)right2 * factor;
pOutputSamples[i*8+6] = (drflac_int32)left3 * factor;
pOutputSamples[i*8+7] = (drflac_int32)right3 * factor;
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 left = pInputSamples0U32[i] << shift0;
drflac_uint32 side = pInputSamples1U32[i] << shift1;
drflac_uint32 right = left - side;
pOutputSamples[i*2+0] = (drflac_int32)left * factor;
pOutputSamples[i*2+1] = (drflac_int32)right * factor;
}
}
#if defined(DRFLAC_SUPPORT_SSE2)
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8;
drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8;
__m128 factor;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
factor = _mm_set1_ps(1.0f / 8388608.0f);
for (i = 0; i < frameCount4; ++i) {
__m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0);
__m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1);
__m128i right = _mm_sub_epi32(left, side);
__m128 leftf = _mm_mul_ps(_mm_cvtepi32_ps(left), factor);
__m128 rightf = _mm_mul_ps(_mm_cvtepi32_ps(right), factor);
_mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf));
_mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 left = pInputSamples0U32[i] << shift0;
drflac_uint32 side = pInputSamples1U32[i] << shift1;
drflac_uint32 right = left - side;
pOutputSamples[i*2+0] = (drflac_int32)left / 8388608.0f;
pOutputSamples[i*2+1] = (drflac_int32)right / 8388608.0f;
}
}
#endif
#if defined(DRFLAC_SUPPORT_NEON)
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8;
drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8;
float32x4_t factor4;
int32x4_t shift0_4;
int32x4_t shift1_4;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
factor4 = vdupq_n_f32(1.0f / 8388608.0f);
shift0_4 = vdupq_n_s32(shift0);
shift1_4 = vdupq_n_s32(shift1);
for (i = 0; i < frameCount4; ++i) {
uint32x4_t left;
uint32x4_t side;
uint32x4_t right;
float32x4_t leftf;
float32x4_t rightf;
left = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4);
side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4);
right = vsubq_u32(left, side);
leftf = vmulq_f32(vcvtq_f32_s32(vreinterpretq_s32_u32(left)), factor4);
rightf = vmulq_f32(vcvtq_f32_s32(vreinterpretq_s32_u32(right)), factor4);
drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 left = pInputSamples0U32[i] << shift0;
drflac_uint32 side = pInputSamples1U32[i] << shift1;
drflac_uint32 right = left - side;
pOutputSamples[i*2+0] = (drflac_int32)left / 8388608.0f;
pOutputSamples[i*2+1] = (drflac_int32)right / 8388608.0f;
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
#if defined(DRFLAC_SUPPORT_SSE2)
if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_f32__decode_left_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_f32__decode_left_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#endif
{
#if 0
drflac_read_pcm_frames_f32__decode_left_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#else
drflac_read_pcm_frames_f32__decode_left_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#endif
}
}
#if 0
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSampl...
{
drflac_uint64 i;
for (i = 0; i < frameCount; ++i) {
drflac_uint32 side = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
drflac_uint32 right = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
drflac_uint32 left = right + side;
pOutputSamples[i*2+0] = (float)((drflac_int32)left / 2147483648.0);
pOutputSamples[i*2+1] = (float)((drflac_int32)right / 2147483648.0);
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
float factor = 1 / 2147483648.0;
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 side0 = pInputSamples0U32[i*4+0] << shift0;
drflac_uint32 side1 = pInputSamples0U32[i*4+1] << shift0;
drflac_uint32 side2 = pInputSamples0U32[i*4+2] << shift0;
drflac_uint32 side3 = pInputSamples0U32[i*4+3] << shift0;
drflac_uint32 right0 = pInputSamples1U32[i*4+0] << shift1;
drflac_uint32 right1 = pInputSamples1U32[i*4+1] << shift1;
drflac_uint32 right2 = pInputSamples1U32[i*4+2] << shift1;
drflac_uint32 right3 = pInputSamples1U32[i*4+3] << shift1;
drflac_uint32 left0 = right0 + side0;
drflac_uint32 left1 = right1 + side1;
drflac_uint32 left2 = right2 + side2;
drflac_uint32 left3 = right3 + side3;
pOutputSamples[i*8+0] = (drflac_int32)left0 * factor;
pOutputSamples[i*8+1] = (drflac_int32)right0 * factor;
pOutputSamples[i*8+2] = (drflac_int32)left1 * factor;
pOutputSamples[i*8+3] = (drflac_int32)right1 * factor;
pOutputSamples[i*8+4] = (drflac_int32)left2 * factor;
pOutputSamples[i*8+5] = (drflac_int32)right2 * factor;
pOutputSamples[i*8+6] = (drflac_int32)left3 * factor;
pOutputSamples[i*8+7] = (drflac_int32)right3 * factor;
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 side = pInputSamples0U32[i] << shift0;
drflac_uint32 right = pInputSamples1U32[i] << shift1;
drflac_uint32 left = right + side;
pOutputSamples[i*2+0] = (drflac_int32)left * factor;
pOutputSamples[i*2+1] = (drflac_int32)right * factor;
}
}
#if defined(DRFLAC_SUPPORT_SSE2)
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8;
drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8;
__m128 factor;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
factor = _mm_set1_ps(1.0f / 8388608.0f);
for (i = 0; i < frameCount4; ++i) {
__m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0);
__m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1);
__m128i left = _mm_add_epi32(right, side);
__m128 leftf = _mm_mul_ps(_mm_cvtepi32_ps(left), factor);
__m128 rightf = _mm_mul_ps(_mm_cvtepi32_ps(right), factor);
_mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf));
_mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 side = pInputSamples0U32[i] << shift0;
drflac_uint32 right = pInputSamples1U32[i] << shift1;
drflac_uint32 left = right + side;
pOutputSamples[i*2+0] = (drflac_int32)left / 8388608.0f;
pOutputSamples[i*2+1] = (drflac_int32)right / 8388608.0f;
}
}
#endif
#if defined(DRFLAC_SUPPORT_NEON)
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8;
drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8;
float32x4_t factor4;
int32x4_t shift0_4;
int32x4_t shift1_4;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
factor4 = vdupq_n_f32(1.0f / 8388608.0f);
shift0_4 = vdupq_n_s32(shift0);
shift1_4 = vdupq_n_s32(shift1);
for (i = 0; i < frameCount4; ++i) {
uint32x4_t side;
uint32x4_t right;
uint32x4_t left;
float32x4_t leftf;
float32x4_t rightf;
side = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4);
right = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4);
left = vaddq_u32(right, side);
leftf = vmulq_f32(vcvtq_f32_s32(vreinterpretq_s32_u32(left)), factor4);
rightf = vmulq_f32(vcvtq_f32_s32(vreinterpretq_s32_u32(right)), factor4);
drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 side = pInputSamples0U32[i] << shift0;
drflac_uint32 right = pInputSamples1U32[i] << shift1;
drflac_uint32 left = right + side;
pOutputSamples[i*2+0] = (drflac_int32)left / 8388608.0f;
pOutputSamples[i*2+1] = (drflac_int32)right / 8388608.0f;
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
#if defined(DRFLAC_SUPPORT_SSE2)
if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_f32__decode_right_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_f32__decode_right_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#endif
{
#if 0
drflac_read_pcm_frames_f32__decode_right_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#else
drflac_read_pcm_frames_f32__decode_right_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#endif
}
}
#if 0
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples...
{
for (drflac_uint64 i = 0; i < frameCount; ++i) {
drflac_uint32 mid = (drflac_uint32)pInputSamples0[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (float)((((drflac_int32)(mid + side) >> 1) << (unusedBitsPerSample)) / 2147483648.0);
pOutputSamples[i*2+1] = (float)((((drflac_int32)(mid - side) >> 1) << (unusedBitsPerSample)) / 2147483648.0);
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift = unusedBitsPerSample;
float factor = 1 / 2147483648.0;
if (shift > 0) {
shift -= 1;
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 temp0L;
drflac_uint32 temp1L;
drflac_uint32 temp2L;
drflac_uint32 temp3L;
drflac_uint32 temp0R;
drflac_uint32 temp1R;
drflac_uint32 temp2R;
drflac_uint32 temp3R;
drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid0 = (mid0 << 1) | (side0 & 0x01);
mid1 = (mid1 << 1) | (side1 & 0x01);
mid2 = (mid2 << 1) | (side2 & 0x01);
mid3 = (mid3 << 1) | (side3 & 0x01);
temp0L = (mid0 + side0) << shift;
temp1L = (mid1 + side1) << shift;
temp2L = (mid2 + side2) << shift;
temp3L = (mid3 + side3) << shift;
temp0R = (mid0 - side0) << shift;
temp1R = (mid1 - side1) << shift;
temp2R = (mid2 - side2) << shift;
temp3R = (mid3 - side3) << shift;
pOutputSamples[i*8+0] = (drflac_int32)temp0L * factor;
pOutputSamples[i*8+1] = (drflac_int32)temp0R * factor;
pOutputSamples[i*8+2] = (drflac_int32)temp1L * factor;
pOutputSamples[i*8+3] = (drflac_int32)temp1R * factor;
pOutputSamples[i*8+4] = (drflac_int32)temp2L * factor;
pOutputSamples[i*8+5] = (drflac_int32)temp2R * factor;
pOutputSamples[i*8+6] = (drflac_int32)temp3L * factor;
pOutputSamples[i*8+7] = (drflac_int32)temp3R * factor;
}
} else {
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 temp0L;
drflac_uint32 temp1L;
drflac_uint32 temp2L;
drflac_uint32 temp3L;
drflac_uint32 temp0R;
drflac_uint32 temp1R;
drflac_uint32 temp2R;
drflac_uint32 temp3R;
drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid0 = (mid0 << 1) | (side0 & 0x01);
mid1 = (mid1 << 1) | (side1 & 0x01);
mid2 = (mid2 << 1) | (side2 & 0x01);
mid3 = (mid3 << 1) | (side3 & 0x01);
temp0L = (drflac_uint32)((drflac_int32)(mid0 + side0) >> 1);
temp1L = (drflac_uint32)((drflac_int32)(mid1 + side1) >> 1);
temp2L = (drflac_uint32)((drflac_int32)(mid2 + side2) >> 1);
temp3L = (drflac_uint32)((drflac_int32)(mid3 + side3) >> 1);
temp0R = (drflac_uint32)((drflac_int32)(mid0 - side0) >> 1);
temp1R = (drflac_uint32)((drflac_int32)(mid1 - side1) >> 1);
temp2R = (drflac_uint32)((drflac_int32)(mid2 - side2) >> 1);
temp3R = (drflac_uint32)((drflac_int32)(mid3 - side3) >> 1);
pOutputSamples[i*8+0] = (drflac_int32)temp0L * factor;
pOutputSamples[i*8+1] = (drflac_int32)temp0R * factor;
pOutputSamples[i*8+2] = (drflac_int32)temp1L * factor;
pOutputSamples[i*8+3] = (drflac_int32)temp1R * factor;
pOutputSamples[i*8+4] = (drflac_int32)temp2L * factor;
pOutputSamples[i*8+5] = (drflac_int32)temp2R * factor;
pOutputSamples[i*8+6] = (drflac_int32)temp3L * factor;
pOutputSamples[i*8+7] = (drflac_int32)temp3R * factor;
}
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample) * factor;
pOutputSamples[i*2+1] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample) * factor;
}
}
#if defined(DRFLAC_SUPPORT_SSE2)
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift = unusedBitsPerSample - 8;
float factor;
__m128 factor128;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
factor = 1.0f / 8388608.0f;
factor128 = _mm_set1_ps(factor);
if (shift == 0) {
for (i = 0; i < frameCount4; ++i) {
__m128i mid;
__m128i side;
__m128i tempL;
__m128i tempR;
__m128 leftf;
__m128 rightf;
mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01)));
tempL = _mm_srai_epi32(_mm_add_epi32(mid, side), 1);
tempR = _mm_srai_epi32(_mm_sub_epi32(mid, side), 1);
leftf = _mm_mul_ps(_mm_cvtepi32_ps(tempL), factor128);
rightf = _mm_mul_ps(_mm_cvtepi32_ps(tempR), factor128);
_mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf));
_mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = ((drflac_int32)(mid + side) >> 1) * factor;
pOutputSamples[i*2+1] = ((drflac_int32)(mid - side) >> 1) * factor;
}
} else {
shift -= 1;
for (i = 0; i < frameCount4; ++i) {
__m128i mid;
__m128i side;
__m128i tempL;
__m128i tempR;
__m128 leftf;
__m128 rightf;
mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01)));
tempL = _mm_slli_epi32(_mm_add_epi32(mid, side), shift);
tempR = _mm_slli_epi32(_mm_sub_epi32(mid, side), shift);
leftf = _mm_mul_ps(_mm_cvtepi32_ps(tempL), factor128);
rightf = _mm_mul_ps(_mm_cvtepi32_ps(tempR), factor128);
_mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf));
_mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int32)((mid + side) << shift) * factor;
pOutputSamples[i*2+1] = (drflac_int32)((mid - side) << shift) * factor;
}
}
}
#endif
#if defined(DRFLAC_SUPPORT_NEON)
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift = unusedBitsPerSample - 8;
float factor;
float32x4_t factor4;
int32x4_t shift4;
int32x4_t wbps0_4;
int32x4_t wbps1_4;
DRFLAC_ASSERT(pFlac->bitsPerSample <= 24);
factor = 1.0f / 8388608.0f;
factor4 = vdupq_n_f32(factor);
wbps0_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample);
wbps1_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample);
if (shift == 0) {
for (i = 0; i < frameCount4; ++i) {
int32x4_t lefti;
int32x4_t righti;
float32x4_t leftf;
float32x4_t rightf;
uint32x4_t mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbps0_4);
uint32x4_t side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbps1_4);
mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, vdupq_n_u32(1)));
lefti = vshrq_n_s32(vreinterpretq_s32_u32(vaddq_u32(mid, side)), 1);
righti = vshrq_n_s32(vreinterpretq_s32_u32(vsubq_u32(mid, side)), 1);
leftf = vmulq_f32(vcvtq_f32_s32(lefti), factor4);
rightf = vmulq_f32(vcvtq_f32_s32(righti), factor4);
drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = ((drflac_int32)(mid + side) >> 1) * factor;
pOutputSamples[i*2+1] = ((drflac_int32)(mid - side) >> 1) * factor;
}
} else {
shift -= 1;
shift4 = vdupq_n_s32(shift);
for (i = 0; i < frameCount4; ++i) {
uint32x4_t mid;
uint32x4_t side;
int32x4_t lefti;
int32x4_t righti;
float32x4_t leftf;
float32x4_t rightf;
mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbps0_4);
side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbps1_4);
mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, vdupq_n_u32(1)));
lefti = vreinterpretq_s32_u32(vshlq_u32(vaddq_u32(mid, side), shift4));
righti = vreinterpretq_s32_u32(vshlq_u32(vsubq_u32(mid, side), shift4));
leftf = vmulq_f32(vcvtq_f32_s32(lefti), factor4);
rightf = vmulq_f32(vcvtq_f32_s32(righti), factor4);
drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
mid = (mid << 1) | (side & 0x01);
pOutputSamples[i*2+0] = (drflac_int32)((mid + side) << shift) * factor;
pOutputSamples[i*2+1] = (drflac_int32)((mid - side) << shift) * factor;
}
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
#if defined(DRFLAC_SUPPORT_SSE2)
if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_f32__decode_mid_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_f32__decode_mid_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#endif
{
#if 0
drflac_read_pcm_frames_f32__decode_mid_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#else
drflac_read_pcm_frames_f32__decode_mid_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#endif
}
}
#if 0
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOut...
{
for (drflac_uint64 i = 0; i < frameCount; ++i) {
pOutputSamples[i*2+0] = (float)((drflac_int32)((drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample)) / 2147483648.0);
pOutputSamples[i*2+1] = (float)((drflac_int32)((drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample)) / 2147483648.0);
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutput...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample;
drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample;
float factor = 1 / 2147483648.0;
for (i = 0; i < frameCount4; ++i) {
drflac_uint32 tempL0 = pInputSamples0U32[i*4+0] << shift0;
drflac_uint32 tempL1 = pInputSamples0U32[i*4+1] << shift0;
drflac_uint32 tempL2 = pInputSamples0U32[i*4+2] << shift0;
drflac_uint32 tempL3 = pInputSamples0U32[i*4+3] << shift0;
drflac_uint32 tempR0 = pInputSamples1U32[i*4+0] << shift1;
drflac_uint32 tempR1 = pInputSamples1U32[i*4+1] << shift1;
drflac_uint32 tempR2 = pInputSamples1U32[i*4+2] << shift1;
drflac_uint32 tempR3 = pInputSamples1U32[i*4+3] << shift1;
pOutputSamples[i*8+0] = (drflac_int32)tempL0 * factor;
pOutputSamples[i*8+1] = (drflac_int32)tempR0 * factor;
pOutputSamples[i*8+2] = (drflac_int32)tempL1 * factor;
pOutputSamples[i*8+3] = (drflac_int32)tempR1 * factor;
pOutputSamples[i*8+4] = (drflac_int32)tempL2 * factor;
pOutputSamples[i*8+5] = (drflac_int32)tempR2 * factor;
pOutputSamples[i*8+6] = (drflac_int32)tempL3 * factor;
pOutputSamples[i*8+7] = (drflac_int32)tempR3 * factor;
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0) * factor;
pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1) * factor;
}
}
#if defined(DRFLAC_SUPPORT_SSE2)
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSa...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8;
drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8;
float factor = 1.0f / 8388608.0f;
__m128 factor128 = _mm_set1_ps(factor);
for (i = 0; i < frameCount4; ++i) {
__m128i lefti;
__m128i righti;
__m128 leftf;
__m128 rightf;
lefti = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0);
righti = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1);
leftf = _mm_mul_ps(_mm_cvtepi32_ps(lefti), factor128);
rightf = _mm_mul_ps(_mm_cvtepi32_ps(righti), factor128);
_mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf));
_mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0) * factor;
pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1) * factor;
}
}
#endif
#if defined(DRFLAC_SUPPORT_NEON)
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSa...
{
drflac_uint64 i;
drflac_uint64 frameCount4 = frameCount >> 2;
const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0;
const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1;
drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8;
drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8;
float factor = 1.0f / 8388608.0f;
float32x4_t factor4 = vdupq_n_f32(factor);
int32x4_t shift0_4 = vdupq_n_s32(shift0);
int32x4_t shift1_4 = vdupq_n_s32(shift1);
for (i = 0; i < frameCount4; ++i) {
int32x4_t lefti;
int32x4_t righti;
float32x4_t leftf;
float32x4_t rightf;
lefti = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4));
righti = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4));
leftf = vmulq_f32(vcvtq_f32_s32(lefti), factor4);
rightf = vmulq_f32(vcvtq_f32_s32(righti), factor4);
drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf));
}
for (i = (frameCount4 << 2); i < frameCount; ++i) {
pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0) * factor;
pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1) * factor;
}
}
#endif
static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples)
{
#if defined(DRFLAC_SUPPORT_SSE2)
if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_f32__decode_independent_stereo__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#elif defined(DRFLAC_SUPPORT_NEON)
if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) {
drflac_read_pcm_frames_f32__decode_independent_stereo__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
} else
#endif
{
#if 0
drflac_read_pcm_frames_f32__decode_independent_stereo__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#else
drflac_read_pcm_frames_f32__decode_independent_stereo__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples);
#endif
}
}
DRFLAC_API drflac_uint64 drflac_read_pcm_frames_f32(drflac* pFlac, drflac_uint64 framesToRead, float* pBufferOut)
{
drflac_uint64 framesRead;
drflac_uint32 unusedBitsPerSample;
if (pFlac == NULL || framesToRead == 0) {
return 0;
}
if (pBufferOut == NULL) {
return drflac__seek_forward_by_pcm_frames(pFlac, framesToRead);
}
DRFLAC_ASSERT(pFlac->bitsPerSample <= 32);
unusedBitsPerSample = 32 - pFlac->bitsPerSample;
framesRead = 0;
while (framesToRead > 0) {
if (pFlac->currentFLACFrame.pcmFramesRemaining == 0) {
if (!drflac__read_and_decode_next_flac_frame(pFlac)) {
break;
}
} else {
unsigned int channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment);
drflac_uint64 iFirstPCMFrame = pFlac->currentFLACFrame.header.blockSizeInPCMFrames - pFlac->currentFLACFrame.pcmFramesRemaining;
drflac_uint64 frameCountThisIteration = framesToRead;
if (frameCountThisIteration > pFlac->currentFLACFrame.pcmFramesRemaining) {
frameCountThisIteration = pFlac->currentFLACFrame.pcmFramesRemaining;
}
if (channelCount == 2) {
const drflac_int32* pDecodedSamples0 = pFlac->currentFLACFrame.subframes[0].pSamplesS32 + iFirstPCMFrame;
const drflac_int32* pDecodedSamples1 = pFlac->currentFLACFrame.subframes[1].pSamplesS32 + iFirstPCMFrame;
switch (pFlac->currentFLACFrame.header.channelAssignment)
{
case DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE:
{
drflac_read_pcm_frames_f32__decode_left_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut);
} break;
case DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE:
{
drflac_read_pcm_frames_f32__decode_right_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut);
} break;
case DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE:
{
drflac_read_pcm_frames_f32__decode_mid_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut);
} break;
case DRFLAC_CHANNEL_ASSIGNMENT_INDEPENDENT:
default:
{
drflac_read_pcm_frames_f32__decode_independent_stereo(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut);
} break;
}
} else {
drflac_uint64 i;
for (i = 0; i < frameCountThisIteration; ++i) {
unsigned int j;
for (j = 0; j < channelCount; ++j) {
drflac_int32 sampleS32 = (drflac_int32)((drflac_uint32)(pFlac->currentFLACFrame.subframes[j].pSamplesS32[iFirstPCMFrame + i]) << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[j].wastedBitsPerSample));
pBufferOut[(i*channelCount)+j] = (float)(sampleS32 / 2147483648.0);
}
}
}
framesRead += frameCountThisIteration;
pBufferOut += frameCountThisIteration * channelCount;
framesToRead -= frameCountThisIteration;
pFlac->currentPCMFrame += frameCountThisIteration;
pFlac->currentFLACFrame.pcmFramesRemaining -= (unsigned int)frameCountThisIteration;
}
}
return framesRead;
}
DRFLAC_API drflac_bool32 drflac_seek_to_pcm_frame(drflac* pFlac, drflac_uint64 pcmFrameIndex)
{
if (pFlac == NULL) {
return DRFLAC_FALSE;
}
if (pFlac->currentPCMFrame == pcmFrameIndex) {
return DRFLAC_TRUE;
}
if (pFlac->firstFLACFramePosInBytes == 0) {
return DRFLAC_FALSE;
}
if (pcmFrameIndex == 0) {
pFlac->currentPCMFrame = 0;
return drflac__seek_to_first_frame(pFlac);
} else {
drflac_bool32 wasSuccessful = DRFLAC_FALSE;
drflac_uint64 originalPCMFrame = pFlac->currentPCMFrame;
if (pcmFrameIndex > pFlac->totalPCMFrameCount) {
pcmFrameIndex = pFlac->totalPCMFrameCount;
}
if (pcmFrameIndex > pFlac->currentPCMFrame) {
drflac_uint32 offset = (drflac_uint32)(pcmFrameIndex - pFlac->currentPCMFrame);
if (pFlac->currentFLACFrame.pcmFramesRemaining > offset) {
pFlac->currentFLACFrame.pcmFramesRemaining -= offset;
pFlac->currentPCMFrame = pcmFrameIndex;
return DRFLAC_TRUE;
}
} else {
drflac_uint32 offsetAbs = (drflac_uint32)(pFlac->currentPCMFrame - pcmFrameIndex);
drflac_uint32 currentFLACFramePCMFrameCount = pFlac->currentFLACFrame.header.blockSizeInPCMFrames;
drflac_uint32 currentFLACFramePCMFramesConsumed = currentFLACFramePCMFrameCount - pFlac->currentFLACFrame.pcmFramesRemaining;
if (currentFLACFramePCMFramesConsumed > offsetAbs) {
pFlac->currentFLACFrame.pcmFramesRemaining += offsetAbs;
pFlac->currentPCMFrame = pcmFrameIndex;
return DRFLAC_TRUE;
}
}
#ifndef DR_FLAC_NO_OGG
if (pFlac->container == drflac_container_ogg)
{
wasSuccessful = drflac_ogg__seek_to_pcm_frame(pFlac, pcmFrameIndex);
}
else
#endif
{
if (!pFlac->_noSeekTableSeek) {
wasSuccessful = drflac__seek_to_pcm_frame__seek_table(pFlac, pcmFrameIndex);
}
#if !defined(DR_FLAC_NO_CRC)
if (!wasSuccessful && !pFlac->_noBinarySearchSeek && pFlac->totalPCMFrameCount > 0) {
wasSuccessful = drflac__seek_to_pcm_frame__binary_search(pFlac, pcmFrameIndex);
}
#endif
if (!wasSuccessful && !pFlac->_noBruteForceSeek) {
wasSuccessful = drflac__seek_to_pcm_frame__brute_force(pFlac, pcmFrameIndex);
}
}
if (wasSuccessful) {
pFlac->currentPCMFrame = pcmFrameIndex;
} else {
if (drflac_seek_to_pcm_frame(pFlac, originalPCMFrame) == DRFLAC_FALSE) {
drflac_seek_to_pcm_frame(pFlac, 0);
}
}
return wasSuccessful;
}
}
#if defined(SIZE_MAX)
#define DRFLAC_SIZE_MAX SIZE_MAX
#else
#if defined(DRFLAC_64BIT)
#define DRFLAC_SIZE_MAX ((drflac_uint64)0xFFFFFFFFFFFFFFFF)
#else
#define DRFLAC_SIZE_MAX 0xFFFFFFFF
#endif
#endif
#define DRFLAC_DEFINE_FULL_READ_AND_CLOSE(extension, type) \
static type* drflac__full_read_and_close_ ## extension (drflac* pFlac, unsigned int* channelsOut, unsigned int* sampleRateOut, drflac_uint64* totalPCMFrameCountOut)\
{ \
type* pSampleData = NULL; \
drflac_uint64 totalPCMFrameCount; \
\
DRFLAC_ASSERT(pFlac != NULL); \
\
totalPCMFrameCount = pFlac->totalPCMFrameCount; \
\
if (totalPCMFrameCount == 0) { \
type buffer[4096]; \
drflac_uint64 pcmFramesRead; \
size_t sampleDataBufferSize = sizeof(buffer); \
\
pSampleData = (type*)drflac__malloc_from_callbacks(sampleDataBufferSize, &pFlac->allocationCallbacks); \
if (pSampleData == NULL) { \
goto on_error; \
} \
\
while ((pcmFramesRead = (drflac_uint64)drflac_read_pcm_frames_##extension(pFlac, sizeof(buffer)/sizeof(buffer[0])/pFlac->channels, buffer)) > 0) { \
if (((totalPCMFrameCount + pcmFramesRead) * pFlac->channels * sizeof(type)) > sampleDataBufferSize) { \
type* pNewSampleData; \
size_t newSampleDataBufferSize; \
\
newSampleDataBufferSize = sampleDataBufferSize * 2; \
pNewSampleData = (type*)drflac__realloc_from_callbacks(pSampleData, newSampleDataBufferSize, sampleDataBufferSize, &pFlac->allocationCallbacks); \
if (pNewSampleData == NULL) { \
drflac__free_from_callbacks(pSampleData, &pFlac->allocationCallbacks); \
goto on_error; \
} \
\
sampleDataBufferSize = newSampleDataBufferSize; \
pSampleData = pNewSampleData; \
} \
\
DRFLAC_COPY_MEMORY(pSampleData + (totalPCMFrameCount*pFlac->channels), buffer, (size_t)(pcmFramesRead*pFlac->channels*sizeof(type))); \
totalPCMFrameCount += pcmFramesRead; \
} \
\
\
DRFLAC_ZERO_MEMORY(pSampleData + (totalPCMFrameCount*pFlac->channels), (size_t)(sampleDataBufferSize - totalPCMFrameCount*pFlac->channels*sizeof(type))); \
} else { \
drflac_uint64 dataSize = totalPCMFrameCount*pFlac->channels*sizeof(type); \
if (dataSize > (drflac_uint64)DRFLAC_SIZE_MAX) { \
goto on_error; \
} \
\
pSampleData = (type*)drflac__malloc_from_callbacks((size_t)dataSize, &pFlac->allocationCallbacks); \
if (pSampleData == NULL) { \
goto on_error; \
} \
\
totalPCMFrameCount = drflac_read_pcm_frames_##extension(pFlac, pFlac->totalPCMFrameCount, pSampleData); \
} \
\
if (sampleRateOut) *sampleRateOut = pFlac->sampleRate; \
if (channelsOut) *channelsOut = pFlac->channels; \
if (totalPCMFrameCountOut) *totalPCMFrameCountOut = totalPCMFrameCount; \
\
drflac_close(pFlac); \
return pSampleData; \
\
on_error: \
drflac_close(pFlac); \
return NULL; \
}
DRFLAC_DEFINE_FULL_READ_AND_CLOSE(s32, drflac_int32)
DRFLAC_DEFINE_FULL_READ_AND_CLOSE(s16, drflac_int16)
DRFLAC_DEFINE_FULL_READ_AND_CLOSE(f32, float)
DRFLAC_API drflac_int32* drflac_open_and_read_pcm_frames_s32(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drflac_uint64* totalPCMFrameCountOut, const drflac_allocation_call...
{
drflac* pFlac;
if (channelsOut) {
*channelsOut = 0;
}
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (totalPCMFrameCountOut) {
*totalPCMFrameCountOut = 0;
}
pFlac = drflac_open(onRead, onSeek, pUserData, pAllocationCallbacks);
if (pFlac == NULL) {
return NULL;
}
return drflac__full_read_and_close_s32(pFlac, channelsOut, sampleRateOut, totalPCMFrameCountOut);
}
DRFLAC_API drflac_int16* drflac_open_and_read_pcm_frames_s16(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drflac_uint64* totalPCMFrameCountOut, const drflac_allocation_call...
{
drflac* pFlac;
if (channelsOut) {
*channelsOut = 0;
}
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (totalPCMFrameCountOut) {
*totalPCMFrameCountOut = 0;
}
pFlac = drflac_open(onRead, onSeek, pUserData, pAllocationCallbacks);
if (pFlac == NULL) {
return NULL;
}
return drflac__full_read_and_close_s16(pFlac, channelsOut, sampleRateOut, totalPCMFrameCountOut);
}
DRFLAC_API float* drflac_open_and_read_pcm_frames_f32(drflac_read_proc onRead, drflac_seek_proc onSeek, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drflac_uint64* totalPCMFrameCountOut, const drflac_allocation_callbacks* ...
{
drflac* pFlac;
if (channelsOut) {
*channelsOut = 0;
}
if (sampleRateOut) {
*sampleRateOut = 0;
}
if (totalPCMFrameCountOut) {
*totalPCMFrameCountOut = 0;
}
pFlac = drflac_open(onRead, onSeek, pUserData, pAllocationCallbacks);
if (pFlac == NULL) {
return NULL;
}
return drflac__full_read_and_close_f32(pFlac, channelsOut, sampleRateOut, totalPCMFrameCountOut);
}
#ifndef DR_FLAC_NO_STDIO
DRFLAC_API drflac_int32* drflac_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks)
{
drflac* pFlac;
if (sampleRate) {
*sampleRate = 0;
}
if (channels) {
*channels = 0;
}
if (totalPCMFrameCount) {
*totalPCMFrameCount = 0;
}
pFlac = drflac_open_file(filename, pAllocationCallbacks);
if (pFlac == NULL) {
return NULL;
}
return drflac__full_read_and_close_s32(pFlac, channels, sampleRate, totalPCMFrameCount);
}
DRFLAC_API drflac_int16* drflac_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks)
{
drflac* pFlac;
if (sampleRate) {
*sampleRate = 0;
}
if (channels) {
*channels = 0;
}
if (totalPCMFrameCount) {
*totalPCMFrameCount = 0;
}
pFlac = drflac_open_file(filename, pAllocationCallbacks);
if (pFlac == NULL) {
return NULL;
}
return drflac__full_read_and_close_s16(pFlac, channels, sampleRate, totalPCMFrameCount);
}
DRFLAC_API float* drflac_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks)
{
drflac* pFlac;
if (sampleRate) {
*sampleRate = 0;
}
if (channels) {
*channels = 0;
}
if (totalPCMFrameCount) {
*totalPCMFrameCount = 0;
}
pFlac = drflac_open_file(filename, pAllocationCallbacks);
if (pFlac == NULL) {
return NULL;
}
return drflac__full_read_and_close_f32(pFlac, channels, sampleRate, totalPCMFrameCount);
}
#endif
DRFLAC_API drflac_int32* drflac_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks)
{
drflac* pFlac;
if (sampleRate) {
*sampleRate = 0;
}
if (channels) {
*channels = 0;
}
if (totalPCMFrameCount) {
*totalPCMFrameCount = 0;
}
pFlac = drflac_open_memory(data, dataSize, pAllocationCallbacks);
if (pFlac == NULL) {
return NULL;
}
return drflac__full_read_and_close_s32(pFlac, channels, sampleRate, totalPCMFrameCount);
}
DRFLAC_API drflac_int16* drflac_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks)
{
drflac* pFlac;
if (sampleRate) {
*sampleRate = 0;
}
if (channels) {
*channels = 0;
}
if (totalPCMFrameCount) {
*totalPCMFrameCount = 0;
}
pFlac = drflac_open_memory(data, dataSize, pAllocationCallbacks);
if (pFlac == NULL) {
return NULL;
}
return drflac__full_read_and_close_s16(pFlac, channels, sampleRate, totalPCMFrameCount);
}
DRFLAC_API float* drflac_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks)
{
drflac* pFlac;
if (sampleRate) {
*sampleRate = 0;
}
if (channels) {
*channels = 0;
}
if (totalPCMFrameCount) {
*totalPCMFrameCount = 0;
}
pFlac = drflac_open_memory(data, dataSize, pAllocationCallbacks);
if (pFlac == NULL) {
return NULL;
}
return drflac__full_read_and_close_f32(pFlac, channels, sampleRate, totalPCMFrameCount);
}
DRFLAC_API void drflac_free(void* p, const drflac_allocation_callbacks* pAllocationCallbacks)
{
if (pAllocationCallbacks != NULL) {
drflac__free_from_callbacks(p, pAllocationCallbacks);
} else {
drflac__free_default(p, NULL);
}
}
DRFLAC_API void drflac_init_vorbis_comment_iterator(drflac_vorbis_comment_iterator* pIter, drflac_uint32 commentCount, const void* pComments)
{
if (pIter == NULL) {
return;
}
pIter->countRemaining = commentCount;
pIter->pRunningData = (const char*)pComments;
}
DRFLAC_API const char* drflac_next_vorbis_comment(drflac_vorbis_comment_iterator* pIter, drflac_uint32* pCommentLengthOut)
{
drflac_int32 length;
const char* pComment;
if (pCommentLengthOut) {
*pCommentLengthOut = 0;
}
if (pIter == NULL || pIter->countRemaining == 0 || pIter->pRunningData == NULL) {
return NULL;
}
length = drflac__le2host_32_ptr_unaligned(pIter->pRunningData);
pIter->pRunningData += 4;
pComment = pIter->pRunningData;
pIter->pRunningData += length;
pIter->countRemaining -= 1;
if (pCommentLengthOut) {
*pCommentLengthOut = length;
}
return pComment;
}
DRFLAC_API void drflac_init_cuesheet_track_iterator(drflac_cuesheet_track_iterator* pIter, drflac_uint32 trackCount, const void* pTrackData)
{
if (pIter == NULL) {
return;
}
pIter->countRemaining = trackCount;
pIter->pRunningData = (const char*)pTrackData;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
drmp3_uint8 table_select[3], region_count[3], subblock_gain[3];
drmp3_uint8 preflag, scalefac_scale, count1_table, scfsi;
} drmp3_L3_gr_info;
typedef struct
{
drmp3_bs bs;
drmp3_uint8 maindata[DRMP3_MAX_BITRESERVOIR_BYTES + DRMP3_MAX_L3_FRAME_PAYLOAD_BYTES];
drmp3_L3_gr_info gr_info[4];
float grbuf[2][576], scf[40], syn[18 + 15][2*32];
drmp3_uint8 ist_pos[2][39];
} drmp3dec_scratch;
static void drmp3_bs_init(drmp3_bs *bs, const drmp3_uint8 *data, int bytes)
{
bs->buf = data;
bs->pos = 0;
bs->limit = bytes*8;
}
static drmp3_uint32 drmp3_bs_get_bits(drmp3_bs *bs, int n)
{
drmp3_uint32 next, cache = 0, s = bs->pos & 7;
int shl = n + s;
const drmp3_uint8 *p = bs->buf + (bs->pos >> 3);
if ((bs->pos += n) > bs->limit)
return 0;
next = *p++ & (255 >> s);
while ((shl -= 8) > 0)
{
cache |= next << shl;
next = *p++;
}
return cache | (next >> -shl);
}
static int drmp3_hdr_valid(const drmp3_uint8 *h)
{
return h[0] == 0xff &&
((h[1] & 0xF0) == 0xf0 || (h[1] & 0xFE) == 0xe2) &&
(DRMP3_HDR_GET_LAYER(h) != 0) &&
(DRMP3_HDR_GET_BITRATE(h) != 15) &&
(DRMP3_HDR_GET_SAMPLE_RATE(h) != 3);
}
static int drmp3_hdr_compare(const drmp3_uint8 *h1, const drmp3_uint8 *h2)
{
return drmp3_hdr_valid(h2) &&
((h1[1] ^ h2[1]) & 0xFE) == 0 &&
((h1[2] ^ h2[2]) & 0x0C) == 0 &&
!(DRMP3_HDR_IS_FREE_FORMAT(h1) ^ DRMP3_HDR_IS_FREE_FORMAT(h2));
}
static unsigned drmp3_hdr_bitrate_kbps(const drmp3_uint8 *h)
{
static const drmp3_uint8 halfrate[2][3][15] = {
{ { 0,4,8,12,16,20,24,28,32,40,48,56,64,72,80 }, { 0,4,8,12,16,20,24,28,32,40,48,56,64,72,80 }, { 0,16,24,28,32,40,48,56,64,72,80,88,96,112,128 } },
{ { 0,16,20,24,28,32,40,48,56,64,80,96,112,128,160 }, { 0,16,24,28,32,40,48,56,64,80,96,112,128,160,192 }, { 0,16,32,48,64,80,96,112,128,144,160,176,192,208,224 } },
};
return 2*halfrate[!!DRMP3_HDR_TEST_MPEG1(h)][DRMP3_HDR_GET_LAYER(h) - 1][DRMP3_HDR_GET_BITRATE(h)];
}
static unsigned drmp3_hdr_sample_rate_hz(const drmp3_uint8 *h)
{
static const unsigned g_hz[3] = { 44100, 48000, 32000 };
return g_hz[DRMP3_HDR_GET_SAMPLE_RATE(h)] >> (int)!DRMP3_HDR_TEST_MPEG1(h) >> (int)!DRMP3_HDR_TEST_NOT_MPEG25(h);
}
static unsigned drmp3_hdr_frame_samples(const drmp3_uint8 *h)
{
return DRMP3_HDR_IS_LAYER_1(h) ? 384 : (1152 >> (int)DRMP3_HDR_IS_FRAME_576(h));
}
static int drmp3_hdr_frame_bytes(const drmp3_uint8 *h, int free_format_size)
{
int frame_bytes = drmp3_hdr_frame_samples(h)*drmp3_hdr_bitrate_kbps(h)*125/drmp3_hdr_sample_rate_hz(h);
if (DRMP3_HDR_IS_LAYER_1(h))
{
frame_bytes &= ~3;
}
return frame_bytes ? frame_bytes : free_format_size;
}
static int drmp3_hdr_padding(const drmp3_uint8 *h)
{
return DRMP3_HDR_TEST_PADDING(h) ? (DRMP3_HDR_IS_LAYER_1(h) ? 4 : 1) : 0;
}
#ifndef DR_MP3_ONLY_MP3
static const drmp3_L12_subband_alloc *drmp3_L12_subband_alloc_table(const drmp3_uint8 *hdr, drmp3_L12_scale_info *sci)
{
const drmp3_L12_subband_alloc *alloc;
int mode = DRMP3_HDR_GET_STEREO_MODE(hdr);
int nbands, stereo_bands = (mode == DRMP3_MODE_MONO) ? 0 : (mode == DRMP3_MODE_JOINT_STEREO) ? (DRMP3_HDR_GET_STEREO_MODE_EXT(hdr) << 2) + 4 : 32;
if (DRMP3_HDR_IS_LAYER_1(hdr))
{
static const drmp3_L12_subband_alloc g_alloc_L1[] = { { 76, 4, 32 } };
alloc = g_alloc_L1;
nbands = 32;
} else if (!DRMP3_HDR_TEST_MPEG1(hdr))
{
static const drmp3_L12_subband_alloc g_alloc_L2M2[] = { { 60, 4, 4 }, { 44, 3, 7 }, { 44, 2, 19 } };
alloc = g_alloc_L2M2;
nbands = 30;
} else
{
static const drmp3_L12_subband_alloc g_alloc_L2M1[] = { { 0, 4, 3 }, { 16, 4, 8 }, { 32, 3, 12 }, { 40, 2, 7 } };
int sample_rate_idx = DRMP3_HDR_GET_SAMPLE_RATE(hdr);
unsigned kbps = drmp3_hdr_bitrate_kbps(hdr) >> (int)(mode != DRMP3_MODE_MONO);
if (!kbps)
{
kbps = 192;
}
alloc = g_alloc_L2M1;
nbands = 27;
if (kbps < 56)
{
static const drmp3_L12_subband_alloc g_alloc_L2M1_lowrate[] = { { 44, 4, 2 }, { 44, 3, 10 } };
alloc = g_alloc_L2M1_lowrate;
nbands = sample_rate_idx == 2 ? 12 : 8;
} else if (kbps >= 96 && sample_rate_idx != 1)
{
nbands = 30;
}
}
sci->total_bands = (drmp3_uint8)nbands;
sci->stereo_bands = (drmp3_uint8)DRMP3_MIN(stereo_bands, nbands);
return alloc;
}
static void drmp3_L12_read_scalefactors(drmp3_bs *bs, drmp3_uint8 *pba, drmp3_uint8 *scfcod, int bands, float *scf)
{
static const float g_deq_L12[18*3] = {
#define DRMP3_DQ(x) 9.53674316e-07f/x, 7.56931807e-07f/x, 6.00777173e-07f/x
DRMP3_DQ(3),DRMP3_DQ(7),DRMP3_DQ(15),DRMP3_DQ(31),DRMP3_DQ(63),DRMP3_DQ(127),DRMP3_DQ(255),DRMP3_DQ(511),DRMP3_DQ(1023),DRMP3_DQ(2047),DRMP3_DQ(4095),DRMP3_DQ(8191),DRMP3_DQ(16383),DRMP3_DQ(32767),DRMP3_DQ(65535),DRMP3_DQ(3),DRMP3_DQ(5),DRMP3...
};
int i, m;
for (i = 0; i < bands; i++)
{
float s = 0;
int ba = *pba++;
int mask = ba ? 4 + ((19 >> scfcod[i]) & 3) : 0;
for (m = 4; m; m >>= 1)
{
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
float ovl = overlap[i];
float sum = co[i]*g_twid3[3 + i] + si[i]*g_twid3[0 + i];
overlap[i] = co[i]*g_twid3[0 + i] - si[i]*g_twid3[3 + i];
dst[i] = ovl*g_twid3[2 - i] - sum*g_twid3[5 - i];
dst[5 - i] = ovl*g_twid3[5 - i] + sum*g_twid3[2 - i];
}
}
static void drmp3_L3_imdct_short(float *grbuf, float *overlap, int nbands)
{
for (;nbands > 0; nbands--, overlap += 9, grbuf += 18)
{
float tmp[18];
DRMP3_COPY_MEMORY(tmp, grbuf, sizeof(tmp));
DRMP3_COPY_MEMORY(grbuf, overlap, 6*sizeof(float));
drmp3_L3_imdct12(tmp, grbuf + 6, overlap + 6);
drmp3_L3_imdct12(tmp + 1, grbuf + 12, overlap + 6);
drmp3_L3_imdct12(tmp + 2, overlap, overlap + 6);
}
}
static void drmp3_L3_change_sign(float *grbuf)
{
int b, i;
for (b = 0, grbuf += 18; b < 32; b += 2, grbuf += 36)
for (i = 1; i < 18; i += 2)
grbuf[i] = -grbuf[i];
}
static void drmp3_L3_imdct_gr(float *grbuf, float *overlap, unsigned block_type, unsigned n_long_bands)
{
static const float g_mdct_window[2][18] = {
{ 0.99904822f,0.99144486f,0.97629601f,0.95371695f,0.92387953f,0.88701083f,0.84339145f,0.79335334f,0.73727734f,0.04361938f,0.13052619f,0.21643961f,0.30070580f,0.38268343f,0.46174861f,0.53729961f,0.60876143f,0.67559021f },
{ 1,1,1,1,1,1,0.99144486f,0.92387953f,0.79335334f,0,0,0,0,0,0,0.13052619f,0.38268343f,0.60876143f }
};
if (n_long_bands)
{
drmp3_L3_imdct36(grbuf, overlap, g_mdct_window[0], n_long_bands);
grbuf += 18*n_long_bands;
overlap += 9*n_long_bands;
}
if (block_type == DRMP3_SHORT_BLOCK_TYPE)
drmp3_L3_imdct_short(grbuf, overlap, 32 - n_long_bands);
else
drmp3_L3_imdct36(grbuf, overlap, g_mdct_window[block_type == DRMP3_STOP_BLOCK_TYPE], 32 - n_long_bands);
}
static void drmp3_L3_save_reservoir(drmp3dec *h, drmp3dec_scratch *s)
{
int pos = (s->bs.pos + 7)/8u;
int remains = s->bs.limit/8u - pos;
if (remains > DRMP3_MAX_BITRESERVOIR_BYTES)
{
pos += remains - DRMP3_MAX_BITRESERVOIR_BYTES;
remains = DRMP3_MAX_BITRESERVOIR_BYTES;
}
if (remains > 0)
{
DRMP3_MOVE_MEMORY(h->reserv_buf, s->maindata + pos, remains);
}
h->reserv = remains;
}
static int drmp3_L3_restore_reservoir(drmp3dec *h, drmp3_bs *bs, drmp3dec_scratch *s, int main_data_begin)
{
int frame_bytes = (bs->limit - bs->pos)/8;
int bytes_have = DRMP3_MIN(h->reserv, main_data_begin);
DRMP3_COPY_MEMORY(s->maindata, h->reserv_buf + DRMP3_MAX(0, h->reserv - main_data_begin), DRMP3_MIN(h->reserv, main_data_begin));
DRMP3_COPY_MEMORY(s->maindata + bytes_have, bs->buf + bs->pos/8, frame_bytes);
drmp3_bs_init(&s->bs, s->maindata, bytes_have + frame_bytes);
return h->reserv >= main_data_begin;
}
static void drmp3_L3_decode(drmp3dec *h, drmp3dec_scratch *s, drmp3_L3_gr_info *gr_info, int nch)
{
int ch;
for (ch = 0; ch < nch; ch++)
{
int layer3gr_limit = s->bs.pos + gr_info[ch].part_23_length;
drmp3_L3_decode_scalefactors(h->header, s->ist_pos[ch], &s->bs, gr_info + ch, s->scf, ch);
drmp3_L3_huffman(s->grbuf[ch], &s->bs, gr_info + ch, s->scf, layer3gr_limit);
}
if (DRMP3_HDR_TEST_I_STEREO(h->header))
{
drmp3_L3_intensity_stereo(s->grbuf[0], s->ist_pos[1], gr_info, h->header);
} else if (DRMP3_HDR_IS_MS_STEREO(h->header))
{
drmp3_L3_midside_stereo(s->grbuf[0], 576);
}
for (ch = 0; ch < nch; ch++, gr_info++)
{
int aa_bands = 31;
int n_long_bands = (gr_info->mixed_block_flag ? 2 : 0) << (int)(DRMP3_HDR_GET_MY_SAMPLE_RATE(h->header) == 2);
if (gr_info->n_short_sfb)
{
aa_bands = n_long_bands - 1;
drmp3_L3_reorder(s->grbuf[ch] + n_long_bands*18, s->syn[0], gr_info->sfbtab + gr_info->n_long_sfb);
}
drmp3_L3_antialias(s->grbuf[ch], aa_bands);
drmp3_L3_imdct_gr(s->grbuf[ch], h->mdct_overlap[ch], gr_info->block_type, n_long_bands);
drmp3_L3_change_sign(s->grbuf[ch]);
}
}
static void drmp3d_DCT_II(float *grbuf, int n)
{
static const float g_sec[24] = {
10.19000816f,0.50060302f,0.50241929f,3.40760851f,0.50547093f,0.52249861f,2.05778098f,0.51544732f,0.56694406f,1.48416460f,0.53104258f,0.64682180f,1.16943991f,0.55310392f,0.78815460f,0.97256821f,0.58293498f,1.06067765f,0.83934963f,0.62250412f,1...
};
int i, k = 0;
#if DRMP3_HAVE_SIMD
if (drmp3_have_simd()) for (; k < n; k += 4)
{
drmp3_f4 t[4][8], *x;
float *y = grbuf + k;
for (x = t[0], i = 0; i < 8; i++, x++)
{
drmp3_f4 x0 = DRMP3_VLD(&y[i*18]);
drmp3_f4 x1 = DRMP3_VLD(&y[(15 - i)*18]);
drmp3_f4 x2 = DRMP3_VLD(&y[(16 + i)*18]);
drmp3_f4 x3 = DRMP3_VLD(&y[(31 - i)*18]);
drmp3_f4 t0 = DRMP3_VADD(x0, x3);
drmp3_f4 t1 = DRMP3_VADD(x1, x2);
drmp3_f4 t2 = DRMP3_VMUL_S(DRMP3_VSUB(x1, x2), g_sec[3*i + 0]);
drmp3_f4 t3 = DRMP3_VMUL_S(DRMP3_VSUB(x0, x3), g_sec[3*i + 1]);
x[0] = DRMP3_VADD(t0, t1);
x[8] = DRMP3_VMUL_S(DRMP3_VSUB(t0, t1), g_sec[3*i + 2]);
x[16] = DRMP3_VADD(t3, t2);
x[24] = DRMP3_VMUL_S(DRMP3_VSUB(t3, t2), g_sec[3*i + 2]);
}
for (x = t[0], i = 0; i < 4; i++, x += 8)
{
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
#endif
#endif
}
} else
#endif
#ifdef DR_MP3_ONLY_SIMD
{}
#else
for (i = 14; i >= 0; i--)
{
#define DRMP3_LOAD(k) float w0 = *w++; float w1 = *w++; float *vz = &zlin[4*i - k*64]; float *vy = &zlin[4*i - (15 - k)*64];
#define DRMP3_S0(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] = vz[j]*w1 + vy[j]*w0, a[j] = vz[j]*w0 - vy[j]*w1; }
#define DRMP3_S1(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] += vz[j]*w1 + vy[j]*w0, a[j] += vz[j]*w0 - vy[j]*w1; }
#define DRMP3_S2(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] += vz[j]*w1 + vy[j]*w0, a[j] += vy[j]*w1 - vz[j]*w0; }
float a[4], b[4];
zlin[4*i] = xl[18*(31 - i)];
zlin[4*i + 1] = xr[18*(31 - i)];
zlin[4*i + 2] = xl[1 + 18*(31 - i)];
zlin[4*i + 3] = xr[1 + 18*(31 - i)];
zlin[4*(i + 16)] = xl[1 + 18*(1 + i)];
zlin[4*(i + 16) + 1] = xr[1 + 18*(1 + i)];
zlin[4*(i - 16) + 2] = xl[18*(1 + i)];
zlin[4*(i - 16) + 3] = xr[18*(1 + i)];
DRMP3_S0(0) DRMP3_S2(1) DRMP3_S1(2) DRMP3_S2(3) DRMP3_S1(4) DRMP3_S2(5) DRMP3_S1(6) DRMP3_S2(7)
dstr[(15 - i)*nch] = drmp3d_scale_pcm(a[1]);
dstr[(17 + i)*nch] = drmp3d_scale_pcm(b[1]);
dstl[(15 - i)*nch] = drmp3d_scale_pcm(a[0]);
dstl[(17 + i)*nch] = drmp3d_scale_pcm(b[0]);
dstr[(47 - i)*nch] = drmp3d_scale_pcm(a[3]);
dstr[(49 + i)*nch] = drmp3d_scale_pcm(b[3]);
dstl[(47 - i)*nch] = drmp3d_scale_pcm(a[2]);
dstl[(49 + i)*nch] = drmp3d_scale_pcm(b[2]);
}
#endif
}
static void drmp3d_synth_granule(float *qmf_state, float *grbuf, int nbands, int nch, drmp3d_sample_t *pcm, float *lins)
{
int i;
for (i = 0; i < nch; i++)
{
drmp3d_DCT_II(grbuf + 576*i, nbands);
}
DRMP3_COPY_MEMORY(lins, qmf_state, sizeof(float)*15*64);
for (i = 0; i < nbands; i += 2)
{
drmp3d_synth(grbuf + i, pcm + 32*nch*i, nch, lins + i*64);
}
#ifndef DR_MP3_NONSTANDARD_BUT_LOGICAL
if (nch == 1)
{
for (i = 0; i < 15*64; i += 2)
{
qmf_state[i] = lins[nbands*64 + i];
}
} else
#endif
{
DRMP3_COPY_MEMORY(qmf_state, lins + nbands*64, sizeof(float)*15*64);
}
}
static int drmp3d_match_frame(const drmp3_uint8 *hdr, int mp3_bytes, int frame_bytes)
{
int i, nmatch;
for (i = 0, nmatch = 0; nmatch < DRMP3_MAX_FRAME_SYNC_MATCHES; nmatch++)
{
i += drmp3_hdr_frame_bytes(hdr + i, frame_bytes) + drmp3_hdr_padding(hdr + i);
if (i + DRMP3_HDR_SIZE > mp3_bytes)
return nmatch > 0;
if (!drmp3_hdr_compare(hdr, hdr + i))
return 0;
}
return 1;
}
static int drmp3d_find_frame(const drmp3_uint8 *mp3, int mp3_bytes, int *free_format_bytes, int *ptr_frame_bytes)
{
int i, k;
for (i = 0; i < mp3_bytes - DRMP3_HDR_SIZE; i++, mp3++)
{
if (drmp3_hdr_valid(mp3))
{
int frame_bytes = drmp3_hdr_frame_bytes(mp3, *free_format_bytes);
int frame_and_padding = frame_bytes + drmp3_hdr_padding(mp3);
for (k = DRMP3_HDR_SIZE; !frame_bytes && k < DRMP3_MAX_FREE_FORMAT_FRAME_SIZE && i + 2*k < mp3_bytes - DRMP3_HDR_SIZE; k++)
{
if (drmp3_hdr_compare(mp3, mp3 + k))
{
int fb = k - drmp3_hdr_padding(mp3);
int nextfb = fb + drmp3_hdr_padding(mp3 + k);
if (i + k + nextfb + DRMP3_HDR_SIZE > mp3_bytes || !drmp3_hdr_compare(mp3, mp3 + k + nextfb))
continue;
frame_and_padding = k;
frame_bytes = fb;
*free_format_bytes = fb;
}
}
if ((frame_bytes && i + frame_and_padding <= mp3_bytes &&
drmp3d_match_frame(mp3, mp3_bytes - i, frame_bytes)) ||
(!i && frame_and_padding == mp3_bytes))
{
*ptr_frame_bytes = frame_and_padding;
return i;
}
*free_format_bytes = 0;
}
}
*ptr_frame_bytes = 0;
return mp3_bytes;
}
DRMP3_API void drmp3dec_init(drmp3dec *dec)
{
dec->header[0] = 0;
}
DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info)
{
int i = 0, igr, frame_size = 0, success = 1;
const drmp3_uint8 *hdr;
drmp3_bs bs_frame[1];
drmp3dec_scratch scratch;
if (mp3_bytes > 4 && dec->header[0] == 0xff && drmp3_hdr_compare(dec->header, mp3))
{
frame_size = drmp3_hdr_frame_bytes(mp3, dec->free_format_bytes) + drmp3_hdr_padding(mp3);
if (frame_size != mp3_bytes && (frame_size + DRMP3_HDR_SIZE > mp3_bytes || !drmp3_hdr_compare(mp3, mp3 + frame_size)))
{
frame_size = 0;
}
}
if (!frame_size)
{
DRMP3_ZERO_MEMORY(dec, sizeof(drmp3dec));
i = drmp3d_find_frame(mp3, mp3_bytes, &dec->free_format_bytes, &frame_size);
if (!frame_size || i + frame_size > mp3_bytes)
{
info->frame_bytes = i;
return 0;
}
}
hdr = mp3 + i;
DRMP3_COPY_MEMORY(dec->header, hdr, DRMP3_HDR_SIZE);
info->frame_bytes = i + frame_size;
info->channels = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2;
info->hz = drmp3_hdr_sample_rate_hz(hdr);
info->layer = 4 - DRMP3_HDR_GET_LAYER(hdr);
info->bitrate_kbps = drmp3_hdr_bitrate_kbps(hdr);
drmp3_bs_init(bs_frame, hdr + DRMP3_HDR_SIZE, frame_size - DRMP3_HDR_SIZE);
if (DRMP3_HDR_IS_CRC(hdr))
{
drmp3_bs_get_bits(bs_frame, 16);
}
if (info->layer == 3)
{
int main_data_begin = drmp3_L3_read_side_info(bs_frame, scratch.gr_info, hdr);
if (main_data_begin < 0 || bs_frame->pos > bs_frame->limit)
{
drmp3dec_init(dec);
return 0;
}
success = drmp3_L3_restore_reservoir(dec, bs_frame, &scratch, main_data_begin);
if (success && pcm != NULL)
{
for (igr = 0; igr < (DRMP3_HDR_TEST_MPEG1(hdr) ? 2 : 1); igr++, pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*576*info->channels))
{
DRMP3_ZERO_MEMORY(scratch.grbuf[0], 576*2*sizeof(float));
drmp3_L3_decode(dec, &scratch, scratch.gr_info + igr*info->channels, info->channels);
drmp3d_synth_granule(dec->qmf_state, scratch.grbuf[0], 18, info->channels, (drmp3d_sample_t*)pcm, scratch.syn[0]);
}
}
drmp3_L3_save_reservoir(dec, &scratch);
} else
{
#ifdef DR_MP3_ONLY_MP3
return 0;
#else
drmp3_L12_scale_info sci[1];
if (pcm == NULL) {
return drmp3_hdr_frame_samples(hdr);
}
drmp3_L12_read_scale_info(hdr, bs_frame, sci);
DRMP3_ZERO_MEMORY(scratch.grbuf[0], 576*2*sizeof(float));
for (i = 0, igr = 0; igr < 3; igr++)
{
if (12 == (i += drmp3_L12_dequantize_granule(scratch.grbuf[0] + i, bs_frame, sci, info->layer | 1)))
{
i = 0;
drmp3_L12_apply_scf_384(sci, sci->scf + igr, scratch.grbuf[0]);
drmp3d_synth_granule(dec->qmf_state, scratch.grbuf[0], 12, info->channels, (drmp3d_sample_t*)pcm, scratch.syn[0]);
DRMP3_ZERO_MEMORY(scratch.grbuf[0], 576*2*sizeof(float));
pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*384*info->channels);
}
if (bs_frame->pos > bs_frame->limit)
{
drmp3dec_init(dec);
return 0;
}
}
#endif
}
return success*drmp3_hdr_frame_samples(dec->header);
}
DRMP3_API void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, size_t num_samples)
{
size_t i = 0;
#if DRMP3_HAVE_SIMD
size_t aligned_count = num_samples & ~7;
for(; i < aligned_count; i+=8)
{
drmp3_f4 scale = DRMP3_VSET(32768.0f);
drmp3_f4 a = DRMP3_VMUL(DRMP3_VLD(&in[i ]), scale);
drmp3_f4 b = DRMP3_VMUL(DRMP3_VLD(&in[i+4]), scale);
#if DRMP3_HAVE_SSE
drmp3_f4 s16max = DRMP3_VSET( 32767.0f);
drmp3_f4 s16min = DRMP3_VSET(-32768.0f);
__m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, s16max), s16min)),
_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, s16max), s16min)));
out[i ] = (drmp3_int16)_mm_extract_epi16(pcm8, 0);
out[i+1] = (drmp3_int16)_mm_extract_epi16(pcm8, 1);
out[i+2] = (drmp3_int16)_mm_extract_epi16(pcm8, 2);
out[i+3] = (drmp3_int16)_mm_extract_epi16(pcm8, 3);
out[i+4] = (drmp3_int16)_mm_extract_epi16(pcm8, 4);
out[i+5] = (drmp3_int16)_mm_extract_epi16(pcm8, 5);
out[i+6] = (drmp3_int16)_mm_extract_epi16(pcm8, 6);
out[i+7] = (drmp3_int16)_mm_extract_epi16(pcm8, 7);
#else
int16x4_t pcma, pcmb;
a = DRMP3_VADD(a, DRMP3_VSET(0.5f));
b = DRMP3_VADD(b, DRMP3_VSET(0.5f));
pcma = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(a), vreinterpretq_s32_u32(vcltq_f32(a, DRMP3_VSET(0)))));
pcmb = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(b), vreinterpretq_s32_u32(vcltq_f32(b, DRMP3_VSET(0)))));
vst1_lane_s16(out+i , pcma, 0);
vst1_lane_s16(out+i+1, pcma, 1);
vst1_lane_s16(out+i+2, pcma, 2);
vst1_lane_s16(out+i+3, pcma, 3);
vst1_lane_s16(out+i+4, pcmb, 0);
vst1_lane_s16(out+i+5, pcmb, 1);
vst1_lane_s16(out+i+6, pcmb, 2);
vst1_lane_s16(out+i+7, pcmb, 3);
#endif
}
#endif
for(; i < num_samples; i++)
{
float sample = in[i] * 32768.0f;
if (sample >= 32766.5)
out[i] = (drmp3_int16) 32767;
else if (sample <= -32767.5)
out[i] = (drmp3_int16)-32768;
else
{
short s = (drmp3_int16)(sample + .5f);
s -= (s < 0);
out[i] = s;
}
}
}
#if defined(SIZE_MAX)
#define DRMP3_SIZE_MAX SIZE_MAX
#else
#if defined(_WIN64) || defined(_LP64) || defined(__LP64__)
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
if (pAllocationCallbacks->onFree != NULL) {
pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData);
}
}
static drmp3_allocation_callbacks drmp3_copy_allocation_callbacks_or_defaults(const drmp3_allocation_callbacks* pAllocationCallbacks)
{
if (pAllocationCallbacks != NULL) {
return *pAllocationCallbacks;
} else {
drmp3_allocation_callbacks allocationCallbacks;
allocationCallbacks.pUserData = NULL;
allocationCallbacks.onMalloc = drmp3__malloc_default;
allocationCallbacks.onRealloc = drmp3__realloc_default;
allocationCallbacks.onFree = drmp3__free_default;
return allocationCallbacks;
}
}
static size_t drmp3__on_read(drmp3* pMP3, void* pBufferOut, size_t bytesToRead)
{
size_t bytesRead = pMP3->onRead(pMP3->pUserData, pBufferOut, bytesToRead);
pMP3->streamCursor += bytesRead;
return bytesRead;
}
static drmp3_bool32 drmp3__on_seek(drmp3* pMP3, int offset, drmp3_seek_origin origin)
{
DRMP3_ASSERT(offset >= 0);
if (!pMP3->onSeek(pMP3->pUserData, offset, origin)) {
return DRMP3_FALSE;
}
if (origin == drmp3_seek_origin_start) {
pMP3->streamCursor = (drmp3_uint64)offset;
} else {
pMP3->streamCursor += offset;
}
return DRMP3_TRUE;
}
static drmp3_bool32 drmp3__on_seek_64(drmp3* pMP3, drmp3_uint64 offset, drmp3_seek_origin origin)
{
if (offset <= 0x7FFFFFFF) {
return drmp3__on_seek(pMP3, (int)offset, origin);
}
if (!drmp3__on_seek(pMP3, 0x7FFFFFFF, drmp3_seek_origin_start)) {
return DRMP3_FALSE;
}
offset -= 0x7FFFFFFF;
while (offset > 0) {
if (offset <= 0x7FFFFFFF) {
if (!drmp3__on_seek(pMP3, (int)offset, drmp3_seek_origin_current)) {
return DRMP3_FALSE;
}
offset = 0;
} else {
if (!drmp3__on_seek(pMP3, 0x7FFFFFFF, drmp3_seek_origin_current)) {
return DRMP3_FALSE;
}
offset -= 0x7FFFFFFF;
}
}
return DRMP3_TRUE;
}
static drmp3_uint32 drmp3_decode_next_frame_ex__callbacks(drmp3* pMP3, drmp3d_sample_t* pPCMFrames)
{
drmp3_uint32 pcmFramesRead = 0;
DRMP3_ASSERT(pMP3 != NULL);
DRMP3_ASSERT(pMP3->onRead != NULL);
if (pMP3->atEnd) {
return 0;
}
for (;;) {
drmp3dec_frame_info info;
if (pMP3->dataSize < DRMP3_MIN_DATA_CHUNK_SIZE) {
size_t bytesRead;
if (pMP3->pData != NULL) {
DRMP3_MOVE_MEMORY(pMP3->pData, pMP3->pData + pMP3->dataConsumed, pMP3->dataSize);
}
pMP3->dataConsumed = 0;
if (pMP3->dataCapacity < DRMP3_DATA_CHUNK_SIZE) {
drmp3_uint8* pNewData;
size_t newDataCap;
newDataCap = DRMP3_DATA_CHUNK_SIZE;
pNewData = (drmp3_uint8*)drmp3__realloc_from_callbacks(pMP3->pData, newDataCap, pMP3->dataCapacity, &pMP3->allocationCallbacks);
if (pNewData == NULL) {
return 0;
}
pMP3->pData = pNewData;
pMP3->dataCapacity = newDataCap;
}
bytesRead = drmp3__on_read(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize));
if (bytesRead == 0) {
if (pMP3->dataSize == 0) {
pMP3->atEnd = DRMP3_TRUE;
return 0;
}
}
pMP3->dataSize += bytesRead;
}
if (pMP3->dataSize > INT_MAX) {
pMP3->atEnd = DRMP3_TRUE;
return 0;
}
DRMP3_ASSERT(pMP3->pData != NULL);
DRMP3_ASSERT(pMP3->dataCapacity > 0);
pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->pData + pMP3->dataConsumed, (int)pMP3->dataSize, pPCMFrames, &info);
if (info.frame_bytes > 0) {
pMP3->dataConsumed += (size_t)info.frame_bytes;
pMP3->dataSize -= (size_t)info.frame_bytes;
}
if (pcmFramesRead > 0) {
pcmFramesRead = drmp3_hdr_frame_samples(pMP3->decoder.header);
pMP3->pcmFramesConsumedInMP3Frame = 0;
pMP3->pcmFramesRemainingInMP3Frame = pcmFramesRead;
pMP3->mp3FrameChannels = info.channels;
pMP3->mp3FrameSampleRate = info.hz;
break;
} else if (info.frame_bytes == 0) {
size_t bytesRead;
DRMP3_MOVE_MEMORY(pMP3->pData, pMP3->pData + pMP3->dataConsumed, pMP3->dataSize);
pMP3->dataConsumed = 0;
if (pMP3->dataCapacity == pMP3->dataSize) {
drmp3_uint8* pNewData;
size_t newDataCap;
newDataCap = pMP3->dataCapacity + DRMP3_DATA_CHUNK_SIZE;
pNewData = (drmp3_uint8*)drmp3__realloc_from_callbacks(pMP3->pData, newDataCap, pMP3->dataCapacity, &pMP3->allocationCallbacks);
if (pNewData == NULL) {
return 0;
}
pMP3->pData = pNewData;
pMP3->dataCapacity = newDataCap;
}
bytesRead = drmp3__on_read(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize));
if (bytesRead == 0) {
pMP3->atEnd = DRMP3_TRUE;
return 0;
}
pMP3->dataSize += bytesRead;
}
};
return pcmFramesRead;
}
static drmp3_uint32 drmp3_decode_next_frame_ex__memory(drmp3* pMP3, drmp3d_sample_t* pPCMFrames)
{
drmp3_uint32 pcmFramesRead = 0;
drmp3dec_frame_info info;
DRMP3_ASSERT(pMP3 != NULL);
DRMP3_ASSERT(pMP3->memory.pData != NULL);
if (pMP3->atEnd) {
return 0;
}
for (;;) {
pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->memory.pData + pMP3->memory.currentReadPos, (int)(pMP3->memory.dataSize - pMP3->memory.currentReadPos), pPCMFrames, &info);
if (pcmFramesRead > 0) {
pcmFramesRead = drmp3_hdr_frame_samples(pMP3->decoder.header);
pMP3->pcmFramesConsumedInMP3Frame = 0;
pMP3->pcmFramesRemainingInMP3Frame = pcmFramesRead;
pMP3->mp3FrameChannels = info.channels;
pMP3->mp3FrameSampleRate = info.hz;
break;
} else if (info.frame_bytes > 0) {
pMP3->memory.currentReadPos += (size_t)info.frame_bytes;
} else {
break;
}
}
pMP3->memory.currentReadPos += (size_t)info.frame_bytes;
return pcmFramesRead;
}
static drmp3_uint32 drmp3_decode_next_frame_ex(drmp3* pMP3, drmp3d_sample_t* pPCMFrames)
{
if (pMP3->memory.pData != NULL && pMP3->memory.dataSize > 0) {
return drmp3_decode_next_frame_ex__memory(pMP3, pPCMFrames);
} else {
return drmp3_decode_next_frame_ex__callbacks(pMP3, pPCMFrames);
}
}
static drmp3_uint32 drmp3_decode_next_frame(drmp3* pMP3)
{
DRMP3_ASSERT(pMP3 != NULL);
return drmp3_decode_next_frame_ex(pMP3, (drmp3d_sample_t*)pMP3->pcmFrames);
}
#if 0
static drmp3_uint32 drmp3_seek_next_frame(drmp3* pMP3)
{
drmp3_uint32 pcmFrameCount;
DRMP3_ASSERT(pMP3 != NULL);
pcmFrameCount = drmp3_decode_next_frame_ex(pMP3, NULL);
if (pcmFrameCount == 0) {
return 0;
}
pMP3->currentPCMFrame += pcmFrameCount;
pMP3->pcmFramesConsumedInMP3Frame = pcmFrameCount;
pMP3->pcmFramesRemainingInMP3Frame = 0;
return pcmFrameCount;
}
#endif
static drmp3_bool32 drmp3_init_internal(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_allocation_callbacks* pAllocationCallbacks)
{
DRMP3_ASSERT(pMP3 != NULL);
DRMP3_ASSERT(onRead != NULL);
drmp3dec_init(&pMP3->decoder);
pMP3->onRead = onRead;
pMP3->onSeek = onSeek;
pMP3->pUserData = pUserData;
pMP3->allocationCallbacks = drmp3_copy_allocation_callbacks_or_defaults(pAllocationCallbacks);
if (pMP3->allocationCallbacks.onFree == NULL || (pMP3->allocationCallbacks.onMalloc == NULL && pMP3->allocationCallbacks.onRealloc == NULL)) {
return DRMP3_FALSE;
}
if (drmp3_decode_next_frame(pMP3) == 0) {
drmp3__free_from_callbacks(pMP3->pData, &pMP3->allocationCallbacks);
return DRMP3_FALSE;
}
pMP3->channels = pMP3->mp3FrameChannels;
pMP3->sampleRate = pMP3->mp3FrameSampleRate;
return DRMP3_TRUE;
}
DRMP3_API drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, const drmp3_allocation_callbacks* pAllocationCallbacks)
{
if (pMP3 == NULL || onRead == NULL) {
return DRMP3_FALSE;
}
DRMP3_ZERO_OBJECT(pMP3);
return drmp3_init_internal(pMP3, onRead, onSeek, pUserData, pAllocationCallbacks);
}
static size_t drmp3__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead)
{
drmp3* pMP3 = (drmp3*)pUserData;
size_t bytesRemaining;
DRMP3_ASSERT(pMP3 != NULL);
DRMP3_ASSERT(pMP3->memory.dataSize >= pMP3->memory.currentReadPos);
bytesRemaining = pMP3->memory.dataSize - pMP3->memory.currentReadPos;
if (bytesToRead > bytesRemaining) {
bytesToRead = bytesRemaining;
}
if (bytesToRead > 0) {
DRMP3_COPY_MEMORY(pBufferOut, pMP3->memory.pData + pMP3->memory.currentReadPos, bytesToRead);
pMP3->memory.currentReadPos += bytesToRead;
}
return bytesToRead;
}
static drmp3_bool32 drmp3__on_seek_memory(void* pUserData, int byteOffset, drmp3_seek_origin origin)
{
drmp3* pMP3 = (drmp3*)pUserData;
DRMP3_ASSERT(pMP3 != NULL);
if (origin == drmp3_seek_origin_current) {
if (byteOffset > 0) {
if (pMP3->memory.currentReadPos + byteOffset > pMP3->memory.dataSize) {
byteOffset = (int)(pMP3->memory.dataSize - pMP3->memory.currentReadPos);
}
} else {
if (pMP3->memory.currentReadPos < (size_t)-byteOffset) {
byteOffset = -(int)pMP3->memory.currentReadPos;
}
}
pMP3->memory.currentReadPos += byteOffset;
} else {
if ((drmp3_uint32)byteOffset <= pMP3->memory.dataSize) {
pMP3->memory.currentReadPos = byteOffset;
} else {
pMP3->memory.currentReadPos = pMP3->memory.dataSize;
}
}
return DRMP3_TRUE;
}
DRMP3_API drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_allocation_callbacks* pAllocationCallbacks)
{
if (pMP3 == NULL) {
return DRMP3_FALSE;
}
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if (pMP3 == NULL) {
return;
}
#ifndef DR_MP3_NO_STDIO
if (pMP3->onRead == drmp3__on_read_stdio) {
FILE* pFile = (FILE*)pMP3->pUserData;
if (pFile != NULL) {
fclose(pFile);
pMP3->pUserData = NULL;
}
}
#endif
drmp3__free_from_callbacks(pMP3->pData, &pMP3->allocationCallbacks);
}
#if defined(DR_MP3_FLOAT_OUTPUT)
static void drmp3_f32_to_s16(drmp3_int16* dst, const float* src, drmp3_uint64 sampleCount)
{
drmp3_uint64 i;
drmp3_uint64 i4;
drmp3_uint64 sampleCount4;
i = 0;
sampleCount4 = sampleCount >> 2;
for (i4 = 0; i4 < sampleCount4; i4 += 1) {
float x0 = src[i+0];
float x1 = src[i+1];
float x2 = src[i+2];
float x3 = src[i+3];
x0 = ((x0 < -1) ? -1 : ((x0 > 1) ? 1 : x0));
x1 = ((x1 < -1) ? -1 : ((x1 > 1) ? 1 : x1));
x2 = ((x2 < -1) ? -1 : ((x2 > 1) ? 1 : x2));
x3 = ((x3 < -1) ? -1 : ((x3 > 1) ? 1 : x3));
x0 = x0 * 32767.0f;
x1 = x1 * 32767.0f;
x2 = x2 * 32767.0f;
x3 = x3 * 32767.0f;
dst[i+0] = (drmp3_int16)x0;
dst[i+1] = (drmp3_int16)x1;
dst[i+2] = (drmp3_int16)x2;
dst[i+3] = (drmp3_int16)x3;
i += 4;
}
for (; i < sampleCount; i += 1) {
float x = src[i];
x = ((x < -1) ? -1 : ((x > 1) ? 1 : x));
x = x * 32767.0f;
dst[i] = (drmp3_int16)x;
}
}
#endif
#if !defined(DR_MP3_FLOAT_OUTPUT)
static void drmp3_s16_to_f32(float* dst, const drmp3_int16* src, drmp3_uint64 sampleCount)
{
drmp3_uint64 i;
for (i = 0; i < sampleCount; i += 1) {
float x = (float)src[i];
x = x * 0.000030517578125f;
dst[i] = x;
}
}
#endif
static drmp3_uint64 drmp3_read_pcm_frames_raw(drmp3* pMP3, drmp3_uint64 framesToRead, void* pBufferOut)
{
drmp3_uint64 totalFramesRead = 0;
DRMP3_ASSERT(pMP3 != NULL);
DRMP3_ASSERT(pMP3->onRead != NULL);
while (framesToRead > 0) {
drmp3_uint32 framesToConsume = (drmp3_uint32)DRMP3_MIN(pMP3->pcmFramesRemainingInMP3Frame, framesToRead);
if (pBufferOut != NULL) {
#if defined(DR_MP3_FLOAT_OUTPUT)
float* pFramesOutF32 = (float*)DRMP3_OFFSET_PTR(pBufferOut, sizeof(float) * totalFramesRead * pMP3->channels);
float* pFramesInF32 = (float*)DRMP3_OFFSET_PTR(&pMP3->pcmFrames[0], sizeof(float) * pMP3->pcmFramesConsumedInMP3Frame * pMP3->mp3FrameChannels);
DRMP3_COPY_MEMORY(pFramesOutF32, pFramesInF32, sizeof(float) * framesToConsume * pMP3->channels);
#else
drmp3_int16* pFramesOutS16 = (drmp3_int16*)DRMP3_OFFSET_PTR(pBufferOut, sizeof(drmp3_int16) * totalFramesRead * pMP3->channels);
drmp3_int16* pFramesInS16 = (drmp3_int16*)DRMP3_OFFSET_PTR(&pMP3->pcmFrames[0], sizeof(drmp3_int16) * pMP3->pcmFramesConsumedInMP3Frame * pMP3->mp3FrameChannels);
DRMP3_COPY_MEMORY(pFramesOutS16, pFramesInS16, sizeof(drmp3_int16) * framesToConsume * pMP3->channels);
#endif
}
pMP3->currentPCMFrame += framesToConsume;
pMP3->pcmFramesConsumedInMP3Frame += framesToConsume;
pMP3->pcmFramesRemainingInMP3Frame -= framesToConsume;
totalFramesRead += framesToConsume;
framesToRead -= framesToConsume;
if (framesToRead == 0) {
break;
}
DRMP3_ASSERT(pMP3->pcmFramesRemainingInMP3Frame == 0);
if (drmp3_decode_next_frame(pMP3) == 0) {
break;
}
}
return totalFramesRead;
}
DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut)
{
if (pMP3 == NULL || pMP3->onRead == NULL) {
return 0;
}
#if defined(DR_MP3_FLOAT_OUTPUT)
return drmp3_read_pcm_frames_raw(pMP3, framesToRead, pBufferOut);
#else
{
drmp3_int16 pTempS16[8192];
drmp3_uint64 totalPCMFramesRead = 0;
while (totalPCMFramesRead < framesToRead) {
drmp3_uint64 framesJustRead;
drmp3_uint64 framesRemaining = framesToRead - totalPCMFramesRead;
drmp3_uint64 framesToReadNow = DRMP3_COUNTOF(pTempS16) / pMP3->channels;
if (framesToReadNow > framesRemaining) {
framesToReadNow = framesRemaining;
}
framesJustRead = drmp3_read_pcm_frames_raw(pMP3, framesToReadNow, pTempS16);
if (framesJustRead == 0) {
break;
}
drmp3_s16_to_f32((float*)DRMP3_OFFSET_PTR(pBufferOut, sizeof(float) * totalPCMFramesRead * pMP3->channels), pTempS16, framesJustRead * pMP3->channels);
totalPCMFramesRead += framesJustRead;
}
return totalPCMFramesRead;
}
#endif
}
DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_s16(drmp3* pMP3, drmp3_uint64 framesToRead, drmp3_int16* pBufferOut)
{
if (pMP3 == NULL || pMP3->onRead == NULL) {
return 0;
}
#if !defined(DR_MP3_FLOAT_OUTPUT)
return drmp3_read_pcm_frames_raw(pMP3, framesToRead, pBufferOut);
#else
{
float pTempF32[4096];
drmp3_uint64 totalPCMFramesRead = 0;
while (totalPCMFramesRead < framesToRead) {
drmp3_uint64 framesJustRead;
drmp3_uint64 framesRemaining = framesToRead - totalPCMFramesRead;
drmp3_uint64 framesToReadNow = DRMP3_COUNTOF(pTempF32) / pMP3->channels;
if (framesToReadNow > framesRemaining) {
framesToReadNow = framesRemaining;
}
framesJustRead = drmp3_read_pcm_frames_raw(pMP3, framesToReadNow, pTempF32);
if (framesJustRead == 0) {
break;
}
drmp3_f32_to_s16((drmp3_int16*)DRMP3_OFFSET_PTR(pBufferOut, sizeof(drmp3_int16) * totalPCMFramesRead * pMP3->channels), pTempF32, framesJustRead * pMP3->channels);
totalPCMFramesRead += framesJustRead;
}
return totalPCMFramesRead;
}
#endif
}
static void drmp3_reset(drmp3* pMP3)
{
DRMP3_ASSERT(pMP3 != NULL);
pMP3->pcmFramesConsumedInMP3Frame = 0;
pMP3->pcmFramesRemainingInMP3Frame = 0;
pMP3->currentPCMFrame = 0;
pMP3->dataSize = 0;
pMP3->atEnd = DRMP3_FALSE;
drmp3dec_init(&pMP3->decoder);
}
static drmp3_bool32 drmp3_seek_to_start_of_stream(drmp3* pMP3)
{
DRMP3_ASSERT(pMP3 != NULL);
DRMP3_ASSERT(pMP3->onSeek != NULL);
if (!drmp3__on_seek(pMP3, 0, drmp3_seek_origin_start)) {
return DRMP3_FALSE;
}
drmp3_reset(pMP3);
return DRMP3_TRUE;
}
static drmp3_bool32 drmp3_seek_forward_by_pcm_frames__brute_force(drmp3* pMP3, drmp3_uint64 frameOffset)
{
drmp3_uint64 framesRead;
#if defined(DR_MP3_FLOAT_OUTPUT)
framesRead = drmp3_read_pcm_frames_f32(pMP3, frameOffset, NULL);
#else
framesRead = drmp3_read_pcm_frames_s16(pMP3, frameOffset, NULL);
#endif
if (framesRead != frameOffset) {
return DRMP3_FALSE;
}
return DRMP3_TRUE;
}
static drmp3_bool32 drmp3_seek_to_pcm_frame__brute_force(drmp3* pMP3, drmp3_uint64 frameIndex)
{
DRMP3_ASSERT(pMP3 != NULL);
if (frameIndex == pMP3->currentPCMFrame) {
return DRMP3_TRUE;
}
if (frameIndex < pMP3->currentPCMFrame) {
if (!drmp3_seek_to_start_of_stream(pMP3)) {
return DRMP3_FALSE;
}
}
DRMP3_ASSERT(frameIndex >= pMP3->currentPCMFrame);
return drmp3_seek_forward_by_pcm_frames__brute_force(pMP3, (frameIndex - pMP3->currentPCMFrame));
}
static drmp3_bool32 drmp3_find_closest_seek_point(drmp3* pMP3, drmp3_uint64 frameIndex, drmp3_uint32* pSeekPointIndex)
{
drmp3_uint32 iSeekPoint;
DRMP3_ASSERT(pSeekPointIndex != NULL);
*pSeekPointIndex = 0;
if (frameIndex < pMP3->pSeekPoints[0].pcmFrameIndex) {
return DRMP3_FALSE;
}
for (iSeekPoint = 0; iSeekPoint < pMP3->seekPointCount; ++iSeekPoint) {
if (pMP3->pSeekPoints[iSeekPoint].pcmFrameIndex > frameIndex) {
break;
}
*pSeekPointIndex = iSeekPoint;
}
return DRMP3_TRUE;
}
static drmp3_bool32 drmp3_seek_to_pcm_frame__seek_table(drmp3* pMP3, drmp3_uint64 frameIndex)
{
drmp3_seek_point seekPoint;
drmp3_uint32 priorSeekPointIndex;
drmp3_uint16 iMP3Frame;
drmp3_uint64 leftoverFrames;
DRMP3_ASSERT(pMP3 != NULL);
DRMP3_ASSERT(pMP3->pSeekPoints != NULL);
DRMP3_ASSERT(pMP3->seekPointCount > 0);
if (drmp3_find_closest_seek_point(pMP3, frameIndex, &priorSeekPointIndex)) {
seekPoint = pMP3->pSeekPoints[priorSeekPointIndex];
} else {
seekPoint.seekPosInBytes = 0;
seekPoint.pcmFrameIndex = 0;
seekPoint.mp3FramesToDiscard = 0;
seekPoint.pcmFramesToDiscard = 0;
}
if (!drmp3__on_seek_64(pMP3, seekPoint.seekPosInBytes, drmp3_seek_origin_start)) {
return DRMP3_FALSE;
}
drmp3_reset(pMP3);
for (iMP3Frame = 0; iMP3Frame < seekPoint.mp3FramesToDiscard; ++iMP3Frame) {
drmp3_uint32 pcmFramesRead;
drmp3d_sample_t* pPCMFrames;
pPCMFrames = NULL;
if (iMP3Frame == seekPoint.mp3FramesToDiscard-1) {
pPCMFrames = (drmp3d_sample_t*)pMP3->pcmFrames;
}
pcmFramesRead = drmp3_decode_next_frame_ex(pMP3, pPCMFrames);
if (pcmFramesRead == 0) {
return DRMP3_FALSE;
}
}
pMP3->currentPCMFrame = seekPoint.pcmFrameIndex - seekPoint.pcmFramesToDiscard;
leftoverFrames = frameIndex - pMP3->currentPCMFrame;
return drmp3_seek_forward_by_pcm_frames__brute_force(pMP3, leftoverFrames);
}
DRMP3_API drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex)
{
if (pMP3 == NULL || pMP3->onSeek == NULL) {
return DRMP3_FALSE;
}
if (frameIndex == 0) {
return drmp3_seek_to_start_of_stream(pMP3);
}
if (pMP3->pSeekPoints != NULL && pMP3->seekPointCount > 0) {
return drmp3_seek_to_pcm_frame__seek_table(pMP3, frameIndex);
} else {
return drmp3_seek_to_pcm_frame__brute_force(pMP3, frameIndex);
}
}
DRMP3_API drmp3_bool32 drmp3_get_mp3_and_pcm_frame_count(drmp3* pMP3, drmp3_uint64* pMP3FrameCount, drmp3_uint64* pPCMFrameCount)
{
drmp3_uint64 currentPCMFrame;
drmp3_uint64 totalPCMFrameCount;
drmp3_uint64 totalMP3FrameCount;
if (pMP3 == NULL) {
return DRMP3_FALSE;
}
if (pMP3->onSeek == NULL) {
return DRMP3_FALSE;
}
currentPCMFrame = pMP3->currentPCMFrame;
if (!drmp3_seek_to_start_of_stream(pMP3)) {
return DRMP3_FALSE;
}
totalPCMFrameCount = 0;
totalMP3FrameCount = 0;
for (;;) {
drmp3_uint32 pcmFramesInCurrentMP3Frame;
pcmFramesInCurrentMP3Frame = drmp3_decode_next_frame_ex(pMP3, NULL);
if (pcmFramesInCurrentMP3Frame == 0) {
break;
}
totalPCMFrameCount += pcmFramesInCurrentMP3Frame;
totalMP3FrameCount += 1;
}
if (!drmp3_seek_to_start_of_stream(pMP3)) {
return DRMP3_FALSE;
}
if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) {
return DRMP3_FALSE;
}
if (pMP3FrameCount != NULL) {
*pMP3FrameCount = totalMP3FrameCount;
}
if (pPCMFrameCount != NULL) {
*pPCMFrameCount = totalPCMFrameCount;
}
return DRMP3_TRUE;
}
DRMP3_API drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3)
{
drmp3_uint64 totalPCMFrameCount;
if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, NULL, &totalPCMFrameCount)) {
return 0;
}
return totalPCMFrameCount;
}
DRMP3_API drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3)
{
drmp3_uint64 totalMP3FrameCount;
if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, &totalMP3FrameCount, NULL)) {
return 0;
}
return totalMP3FrameCount;
}
static void drmp3__accumulate_running_pcm_frame_count(drmp3* pMP3, drmp3_uint32 pcmFrameCountIn, drmp3_uint64* pRunningPCMFrameCount, float* pRunningPCMFrameCountFractionalPart)
{
float srcRatio;
float pcmFrameCountOutF;
drmp3_uint32 pcmFrameCountOut;
srcRatio = (float)pMP3->mp3FrameSampleRate / (float)pMP3->sampleRate;
DRMP3_ASSERT(srcRatio > 0);
pcmFrameCountOutF = *pRunningPCMFrameCountFractionalPart + (pcmFrameCountIn / srcRatio);
pcmFrameCountOut = (drmp3_uint32)pcmFrameCountOutF;
*pRunningPCMFrameCountFractionalPart = pcmFrameCountOutF - pcmFrameCountOut;
*pRunningPCMFrameCount += pcmFrameCountOut;
}
typedef struct
{
drmp3_uint64 bytePos;
drmp3_uint64 pcmFrameIndex;
} drmp3__seeking_mp3_frame_info;
DRMP3_API drmp3_bool32 drmp3_calculate_seek_points(drmp3* pMP3, drmp3_uint32* pSeekPointCount, drmp3_seek_point* pSeekPoints)
{
drmp3_uint32 seekPointCount;
drmp3_uint64 currentPCMFrame;
drmp3_uint64 totalMP3FrameCount;
drmp3_uint64 totalPCMFrameCount;
if (pMP3 == NULL || pSeekPointCount == NULL || pSeekPoints == NULL) {
return DRMP3_FALSE;
}
seekPointCount = *pSeekPointCount;
if (seekPointCount == 0) {
return DRMP3_FALSE;
}
currentPCMFrame = pMP3->currentPCMFrame;
if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, &totalMP3FrameCount, &totalPCMFrameCount)) {
return DRMP3_FALSE;
}
if (totalMP3FrameCount < DRMP3_SEEK_LEADING_MP3_FRAMES+1) {
seekPointCount = 1;
pSeekPoints[0].seekPosInBytes = 0;
pSeekPoints[0].pcmFrameIndex = 0;
pSeekPoints[0].mp3FramesToDiscard = 0;
pSeekPoints[0].pcmFramesToDiscard = 0;
} else {
drmp3_uint64 pcmFramesBetweenSeekPoints;
drmp3__seeking_mp3_frame_info mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES+1];
drmp3_uint64 runningPCMFrameCount = 0;
float runningPCMFrameCountFractionalPart = 0;
drmp3_uint64 nextTargetPCMFrame;
drmp3_uint32 iMP3Frame;
drmp3_uint32 iSeekPoint;
if (seekPointCount > totalMP3FrameCount-1) {
seekPointCount = (drmp3_uint32)totalMP3FrameCount-1;
}
pcmFramesBetweenSeekPoints = totalPCMFrameCount / (seekPointCount+1);
if (!drmp3_seek_to_start_of_stream(pMP3)) {
return DRMP3_FALSE;
}
for (iMP3Frame = 0; iMP3Frame < DRMP3_SEEK_LEADING_MP3_FRAMES+1; ++iMP3Frame) {
drmp3_uint32 pcmFramesInCurrentMP3FrameIn;
DRMP3_ASSERT(pMP3->streamCursor >= pMP3->dataSize);
mp3FrameInfo[iMP3Frame].bytePos = pMP3->streamCursor - pMP3->dataSize;
mp3FrameInfo[iMP3Frame].pcmFrameIndex = runningPCMFrameCount;
pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL);
if (pcmFramesInCurrentMP3FrameIn == 0) {
return DRMP3_FALSE;
}
drmp3__accumulate_running_pcm_frame_count(pMP3, pcmFramesInCurrentMP3FrameIn, &runningPCMFrameCount, &runningPCMFrameCountFractionalPart);
}
nextTargetPCMFrame = 0;
for (iSeekPoint = 0; iSeekPoint < seekPointCount; ++iSeekPoint) {
nextTargetPCMFrame += pcmFramesBetweenSeekPoints;
for (;;) {
if (nextTargetPCMFrame < runningPCMFrameCount) {
pSeekPoints[iSeekPoint].seekPosInBytes = mp3FrameInfo[0].bytePos;
pSeekPoints[iSeekPoint].pcmFrameIndex = nextTargetPCMFrame;
pSeekPoints[iSeekPoint].mp3FramesToDiscard = DRMP3_SEEK_LEADING_MP3_FRAMES;
pSeekPoints[iSeekPoint].pcmFramesToDiscard = (drmp3_uint16)(nextTargetPCMFrame - mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES-1].pcmFrameIndex);
break;
} else {
size_t i;
drmp3_uint32 pcmFramesInCurrentMP3FrameIn;
for (i = 0; i < DRMP3_COUNTOF(mp3FrameInfo)-1; ++i) {
mp3FrameInfo[i] = mp3FrameInfo[i+1];
}
mp3FrameInfo[DRMP3_COUNTOF(mp3FrameInfo)-1].bytePos = pMP3->streamCursor - pMP3->dataSize;
mp3FrameInfo[DRMP3_COUNTOF(mp3FrameInfo)-1].pcmFrameIndex = runningPCMFrameCount;
pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL);
if (pcmFramesInCurrentMP3FrameIn == 0) {
pSeekPoints[iSeekPoint].seekPosInBytes = mp3FrameInfo[0].bytePos;
pSeekPoints[iSeekPoint].pcmFrameIndex = nextTargetPCMFrame;
pSeekPoints[iSeekPoint].mp3FramesToDiscard = DRMP3_SEEK_LEADING_MP3_FRAMES;
pSeekPoints[iSeekPoint].pcmFramesToDiscard = (drmp3_uint16)(nextTargetPCMFrame - mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES-1].pcmFrameIndex);
break;
}
drmp3__accumulate_running_pcm_frame_count(pMP3, pcmFramesInCurrentMP3FrameIn, &runningPCMFrameCount, &runningPCMFrameCountFractionalPart);
}
}
}
if (!drmp3_seek_to_start_of_stream(pMP3)) {
return DRMP3_FALSE;
}
if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) {
return DRMP3_FALSE;
}
}
*pSeekPointCount = seekPointCount;
return DRMP3_TRUE;
}
DRMP3_API drmp3_bool32 drmp3_bind_seek_table(drmp3* pMP3, drmp3_uint32 seekPointCount, drmp3_seek_point* pSeekPoints)
{
if (pMP3 == NULL) {
return DRMP3_FALSE;
}
if (seekPointCount == 0 || pSeekPoints == NULL) {
pMP3->seekPointCount = 0;
pMP3->pSeekPoints = NULL;
} else {
pMP3->seekPointCount = seekPointCount;
pMP3->pSeekPoints = pSeekPoints;
}
return DRMP3_TRUE;
}
static float* drmp3__full_read_and_close_f32(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
{
drmp3_uint64 totalFramesRead = 0;
drmp3_uint64 framesCapacity = 0;
float* pFrames = NULL;
float temp[4096];
DRMP3_ASSERT(pMP3 != NULL);
for (;;) {
drmp3_uint64 framesToReadRightNow = DRMP3_COUNTOF(temp) / pMP3->channels;
drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadRightNow, temp);
if (framesJustRead == 0) {
break;
}
if (framesCapacity < totalFramesRead + framesJustRead) {
drmp3_uint64 oldFramesBufferSize;
drmp3_uint64 newFramesBufferSize;
drmp3_uint64 newFramesCap;
float* pNewFrames;
newFramesCap = framesCapacity * 2;
if (newFramesCap < totalFramesRead + framesJustRead) {
newFramesCap = totalFramesRead + framesJustRead;
}
oldFramesBufferSize = framesCapacity * pMP3->channels * sizeof(float);
newFramesBufferSize = newFramesCap * pMP3->channels * sizeof(float);
if (newFramesBufferSize > (drmp3_uint64)DRMP3_SIZE_MAX) {
break;
}
pNewFrames = (float*)drmp3__realloc_from_callbacks(pFrames, (size_t)newFramesBufferSize, (size_t)oldFramesBufferSize, &pMP3->allocationCallbacks);
if (pNewFrames == NULL) {
drmp3__free_from_callbacks(pFrames, &pMP3->allocationCallbacks);
break;
}
pFrames = pNewFrames;
framesCapacity = newFramesCap;
}
DRMP3_COPY_MEMORY(pFrames + totalFramesRead*pMP3->channels, temp, (size_t)(framesJustRead*pMP3->channels*sizeof(float)));
totalFramesRead += framesJustRead;
if (framesJustRead != framesToReadRightNow) {
break;
}
}
if (pConfig != NULL) {
pConfig->channels = pMP3->channels;
pConfig->sampleRate = pMP3->sampleRate;
}
drmp3_uninit(pMP3);
if (pTotalFrameCount) {
*pTotalFrameCount = totalFramesRead;
}
return pFrames;
}
static drmp3_int16* drmp3__full_read_and_close_s16(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount)
{
drmp3_uint64 totalFramesRead = 0;
drmp3_uint64 framesCapacity = 0;
drmp3_int16* pFrames = NULL;
drmp3_int16 temp[4096];
DRMP3_ASSERT(pMP3 != NULL);
for (;;) {
drmp3_uint64 framesToReadRightNow = DRMP3_COUNTOF(temp) / pMP3->channels;
drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_s16(pMP3, framesToReadRightNow, temp);
if (framesJustRead == 0) {
break;
}
if (framesCapacity < totalFramesRead + framesJustRead) {
drmp3_uint64 newFramesBufferSize;
drmp3_uint64 oldFramesBufferSize;
drmp3_uint64 newFramesCap;
drmp3_int16* pNewFrames;
newFramesCap = framesCapacity * 2;
if (newFramesCap < totalFramesRead + framesJustRead) {
newFramesCap = totalFramesRead + framesJustRead;
}
oldFramesBufferSize = framesCapacity * pMP3->channels * sizeof(drmp3_int16);
newFramesBufferSize = newFramesCap * pMP3->channels * sizeof(drmp3_int16);
if (newFramesBufferSize > (drmp3_uint64)DRMP3_SIZE_MAX) {
break;
}
pNewFrames = (drmp3_int16*)drmp3__realloc_from_callbacks(pFrames, (size_t)newFramesBufferSize, (size_t)oldFramesBufferSize, &pMP3->allocationCallbacks);
if (pNewFrames == NULL) {
drmp3__free_from_callbacks(pFrames, &pMP3->allocationCallbacks);
break;
}
pFrames = pNewFrames;
framesCapacity = newFramesCap;
}
DRMP3_COPY_MEMORY(pFrames + totalFramesRead*pMP3->channels, temp, (size_t)(framesJustRead*pMP3->channels*sizeof(drmp3_int16)));
totalFramesRead += framesJustRead;
if (framesJustRead != framesToReadRightNow) {
break;
}
}
if (pConfig != NULL) {
pConfig->channels = pMP3->channels;
pConfig->sampleRate = pMP3->sampleRate;
}
drmp3_uninit(pMP3);
if (pTotalFrameCount) {
*pTotalFrameCount = totalFramesRead;
}
return pFrames;
}
DRMP3_API float* drmp3_open_and_read_pcm_frames_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks)
{
drmp3 mp3;
if (!drmp3_init(&mp3, onRead, onSeek, pUserData, pAllocationCallbacks)) {
return NULL;
}
return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount);
}
DRMP3_API drmp3_int16* drmp3_open_and_read_pcm_frames_s16(drmp3_read_proc onRead, drmp3_seek_proc onSeek, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks)
{
drmp3 mp3;
if (!drmp3_init(&mp3, onRead, onSeek, pUserData, pAllocationCallbacks)) {
return NULL;
}
return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount);
}
DRMP3_API float* drmp3_open_memory_and_read_pcm_frames_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks)
{
drmp3 mp3;
if (!drmp3_init_memory(&mp3, pData, dataSize, pAllocationCallbacks)) {
return NULL;
}
return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount);
}
DRMP3_API drmp3_int16* drmp3_open_memory_and_read_pcm_frames_s16(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks)
{
drmp3 mp3;
if (!drmp3_init_memory(&mp3, pData, dataSize, pAllocationCallbacks)) {
return NULL;
}
return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount);
}
#ifndef DR_MP3_NO_STDIO
DRMP3_API float* drmp3_open_file_and_read_pcm_frames_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks)
{
drmp3 mp3;
if (!drmp3_init_file(&mp3, filePath, pAllocationCallbacks)) {
return NULL;
}
return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount);
}
DRMP3_API drmp3_int16* drmp3_open_file_and_read_pcm_frames_s16(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks)
{
drmp3 mp3;
if (!drmp3_init_file(&mp3, filePath, pAllocationCallbacks)) {
return NULL;
}
return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount);
}
#endif
DRMP3_API void* drmp3_malloc(size_t sz, const drmp3_allocation_callbacks* pAllocationCallbacks)
{
if (pAllocationCallbacks != NULL) {
return drmp3__malloc_from_callbacks(sz, pAllocationCallbacks);
} else {
return drmp3__malloc_default(sz, NULL);
}
}
DRMP3_API void drmp3_free(void* p, const drmp3_allocation_callbacks* pAllocationCallbacks)
{
if (pAllocationCallbacks != NULL) {
drmp3__free_from_callbacks(p, pAllocationCallbacks);
} else {
drmp3__free_default(p, NULL);
}
}
#endif
/* dr_mp3_c end */
#endif /* DRMP3_IMPLEMENTATION */
#endif /* MA_NO_MP3 */
/* End globally disabled warnings. */
#if defined(_MSC_VER)
#pragma warning(pop)
#endif
#endif /* miniaudio_c */
#endif /* MINIAUDIO_IMPLEMENTATION */
/*
This software is available as a choice of the following licenses. Choose
whichever you prefer.
===============================================================================
ALTERNATIVE 1 - Public Domain (www.unlicense.org)
===============================================================================
This is free and unencumbered software released into the public domain.
Anyone is free to copy, modify, publish, use, compile, sell, or distribute this
software, either in source code form or as a compiled binary, for any purpose,
commercial or non-commercial, and by any means.
In jurisdictions that recognize copyright laws, the author or authors of this
software dedicate any and all copyright interest in the software to the public
domain. We make this dedication for the benefit of the public at large and to
the detriment of our heirs and successors. We intend this dedication to be an
overt act of relinquishment in perpetuity of all present and future rights to
this software under copyright law.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
( run in 1.599 second using v1.01-cache-2.11-cpan-e1769b4cff6 )