App-MHFS
view release on metacpan or search on metacpan
lib/App/MHFS.pm view on Meta::CPAN
my $is_unicode = $1;
$torrent{'name'} = $2;
$torrent{'hash'} = $3;
$torrent{'size_bytes'} = $4;
$torrent{'bytes_done'} = $5;
$torrent{'private'} = $6;
if($is_unicode) {
my $escaped_unicode = $torrent{'name'};
$torrent{'name'} =~ s/\\u(.{4})/chr(hex($1))/eg;
$torrent{'name'} =~ s/\\x(.{2})/chr(hex($1))/eg;
my $decoded_as = $torrent{'name'};
$torrent{'name'} = ${escape_html($torrent{'name'})};
if($qs->{'logunicode'}) {
say 'unicode escaped: ' . $escaped_unicode;
say 'decoded as: ' . $decoded_as;
say 'html escaped ' . $torrent{'name'};
}
}
$buf .= '<tr><td>' . $torrent{'name'} . '</td><td>' . $torrent{'hash'} . '</td><td>' . $torrent{'size_bytes'} . '</td><td>' . $torrent{'bytes_done'} . '</td><td>' . $torrent{'private'} . '</td></tr>';
$curtor = '';
}
else {
my $line = shift @lines;
if(! $line) {
last;
share/public_html/static/hls.js view on Meta::CPAN
if (!sync) {
return;
}
}
};
// AVC - encrypt one 16 bytes block out of ten, starting from offset 32
SampleAesDecrypter.prototype.getAvcEncryptedData = function getAvcEncryptedData(decodedData) {
var encryptedDataLen = Math.floor((decodedData.length - 48) / 160) * 16 + 16;
var encryptedData = new Int8Array(encryptedDataLen);
var outputPos = 0;
for (var inputPos = 32; inputPos <= decodedData.length - 16; inputPos += 160, outputPos += 16) {
encryptedData.set(decodedData.subarray(inputPos, inputPos + 16), outputPos);
}
return encryptedData;
};
SampleAesDecrypter.prototype.getAvcDecryptedUnit = function getAvcDecryptedUnit(decodedData, decryptedData) {
decryptedData = new Uint8Array(decryptedData);
var inputPos = 0;
for (var outputPos = 32; outputPos <= decodedData.length - 16; outputPos += 160, inputPos += 16) {
decodedData.set(decryptedData.subarray(inputPos, inputPos + 16), outputPos);
}
return decodedData;
};
SampleAesDecrypter.prototype.decryptAvcSample = function decryptAvcSample(samples, sampleIndex, unitIndex, callback, curUnit, sync) {
var decodedData = this.discardEPB(curUnit.data);
var encryptedData = this.getAvcEncryptedData(decodedData);
var localthis = this;
this.decryptBuffer(encryptedData.buffer, function (decryptedData) {
curUnit.data = localthis.getAvcDecryptedUnit(decodedData, decryptedData);
if (!sync) {
localthis.decryptAvcSamples(samples, sampleIndex, unitIndex + 1, callback);
}
});
};
SampleAesDecrypter.prototype.decryptAvcSamples = function decryptAvcSamples(samples, sampleIndex, unitIndex, callback) {
for (;; sampleIndex++, unitIndex = 0) {
if (sampleIndex >= samples.length) {
share/public_html/static/hls.js view on Meta::CPAN
}
avcSample.frame = true;
var data = unit.data;
// only check slice type to detect KF in case SPS found in same packet (any keyframe is preceded by SPS ...)
if (spsfound && data.length > 4) {
// retrieve slice type by parsing beginning of NAL unit (follow H264 spec, slice_header definition) to detect keyframe embedded in NDR
var sliceType = new exp_golomb(data).readSliceType();
// 2 : I slice, 4 : SI slice, 7 : I slice, 9: SI slice
// SI slice : A slice that is coded using intra prediction only and using quantisation of the prediction samples.
// An SI slice can be coded such that its decoded samples can be constructed identically to an SP slice.
// I slice: A slice that is not an SI slice that is decoded using intra prediction only.
// if (sliceType === 2 || sliceType === 7) {
if (sliceType === 2 || sliceType === 4 || sliceType === 7 || sliceType === 9) {
avcSample.key = true;
}
}
break;
// IDR
case 5:
push = true;
// handle PES not starting with AUD
share/public_html/static/hls.js view on Meta::CPAN
var fragment = data.frag;
var samples = data.samples;
// create track dynamically
if (!this.id3Track) {
this.id3Track = this.getID3Track(this.media.textTracks);
this.id3Track.mode = 'hidden';
}
// Attempt to recreate Safari functionality by creating
// WebKitDataCue objects when available and store the decoded
// ID3 data in the value property of the cue
var Cue = window.WebKitDataCue || window.VTTCue || window.TextTrackCue;
for (var i = 0; i < samples.length; i++) {
var frames = id3["a" /* default */].getID3Frames(samples[i].data);
if (frames) {
var startTime = samples[i].pts;
var endTime = i < samples.length - 1 ? samples[i + 1].pts : fragment.endPTS;
// Give a slight bump to the endTime if it's equal to startTime to avoid a SyntaxError in IE
share/public_html/static/hls.js view on Meta::CPAN
var video = this.video = data.media instanceof window.HTMLVideoElement ? data.media : null;
if (typeof video.getVideoPlaybackQuality === 'function') {
this.isVideoPlaybackQualityAvailable = true;
}
clearInterval(this.timer);
this.timer = setInterval(this.checkFPSInterval.bind(this), config.fpsDroppedMonitoringPeriod);
}
};
FPSController.prototype.checkFPS = function checkFPS(video, decodedFrames, droppedFrames) {
var currentTime = fps_controller_performance.now();
if (decodedFrames) {
if (this.lastTime) {
var currentPeriod = currentTime - this.lastTime,
currentDropped = droppedFrames - this.lastDroppedFrames,
currentDecoded = decodedFrames - this.lastDecodedFrames,
droppedFPS = 1000 * currentDropped / currentPeriod,
hls = this.hls;
hls.trigger(events["a" /* default */].FPS_DROP, { currentDropped: currentDropped, currentDecoded: currentDecoded, totalDroppedFrames: droppedFrames });
if (droppedFPS > 0) {
// logger.log('checkFPS : droppedFPS/decodedFPS:' + droppedFPS/(1000 * currentDecoded / currentPeriod));
if (currentDropped > hls.config.fpsDroppedMonitoringThreshold * currentDecoded) {
var currentLevel = hls.currentLevel;
logger["b" /* logger */].warn('drop FPS ratio greater than max allowed value for currentLevel: ' + currentLevel);
if (currentLevel > 0 && (hls.autoLevelCapping === -1 || hls.autoLevelCapping >= currentLevel)) {
currentLevel = currentLevel - 1;
hls.trigger(events["a" /* default */].FPS_DROP_LEVEL_CAPPING, { level: currentLevel, droppedLevel: hls.currentLevel });
hls.autoLevelCapping = currentLevel;
hls.streamController.nextLevelSwitch();
}
}
}
}
this.lastTime = currentTime;
this.lastDroppedFrames = droppedFrames;
this.lastDecodedFrames = decodedFrames;
}
};
FPSController.prototype.checkFPSInterval = function checkFPSInterval() {
var video = this.video;
if (video) {
if (this.isVideoPlaybackQualityAvailable) {
var videoPlaybackQuality = video.getVideoPlaybackQuality();
this.checkFPS(video, videoPlaybackQuality.totalVideoFrames, videoPlaybackQuality.droppedVideoFrames);
} else {
share/public_html/static/jsmpeg.min.js view on Meta::CPAN
var JSMpeg={Player:null,VideoElement:null,BitBuffer:null,Source:{},Demuxer:{},Decoder:{},Renderer:{},AudioOutput:{},Now:function(){return window.performance?window.performance.now()/1e3:Date.now()/1e3},CreateVideoElements:function(){var elements=docu...
src++;y|=sY[src]<<16;src++;y|=sY[src]<<24;src++;dY[dest++]=y}dest+=scan>>2;src+=scan}}}width=this.halfWidth;scan=width-8;H=motionH/2>>1;V=motionV/2>>1;oddH=(motionH/2&1)===1;oddV=(motionV/2&1)===1;src=((this.mbRow<<3)+V)*width+(this.mbCol<<3)+H;dest=...
gl.uniform1i(gl.getUniformLocation(this.program,name),index);return texture};WebGLRenderer.prototype.createProgram=function(vsh,fsh){var gl=this.gl;var program=gl.createProgram();gl.attachShader(program,this.compileShader(gl.VERTEX_SHADER,vsh));gl.at...
share/public_html/static/music_inc/music_inc_module.js view on Meta::CPAN
//import {default as NetworkDrFlac} from './music_drflac_module.js'
import {default as NetworkDrFlac} from './music_drflac_module.cache.js'
// times in seconds
const AQMaxDecodedTime = 20; // maximum time decoded, but not queued
const AQStartLookahead = 0.100; // minimum time to buffer before starting playback
const AQLookahead = 5; // buffer as long as less than AQLookahead is buffered
let MainAudioContext;
let GainNode;
let AQID = -1;
let AudioQueue = [];
let Tracks_HEAD;
let Tracks_TAIL;
let Tracks_QueueCurrent;
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
{
/* The type of the subframe: SUBFRAME_CONSTANT, SUBFRAME_VERBATIM, SUBFRAME_FIXED or SUBFRAME_LPC. */
drflac_uint8 subframeType;
/* The number of wasted bits per sample as specified by the sub-frame header. */
drflac_uint8 wastedBitsPerSample;
/* The order to use for the prediction stage for SUBFRAME_FIXED and SUBFRAME_LPC. */
drflac_uint8 lpcOrder;
/* A pointer to the buffer containing the decoded samples in the subframe. This pointer is an offset from drflac::pExtraData. */
drflac_int32* pSamplesS32;
} drflac_subframe;
typedef struct
{
/*
If the stream uses variable block sizes, this will be set to the index of the first PCM frame. If fixed block sizes are used, this will
always be set to 0. This is 64-bit because the decoded PCM frame number will be 36 bits.
*/
drflac_uint64 pcmFrameNumber;
/*
If the stream uses fixed block sizes, this will be set to the frame number. If variable block sizes are used, this will always be 0. This
is 32-bit because in fixed block sizes, the maximum frame number will be 31 bits.
*/
drflac_uint32 flacFrameNumber;
/* The sample rate of this frame. */
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
drflac_uint64 currentPCMFrame;
/* The position of the first FLAC frame in the stream. This is only ever used for seeking. */
drflac_uint64 firstFLACFramePosInBytes;
/* A hack to avoid a malloc() when opening a decoder with drflac_open_memory(). */
drflac__memory_stream memoryStream;
/* A pointer to the decoded sample data. This is an offset of pExtraData. */
drflac_int32* pDecodedSamples;
/* A pointer to the seek table. This is an offset of pExtraData, or NULL if there is no seek table. */
drflac_seekpoint* pSeekpoints;
/* Internal use only. Only used with Ogg containers. Points to a drflac_oggbs object. This is an offset of pExtraData. */
void* _oggbs;
/* Internal use only. Used for profiling and testing different seeking modes. */
drflac_bool32 _noSeekTableSeek : 1;
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
Parameters
----------
pFlac (in)
The decoder.
framesToRead (in)
The number of PCM frames to read.
pBufferOut (out, optional)
A pointer to the buffer that will receive the decoded samples.
Return Value
------------
Returns the number of PCM frames actually read. If the return value is less than `framesToRead` it has reached the end.
Remarks
-------
pBufferOut can be null, in which case the call will act as a seek, and the return value will be the number of frames seeked.
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
Parameters
----------
pFlac (in)
The decoder.
framesToRead (in)
The number of PCM frames to read.
pBufferOut (out, optional)
A pointer to the buffer that will receive the decoded samples.
Return Value
------------
Returns the number of PCM frames actually read. If the return value is less than `framesToRead` it has reached the end.
Remarks
-------
pBufferOut can be null, in which case the call will act as a seek, and the return value will be the number of frames seeked.
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
Parameters
----------
pFlac (in)
The decoder.
framesToRead (in)
The number of PCM frames to read.
pBufferOut (out, optional)
A pointer to the buffer that will receive the decoded samples.
Return Value
------------
Returns the number of PCM frames actually read. If the return value is less than `framesToRead` it has reached the end.
Remarks
-------
pBufferOut can be null, in which case the call will act as a seek, and the return value will be the number of frames seeked.
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
return DRFLAC_FALSE;
}
if (bit == 0) {
zeroCounter += 1;
} else {
break;
}
}
drflac_uint32 decodedRice;
if (riceParam > 0) {
if (!drflac__read_uint32(bs, riceParam, &decodedRice)) {
return DRFLAC_FALSE;
}
} else {
decodedRice = 0;
}
decodedRice |= (zeroCounter << riceParam);
if ((decodedRice & 0x01)) {
decodedRice = ~(decodedRice >> 1);
} else {
decodedRice = (decodedRice >> 1);
}
if (bitsPerSample+shift >= 32) {
pSamplesOut[i] = decodedRice + drflac__calculate_prediction_64(order, shift, coefficients, pSamplesOut + i);
} else {
pSamplesOut[i] = decodedRice + drflac__calculate_prediction_32(order, shift, coefficients, pSamplesOut + i);
}
}
return DRFLAC_TRUE;
}
#endif
#if 0
static drflac_bool32 drflac__read_rice_parts__reference(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut)
{
drflac_uint32 zeroCounter = 0;
drflac_uint32 decodedRice;
for (;;) {
drflac_uint8 bit;
if (!drflac__read_uint8(bs, 1, &bit)) {
return DRFLAC_FALSE;
}
if (bit == 0) {
zeroCounter += 1;
} else {
break;
}
}
if (riceParam > 0) {
if (!drflac__read_uint32(bs, riceParam, &decodedRice)) {
return DRFLAC_FALSE;
}
} else {
decodedRice = 0;
}
*pZeroCounterOut = zeroCounter;
*pRiceParamPartOut = decodedRice;
return DRFLAC_TRUE;
}
#endif
#if 0
static DRFLAC_INLINE drflac_bool32 drflac__read_rice_parts(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut)
{
drflac_cache_t riceParamMask;
drflac_uint32 zeroCounter;
drflac_uint32 setBitOffsetPlus1;
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
}
}
return DRFLAC_TRUE;
}
/*
Reads and decodes the residual for the sub-frame the decoder is currently sitting on. This function should be called
when the decoder is sitting at the very start of the RESIDUAL block. The first <order> residuals will be ignored. The
<blockSize> and <order> parameters are used to determine how many residual values need to be decoded.
*/
static drflac_bool32 drflac__decode_samples_with_residual(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 blockSize, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pDecodedSamples)
{
drflac_uint8 residualMethod;
drflac_uint8 partitionOrder;
drflac_uint32 samplesInPartition;
drflac_uint32 partitionsRemaining;
DRFLAC_ASSERT(bs != NULL);
DRFLAC_ASSERT(blockSize != 0);
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
samplesInPartition = blockSize / (1 << partitionOrder);
}
}
return DRFLAC_TRUE;
}
/*
Reads and seeks past the residual for the sub-frame the decoder is currently sitting on. This function should be called
when the decoder is sitting at the very start of the RESIDUAL block. The first <order> residuals will be set to 0. The
<blockSize> and <order> parameters are used to determine how many residual values need to be decoded.
*/
static drflac_bool32 drflac__read_and_seek_residual(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 order)
{
drflac_uint8 residualMethod;
drflac_uint8 partitionOrder;
drflac_uint32 samplesInPartition;
drflac_uint32 partitionsRemaining;
DRFLAC_ASSERT(bs != NULL);
DRFLAC_ASSERT(blockSize != 0);
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
}
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;
/* Only a single sample needs to be decoded here. */
drflac_int32 sample;
if (!drflac__read_int32(bs, subframeBitsPerSample, &sample)) {
return DRFLAC_FALSE;
}
/*
We don't really need to expand this, but it does simplify the process of reading samples. If this becomes a performance issue (unlikely)
we'll want to look at a more efficient way.
*/
for (i = 0; i < blockSize; ++i) {
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
pFlac->totalPCMFrameCount = pInit->totalPCMFrameCount;
pFlac->container = pInit->container;
}
static drflac* drflac_open_with_metadata_private(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, drflac_container container, void* pUserData, void* pUserDataMD, const drflac_allocation_callbacks* pAllocationCallbacks)
{
drflac_init_info init;
drflac_uint32 allocationSize;
drflac_uint32 wholeSIMDVectorCountPerChannel;
drflac_uint32 decodedSamplesAllocationSize;
#ifndef DR_FLAC_NO_OGG
drflac_oggbs oggbs;
#endif
drflac_uint64 firstFramePos;
drflac_uint64 seektablePos;
drflac_uint32 seektableSize;
drflac_allocation_callbacks allocationCallbacks;
drflac* pFlac;
/* CPU support first. */
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
allocationCallbacks.pUserData = NULL;
allocationCallbacks.onMalloc = drflac__malloc_default;
allocationCallbacks.onRealloc = drflac__realloc_default;
allocationCallbacks.onFree = drflac__free_default;
}
/*
The size of the allocation for the drflac object needs to be large enough to fit the following:
1) The main members of the drflac structure
2) A block of memory large enough to store the decoded samples of the largest frame in the stream
3) If the container is Ogg, a drflac_oggbs object
The complicated part of the allocation is making sure there's enough room the decoded samples, taking into consideration
the different SIMD instruction sets.
*/
allocationSize = sizeof(drflac);
/*
The allocation size for decoded frames depends on the number of 32-bit integers that fit inside the largest SIMD vector
we are supporting.
*/
if ((init.maxBlockSizeInPCMFrames % (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))) == 0) {
wholeSIMDVectorCountPerChannel = (init.maxBlockSizeInPCMFrames / (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32)));
} else {
wholeSIMDVectorCountPerChannel = (init.maxBlockSizeInPCMFrames / (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))) + 1;
}
decodedSamplesAllocationSize = wholeSIMDVectorCountPerChannel * DRFLAC_MAX_SIMD_VECTOR_SIZE * init.channels;
allocationSize += decodedSamplesAllocationSize;
allocationSize += DRFLAC_MAX_SIMD_VECTOR_SIZE; /* Allocate extra bytes to ensure we have enough for alignment. */
#ifndef DR_FLAC_NO_OGG
/* There's additional data required for Ogg streams. */
if (init.container == drflac_container_ogg) {
allocationSize += sizeof(drflac_oggbs);
}
DRFLAC_ZERO_MEMORY(&oggbs, sizeof(oggbs));
if (init.container == drflac_container_ogg) {
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
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);
*pInternalOggbs = oggbs;
/* The Ogg bistream needs to be layered on top of the original bitstream. */
pFlac->bs.onRead = drflac__on_read_ogg;
pFlac->bs.onSeek = drflac__on_seek_ogg;
pFlac->bs.pUserData = (void*)pInternalOggbs;
pFlac->_oggbs = (void*)pInternalOggbs;
}
#endif
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
{
pFlac->pSeekpoints = NULL;
pFlac->seekpointCount = 0;
}
else
#endif
{
/* If we have a seektable we need to load it now, making sure we move back to where we were previously. */
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);
/* Seek to the seektable, then just read directly into our seektable buffer. */
if (pFlac->bs.onSeek(pFlac->bs.pUserData, (int)seektablePos, drflac_seek_origin_start)) {
if (pFlac->bs.onRead(pFlac->bs.pUserData, pFlac->pSeekpoints, seektableSize) == seektableSize) {
/* Endian swap. */
drflac_uint32 iSeekpoint;
for (iSeekpoint = 0; iSeekpoint < pFlac->seekpointCount; ++iSeekpoint) {
share/public_html/static/music_inc/src/dr_flac.h view on Meta::CPAN
} \
\
sampleDataBufferSize = newSampleDataBufferSize; \
pSampleData = pNewSampleData; \
} \
\
DRFLAC_COPY_MEMORY(pSampleData + (totalPCMFrameCount*pFlac->channels), buffer, (size_t)(pcmFramesRead*pFlac->channels*sizeof(type))); \
totalPCMFrameCount += pcmFramesRead; \
} \
\
/* At this point everything should be decoded, but we just want to fill the unused part buffer with silence - need to \
protect those ears from random noise! */ \
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_SIZE_MAX) { \
goto on_error; /* The decoded data is too big. */ \
} \
\
pSampleData = (type*)drflac__malloc_from_callbacks((size_t)dataSize, &pFlac->allocationCallbacks); /* <-- Safe cast as per the check above. */ \
if (pSampleData == NULL) { \
goto on_error; \
} \
\
totalPCMFrameCount = drflac_read_pcm_frames_##extension(pFlac, pFlac->totalPCMFrameCount, pSampleData); \
} \
\
share/public_html/static/music_inc/src/drflac_cache.c view on Meta::CPAN
}
else if(!seekres)
{
printf("network_drflac_read_pcm_frames_f32_mem: seek failed current: %u desired: %u\n", currentPCMFrame32, start_pcm_frame);
error->code = NDRFLAC_GENERIC_ERROR;
return 0;
}
// decode to pcm
float32_t *data = malloc(pFlac->channels*sizeof(float32_t)*desired_pcm_frames);
uint32_t frames_decoded = drflac_read_pcm_frames_f32(pFlac, desired_pcm_frames, data);
if(frames_decoded != desired_pcm_frames)
{
printf("network_drflac_read_pcm_frames_f32_mem: expected %u decoded %u\n", desired_pcm_frames, frames_decoded);
}
if(!NDRFLAC_OK(ndrflac))
{
printf("network_drflac_read_pcm_frames_f32_mem: failed read_pcm_frames_f32\n");
free(data);
return 0;
}
#ifdef NETWORK_DR_FLAC_FORCE_REDBOOK
// resample
if(pFlac->sampleRate != 44100)
{
printf("Converting samplerate from %u to %u\n", pFlac->sampleRate, 44100);
ma_resampler_config config = ma_resampler_config_init(ma_format_f32, pFlac->channels, pFlac->sampleRate, 44100, ma_resample_algorithm_linear);
ma_resampler resampler;
ma_result result = ma_resampler_init(&config, &resampler);
ma_uint64 frameCountIn = frames_decoded;
ma_uint64 frameCountOut = ma_resampler_get_expected_output_frame_count(&resampler, frameCountIn);
float32_t *pFramesOut = malloc(frameCountOut*pFlac->channels*sizeof(float32_t));
ma_result result_process = ma_resampler_process_pcm_frames(&resampler, data, &frameCountIn, pFramesOut, &frameCountOut);
free(data);
if (result_process != MA_SUCCESS)
{
printf("network_drflac_read_pcm_frames_f32_mem: failed read_pcm_frames_f32 resample\n");
return 0;
}
data = pFramesOut;
frames_decoded = frameCountOut;
}
#endif
// deinterleave
for(unsigned i = 0; i < frames_decoded; i++)
{
for(unsigned j = 0; j < pFlac->channels; j++)
{
unsigned chanIndex = j*frames_decoded;
float32_t sample = data[(i*pFlac->channels) + j];
outFloat[chanIndex+i] = sample;
}
}
printf("returning from start_pcm_frame: %u frames_decoded %u data %p\n", start_pcm_frame, frames_decoded, data);
free(data);
// return number of samples
return frames_decoded;
}
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
if (framesToReadThisIterationIn > requiredInputFrameCount) {
framesToReadThisIterationIn = requiredInputFrameCount;
}
if (framesToReadThisIterationIn > 0) {
ma_device__on_data(pDevice, pIntermediaryBuffer, NULL, (ma_uint32)framesToReadThisIterationIn);
totalFramesReadIn += 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;
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
if (framesToReadThisIterationIn > requiredInputFrameCount) {
framesToReadThisIterationIn = requiredInputFrameCount;
}
if (requiredInputFrameCount > 0) {
framesReadThisIterationIn = pDecoder->onReadPCMFrames(pDecoder, pIntermediaryBuffer, framesToReadThisIterationIn);
totalFramesReadIn += framesReadThisIterationIn;
}
/*
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;
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
drflac_uint8 bit;
if (!drflac__read_uint8(bs, 1, &bit)) {
return DRFLAC_FALSE;
}
if (bit == 0) {
zeroCounter += 1;
} else {
break;
}
}
drflac_uint32 decodedRice;
if (riceParam > 0) {
if (!drflac__read_uint32(bs, riceParam, &decodedRice)) {
return DRFLAC_FALSE;
}
} else {
decodedRice = 0;
}
decodedRice |= (zeroCounter << riceParam);
if ((decodedRice & 0x01)) {
decodedRice = ~(decodedRice >> 1);
} else {
decodedRice = (decodedRice >> 1);
}
if (bitsPerSample+shift >= 32) {
pSamplesOut[i] = decodedRice + drflac__calculate_prediction_64(order, shift, coefficients, pSamplesOut + i);
} else {
pSamplesOut[i] = decodedRice + drflac__calculate_prediction_32(order, shift, coefficients, pSamplesOut + i);
}
}
return DRFLAC_TRUE;
}
#endif
#if 0
static drflac_bool32 drflac__read_rice_parts__reference(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut)
{
drflac_uint32 zeroCounter = 0;
drflac_uint32 decodedRice;
for (;;) {
drflac_uint8 bit;
if (!drflac__read_uint8(bs, 1, &bit)) {
return DRFLAC_FALSE;
}
if (bit == 0) {
zeroCounter += 1;
} else {
break;
}
}
if (riceParam > 0) {
if (!drflac__read_uint32(bs, riceParam, &decodedRice)) {
return DRFLAC_FALSE;
}
} else {
decodedRice = 0;
}
*pZeroCounterOut = zeroCounter;
*pRiceParamPartOut = decodedRice;
return DRFLAC_TRUE;
}
#endif
#if 0
static DRFLAC_INLINE drflac_bool32 drflac__read_rice_parts(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut)
{
drflac_cache_t riceParamMask;
drflac_uint32 zeroCounter;
drflac_uint32 setBitOffsetPlus1;
drflac_uint32 riceParamPart;
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
pFlac->channels = (drflac_uint8)pInit->channels;
pFlac->bitsPerSample = (drflac_uint8)pInit->bitsPerSample;
pFlac->totalPCMFrameCount = pInit->totalPCMFrameCount;
pFlac->container = pInit->container;
}
static drflac* drflac_open_with_metadata_private(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, drflac_container container, void* pUserData, void* pUserDataMD, const drflac_allocation_callbacks* pAllocationCallbacks)
{
drflac_init_info init;
drflac_uint32 allocationSize;
drflac_uint32 wholeSIMDVectorCountPerChannel;
drflac_uint32 decodedSamplesAllocationSize;
#ifndef DR_FLAC_NO_OGG
drflac_oggbs oggbs;
#endif
drflac_uint64 firstFramePos;
drflac_uint64 seektablePos;
drflac_uint32 seektableSize;
drflac_allocation_callbacks allocationCallbacks;
drflac* pFlac;
drflac__init_cpu_caps();
if (!drflac__init_private(&init, onRead, onSeek, onMeta, container, pUserData, pUserDataMD)) {
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
allocationCallbacks.onMalloc = drflac__malloc_default;
allocationCallbacks.onRealloc = drflac__realloc_default;
allocationCallbacks.onFree = drflac__free_default;
}
allocationSize = sizeof(drflac);
if ((init.maxBlockSizeInPCMFrames % (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))) == 0) {
wholeSIMDVectorCountPerChannel = (init.maxBlockSizeInPCMFrames / (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32)));
} else {
wholeSIMDVectorCountPerChannel = (init.maxBlockSizeInPCMFrames / (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))) + 1;
}
decodedSamplesAllocationSize = wholeSIMDVectorCountPerChannel * DRFLAC_MAX_SIMD_VECTOR_SIZE * init.channels;
allocationSize += decodedSamplesAllocationSize;
allocationSize += DRFLAC_MAX_SIMD_VECTOR_SIZE;
#ifndef DR_FLAC_NO_OGG
if (init.container == drflac_container_ogg) {
allocationSize += sizeof(drflac_oggbs);
}
DRFLAC_ZERO_MEMORY(&oggbs, sizeof(oggbs));
if (init.container == drflac_container_ogg) {
oggbs.onRead = onRead;
oggbs.onSeek = onSeek;
oggbs.pUserData = pUserData;
share/public_html/static/music_inc/src/miniaudio.h view on Meta::CPAN
}
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);
*pInternalOggbs = 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);
}
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
{
/* The type of the subframe: SUBFRAME_CONSTANT, SUBFRAME_VERBATIM, SUBFRAME_FIXED or SUBFRAME_LPC. */
drflac_uint8 subframeType;
/* The number of wasted bits per sample as specified by the sub-frame header. */
drflac_uint8 wastedBitsPerSample;
/* The order to use for the prediction stage for SUBFRAME_FIXED and SUBFRAME_LPC. */
drflac_uint8 lpcOrder;
/* A pointer to the buffer containing the decoded samples in the subframe. This pointer is an offset from drflac::pExtraData. */
drflac_int32* pSamplesS32;
} drflac_subframe;
typedef struct
{
/*
If the stream uses variable block sizes, this will be set to the index of the first PCM frame. If fixed block sizes are used, this will
always be set to 0. This is 64-bit because the decoded PCM frame number will be 36 bits.
*/
drflac_uint64 pcmFrameNumber;
/*
If the stream uses fixed block sizes, this will be set to the frame number. If variable block sizes are used, this will always be 0. This
is 32-bit because in fixed block sizes, the maximum frame number will be 31 bits.
*/
drflac_uint32 flacFrameNumber;
/* The sample rate of this frame. */
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
drflac_uint64 currentPCMFrame;
/* The position of the first FLAC frame in the stream. This is only ever used for seeking. */
drflac_uint64 firstFLACFramePosInBytes;
/* A hack to avoid a malloc() when opening a decoder with drflac_open_memory(). */
drflac__memory_stream memoryStream;
/* A pointer to the decoded sample data. This is an offset of pExtraData. */
drflac_int32* pDecodedSamples;
/* A pointer to the seek table. This is an offset of pExtraData, or NULL if there is no seek table. */
drflac_seekpoint* pSeekpoints;
/* Internal use only. Only used with Ogg containers. Points to a drflac_oggbs object. This is an offset of pExtraData. */
void* _oggbs;
/* Internal use only. Used for profiling and testing different seeking modes. */
drflac_bool32 _noSeekTableSeek : 1;
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
Parameters
----------
pFlac (in)
The decoder.
framesToRead (in)
The number of PCM frames to read.
pBufferOut (out, optional)
A pointer to the buffer that will receive the decoded samples.
Return Value
------------
Returns the number of PCM frames actually read. If the return value is less than `framesToRead` it has reached the end.
Remarks
-------
pBufferOut can be null, in which case the call will act as a seek, and the return value will be the number of frames seeked.
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
Parameters
----------
pFlac (in)
The decoder.
framesToRead (in)
The number of PCM frames to read.
pBufferOut (out, optional)
A pointer to the buffer that will receive the decoded samples.
Return Value
------------
Returns the number of PCM frames actually read. If the return value is less than `framesToRead` it has reached the end.
Remarks
-------
pBufferOut can be null, in which case the call will act as a seek, and the return value will be the number of frames seeked.
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
Parameters
----------
pFlac (in)
The decoder.
framesToRead (in)
The number of PCM frames to read.
pBufferOut (out, optional)
A pointer to the buffer that will receive the decoded samples.
Return Value
------------
Returns the number of PCM frames actually read. If the return value is less than `framesToRead` it has reached the end.
Remarks
-------
pBufferOut can be null, in which case the call will act as a seek, and the return value will be the number of frames seeked.
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
return DRFLAC_FALSE;
}
if (bit == 0) {
zeroCounter += 1;
} else {
break;
}
}
drflac_uint32 decodedRice;
if (riceParam > 0) {
if (!drflac__read_uint32(bs, riceParam, &decodedRice)) {
return DRFLAC_FALSE;
}
} else {
decodedRice = 0;
}
decodedRice |= (zeroCounter << riceParam);
if ((decodedRice & 0x01)) {
decodedRice = ~(decodedRice >> 1);
} else {
decodedRice = (decodedRice >> 1);
}
if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) {
pSamplesOut[i] = decodedRice + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + i);
} else {
pSamplesOut[i] = decodedRice + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + i);
}
}
return DRFLAC_TRUE;
}
#endif
#if 0
static drflac_bool32 drflac__read_rice_parts__reference(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut)
{
drflac_uint32 zeroCounter = 0;
drflac_uint32 decodedRice;
for (;;) {
drflac_uint8 bit;
if (!drflac__read_uint8(bs, 1, &bit)) {
return DRFLAC_FALSE;
}
if (bit == 0) {
zeroCounter += 1;
} else {
break;
}
}
if (riceParam > 0) {
if (!drflac__read_uint32(bs, riceParam, &decodedRice)) {
return DRFLAC_FALSE;
}
} else {
decodedRice = 0;
}
*pZeroCounterOut = zeroCounter;
*pRiceParamPartOut = decodedRice;
return DRFLAC_TRUE;
}
#endif
#if 0
static DRFLAC_INLINE drflac_bool32 drflac__read_rice_parts(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut)
{
drflac_cache_t riceParamMask;
drflac_uint32 zeroCounter;
drflac_uint32 setBitOffsetPlus1;
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
}
}
return DRFLAC_TRUE;
}
/*
Reads and decodes the residual for the sub-frame the decoder is currently sitting on. This function should be called
when the decoder is sitting at the very start of the RESIDUAL block. The first <order> residuals will be ignored. The
<blockSize> and <order> parameters are used to determine how many residual values need to be decoded.
*/
static drflac_bool32 drflac__decode_samples_with_residual(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 blockSize, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* ...
{
drflac_uint8 residualMethod;
drflac_uint8 partitionOrder;
drflac_uint32 samplesInPartition;
drflac_uint32 partitionsRemaining;
DRFLAC_ASSERT(bs != NULL);
DRFLAC_ASSERT(blockSize != 0);
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
samplesInPartition = blockSize / (1 << partitionOrder);
}
}
return DRFLAC_TRUE;
}
/*
Reads and seeks past the residual for the sub-frame the decoder is currently sitting on. This function should be called
when the decoder is sitting at the very start of the RESIDUAL block. The first <order> residuals will be set to 0. The
<blockSize> and <order> parameters are used to determine how many residual values need to be decoded.
*/
static drflac_bool32 drflac__read_and_seek_residual(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 order)
{
drflac_uint8 residualMethod;
drflac_uint8 partitionOrder;
drflac_uint32 samplesInPartition;
drflac_uint32 partitionsRemaining;
DRFLAC_ASSERT(bs != NULL);
DRFLAC_ASSERT(blockSize != 0);
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
}
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;
/* Only a single sample needs to be decoded here. */
drflac_int32 sample;
if (!drflac__read_int32(bs, subframeBitsPerSample, &sample)) {
return DRFLAC_FALSE;
}
/*
We don't really need to expand this, but it does simplify the process of reading samples. If this becomes a performance issue (unlikely)
we'll want to look at a more efficient way.
*/
for (i = 0; i < blockSize; ++i) {
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
pFlac->totalPCMFrameCount = pInit->totalPCMFrameCount;
pFlac->container = pInit->container;
}
static drflac* drflac_open_with_metadata_private(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, drflac_container container, void* pUserData, void* pUserDataMD, const drflac_allocation_callbacks* pAllocationCallbacks)
{
drflac_init_info init;
drflac_uint32 allocationSize;
drflac_uint32 wholeSIMDVectorCountPerChannel;
drflac_uint32 decodedSamplesAllocationSize;
#ifndef DR_FLAC_NO_OGG
drflac_oggbs oggbs;
#endif
drflac_uint64 firstFramePos;
drflac_uint64 seektablePos;
drflac_uint32 seektableSize;
drflac_allocation_callbacks allocationCallbacks;
drflac* pFlac;
/* CPU support first. */
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
allocationCallbacks.pUserData = NULL;
allocationCallbacks.onMalloc = drflac__malloc_default;
allocationCallbacks.onRealloc = drflac__realloc_default;
allocationCallbacks.onFree = drflac__free_default;
}
/*
The size of the allocation for the drflac object needs to be large enough to fit the following:
1) The main members of the drflac structure
2) A block of memory large enough to store the decoded samples of the largest frame in the stream
3) If the container is Ogg, a drflac_oggbs object
The complicated part of the allocation is making sure there's enough room the decoded samples, taking into consideration
the different SIMD instruction sets.
*/
allocationSize = sizeof(drflac);
/*
The allocation size for decoded frames depends on the number of 32-bit integers that fit inside the largest SIMD vector
we are supporting.
*/
if ((init.maxBlockSizeInPCMFrames % (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))) == 0) {
wholeSIMDVectorCountPerChannel = (init.maxBlockSizeInPCMFrames / (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32)));
} else {
wholeSIMDVectorCountPerChannel = (init.maxBlockSizeInPCMFrames / (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))) + 1;
}
decodedSamplesAllocationSize = wholeSIMDVectorCountPerChannel * DRFLAC_MAX_SIMD_VECTOR_SIZE * init.channels;
allocationSize += decodedSamplesAllocationSize;
allocationSize += DRFLAC_MAX_SIMD_VECTOR_SIZE; /* Allocate extra bytes to ensure we have enough for alignment. */
#ifndef DR_FLAC_NO_OGG
/* There's additional data required for Ogg streams. */
if (init.container == drflac_container_ogg) {
allocationSize += sizeof(drflac_oggbs);
}
DRFLAC_ZERO_MEMORY(&oggbs, sizeof(oggbs));
if (init.container == drflac_container_ogg) {
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
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));
/* The Ogg bistream needs to be layered on top of the original bitstream. */
pFlac->bs.onRead = drflac__on_read_ogg;
pFlac->bs.onSeek = drflac__on_seek_ogg;
pFlac->bs.pUserData = (void*)pInternalOggbs;
pFlac->_oggbs = (void*)pInternalOggbs;
}
#endif
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
{
pFlac->pSeekpoints = NULL;
pFlac->seekpointCount = 0;
}
else
#endif
{
/* If we have a seektable we need to load it now, making sure we move back to where we were previously. */
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);
/* Seek to the seektable, then just read directly into our seektable buffer. */
if (pFlac->bs.onSeek(pFlac->bs.pUserData, (int)seektablePos, drflac_seek_origin_start)) {
if (pFlac->bs.onRead(pFlac->bs.pUserData, pFlac->pSeekpoints, seektableSize) == seektableSize) {
/* Endian swap. */
drflac_uint32 iSeekpoint;
for (iSeekpoint = 0; iSeekpoint < pFlac->seekpointCount; ++iSeekpoint) {
share/public_html/static/music_worklet_inprogress/decoder/deps/dr_libs/dr_flac.h view on Meta::CPAN
} \
\
sampleDataBufferSize = newSampleDataBufferSize; \
pSampleData = pNewSampleData; \
} \
\
DRFLAC_COPY_MEMORY(pSampleData + (totalPCMFrameCount*pFlac->channels), buffer, (size_t)(pcmFramesRead*pFlac->channels*sizeof(type))); \
totalPCMFrameCount += pcmFramesRead; \
} \
\
/* At this point everything should be decoded, but we just want to fill the unused part buffer with silence - need to \
protect those ears from random noise! */ \
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; /* The decoded data is too big. */ \
} \
\
pSampleData = (type*)drflac__malloc_from_callbacks((size_t)dataSize, &pFlac->allocationCallbacks); /* <-- Safe cast as per the check above. */ \
if (pSampleData == NULL) { \
goto on_error; \
} \
\
totalPCMFrameCount = drflac_read_pcm_frames_##extension(pFlac, pFlac->totalPCMFrameCount, pSampleData); \
} \
\
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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.
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
In the example above, the sound is being initialized without a file nor a data source. This is
valid, in which case the sound acts as a node in the middle of the node graph. This means you can
connect other sounds to this sound and allow it to act like a sound group. Indeed, this is exactly
what a `ma_sound_group` is.
When loading a sound, you specify a set of flags that control how the sound is loaded and what
features are enabled for that sound. When no flags are set, the sound will be fully loaded into
memory in exactly the same format as how it's stored on the file system. The resource manager will
allocate a block of memory and then load the file directly into it. When reading audio data, it
will be decoded dynamically on the fly. In order to save processing time on the audio thread, it
might be beneficial to pre-decode the sound. You can do this with the `MA_SOUND_FLAG_DECODE` flag:
```c
ma_sound_init_from_file(&engine, "my_sound.wav", MA_SOUND_FLAG_DECODE, pGroup, NULL, &sound);
```
By default, sounds will be loaded synchronously, meaning `ma_sound_init_*()` will not return until
the sound has been fully loaded. If this is prohibitive you can instead load sounds asynchronously
by specificying the `MA_SOUND_FLAG_ASYNC` flag:
```c
ma_sound_init_from_file(&engine, "my_sound.wav", MA_SOUND_FLAG_DECODE | MA_SOUND_FLAG_ASYNC, pGroup, NULL, &sound);
```
This will result in `ma_sound_init_*()` returning quickly, but the sound won't yet have been fully
loaded. When you start the sound, it won't output anything until some sound is available. The sound
will start outputting audio before the sound has been fully decoded when the `MA_SOUND_FLAG_DECODE`
is specified.
If you need to wait for an asynchronously loaded sound to be fully loaded, you can use a fence. A
fence in miniaudio is a simple synchronization mechanism which simply blocks until it's internal
counter hit's zero. You can specify a fence like so:
```c
ma_result result;
ma_fence fence;
ma_sound sounds[4];
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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
data at mixing time which may be prohibitive in high-performance and large scale scenarios like
games.
Internally the resource manager uses the `ma_decoder` API to load sounds. This means by default it
only supports decoders that are built into miniaudio. It's possible to support additional encoding
formats through the use of custom decoders. To do so, pass in your `ma_decoding_backend_vtable`
vtables into the resource manager config:
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
for (iSound = 0; iSound < soundCount; iSound += 1) {
ma_resource_manager_data_source_init(pResourceManager, pSoundFilePaths[iSound], flags, ¬ifications, &pSoundSources[iSound]);
}
// ... DO SOMETHING ELSE WHILE SOUNDS ARE LOADING ...
// Wait for loading of sounds to finish.
ma_fence_wait(&fence);
```
In the example above we used a fence for waiting until the entire file has been fully decoded. If
you only need to wait for the initialization of the internal decoder to complete, you can use the
`init` member of the `ma_resource_manager_pipeline_notifications` object:
```c
notifications.init.pFence = &fence;
```
If a fence is not appropriate for your situation, you can instead use a callback that is fired on
an individual sound basis. This is done in a very similar way to fences:
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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.
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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.
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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
{
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
ma_uint32 flags;
} ma_resource_manager_data_source_config;
MA_API ma_resource_manager_data_source_config ma_resource_manager_data_source_config_init(void);
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. */
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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. */
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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
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
};
/* 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);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
static void ma_resource_manager_data_buffer_node_free(ma_resource_manager* pResourceManager, ma_resource_manager_data_buffer_node* pDataBufferNode)
{
MA_ASSERT(pResourceManager != NULL);
MA_ASSERT(pDataBufferNode != NULL);
if (pDataBufferNode->isDataOwnedByResourceManager) {
if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBufferNode) == ma_resource_manager_data_supply_type_encoded) {
ma_free((void*)pDataBufferNode->data.backend.encoded.pData, &pResourceManager->config.allocationCallbacks);
pDataBufferNode->data.backend.encoded.pData = NULL;
pDataBufferNode->data.backend.encoded.sizeInBytes = 0;
} else if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBufferNode) == ma_resource_manager_data_supply_type_decoded) {
ma_free((void*)pDataBufferNode->data.backend.decoded.pData, &pResourceManager->config.allocationCallbacks);
pDataBufferNode->data.backend.decoded.pData = NULL;
pDataBufferNode->data.backend.decoded.totalFrameCount = 0;
} else if (ma_resource_manager_data_buffer_node_get_data_supply_type(pDataBufferNode) == ma_resource_manager_data_supply_type_decoded_paged) {
ma_paged_audio_buffer_data_uninit(&pDataBufferNode->data.backend.decodedPaged.data, &pResourceManager->config.allocationCallbacks);
} else {
/* Should never hit this if the node was successfully initialized. */
MA_ASSERT(pDataBufferNode->result != MA_SUCCESS);
}
}
/* The data buffer itself needs to be freed. */
ma_free(pDataBufferNode, &pResourceManager->config.allocationCallbacks);
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
return (ma_thread_result)0;
}
#endif
MA_API ma_resource_manager_config ma_resource_manager_config_init(void)
{
ma_resource_manager_config config;
MA_ZERO_OBJECT(&config);
config.decodedFormat = ma_format_unknown;
config.decodedChannels = 0;
config.decodedSampleRate = 0;
config.jobThreadCount = 1; /* A single miniaudio-managed job thread by default. */
config.jobQueueCapacity = MA_JOB_TYPE_RESOURCE_MANAGER_QUEUE_CAPACITY;
/* Flags. */
config.flags = 0;
#ifdef MA_NO_THREADING
{
/* Threading is disabled at compile time so disable threading at runtime as well by default. */
config.flags |= MA_RESOURCE_MANAGER_FLAG_NO_THREADING;
config.jobThreadCount = 0;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
config.loopPointEndInPCMFrames = ~((ma_uint64)0);
return config;
}
static ma_decoder_config ma_resource_manager__init_decoder_config(ma_resource_manager* pResourceManager)
{
ma_decoder_config config;
config = ma_decoder_config_init(pResourceManager->config.decodedFormat, pResourceManager->config.decodedChannels, pResourceManager->config.decodedSampleRate);
config.allocationCallbacks = pResourceManager->config.allocationCallbacks;
config.ppCustomBackendVTables = pResourceManager->config.ppCustomDecodingBackendVTables;
config.customBackendCount = pResourceManager->config.customDecodingBackendCount;
config.pCustomBackendUserData = pResourceManager->config.pCustomDecodingBackendUserData;
return config;
}
static ma_result ma_resource_manager__init_decoder(ma_resource_manager* pResourceManager, const char* pFilePath, const wchar_t* pFilePathW, ma_decoder* pDecoder)
{
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
return MA_SUCCESS;
}
static ma_data_source* ma_resource_manager_data_buffer_get_connector(ma_resource_manager_data_buffer* pDataBuffer)
{
switch (pDataBuffer->pNode->data.type)
{
case ma_resource_manager_data_supply_type_encoded: return &pDataBuffer->connector.decoder;
case ma_resource_manager_data_supply_type_decoded: return &pDataBuffer->connector.buffer;
case ma_resource_manager_data_supply_type_decoded_paged: return &pDataBuffer->connector.pagedBuffer;
case ma_resource_manager_data_supply_type_unknown:
default:
{
ma_log_postf(ma_resource_manager_get_log(pDataBuffer->pResourceManager), MA_LOG_LEVEL_ERROR, "Failed to retrieve data buffer connector. Unknown data supply type.\n");
return NULL;
};
};
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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;
};
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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;
};
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
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;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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));
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
/* Loading synchronously. Load the sound in it's entirety here. */
if ((flags & MA_RESOURCE_MANAGER_DATA_SOURCE_FLAG_DECODE) == 0) {
/* No decoding. This is the simple case - just store the file contents in memory. */
result = ma_resource_manager_data_buffer_node_init_supply_encoded(pResourceManager, pDataBufferNode, pFilePath, pFilePathW);
if (result != MA_SUCCESS) {
goto done;
}
} else {
/* Decoding. We do this the same way as we do when loading asynchronously. */
ma_decoder* pDecoder;
result = ma_resource_manager_data_buffer_node_init_supply_decoded(pResourceManager, pDataBufferNode, pFilePath, pFilePathW, flags, &pDecoder);
if (result != MA_SUCCESS) {
goto done;
}
/* We have the decoder, now decode page by page just like we do when loading asynchronously. */
for (;;) {
/* Decode next page. */
result = ma_resource_manager_data_buffer_node_decode_next_page(pResourceManager, pDataBufferNode, pDecoder);
if (result != MA_SUCCESS) {
break; /* Will return MA_AT_END when the last page has been decoded. */
}
}
/* Reaching the end needs to be considered successful. */
if (result == MA_AT_END) {
result = MA_SUCCESS;
}
/*
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. */
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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;
/*
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
/* 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:
{
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
*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:
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
}
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:
{
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
{
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;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
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)
{
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
around to the paging jobs. When the last paging job has completed it's processing, it'll
free the decoder for us.
This job does not do any actual decoding. It instead just posts a PAGE_DATA_BUFFER_NODE job
which is where the actual decoding work will be done. However, once this job is complete,
the node will be in a state where data buffer connectors can be initialized.
*/
ma_decoder* pDecoder; /* <-- Free'd on the last page decode. */
ma_job pageDataBufferNodeJob;
/* Allocate the decoder by initializing a decoded data supply. */
result = ma_resource_manager_data_buffer_node_init_supply_decoded(pResourceManager, pDataBufferNode, pJob->data.resourceManager.loadDataBufferNode.pFilePath, pJob->data.resourceManager.loadDataBufferNode.pFilePathW, pJob->data.resourceManager...
/*
Don't ever propagate an MA_BUSY result code or else the resource manager will think the
node is just busy decoding rather than in an error state. This should never happen, but
including this logic for safety just in case.
*/
if (result == MA_BUSY) {
result = MA_ERROR;
}
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
If we have a success code by this point, we want to post another job. We're going to set the
result back to MA_BUSY to make it clear that there's still more to load.
*/
if (result == MA_SUCCESS) {
ma_job newJob;
newJob = *pJob; /* Everything is the same as the input job, except the execution order. */
newJob.order = ma_resource_manager_data_buffer_node_next_execution_order(pDataBufferNode); /* We need a fresh execution order. */
result = ma_resource_manager_post_job(pResourceManager, &newJob);
/* Since the sound isn't yet fully decoded we want the status to be set to busy. */
if (result == MA_SUCCESS) {
result = MA_BUSY;
}
}
done:
/* If there's still more to decode the result will be set to MA_BUSY. Otherwise we can free the decoder. */
if (result != MA_BUSY) {
ma_decoder_uninit((ma_decoder*)pJob->data.resourceManager.pageDataBufferNode.pDecoder);
ma_free(pJob->data.resourceManager.pageDataBufferNode.pDecoder, &pResourceManager->config.allocationCallbacks);
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
ma_resource_manager_config resourceManagerConfig;
pEngine->pResourceManager = (ma_resource_manager*)ma_malloc(sizeof(*pEngine->pResourceManager), &pEngine->allocationCallbacks);
if (pEngine->pResourceManager == NULL) {
result = MA_OUT_OF_MEMORY;
goto on_error_2;
}
resourceManagerConfig = ma_resource_manager_config_init();
resourceManagerConfig.pLog = pEngine->pLog; /* Always use the engine's log for internally-managed resource managers. */
resourceManagerConfig.decodedFormat = ma_format_f32;
resourceManagerConfig.decodedChannels = 0; /* Leave the decoded channel count as 0 so we can get good spatialization. */
resourceManagerConfig.decodedSampleRate = ma_engine_get_sample_rate(pEngine);
ma_allocation_callbacks_init_copy(&resourceManagerConfig.allocationCallbacks, &pEngine->allocationCallbacks);
resourceManagerConfig.pVFS = engineConfig.pResourceManagerVFS;
/* The Emscripten build cannot use threads. */
#if defined(MA_EMSCRIPTEN)
{
resourceManagerConfig.jobThreadCount = 0;
resourceManagerConfig.flags |= MA_RESOURCE_MANAGER_FLAG_NO_THREADING;
}
#endif
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
drflac_uint8 bit;
if (!drflac__read_uint8(bs, 1, &bit)) {
return DRFLAC_FALSE;
}
if (bit == 0) {
zeroCounter += 1;
} else {
break;
}
}
drflac_uint32 decodedRice;
if (riceParam > 0) {
if (!drflac__read_uint32(bs, riceParam, &decodedRice)) {
return DRFLAC_FALSE;
}
} else {
decodedRice = 0;
}
decodedRice |= (zeroCounter << riceParam);
if ((decodedRice & 0x01)) {
decodedRice = ~(decodedRice >> 1);
} else {
decodedRice = (decodedRice >> 1);
}
if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) {
pSamplesOut[i] = decodedRice + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + i);
} else {
pSamplesOut[i] = decodedRice + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + i);
}
}
return DRFLAC_TRUE;
}
#endif
#if 0
static drflac_bool32 drflac__read_rice_parts__reference(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut)
{
drflac_uint32 zeroCounter = 0;
drflac_uint32 decodedRice;
for (;;) {
drflac_uint8 bit;
if (!drflac__read_uint8(bs, 1, &bit)) {
return DRFLAC_FALSE;
}
if (bit == 0) {
zeroCounter += 1;
} else {
break;
}
}
if (riceParam > 0) {
if (!drflac__read_uint32(bs, riceParam, &decodedRice)) {
return DRFLAC_FALSE;
}
} else {
decodedRice = 0;
}
*pZeroCounterOut = zeroCounter;
*pRiceParamPartOut = decodedRice;
return DRFLAC_TRUE;
}
#endif
#if 0
static DRFLAC_INLINE drflac_bool32 drflac__read_rice_parts(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut)
{
drflac_cache_t riceParamMask;
drflac_uint32 zeroCounter;
drflac_uint32 setBitOffsetPlus1;
drflac_uint32 riceParamPart;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
pFlac->channels = (drflac_uint8)pInit->channels;
pFlac->bitsPerSample = (drflac_uint8)pInit->bitsPerSample;
pFlac->totalPCMFrameCount = pInit->totalPCMFrameCount;
pFlac->container = pInit->container;
}
static drflac* drflac_open_with_metadata_private(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, drflac_container container, void* pUserData, void* pUserDataMD, const drflac_allocation_callbacks* pAllocationCallbacks)
{
drflac_init_info init;
drflac_uint32 allocationSize;
drflac_uint32 wholeSIMDVectorCountPerChannel;
drflac_uint32 decodedSamplesAllocationSize;
#ifndef DR_FLAC_NO_OGG
drflac_oggbs oggbs;
#endif
drflac_uint64 firstFramePos;
drflac_uint64 seektablePos;
drflac_uint32 seektableSize;
drflac_allocation_callbacks allocationCallbacks;
drflac* pFlac;
drflac__init_cpu_caps();
if (!drflac__init_private(&init, onRead, onSeek, onMeta, container, pUserData, pUserDataMD)) {
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
allocationCallbacks.onMalloc = drflac__malloc_default;
allocationCallbacks.onRealloc = drflac__realloc_default;
allocationCallbacks.onFree = drflac__free_default;
}
allocationSize = sizeof(drflac);
if ((init.maxBlockSizeInPCMFrames % (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))) == 0) {
wholeSIMDVectorCountPerChannel = (init.maxBlockSizeInPCMFrames / (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32)));
} else {
wholeSIMDVectorCountPerChannel = (init.maxBlockSizeInPCMFrames / (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))) + 1;
}
decodedSamplesAllocationSize = wholeSIMDVectorCountPerChannel * DRFLAC_MAX_SIMD_VECTOR_SIZE * init.channels;
allocationSize += decodedSamplesAllocationSize;
allocationSize += DRFLAC_MAX_SIMD_VECTOR_SIZE;
#ifndef DR_FLAC_NO_OGG
if (init.container == drflac_container_ogg) {
allocationSize += sizeof(drflac_oggbs);
}
DRFLAC_ZERO_MEMORY(&oggbs, sizeof(oggbs));
if (init.container == drflac_container_ogg) {
oggbs.onRead = onRead;
oggbs.onSeek = onSeek;
oggbs.pUserData = pUserData;
share/public_html/static/music_worklet_inprogress/decoder/deps/miniaudio/miniaudio.h view on Meta::CPAN
}
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);
}
share/public_html/static/music_worklet_inprogress/decoder/src/mhfs_cl_decoder.h view on Meta::CPAN
return MHFS_CL_ERROR;
}
mhfs_d->dcTempOutSize = reqBytes;
mhfs_d->pDCTempOut = tempOut;
}
const mhfs_cl_error readCode = mhfs_cl_track_read_pcm_frames_f32(pTrack, dec_frames_req, mhfs_d->pDCTempOut, pReturnData);
if((readCode != MHFS_CL_SUCCESS) || (pReturnData->frames_read == 0))
{
return readCode;
}
uint64_t decoded_frames = pReturnData->frames_read;
// resample
uint64_t frameCountOut = desired_pcm_frames;
ma_result result = ma_data_converter_process_pcm_frames(&mhfs_d->madc, mhfs_d->pDCTempOut, &decoded_frames, outFloat, &frameCountOut);
if(result != MA_SUCCESS)
{
MHFSCLDEC_PRINT("resample failed\n");
return MHFS_CL_ERROR;
}
pReturnData->frames_read = frameCountOut;
return MHFS_CL_SUCCESS;
}
}
share/public_html/static/music_worklet_inprogress/decoder/src/mhfs_cl_track.h view on Meta::CPAN
return seekBlockRes;
}
if(seekRes != MA_SUCCESS)
{
MHFSCLTR_PRINT("%s: seek failed current: %u desired: %u ma_result %d\n", __func__, currentPCMFrame32, pTrack->currentFrame, seekRes);
retval = MHFS_CL_ERROR;
goto mhfs_cl_track_read_pcm_frames_f32_FAIL;
}
// finally read
uint64_t frames_decoded = 0;
if(desired_pcm_frames != 0)
{
uint64_t toread = desired_pcm_frames;
// decode to pcm
mhfs_cl_track_blockvf_ma_decoder_call_before(pTrack, true);
ma_result decRes = ma_decoder_read_pcm_frames(&pTrack->decoder, outFloat, toread, &frames_decoded);
const mhfs_cl_error decBlockRes = mhfs_cl_track_blockvf_ma_decoder_call_after(pTrack, true, &pReturnData->needed_offset);
if(decBlockRes != MHFS_CL_SUCCESS)
{
MHFSCLTR_PRINT("mhfs_cl_track_read_pcm_frames_f32_mem: failed read_pcm_frames_f32\n");
return decBlockRes;
}
if(decRes != MA_SUCCESS)
{
MHFSCLTR_PRINT("mhfs_cl_track_read_pcm_frames_f32_mem: failed read_pcm_frames_f32(decode), ma_result %d\n", decRes);
retval = MHFS_CL_ERROR;
if(decRes == MA_AT_END)
{
MHFSCLTR_PRINT("MA_AT_END\n"); // not a real error
}
goto mhfs_cl_track_read_pcm_frames_f32_FAIL;
}
if(frames_decoded != desired_pcm_frames)
{
MHFSCLTR_PRINT("mhfs_cl_track_read_pcm_frames_f32_mem: expected %u decoded %"PRIu64"\n", desired_pcm_frames, frames_decoded);
}
pTrack->currentFrame += frames_decoded;
}
MHFSCLTR_PRINT("returning from pTrack->currentFrame: %u, totalFrames %"PRIu64" frames_decoded %"PRIu64" desired %u\n", pTrack->currentFrame, pTrack->meta.totalPCMFrameCount, frames_decoded, desired_pcm_frames);
pReturnData->frames_read = frames_decoded;
return MHFS_CL_SUCCESS;
mhfs_cl_track_read_pcm_frames_f32_FAIL:
if(pTrack->dec_initialized)
{
ma_decoder_uninit(&pTrack->decoder);
pTrack->dec_initialized = false;
}
return retval;
}
share/public_html/static/music_worklet_inprogress/music_inc_module.js view on Meta::CPAN
import {default as MHFSPlayer} from './player/mhfsplayer.js'
// times in seconds
const AQMaxDecodedTime = 20; // maximum time decoded, but not queued
const DesiredChannels = 2;
const DesiredSampleRate = 44100;
let SBAR_UPDATING = 0;
(async function () {
Number.prototype.toHHMMSS = function () {
var sec_num = Math.floor(this); //parseInt(this, 10); // don't forget the second param
var hours = Math.floor(sec_num / 3600);
share/public_html/static/music_worklet_inprogress/player/mhfsplayer.js view on Meta::CPAN
console.log('SEEK ' + stime);
await that.StopQueue();
that.StopAudio();
that.StartQueue(track, stime);
};
that._pborderchanged = async function(pbstate) {
that.pborder = pbstate;
// we need either the last decoded but not queued track or the last track if everything is queued
let ti;
for(ti = 0; ;ti++) {
if(!that.AudioQueue[ti]) {
if(ti === 0) return;
ti--;
break;
}
if(!that.AudioQueue[ti].donedecode) return;
if(!that.AudioQueue[ti].queued) break;
}
// make ti our last track
that.AudioQueue.length = ti+1;
// determine the next track we want to queue
const track = getNextTrack(that.AudioQueue[ti].track);
await that.StopQueue();
// cancel cached decoded audio that's not apart of AQ
that.truncateDecoded();
if(!that.AudioQueue[ti]) {
console.log('no track');
}
// queue following the playback order
that.StartQueue(track);
};
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