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Bug 1061046. Part 12: Simplify AudioNodeExternalInputStream since no resampling is needed. r=karlt

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Since resampling is not needed, we can preserve the number of input channels in each
input audio block.
This commit is contained in:
Robert O'Callahan 2014-09-18 11:51:21 +12:00
parent 4b3e1fddd3
commit 8f79162162
3 changed files with 70 additions and 402 deletions
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5
dom/media/AudioSegment.h
View File

@ -289,8 +289,9 @@ public:
return 0;
}
bool IsNull() {
for (ChunkIterator ci(*this); !ci.IsEnded(); ci.Next()) {
bool IsNull() const {
for (ChunkIterator ci(*const_cast<AudioSegment*>(this)); !ci.IsEnded();
ci.Next()) {
if (!ci->IsNull()) {
return false;
}

393
dom/media/webaudio/AudioNodeExternalInputStream.cpp
View File

@ -14,7 +14,6 @@ namespace mozilla {
AudioNodeExternalInputStream::AudioNodeExternalInputStream(AudioNodeEngine* aEngine, TrackRate aSampleRate)
: AudioNodeStream(aEngine, MediaStreamGraph::INTERNAL_STREAM, aSampleRate)
, mCurrentOutputPosition(0)
{
MOZ_COUNT_CTOR(AudioNodeExternalInputStream);
}
@ -24,294 +23,83 @@ AudioNodeExternalInputStream::~AudioNodeExternalInputStream()
MOZ_COUNT_DTOR(AudioNodeExternalInputStream);
}
AudioNodeExternalInputStream::TrackMapEntry::~TrackMapEntry()
/**
* Copies the data in aInput to aOffsetInBlock within aBlock.
* aBlock must have been allocated with AllocateInputBlock and have a channel
* count that's a superset of the channels in aInput.
*/
static void
CopyChunkToBlock(const AudioChunk& aInput, AudioChunk *aBlock,
uint32_t aOffsetInBlock)
{
if (mResampler) {
speex_resampler_destroy(mResampler);
}
}
size_t
AudioNodeExternalInputStream::GetTrackMapEntry(const StreamBuffer::Track& aTrack,
GraphTime aFrom)
{
AudioSegment* segment = aTrack.Get<AudioSegment>();
// Check the map for an existing entry corresponding to the input track.
for (size_t i = 0; i < mTrackMap.Length(); ++i) {
TrackMapEntry* map = &mTrackMap[i];
if (map->mTrackID == aTrack.GetID()) {
return i;
uint32_t blockChannels = aBlock->ChannelCount();
nsAutoTArray<const void*,2> channels;
if (aInput.IsNull()) {
channels.SetLength(blockChannels);
PodZero(channels.Elements(), blockChannels);
} else {
channels.SetLength(aInput.ChannelCount());
PodCopy(channels.Elements(), aInput.mChannelData.Elements(), channels.Length());
if (channels.Length() != blockChannels) {
// We only need to upmix here because aBlock's channel count has been
// chosen to be a superset of the channel count of every chunk.
AudioChannelsUpMix(&channels, blockChannels, nullptr);
}
}
// Determine channel count by finding the first entry with non-silent data.
AudioSegment::ChunkIterator ci(*segment);
while (!ci.IsEnded() && ci->IsNull()) {
ci.Next();
}
if (ci.IsEnded()) {
// The track is entirely silence so far, we can ignore it for now.
return nsTArray<TrackMapEntry>::NoIndex;
}
// Create a speex resampler with the same sample rate and number of channels
// as the track.
SpeexResamplerState* resampler = nullptr;
size_t channelCount = std::min((*ci).mChannelData.Length(),
WebAudioUtils::MaxChannelCount);
if (aTrack.GetRate() != mSampleRate) {
resampler = speex_resampler_init(channelCount,
aTrack.GetRate(), mSampleRate, SPEEX_RESAMPLER_QUALITY_DEFAULT, nullptr);
speex_resampler_skip_zeros(resampler);
}
TrackMapEntry* map = mTrackMap.AppendElement();
map->mEndOfConsumedInputTicks = 0;
map->mEndOfLastInputIntervalInInputStream = -1;
map->mEndOfLastInputIntervalInOutputStream = -1;
map->mSamplesPassedToResampler =
TimeToTicksRoundUp(aTrack.GetRate(), GraphTimeToStreamTime(aFrom));
map->mResampler = resampler;
map->mResamplerChannelCount = channelCount;
map->mTrackID = aTrack.GetID();
return mTrackMap.Length() - 1;
}
static const uint32_t SPEEX_RESAMPLER_PROCESS_MAX_OUTPUT = 1000;
template <typename T> static void
ResampleChannelBuffer(SpeexResamplerState* aResampler, uint32_t aChannel,
const T* aInput, uint32_t aInputDuration,
nsTArray<float>* aOutput)
{
if (!aResampler) {
float* out = aOutput->AppendElements(aInputDuration);
for (uint32_t i = 0; i < aInputDuration; ++i) {
out[i] = AudioSampleToFloat(aInput[i]);
}
return;
}
uint32_t processed = 0;
while (processed < aInputDuration) {
uint32_t prevLength = aOutput->Length();
float* output = aOutput->AppendElements(SPEEX_RESAMPLER_PROCESS_MAX_OUTPUT);
uint32_t in = aInputDuration - processed;
uint32_t out = aOutput->Length() - prevLength;
WebAudioUtils::SpeexResamplerProcess(aResampler, aChannel,
aInput + processed, &in,
output, &out);
processed += in;
aOutput->SetLength(prevLength + out);
}
}
void
AudioNodeExternalInputStream::TrackMapEntry::ResampleChannels(const nsTArray<const void*>& aBuffers,
uint32_t aInputDuration,
AudioSampleFormat aFormat,
float aVolume)
{
NS_ASSERTION(aBuffers.Length() == mResamplerChannelCount,
"Channel count must be correct here");
nsAutoTArray<nsTArray<float>,2> resampledBuffers;
resampledBuffers.SetLength(aBuffers.Length());
nsTArray<float> samplesAdjustedForVolume;
nsAutoTArray<const float*,2> bufferPtrs;
bufferPtrs.SetLength(aBuffers.Length());
for (uint32_t i = 0; i < aBuffers.Length(); ++i) {
AudioSampleFormat format = aFormat;
const void* buffer = aBuffers[i];
if (aVolume != 1.0f) {
format = AUDIO_FORMAT_FLOAT32;
samplesAdjustedForVolume.SetLength(aInputDuration);
switch (aFormat) {
uint32_t duration = aInput.GetDuration();
for (uint32_t c = 0; c < blockChannels; ++c) {
float* outputData =
static_cast<float*>(const_cast<void*>(aBlock->mChannelData[c])) + aOffsetInBlock;
if (channels[c]) {
switch (aInput.mBufferFormat) {
case AUDIO_FORMAT_FLOAT32:
ConvertAudioSamplesWithScale(static_cast<const float*>(buffer),
samplesAdjustedForVolume.Elements(),
aInputDuration, aVolume);
ConvertAudioSamplesWithScale(
static_cast<const float*>(channels[c]), outputData, duration,
aInput.mVolume);
break;
case AUDIO_FORMAT_S16:
ConvertAudioSamplesWithScale(static_cast<const int16_t*>(buffer),
samplesAdjustedForVolume.Elements(),
aInputDuration, aVolume);
ConvertAudioSamplesWithScale(
static_cast<const int16_t*>(channels[c]), outputData, duration,
aInput.mVolume);
break;
default:
MOZ_ASSERT(false);
return;
NS_ERROR("Unhandled format");
}
buffer = samplesAdjustedForVolume.Elements();
}
switch (format) {
case AUDIO_FORMAT_FLOAT32:
ResampleChannelBuffer(mResampler, i,
static_cast<const float*>(buffer),
aInputDuration, &resampledBuffers[i]);
break;
case AUDIO_FORMAT_S16:
ResampleChannelBuffer(mResampler, i,
static_cast<const int16_t*>(buffer),
aInputDuration, &resampledBuffers[i]);
break;
default:
MOZ_ASSERT(false);
return;
}
bufferPtrs[i] = resampledBuffers[i].Elements();
NS_ASSERTION(i == 0 ||
resampledBuffers[i].Length() == resampledBuffers[0].Length(),
"Resampler made different decisions for different channels!");
}
uint32_t length = resampledBuffers[0].Length();
nsRefPtr<ThreadSharedObject> buf = new SharedChannelArrayBuffer<float>(&resampledBuffers);
mResampledData.AppendFrames(buf.forget(), bufferPtrs, length);
}
void
AudioNodeExternalInputStream::TrackMapEntry::ResampleInputData(AudioSegment* aSegment)
{
AudioSegment::ChunkIterator ci(*aSegment);
while (!ci.IsEnded()) {
const AudioChunk& chunk = *ci;
nsAutoTArray<const void*,2> channels;
if (chunk.GetDuration() > UINT32_MAX) {
// This will cause us to OOM or overflow below. So let's just bail.
NS_ERROR("Chunk duration out of bounds");
return;
}
uint32_t duration = uint32_t(chunk.GetDuration());
if (chunk.IsNull()) {
nsAutoTArray<AudioDataValue,1024> silence;
silence.SetLength(duration);
PodZero(silence.Elements(), silence.Length());
channels.SetLength(mResamplerChannelCount);
for (uint32_t i = 0; i < channels.Length(); ++i) {
channels[i] = silence.Elements();
}
ResampleChannels(channels, duration, AUDIO_OUTPUT_FORMAT, 0.0f);
} else if (chunk.mChannelData.Length() == mResamplerChannelCount) {
// Common case, since mResamplerChannelCount is set to the first chunk's
// number of channels.
channels.AppendElements(chunk.mChannelData);
ResampleChannels(channels, duration, chunk.mBufferFormat, chunk.mVolume);
} else {
// Uncommon case. Since downmixing requires channels to be floats,
// convert everything to floats now.
uint32_t upChannels = GetAudioChannelsSuperset(chunk.mChannelData.Length(), mResamplerChannelCount);
nsTArray<float> buffer;
if (chunk.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
channels.AppendElements(chunk.mChannelData);
} else {
NS_ASSERTION(chunk.mBufferFormat == AUDIO_FORMAT_S16, "Unknown format");
if (duration > UINT32_MAX/chunk.mChannelData.Length()) {
NS_ERROR("Chunk duration out of bounds");
return;
}
buffer.SetLength(chunk.mChannelData.Length()*duration);
for (uint32_t i = 0; i < chunk.mChannelData.Length(); ++i) {
const int16_t* samples = static_cast<const int16_t*>(chunk.mChannelData[i]);
float* converted = &buffer[i*duration];
for (uint32_t j = 0; j < duration; ++j) {
converted[j] = AudioSampleToFloat(samples[j]);
}
channels.AppendElement(converted);
}
}
nsTArray<float> zeroes;
if (channels.Length() < upChannels) {
zeroes.SetLength(duration);
PodZero(zeroes.Elements(), zeroes.Length());
AudioChannelsUpMix(&channels, upChannels, zeroes.Elements());
}
if (channels.Length() == mResamplerChannelCount) {
ResampleChannels(channels, duration, AUDIO_FORMAT_FLOAT32, chunk.mVolume);
} else {
nsTArray<float> output;
if (duration > UINT32_MAX/mResamplerChannelCount) {
NS_ERROR("Chunk duration out of bounds");
return;
}
output.SetLength(duration*mResamplerChannelCount);
nsAutoTArray<float*,2> outputPtrs;
nsAutoTArray<const void*,2> outputPtrsConst;
for (uint32_t i = 0; i < mResamplerChannelCount; ++i) {
outputPtrs.AppendElement(output.Elements() + i*duration);
outputPtrsConst.AppendElement(outputPtrs[i]);
}
AudioChannelsDownMix(channels, outputPtrs.Elements(), outputPtrs.Length(), duration);
ResampleChannels(outputPtrsConst, duration, AUDIO_FORMAT_FLOAT32, chunk.mVolume);
}
}
ci.Next();
}
}
/**
* Copies the data in aInput to aOffsetInBlock within aBlock. All samples must
* be float. Both chunks must have the same number of channels (or else
* aInput is null). aBlock must have been allocated with AllocateInputBlock.
*/
static void
CopyChunkToBlock(const AudioChunk& aInput, AudioChunk *aBlock, uint32_t aOffsetInBlock)
{
uint32_t d = aInput.GetDuration();
for (uint32_t i = 0; i < aBlock->mChannelData.Length(); ++i) {
float* out = static_cast<float*>(const_cast<void*>(aBlock->mChannelData[i])) +
aOffsetInBlock;
if (aInput.IsNull()) {
PodZero(out, d);
} else {
const float* in = static_cast<const float*>(aInput.mChannelData[i]);
ConvertAudioSamplesWithScale(in, out, d, aInput.mVolume);
PodZero(outputData, duration);
}
}
}
/**
* Converts the data in aSegment to a single chunk aChunk. Every chunk in
* aSegment must have the same number of channels (or be null). aSegment must have
* duration WEBAUDIO_BLOCK_SIZE. Every chunk in aSegment must be in float format.
* Converts the data in aSegment to a single chunk aBlock. aSegment must have
* duration WEBAUDIO_BLOCK_SIZE. aFallbackChannelCount is a superset of the
* channels in every chunk of aSegment. aBlock must be float format or null.
*/
static void
ConvertSegmentToAudioBlock(AudioSegment* aSegment, AudioChunk* aBlock)
static void ConvertSegmentToAudioBlock(AudioSegment* aSegment,
AudioChunk* aBlock,
int32_t aFallbackChannelCount)
{
NS_ASSERTION(aSegment->GetDuration() == WEBAUDIO_BLOCK_SIZE, "Bad segment duration");
{
AudioSegment::ChunkIterator ci(*aSegment);
NS_ASSERTION(!ci.IsEnded(), "Segment must have at least one chunk");
AudioChunk& firstChunk = *ci;
ci.Next();
if (ci.IsEnded()) {
*aBlock = firstChunk;
NS_ASSERTION(!ci.IsEnded(), "Should be at least one chunk!");
if (ci->GetDuration() == WEBAUDIO_BLOCK_SIZE &&
(ci->IsNull() || ci->mBufferFormat == AUDIO_FORMAT_FLOAT32)) {
// Return this chunk directly to avoid copying data.
*aBlock = *ci;
return;
}
while (ci->IsNull() && !ci.IsEnded()) {
ci.Next();
}
if (ci.IsEnded()) {
// All null.
aBlock->SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
AllocateAudioBlock(ci->mChannelData.Length(), aBlock);
}
AudioSegment::ChunkIterator ci(*aSegment);
AllocateAudioBlock(aFallbackChannelCount, aBlock);
uint32_t duration = 0;
while (!ci.IsEnded()) {
for (AudioSegment::ChunkIterator ci(*aSegment); !ci.IsEnded(); ci.Next()) {
CopyChunkToBlock(*ci, aBlock, duration);
duration += ci->GetDuration();
ci.Next();
}
}
@ -334,29 +122,18 @@ AudioNodeExternalInputStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
MediaStream* source = mInputs[0]->GetSource();
nsAutoTArray<AudioSegment,1> audioSegments;
nsAutoTArray<bool,1> trackMapEntriesUsed;
uint32_t inputChannels = 0;
for (StreamBuffer::TrackIter tracks(source->mBuffer, MediaSegment::AUDIO);
!tracks.IsEnded(); tracks.Next()) {
const StreamBuffer::Track& inputTrack = *tracks;
// Create a TrackMapEntry if necessary.
size_t trackMapIndex = GetTrackMapEntry(inputTrack, aFrom);
// Maybe there's nothing in this track yet. If so, ignore it. (While the
// track is only playing silence, we may not be able to determine the
// correct number of channels to start resampling.)
if (trackMapIndex == nsTArray<TrackMapEntry>::NoIndex) {
const AudioSegment& inputSegment =
*static_cast<AudioSegment*>(inputTrack.GetSegment());
if (inputSegment.IsNull()) {
continue;
}
while (trackMapEntriesUsed.Length() <= trackMapIndex) {
trackMapEntriesUsed.AppendElement(false);
}
trackMapEntriesUsed[trackMapIndex] = true;
TrackMapEntry* trackMap = &mTrackMap[trackMapIndex];
AudioSegment segment;
AudioSegment& segment = *audioSegments.AppendElement();
GraphTime next;
TrackRate inputTrackRate = inputTrack.GetRate();
for (GraphTime t = aFrom; t < aTo; t = next) {
MediaInputPort::InputInterval interval = mInputs[0]->GetNextInputInterval(t);
interval.mEnd = std::min(interval.mEnd, aTo);
@ -364,67 +141,28 @@ AudioNodeExternalInputStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
break;
next = interval.mEnd;
// Ticks >= startTicks and < endTicks are in the interval
StreamTime outputEnd = GraphTimeToStreamTime(interval.mEnd);
TrackTicks startTicks = trackMap->mSamplesPassedToResampler + segment.GetDuration();
StreamTime outputStart = GraphTimeToStreamTime(interval.mStart);
NS_ASSERTION(startTicks == TimeToTicksRoundUp(inputTrackRate, outputStart),
"Samples missing");
TrackTicks endTicks = TimeToTicksRoundUp(inputTrackRate, outputEnd);
TrackTicks ticks = endTicks - startTicks;
StreamTime outputEnd = GraphTimeToStreamTime(interval.mEnd);
TrackTicks ticks = outputEnd - outputStart;
if (interval.mInputIsBlocked) {
segment.AppendNullData(ticks);
} else {
// See comments in TrackUnionStream::CopyTrackData
StreamTime inputStart = source->GraphTimeToStreamTime(interval.mStart);
StreamTime inputEnd = source->GraphTimeToStreamTime(interval.mEnd);
TrackTicks inputTrackEndPoint =
inputTrack.IsEnded() ? inputTrack.GetEnd() : TRACK_TICKS_MAX;
StreamTime inputStart =
std::min(inputSegment.GetDuration(),
source->GraphTimeToStreamTime(interval.mStart));
StreamTime inputEnd =
std::min(inputSegment.GetDuration(),
source->GraphTimeToStreamTime(interval.mEnd));
if (trackMap->mEndOfLastInputIntervalInInputStream != inputStart ||
trackMap->mEndOfLastInputIntervalInOutputStream != outputStart) {
// Start of a new series of intervals where neither stream is blocked.
trackMap->mEndOfConsumedInputTicks = TimeToTicksRoundDown(inputTrackRate, inputStart) - 1;
}
TrackTicks inputStartTicks = trackMap->mEndOfConsumedInputTicks;
TrackTicks inputEndTicks = inputStartTicks + ticks;
trackMap->mEndOfConsumedInputTicks = inputEndTicks;
trackMap->mEndOfLastInputIntervalInInputStream = inputEnd;
trackMap->mEndOfLastInputIntervalInOutputStream = outputEnd;
if (inputStartTicks < 0) {
// Data before the start of the track is just null.
segment.AppendNullData(-inputStartTicks);
inputStartTicks = 0;
}
if (inputEndTicks > inputStartTicks) {
segment.AppendSlice(*inputTrack.GetSegment(),
std::min(inputTrackEndPoint, inputStartTicks),
std::min(inputTrackEndPoint, inputEndTicks));
}
segment.AppendSlice(inputSegment, inputStart, inputEnd);
// Pad if we're looking past the end of the track
segment.AppendNullData(ticks - segment.GetDuration());
segment.AppendNullData(ticks - (inputEnd - inputStart));
}
}
trackMap->mSamplesPassedToResampler += segment.GetDuration();
trackMap->ResampleInputData(&segment);
if (trackMap->mResampledData.GetDuration() < mCurrentOutputPosition + WEBAUDIO_BLOCK_SIZE) {
// We don't have enough data. Delay it.
trackMap->mResampledData.InsertNullDataAtStart(
mCurrentOutputPosition + WEBAUDIO_BLOCK_SIZE - trackMap->mResampledData.GetDuration());
}
audioSegments.AppendElement()->AppendSlice(trackMap->mResampledData,
mCurrentOutputPosition, mCurrentOutputPosition + WEBAUDIO_BLOCK_SIZE);
trackMap->mResampledData.ForgetUpTo(mCurrentOutputPosition + WEBAUDIO_BLOCK_SIZE);
inputChannels = GetAudioChannelsSuperset(inputChannels, trackMap->mResamplerChannelCount);
}
for (int32_t i = mTrackMap.Length() - 1; i >= 0; --i) {
if (i >= int32_t(trackMapEntriesUsed.Length()) || !trackMapEntriesUsed[i]) {
mTrackMap.RemoveElementAt(i);
for (AudioSegment::ChunkIterator iter(segment); !iter.IsEnded(); iter.Next()) {
inputChannels = GetAudioChannelsSuperset(inputChannels, iter->ChannelCount());
}
}
@ -433,7 +171,7 @@ AudioNodeExternalInputStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
nsAutoTArray<float,GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
for (uint32_t i = 0; i < audioSegments.Length(); ++i) {
AudioChunk tmpChunk;
ConvertSegmentToAudioBlock(&audioSegments[i], &tmpChunk);
ConvertSegmentToAudioBlock(&audioSegments[i], &tmpChunk, inputChannels);
if (!tmpChunk.IsNull()) {
if (accumulateIndex == 0) {
AllocateAudioBlock(inputChannels, &mLastChunks[0]);
@ -446,7 +184,6 @@ AudioNodeExternalInputStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
if (accumulateIndex == 0) {
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
}
mCurrentOutputPosition += WEBAUDIO_BLOCK_SIZE;
// Using AudioNodeStream's AdvanceOutputSegment to push the media stream graph along with null data.
AdvanceOutputSegment();

74
dom/media/webaudio/AudioNodeExternalInputStream.h
View File

@ -10,16 +10,13 @@
#include "AudioNodeStream.h"
#include "mozilla/Atomics.h"
// Forward declaration for mResamplerMap
typedef struct SpeexResamplerState_ SpeexResamplerState;
namespace mozilla {
/**
* This is a MediaStream implementation that acts for a Web Audio node but
* unlike other AudioNodeStreams, supports any kind of MediaStream as an
* input --- handling any number of audio tracks, resampling them from whatever
* sample rate they're using, and handling blocking of the input MediaStream.
* input --- handling any number of audio tracks and handling blocking of
* the input MediaStream.
*/
class AudioNodeExternalInputStream : public AudioNodeStream {
public:
@ -31,73 +28,6 @@ public:
virtual void ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags) MOZ_OVERRIDE;
private:
// For storing pointers and data about input tracks, like the last TrackTick which
// was read, and the associated speex resampler.
struct TrackMapEntry {
~TrackMapEntry();
/**
* Resamples data from all chunks in aIterator and following, using mResampler,
* adding the results to mResampledData.
*/
void ResampleInputData(AudioSegment* aSegment);
/**
* Resamples a set of channel buffers using mResampler, adding the results
* to mResampledData.
*/
void ResampleChannels(const nsTArray<const void*>& aBuffers,
uint32_t aInputDuration,
AudioSampleFormat aFormat,
float aVolume);
// mEndOfConsumedInputTicks is the end of the input ticks that we've consumed.
// 0 if we haven't consumed any yet.
TrackTicks mEndOfConsumedInputTicks;
// mEndOfLastInputIntervalInInputStream is the timestamp for the end of the
// previous interval which was unblocked for both the input and output
// stream, in the input stream's timeline, or -1 if there wasn't one.
StreamTime mEndOfLastInputIntervalInInputStream;
// mEndOfLastInputIntervalInOutputStream is the timestamp for the end of the
// previous interval which was unblocked for both the input and output
// stream, in the output stream's timeline, or -1 if there wasn't one.
StreamTime mEndOfLastInputIntervalInOutputStream;
/**
* Number of samples passed to the resampler so far.
*/
TrackTicks mSamplesPassedToResampler;
/**
* Resampler being applied to this track.
*/
SpeexResamplerState* mResampler;
/**
* The track data that has been resampled to the rate of the
* AudioNodeExternalInputStream. All data in these chunks is in floats (or null),
* and has the number of channels given in mResamplerChannelCount.
* mResampledData starts at zero in the stream's output track (so generally
* it will consist of null data followed by actual data).
*/
AudioSegment mResampledData;
/**
* Number of channels used to create mResampler.
*/
uint32_t mResamplerChannelCount;
/**
* The ID for the track of the input stream this entry is for.
*/
TrackID mTrackID;
};
nsTArray<TrackMapEntry> mTrackMap;
// Amount of track data produced so far. A multiple of WEBAUDIO_BLOCK_SIZE.
TrackTicks mCurrentOutputPosition;
/**
* Creates a TrackMapEntry for the track, if needed. Returns the index
* of the TrackMapEntry or NoIndex if no entry is needed yet.
*/
size_t GetTrackMapEntry(const StreamBuffer::Track& aTrack,
GraphTime aFrom);
/**
* Determines if this is enabled or not. Disabled nodes produce silence.
* This node becomes disabled if the document principal does not subsume the