App-MHFS
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lib/App/MHFS.pm view on Meta::CPAN
# save references to before
my @potentialupdates = ('html', 'musicdbhtml', 'musicdbjson');
my %before;
foreach my $pupdate (@potentialupdates) {
$before{$pupdate} = $self->{$pupdate};
}
# build the new libraries
$self->BuildLibraries();
# determine what needs to be updated
my @updates = (['sources', $self->{'sources'}]);
foreach my $pupdate(@potentialupdates) {
if($before{$pupdate} ne $self->{$pupdate}) {
push @updates, [$pupdate, $self->{$pupdate}];
}
}
# serialize and output
my $pipedata = freeze(\@updates);
print $datachannel $pipedata;
lib/App/MHFS.pm view on Meta::CPAN
warn("thaw threw exception");
}
}
if($unthawed){
foreach my $update (@$unthawed) {
say "Updating " . $update->[0];
$self->{$update->[0]} = $update->[1];
}
}
else {
say "failed to thaw, library not updated.";
}
$onUpdateEnd->();
});
}
sub new {
my ($class, $settings) = @_;
my $self = {'settings' => $settings};
bless $self, $class;
my $pstart = __PACKAGE__.":";
share/public_html/static/hls.js view on Meta::CPAN
// fired after manifest has been parsed - data: { levels : [available quality levels], firstLevel : index of first quality level appearing in Manifest}
MANIFEST_PARSED: 'hlsManifestParsed',
// fired when a level switch is requested - data: { level : id of new level }
LEVEL_SWITCHING: 'hlsLevelSwitching',
// fired when a level switch is effective - data: { level : id of new level }
LEVEL_SWITCHED: 'hlsLevelSwitched',
// fired when a level playlist loading starts - data: { url : level URL, level : id of level being loaded}
LEVEL_LOADING: 'hlsLevelLoading',
// fired when a level playlist loading finishes - data: { details : levelDetails object, level : id of loaded level, stats : { trequest, tfirst, tload, mtime} }
LEVEL_LOADED: 'hlsLevelLoaded',
// fired when a level's details have been updated based on previous details, after it has been loaded - data: { details : levelDetails object, level : id of updated level }
LEVEL_UPDATED: 'hlsLevelUpdated',
// fired when a level's PTS information has been updated after parsing a fragment - data: { details : levelDetails object, level : id of updated level, drift: PTS drift observed when parsing last fragment }
LEVEL_PTS_UPDATED: 'hlsLevelPtsUpdated',
// fired to notify that audio track lists has been updated - data: { audioTracks : audioTracks }
AUDIO_TRACKS_UPDATED: 'hlsAudioTracksUpdated',
// fired when an audio track switching is requested - data: { id : audio track id }
AUDIO_TRACK_SWITCHING: 'hlsAudioTrackSwitching',
// fired when an audio track switch actually occurs - data: { id : audio track id }
AUDIO_TRACK_SWITCHED: 'hlsAudioTrackSwitched',
// fired when an audio track loading starts - data: { url : audio track URL, id : audio track id }
AUDIO_TRACK_LOADING: 'hlsAudioTrackLoading',
// fired when an audio track loading finishes - data: { details : levelDetails object, id : audio track id, stats : { trequest, tfirst, tload, mtime } }
AUDIO_TRACK_LOADED: 'hlsAudioTrackLoaded',
// fired to notify that subtitle track lists has been updated - data: { subtitleTracks : subtitleTracks }
SUBTITLE_TRACKS_UPDATED: 'hlsSubtitleTracksUpdated',
// fired when an subtitle track switch occurs - data: { id : subtitle track id }
SUBTITLE_TRACK_SWITCH: 'hlsSubtitleTrackSwitch',
// fired when a subtitle track loading starts - data: { url : subtitle track URL, id : subtitle track id }
SUBTITLE_TRACK_LOADING: 'hlsSubtitleTrackLoading',
// fired when a subtitle track loading finishes - data: { details : levelDetails object, id : subtitle track id, stats : { trequest, tfirst, tload, mtime } }
SUBTITLE_TRACK_LOADED: 'hlsSubtitleTrackLoaded',
// fired when a subtitle fragment has been processed - data: { success : boolean, frag : the processed frag }
SUBTITLE_FRAG_PROCESSED: 'hlsSubtitleFragProcessed',
// fired when the first timestamp is found - data: { id : demuxer id, initPTS: initPTS, frag : fragment object }
share/public_html/static/hls.js view on Meta::CPAN
var fragmentEntity = {
body: fragment,
range: Object.create(null),
buffered: false
};
this.fragments[fragKey] = fragmentEntity;
}
};
/**
* Fires when the buffer is updated
*/
FragmentTracker.prototype.onBufferAppended = function onBufferAppended(e) {
var _this5 = this;
// Store the latest timeRanges loaded in the buffer
this.timeRanges = e.timeRanges;
Object.keys(this.timeRanges).forEach(function (elementaryStream) {
var timeRange = _this5.timeRanges[elementaryStream];
share/public_html/static/hls.js view on Meta::CPAN
// tick to speed up processing
this.tick();
};
AudioStreamController.prototype.onMediaEnded = function onMediaEnded() {
// reset startPosition and lastCurrentTime to restart playback @ stream beginning
this.startPosition = this.lastCurrentTime = 0;
};
AudioStreamController.prototype.onAudioTracksUpdated = function onAudioTracksUpdated(data) {
logger["b" /* logger */].log('audio tracks updated');
this.tracks = data.audioTracks;
};
AudioStreamController.prototype.onAudioTrackSwitching = function onAudioTrackSwitching(data) {
// if any URL found on new audio track, it is an alternate audio track
var altAudio = !!data.url;
this.trackId = data.id;
this.fragCurrent = null;
this.state = audio_stream_controller_State.PAUSED;
share/public_html/static/hls.js view on Meta::CPAN
});
}
};
// Got all new subtitle tracks.
SubtitleStreamController.prototype.onSubtitleTracksUpdated = function onSubtitleTracksUpdated(data) {
var _this4 = this;
logger["b" /* logger */].log('subtitle tracks updated');
this.tracks = data.subtitleTracks;
this.clearVttFragQueues();
this.vttFragSNsProcessed = {};
this.tracks.forEach(function (track) {
_this4.vttFragSNsProcessed[track.id] = [];
});
};
SubtitleStreamController.prototype.onSubtitleTrackSwitch = function onSubtitleTrackSwitch(data) {
this.currentTrackId = data.id;
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
drflac_uint32 consumedBits;
/*
The cached data which was most recently read from the client. There are two levels of cache. Data flows as such:
Client -> L2 -> L1. The L2 -> L1 movement is aligned and runs on a fast path in just a few instructions.
*/
drflac_cache_t cacheL2[DR_FLAC_BUFFER_SIZE/sizeof(drflac_cache_t)];
drflac_cache_t cache;
/*
CRC-16. This is updated whenever bits are read from the bit stream. Manually set this to 0 to reset the CRC. For FLAC, this
is reset to 0 at the beginning of each frame.
*/
drflac_uint16 crc16;
drflac_cache_t crc16Cache; /* A cache for optimizing CRC calculations. This is filled when when the L1 cache is reloaded. */
drflac_uint32 crc16CacheIgnoredBytes; /* The number of bytes to ignore when updating the CRC-16 from the CRC-16 cache. */
} drflac_bs;
typedef struct
{
/* The type of the subframe: SUBFRAME_CONSTANT, SUBFRAME_VERBATIM, SUBFRAME_FIXED or SUBFRAME_LPC. */
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
#endif
}
// we already tried this byte and there are no more to try, break out
if(targetByte == lastTargetByte)
{
return DRFLAC_FALSE;
}
}
/* The current PCM frame needs to be updated based on the frame we just seeked to. */
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)
{
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
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.
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.
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
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.
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
Return Value
------------
A `ma_context_config` initialized to defaults.
Remarks
-------
You must always use this to initialize the default state of the `ma_context_config` object. Not using this will result in your program breaking when miniaudio
is updated and new members are added to `ma_context_config`. It also sets logical defaults.
You can override members of the returned object by changing it's members directly.
See Also
--------
ma_context_init()
*/
MA_API ma_context_config ma_context_config_init(void);
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
AudioChannelBitmap bit = bitmap & (1 << iBit);
if (bit != 0) {
pChannelMap[iChannel++] = ma_channel_from_AudioChannelBit(bit);
}
}
} else
#endif
{
/*
Need to use the tag to determine the channel map. For now I'm just assuming a default channel map, but later on this should
be updated to determine the mapping based on the tag.
*/
UInt32 channelCount;
/* Our channel map retrieval APIs below take 32-bit integers, so we'll want to clamp the channel map capacity. */
if (channelMapCap > 0xFFFFFFFF) {
channelMapCap = 0xFFFFFFFF;
}
channelCount = ma_min(AudioChannelLayoutTag_GetNumberOfChannels(pChannelLayout->mChannelLayoutTag), (UInt32)channelMapCap);
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
if (pDecoder->onReadPCMFrames == NULL) {
return 0;
}
/* Fast path. */
if (pDecoder->converter.isPassthrough) {
totalFramesReadOut = pDecoder->onReadPCMFrames(pDecoder, pFramesOut, frameCount);
} 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) {
totalFramesReadOut = pDecoder->onReadPCMFrames(pDecoder, NULL, frameCount); /* All decoder backends must support passing in NULL for the output buffer. */
} else {
/* Slow path. Need to run everything through the data converter. */
totalFramesReadOut = 0;
totalFramesReadIn = 0;
pRunningFramesOut = pFramesOut;
while (totalFramesReadOut < frameCount) {
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
`lpfOrder` option, which has a maximum value of MA_MAX_FILTER_ORDER.
- Data conversion now supports s16 natively which runs through a fixed point pipeline. Previously everything needed to be converted to floating point before
processing, whereas now both s16 and f32 are natively supported. Other formats still require conversion to either s16 or f32 prior to processing, however
`ma_data_converter` will handle this for you.
Custom Memory Allocators
------------------------
miniaudio has always supported macro level customization for memory allocation via MA_MALLOC, MA_REALLOC and MA_FREE, however some scenarios require more
flexibility by allowing a user data pointer to be passed to the custom allocation routines. Support for this has been added to version 0.10 via the
`ma_allocation_callbacks` structure. Anything making use of heap allocations has been updated to accept this new structure.
The `ma_context_config` structure has been updated with a new member called `allocationCallbacks`. Leaving this set to it's defaults returned by
`ma_context_config_init()` will cause it to use MA_MALLOC, MA_REALLOC and MA_FREE. Likewise, The `ma_decoder_config` structure has been updated in the same
way, and leaving everything as-is after `ma_decoder_config_init()` will cause it to use the same defaults.
The following APIs have been updated to take a pointer to a `ma_allocation_callbacks` object. Setting this parameter to NULL will cause it to use defaults.
Otherwise they will use the relevant callback in the structure.
- ma_malloc()
- ma_realloc()
- ma_free()
- ma_aligned_malloc()
- ma_aligned_free()
- ma_rb_init() / ma_rb_init_ex()
- ma_pcm_rb_init() / ma_pcm_rb_init_ex()
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
- Fix a bug where opening a file would return a success code when in fact it failed.
- Fix compilation errors with Visual Studio 6 and 2003.
- Fix warnings on macOS.
v0.10.0 - 2020-03-07
- API CHANGE: Refactor data conversion APIs
- ma_format_converter has been removed. Use ma_convert_pcm_frames_format() instead.
- ma_channel_router has been replaced with ma_channel_converter.
- ma_src has been replaced with ma_resampler
- ma_pcm_converter has been replaced with ma_data_converter
- API CHANGE: Add support for custom memory allocation callbacks. The following APIs have been updated to take an extra parameter for the allocation
callbacks:
- ma_malloc()
- ma_realloc()
- ma_free()
- ma_aligned_malloc()
- ma_aligned_free()
- ma_rb_init() / ma_rb_init_ex()
- ma_pcm_rb_init() / ma_pcm_rb_init_ex()
- API CHANGE: Simplify latency specification in device configurations. The bufferSizeInFrames and bufferSizeInMilliseconds parameters have been replaced with
periodSizeInFrames and periodSizeInMilliseconds respectively. The previous variables defined the size of the entire buffer, whereas the new ones define the
size of a period. The following APIs have been removed since they are no longer relevant:
- ma_get_default_buffer_size_in_milliseconds()
- ma_get_default_buffer_size_in_frames()
- API CHANGE: ma_device_set_stop_callback() has been removed. If you require a stop callback, you must now set it via the device config just like the data
callback.
- API CHANGE: The ma_sine_wave API has been replaced with ma_waveform. The following APIs have been removed:
- ma_sine_wave_init()
- ma_sine_wave_read_f32()
- ma_sine_wave_read_f32_ex()
- API CHANGE: ma_convert_frames() has been updated to take an extra parameter which is the size of the output buffer in PCM frames. Parameters have also been
reordered.
- API CHANGE: ma_convert_frames_ex() has been changed to take a pointer to a ma_data_converter_config object to specify the input and output formats to
convert between.
- API CHANGE: ma_calculate_frame_count_after_src() has been renamed to ma_calculate_frame_count_after_resampling().
- Add support for the following filters:
- Biquad (ma_biquad)
- First order low-pass (ma_lpf1)
- Second order low-pass (ma_lpf2)
- Low-pass with configurable order (ma_lpf)
- First order high-pass (ma_hpf1)
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
- Add AAudio backend (Android Audio). This is the new priority backend for Android. Support for AAudio starts
with Android 8. OpenSL|ES is used as a fallback for older versions of Android.
- Remove OpenAL and SDL backends.
- Fix a possible deadlock when rapidly stopping the device after it has started.
- Update documentation.
- Change licensing to a choice of public domain _or_ MIT-0 (No Attribution).
v0.8.14 - 2018-12-16
- Core Audio: Fix a bug where the device state is not set correctly after stopping.
- Add support for custom weights to the channel router.
- Update decoders to use updated APIs in dr_flac, dr_mp3 and dr_wav.
v0.8.13 - 2018-12-04
- Core Audio: Fix a bug with channel mapping.
- Fix a bug with channel routing where the back/left and back/right channels have the wrong weight.
v0.8.12 - 2018-11-27
- Drop support for SDL 1.2. The Emscripten build now requires "-s USE_SDL=2".
- Fix a linking error with ALSA.
- Fix a bug on iOS where the device name is not set correctly.
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
drflac_uint32 consumedBits;
/*
The cached data which was most recently read from the client. There are two levels of cache. Data flows as such:
Client -> L2 -> L1. The L2 -> L1 movement is aligned and runs on a fast path in just a few instructions.
*/
drflac_cache_t cacheL2[DR_FLAC_BUFFER_SIZE/sizeof(drflac_cache_t)];
drflac_cache_t cache;
/*
CRC-16. This is updated whenever bits are read from the bit stream. Manually set this to 0 to reset the CRC. For FLAC, this
is reset to 0 at the beginning of each frame.
*/
drflac_uint16 crc16;
drflac_cache_t crc16Cache; /* A cache for optimizing CRC calculations. This is filled when when the L1 cache is reloaded. */
drflac_uint32 crc16CacheIgnoredBytes; /* The number of bytes to ignore when updating the CRC-16 from the CRC-16 cache. */
} drflac_bs;
typedef struct
{
/* The type of the subframe: SUBFRAME_CONSTANT, SUBFRAME_VERBATIM, SUBFRAME_FIXED or SUBFRAME_LPC. */
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
}
#endif
}
/* We already tried this byte and there are no more to try, break out. */
if(targetByte == lastTargetByte) {
return DRFLAC_FALSE;
}
}
/* The current PCM frame needs to be updated based on the frame we just seeked to. */
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)
{
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
`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
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
ma_standard_sample_rate_16000 = 16000, /* Extreme lows */
ma_standard_sample_rate_11025 = 11250,
ma_standard_sample_rate_8000 = 8000,
ma_standard_sample_rate_352800 = 352800, /* Extreme highs */
ma_standard_sample_rate_384000 = 384000,
ma_standard_sample_rate_min = ma_standard_sample_rate_8000,
ma_standard_sample_rate_max = ma_standard_sample_rate_384000,
ma_standard_sample_rate_count = 14 /* Need to maintain the count manually. Make sure this is updated if items are added to enum. */
} ma_standard_sample_rate;
typedef enum
{
ma_channel_mix_mode_rectangular = 0, /* Simple averaging based on the plane(s) the channel is sitting on. */
ma_channel_mix_mode_simple, /* Drop excess channels; zeroed out extra channels. */
ma_channel_mix_mode_custom_weights, /* Use custom weights specified in ma_channel_router_config. */
ma_channel_mix_mode_default = ma_channel_mix_mode_rectangular
} ma_channel_mix_mode;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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.
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
Return Value
------------
A `ma_context_config` initialized to defaults.
Remarks
-------
You must always use this to initialize the default state of the `ma_context_config` object. Not using this will result in your program breaking when miniaudio
is updated and new members are added to `ma_context_config`. It also sets logical defaults.
You can override members of the returned object by changing it's members directly.
See Also
--------
ma_context_init()
*/
MA_API ma_context_config ma_context_config_init(void);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
};
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. */
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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);
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.pAudioClientCapture, pDevice->wasapi.hEventCapture);
pDevice->wasapi.periodSizeInFramesCapture = data.periodSizeInFramesOut;
ma_IAudioClient_GetBufferSize((ma_IAudioClient*)pDevice->wasapi.pAudioClientCapture, &pDevice->wasapi.actualBufferSizeInFramesCapture);
/* We must always have a valid ID. */
ma_wcscpy_s(pDevice->capture.id.wasapi, sizeof(pDevice->capture.id.wasapi), data.id.wasapi);
/* The descriptor needs to be updated with actual values. */
pDescriptorCapture->format = data.formatOut;
pDescriptorCapture->channels = data.channelsOut;
pDescriptorCapture->sampleRate = data.sampleRateOut;
MA_COPY_MEMORY(pDescriptorCapture->channelMap, data.channelMapOut, sizeof(data.channelMapOut));
pDescriptorCapture->periodSizeInFrames = data.periodSizeInFramesOut;
pDescriptorCapture->periodCount = data.periodsOut;
}
if (pConfig->deviceType == ma_device_type_playback || pConfig->deviceType == ma_device_type_duplex) {
ma_device_init_internal_data__wasapi data;
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
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
AudioChannelBitmap bit = bitmap & (1 << iBit);
if (bit != 0) {
pChannelMap[iChannel++] = ma_channel_from_AudioChannelBit(bit);
}
}
} else
#endif
{
/*
Need to use the tag to determine the channel map. For now I'm just assuming a default channel map, but later on this should
be updated to determine the mapping based on the tag.
*/
UInt32 channelCount;
/* Our channel map retrieval APIs below take 32-bit integers, so we'll want to clamp the channel map capacity. */
if (channelMapCap > 0xFFFFFFFF) {
channelMapCap = 0xFFFFFFFF;
}
channelCount = ma_min(AudioChannelLayoutTag_GetNumberOfChannels(pChannelLayout->mChannelLayoutTag), (UInt32)channelMapCap);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
{
NSInteger type = [[[pNotification userInfo] objectForKey:AVAudioSessionInterruptionTypeKey] integerValue];
switch (type)
{
case AVAudioSessionInterruptionTypeBegan:
{
ma_log_postf(ma_device_get_log(m_pDevice), MA_LOG_LEVEL_INFO, "[Core Audio] Interruption: AVAudioSessionInterruptionTypeBegan\n");
/*
Core Audio will have stopped the internal device automatically, but we need explicitly
stop it at a higher level to ensure miniaudio-specific state is updated for consistency.
*/
ma_device_stop(m_pDevice);
/*
Fire the notification after the device has been stopped to ensure it's in the correct
state when the notification handler is invoked.
*/
ma_device__on_notification_interruption_began(m_pDevice);
} break;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
} else {
pDataBufferNode->pParent->pChildHi = pReplacementDataBufferNode;
}
}
/* Now need to update the replacement node's pointers. */
pReplacementDataBufferNode->pParent = pDataBufferNode->pParent;
pReplacementDataBufferNode->pChildLo = pDataBufferNode->pChildLo;
pReplacementDataBufferNode->pChildHi = pDataBufferNode->pChildHi;
/* Now the children of the replacement node need to have their parent pointers updated. */
if (pReplacementDataBufferNode->pChildLo != NULL) {
pReplacementDataBufferNode->pChildLo->pParent = pReplacementDataBufferNode;
}
if (pReplacementDataBufferNode->pChildHi != NULL) {
pReplacementDataBufferNode->pChildHi->pParent = pReplacementDataBufferNode;
}
/* Now the root node needs to be updated. */
if (pResourceManager->pRootDataBufferNode == pDataBufferNode) {
pResourceManager->pRootDataBufferNode = pReplacementDataBufferNode;
}
}
}
return MA_SUCCESS;
}
#if 0 /* Unused for now. */
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
/*
At this point the data buffer is either fully decoded or some error occurred. Either
way, the decoder is no longer necessary.
*/
ma_decoder_uninit(pDecoder);
ma_free(pDecoder, &pResourceManager->config.allocationCallbacks);
}
/* Getting here means we were successful. Make sure the status of the node is updated accordingly. */
c89atomic_exchange_i32(&pDataBufferNode->result, result);
} else {
/* Loading asynchronously. We may need to wait for initialization. */
if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_WAIT_INIT) != 0) {
ma_resource_manager_inline_notification_wait(&initNotification);
}
}
} else {
/* The data is not managed by the resource manager so there's nothing else to do. */
MA_ASSERT(pExistingData != NULL);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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. */
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
/*
We cannot use the output bus lock here since it'll be getting used at a higher level, but we do
still need to use the input bus lock since we'll be updating pointers on two different output
buses. The same rules apply here as the attaching case. Although we're using a lock here, we're
*not* using a lock when iterating over the list in the audio thread. We therefore need to craft
this in a way such that the iteration on the audio thread doesn't break.
The the first thing to do is swap out the "next" pointer of the previous output bus with the
new "next" output bus. This is the operation that matters for iteration on the audio thread.
After that, the previous pointer on the new "next" pointer needs to be updated, after which
point the linked list will be in a good state.
*/
ma_node_input_bus_lock(pInputBus);
{
ma_node_output_bus* pOldPrev = (ma_node_output_bus*)c89atomic_load_ptr(&pOutputBus->pPrev);
ma_node_output_bus* pOldNext = (ma_node_output_bus*)c89atomic_load_ptr(&pOutputBus->pNext);
if (pOldPrev != NULL) {
c89atomic_exchange_ptr(&pOldPrev->pNext, pOldNext); /* <-- This is where the output bus is detached from the list. */
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
{
ma_node_output_bus* pOldInputNode = (ma_node_output_bus*)c89atomic_load_ptr(&pOutputBus->pInputNode);
/* Detach from any existing attachment first if necessary. */
if (pOldInputNode != NULL) {
ma_node_input_bus_detach__no_output_bus_lock(pInputBus, pOutputBus);
}
/*
At this point we can be sure the output bus is not attached to anything. The linked list in the
old input bus has been updated so that pOutputBus will not get iterated again.
*/
pOutputBus->pInputNode = pNewInputNode; /* No need for an atomic assignment here because modification of this variable always happens within a lock. */
pOutputBus->inputNodeInputBusIndex = (ma_uint8)inputNodeInputBusIndex; /* As above. */
/*
Now we need to attach the output bus to the linked list. This involves updating two pointers on
two different output buses so I'm going to go ahead and keep this simple and just use a lock.
There are ways to do this without a lock, but it's just too hard to maintain for it's value.
Although we're locking here, it's important to remember that we're *not* locking when iterating
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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,
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
}
}
*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);
}
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