/**
* 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,
int32_t aFallbackChannelCount)
{
NS_ASSERTION(aSegment->GetDuration() == WEBAUDIO_BLOCK_SIZE, "Bad segment duration");
{
AudioSegment::ChunkIterator ci(*aSegment);
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;
}
}
AllocateAudioBlock(aFallbackChannelCount, aBlock);
uint32_t duration = 0;
for (AudioSegment::ChunkIterator ci(*aSegment); !ci.IsEnded(); ci.Next()) {
CopyChunkToBlock(*ci, aBlock, duration);
duration += ci->GetDuration();
}
}
void Reverb::initialize(const nsTArray<const float*>& impulseResponseBuffer,
size_t impulseResponseBufferLength, size_t renderSliceSize,
size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
{
m_impulseResponseLength = impulseResponseBufferLength;
// The reverb can handle a mono impulse response and still do stereo processing
size_t numResponseChannels = impulseResponseBuffer.Length();
m_convolvers.SetCapacity(numberOfChannels);
int convolverRenderPhase = 0;
for (size_t i = 0; i < numResponseChannels; ++i) {
const float* channel = impulseResponseBuffer[i];
size_t length = impulseResponseBufferLength;
nsAutoPtr<ReverbConvolver> convolver(new ReverbConvolver(channel, length, renderSliceSize, maxFFTSize, convolverRenderPhase, useBackgroundThreads));
m_convolvers.AppendElement(convolver.forget());
convolverRenderPhase += renderSliceSize;
}
// For "True" stereo processing we allocate a temporary buffer to avoid repeatedly allocating it in the process() method.
// It can be bad to allocate memory in a real-time thread.
if (numResponseChannels == 4) {
AllocateAudioBlock(2, &m_tempBuffer);
WriteZeroesToAudioBlock(&m_tempBuffer, 0, WEBAUDIO_BLOCK_SIZE);
}
}
/**
* Copy as many frames as possible from the source buffer to aOutput, and
* advance aOffsetWithinBlock and aCurrentPosition based on how many frames
* we write. This will never advance aOffsetWithinBlock past
* WEBAUDIO_BLOCK_SIZE, or aCurrentPosition past mStop. It takes data from
* the buffer at aBufferOffset, and never takes more data than aBufferMax.
* This function knows when it needs to allocate the output buffer, and also
* optimizes the case where it can avoid memory allocations.
*/
void CopyFromBuffer(AudioNodeStream* aStream,
AudioChunk* aOutput,
uint32_t aChannels,
uint32_t* aOffsetWithinBlock,
TrackTicks* aCurrentPosition,
int32_t aBufferMax)
{
MOZ_ASSERT(*aCurrentPosition < mStop);
uint32_t numFrames =
std::min(std::min<TrackTicks>(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
aBufferMax - mBufferPosition),
mStop - *aCurrentPosition);
if (numFrames == WEBAUDIO_BLOCK_SIZE && !mResampler) {
MOZ_ASSERT(mBufferPosition < aBufferMax);
BorrowFromInputBuffer(aOutput, aChannels);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mBufferPosition += numFrames;
} else {
if (*aOffsetWithinBlock == 0) {
AllocateAudioBlock(aChannels, aOutput);
}
if (!mResampler) {
MOZ_ASSERT(mBufferPosition < aBufferMax);
CopyFromInputBuffer(aOutput, aChannels, *aOffsetWithinBlock, numFrames);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mBufferPosition += numFrames;
} else {
CopyFromInputBufferWithResampling(aStream, aOutput, aChannels, aOffsetWithinBlock, aCurrentPosition, aBufferMax);
}
}
}
/**
* 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.
*/
static void
ConvertSegmentToAudioBlock(AudioSegment* aSegment, AudioChunk* aBlock)
{
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;
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);
uint32_t duration = 0;
while (!ci.IsEnded()) {
CopyChunkToBlock(*ci, aBlock, duration);
duration += ci->GetDuration();
ci.Next();
}
}
void
AudioNodeStream::ObtainInputBlock(AudioChunk& aTmpChunk, uint32_t aPortIndex)
{
uint32_t inputCount = mInputs.Length();
uint32_t outputChannelCount = 1;
nsAutoTArray<AudioChunk*,250> inputChunks;
for (uint32_t i = 0; i < inputCount; ++i) {
if (aPortIndex != mInputs[i]->InputNumber()) {
// This input is connected to a different port
continue;
}
MediaStream* s = mInputs[i]->GetSource();
AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
MOZ_ASSERT(a == s->AsAudioNodeStream());
if (a->IsAudioParamStream()) {
continue;
}
AudioChunk* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()];
MOZ_ASSERT(chunk);
if (chunk->IsNull() || chunk->mChannelData.IsEmpty()) {
continue;
}
inputChunks.AppendElement(chunk);
outputChannelCount =
GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length());
}
outputChannelCount = ComputedNumberOfChannels(outputChannelCount);
uint32_t inputChunkCount = inputChunks.Length();
if (inputChunkCount == 0 ||
(inputChunkCount == 1 && inputChunks[0]->mChannelData.Length() == 0)) {
aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
if (inputChunkCount == 1 &&
inputChunks[0]->mChannelData.Length() == outputChannelCount) {
aTmpChunk = *inputChunks[0];
return;
}
if (outputChannelCount == 0) {
aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
AllocateAudioBlock(outputChannelCount, &aTmpChunk);
// The static storage here should be 1KB, so it's fine
nsAutoTArray<float, GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
for (uint32_t i = 0; i < inputChunkCount; ++i) {
AccumulateInputChunk(i, *inputChunks[i], &aTmpChunk, &downmixBuffer);
}
}
void
DelayBuffer::Read(const double aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
AudioChunk* aOutputChunk,
ChannelInterpretation aChannelInterpretation)
{
int chunkCount = mChunks.Length();
if (!chunkCount) {
aOutputChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
// Find the maximum number of contributing channels to determine the output
// channel count that retains all signal information. Buffered blocks will
// be upmixed if necessary.
//
// First find the range of "delay" offsets backwards from the current
// position. Note that these may be negative for frames that are after the
// current position (including i).
double minDelay = aPerFrameDelays[0];
double maxDelay = minDelay;
for (unsigned i = 1; i < WEBAUDIO_BLOCK_SIZE; ++i) {
minDelay = std::min(minDelay, aPerFrameDelays[i] - i);
maxDelay = std::max(maxDelay, aPerFrameDelays[i] - i);
}
// Now find the chunks touched by this range and check their channel counts.
int oldestChunk = ChunkForDelay(int(maxDelay) + 1);
int youngestChunk = ChunkForDelay(minDelay);
uint32_t channelCount = 0;
for (int i = oldestChunk; true; i = (i + 1) % chunkCount) {
channelCount = GetAudioChannelsSuperset(channelCount,
mChunks[i].ChannelCount());
if (i == youngestChunk) {
break;
}
}
if (channelCount) {
AllocateAudioBlock(channelCount, aOutputChunk);
ReadChannels(aPerFrameDelays, aOutputChunk,
0, channelCount, aChannelInterpretation);
} else {
aOutputChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
}
// Remember currentDelayFrames for the next ProcessBlock call
mCurrentDelay = aPerFrameDelays[WEBAUDIO_BLOCK_SIZE - 1];
}
/**
* Fill aOutput with as many zero frames as we can, and advance
* aOffsetWithinBlock and aCurrentPosition based on how many frames we write.
* This will never advance aOffsetWithinBlock past WEBAUDIO_BLOCK_SIZE or
* aCurrentPosition past aMaxPos. This function knows when it needs to
* allocate the output buffer, and also optimizes the case where it can avoid
* memory allocations.
*/
void FillWithZeroes(AudioChunk* aOutput,
uint32_t aChannels,
uint32_t* aOffsetWithinBlock,
TrackTicks* aCurrentPosition,
TrackTicks aMaxPos)
{
uint32_t numFrames = std::min(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
uint32_t(aMaxPos - *aCurrentPosition));
if (numFrames == WEBAUDIO_BLOCK_SIZE) {
aOutput->SetNull(numFrames);
} else {
if (aOutput->IsNull()) {
AllocateAudioBlock(aChannels, aOutput);
}
WriteZeroesToAudioBlock(aOutput, *aOffsetWithinBlock, numFrames);
}
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
}
virtual void ProduceAudioBlock(AudioNodeStream* aStream,
const AudioChunk& aInput,
AudioChunk* aOutput,
bool* aFinished)
{
uint32_t channelCount = aInput.mChannelData.Length();
if (!mCurve.Length() || !channelCount) {
// Optimize the case where we don't have a curve buffer,
// or the input is null.
*aOutput = aInput;
return;
}
AllocateAudioBlock(channelCount, aOutput);
for (uint32_t i = 0; i < channelCount; ++i) {
const float* inputBuffer = static_cast<const float*>(aInput.mChannelData[i]);
float* outputBuffer = const_cast<float*> (static_cast<const float*>(aOutput->mChannelData[i]));
float* sampleBuffer;
switch (mType) {
case OverSampleType::None:
mResampler.Reset(channelCount, aStream->SampleRate(), OverSampleType::None);
ProcessCurve<1>(inputBuffer, outputBuffer);
break;
case OverSampleType::_2x:
mResampler.Reset(channelCount, aStream->SampleRate(), OverSampleType::_2x);
sampleBuffer = mResampler.UpSample(i, inputBuffer, 2);
ProcessCurve<2>(sampleBuffer, sampleBuffer);
mResampler.DownSample(i, outputBuffer, 2);
break;
case OverSampleType::_4x:
mResampler.Reset(channelCount, aStream->SampleRate(), OverSampleType::_4x);
sampleBuffer = mResampler.UpSample(i, inputBuffer, 4);
ProcessCurve<4>(sampleBuffer, sampleBuffer);
mResampler.DownSample(i, outputBuffer, 4);
break;
default:
NS_NOTREACHED("We should never reach here");
}
}
}
/**
* Copy as many frames as possible from the source buffer to aOutput, and
* advance aOffsetWithinBlock and aCurrentPosition based on how many frames
* we copy. This will never advance aOffsetWithinBlock past
* WEBAUDIO_BLOCK_SIZE, or aCurrentPosition past mStop. It takes data from
* the buffer at aBufferOffset, and never takes more data than aBufferMax.
* This function knows when it needs to allocate the output buffer, and also
* optimizes the case where it can avoid memory allocations.
*/
void CopyFromBuffer(AudioNodeStream* aStream,
AudioChunk* aOutput,
uint32_t aChannels,
uint32_t* aOffsetWithinBlock,
TrackTicks* aCurrentPosition,
uint32_t aBufferOffset,
uint32_t aBufferMax)
{
MOZ_ASSERT(*aCurrentPosition < mStop);
uint32_t numFrames =
std::min<TrackTicks>(std::min(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
aBufferMax - aBufferOffset),
mStop - *aCurrentPosition);
if (numFrames == WEBAUDIO_BLOCK_SIZE && !ShouldResample(aStream->SampleRate())) {
BorrowFromInputBuffer(aOutput, aChannels, aBufferOffset);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mPosition += numFrames;
} else {
if (aOutput->IsNull()) {
MOZ_ASSERT(*aOffsetWithinBlock == 0);
AllocateAudioBlock(aChannels, aOutput);
}
if (!ShouldResample(aStream->SampleRate())) {
CopyFromInputBuffer(aOutput, aChannels, aBufferOffset, *aOffsetWithinBlock, numFrames);
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
mPosition += numFrames;
} else {
uint32_t framesRead, framesWritten, availableInInputBuffer;
availableInInputBuffer = aBufferMax - aBufferOffset;
CopyFromInputBufferWithResampling(aStream, aOutput, aChannels, aBufferOffset, *aOffsetWithinBlock, availableInInputBuffer, framesRead, framesWritten);
*aOffsetWithinBlock += framesWritten;
*aCurrentPosition += framesRead;
mPosition += framesRead;
}
}
}
void
AudioNodeStream::ObtainInputBlock(AudioChunk& aTmpChunk, uint32_t aPortIndex)
{
uint32_t inputCount = mInputs.Length();
uint32_t outputChannelCount = 1;
nsAutoTArray<AudioChunk*,250> inputChunks;
for (uint32_t i = 0; i < inputCount; ++i) {
if (aPortIndex != mInputs[i]->InputNumber()) {
// This input is connected to a different port
continue;
}
MediaStream* s = mInputs[i]->GetSource();
AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
MOZ_ASSERT(a == s->AsAudioNodeStream());
if (a->IsAudioParamStream()) {
continue;
}
// It is possible for mLastChunks to be empty here, because `a` might be a
// AudioNodeStream that has not been scheduled yet, because it is further
// down the graph _but_ as a connection to this node. Because we enforce the
// presence of at least one DelayNode, with at least one block of delay, and
// because the output of a DelayNode when it has been fed less that
// `delayTime` amount of audio is silence, we can simply continue here,
// because this input would not influence the output of this node. Next
// iteration, a->mLastChunks.IsEmpty() will be false, and everthing will
// work as usual.
if (a->mLastChunks.IsEmpty()) {
continue;
}
AudioChunk* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()];
MOZ_ASSERT(chunk);
if (chunk->IsNull() || chunk->mChannelData.IsEmpty()) {
continue;
}
inputChunks.AppendElement(chunk);
outputChannelCount =
GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length());
}
outputChannelCount = ComputedNumberOfChannels(outputChannelCount);
uint32_t inputChunkCount = inputChunks.Length();
if (inputChunkCount == 0 ||
(inputChunkCount == 1 && inputChunks[0]->mChannelData.Length() == 0)) {
aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
if (inputChunkCount == 1 &&
inputChunks[0]->mChannelData.Length() == outputChannelCount) {
aTmpChunk = *inputChunks[0];
return;
}
if (outputChannelCount == 0) {
aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
AllocateAudioBlock(outputChannelCount, &aTmpChunk);
// The static storage here should be 1KB, so it's fine
nsAutoTArray<float, GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
for (uint32_t i = 0; i < inputChunkCount; ++i) {
AccumulateInputChunk(i, *inputChunks[i], &aTmpChunk, &downmixBuffer);
}
}
AudioChunk*
AudioNodeStream::ObtainInputBlock(AudioChunk* aTmpChunk)
{
uint32_t inputCount = mInputs.Length();
uint32_t outputChannelCount = 0;
nsAutoTArray<AudioChunk*,250> inputChunks;
for (uint32_t i = 0; i < inputCount; ++i) {
MediaStream* s = mInputs[i]->GetSource();
AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
MOZ_ASSERT(a == s->AsAudioNodeStream());
if (a->IsFinishedOnGraphThread()) {
continue;
}
AudioChunk* chunk = &a->mLastChunk;
// XXX when we implement DelayNode, this will no longer be true and we'll
// need to treat a null chunk (when the DelayNode hasn't had a chance
// to produce data yet) as silence here.
MOZ_ASSERT(chunk);
if (chunk->IsNull()) {
continue;
}
inputChunks.AppendElement(chunk);
outputChannelCount =
GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length());
}
uint32_t inputChunkCount = inputChunks.Length();
if (inputChunkCount == 0) {
aTmpChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
return aTmpChunk;
}
if (inputChunkCount == 1) {
return inputChunks[0];
}
AllocateAudioBlock(outputChannelCount, aTmpChunk);
for (uint32_t i = 0; i < inputChunkCount; ++i) {
AudioChunk* chunk = inputChunks[i];
nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channels;
channels.AppendElements(chunk->mChannelData);
if (channels.Length() < outputChannelCount) {
AudioChannelsUpMix(&channels, outputChannelCount, nullptr);
NS_ASSERTION(outputChannelCount == channels.Length(),
"We called GetAudioChannelsSuperset to avoid this");
}
for (uint32_t c = 0; c < channels.Length(); ++c) {
const float* inputData = static_cast<const float*>(channels[c]);
float* outputData = static_cast<float*>(const_cast<void*>(aTmpChunk->mChannelData[c]));
if (inputData) {
if (i == 0) {
AudioBlockCopyChannelWithScale(inputData, chunk->mVolume, outputData);
} else {
AudioBlockAddChannelWithScale(inputData, chunk->mVolume, outputData);
}
} else {
if (i == 0) {
memset(outputData, 0, WEBAUDIO_BLOCK_SIZE*sizeof(float));
}
}
}
}
return aTmpChunk;
}
void
AudioNodeExternalInputStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
uint32_t aFlags)
{
// According to spec, number of outputs is always 1.
MOZ_ASSERT(mLastChunks.Length() == 1);
// GC stuff can result in our input stream being destroyed before this stream.
// Handle that.
if (!IsEnabled() || mInputs.IsEmpty() || mPassThrough) {
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
AdvanceOutputSegment();
return;
}
MOZ_ASSERT(mInputs.Length() == 1);
MediaStream* source = mInputs[0]->GetSource();
nsAutoTArray<AudioSegment,1> audioSegments;
uint32_t inputChannels = 0;
for (StreamBuffer::TrackIter tracks(source->mBuffer, MediaSegment::AUDIO);
!tracks.IsEnded(); tracks.Next()) {
const StreamBuffer::Track& inputTrack = *tracks;
const AudioSegment& inputSegment =
*static_cast<AudioSegment*>(inputTrack.GetSegment());
if (inputSegment.IsNull()) {
continue;
}
AudioSegment& segment = *audioSegments.AppendElement();
GraphTime next;
for (GraphTime t = aFrom; t < aTo; t = next) {
MediaInputPort::InputInterval interval = mInputs[0]->GetNextInputInterval(t);
interval.mEnd = std::min(interval.mEnd, aTo);
if (interval.mStart >= interval.mEnd)
break;
next = interval.mEnd;
StreamTime outputStart = GraphTimeToStreamTime(interval.mStart);
StreamTime outputEnd = GraphTimeToStreamTime(interval.mEnd);
StreamTime ticks = outputEnd - outputStart;
if (interval.mInputIsBlocked) {
segment.AppendNullData(ticks);
} else {
StreamTime inputStart =
std::min(inputSegment.GetDuration(),
source->GraphTimeToStreamTime(interval.mStart));
StreamTime inputEnd =
std::min(inputSegment.GetDuration(),
source->GraphTimeToStreamTime(interval.mEnd));
segment.AppendSlice(inputSegment, inputStart, inputEnd);
// Pad if we're looking past the end of the track
segment.AppendNullData(ticks - (inputEnd - inputStart));
}
}
for (AudioSegment::ChunkIterator iter(segment); !iter.IsEnded(); iter.Next()) {
inputChannels = GetAudioChannelsSuperset(inputChannels, iter->ChannelCount());
}
}
uint32_t accumulateIndex = 0;
if (inputChannels) {
nsAutoTArray<float,GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
for (uint32_t i = 0; i < audioSegments.Length(); ++i) {
AudioChunk tmpChunk;
ConvertSegmentToAudioBlock(&audioSegments[i], &tmpChunk, inputChannels);
if (!tmpChunk.IsNull()) {
if (accumulateIndex == 0) {
AllocateAudioBlock(inputChannels, &mLastChunks[0]);
}
AccumulateInputChunk(accumulateIndex, tmpChunk, &mLastChunks[0], &downmixBuffer);
accumulateIndex++;
}
}
}
if (accumulateIndex == 0) {
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
}
// Using AudioNodeStream's AdvanceOutputSegment to push the media stream graph along with null data.
AdvanceOutputSegment();
}
void
AudioNodeStream::ObtainInputBlock(AudioChunk& aTmpChunk, uint32_t aPortIndex)
{
uint32_t inputCount = mInputs.Length();
uint32_t outputChannelCount = 1;
nsAutoTArray<AudioChunk*,250> inputChunks;
for (uint32_t i = 0; i < inputCount; ++i) {
if (aPortIndex != mInputs[i]->InputNumber()) {
// This input is connected to a different port
continue;
}
MediaStream* s = mInputs[i]->GetSource();
AudioNodeStream* a = static_cast<AudioNodeStream*>(s);
MOZ_ASSERT(a == s->AsAudioNodeStream());
if (a->IsFinishedOnGraphThread() ||
a->IsAudioParamStream()) {
continue;
}
AudioChunk* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()];
MOZ_ASSERT(chunk);
if (chunk->IsNull()) {
continue;
}
inputChunks.AppendElement(chunk);
outputChannelCount =
GetAudioChannelsSuperset(outputChannelCount, chunk->mChannelData.Length());
}
switch (mChannelCountMode) {
case ChannelCountMode::Explicit:
// Disregard the output channel count that we've calculated, and just use
// mNumberOfInputChannels.
outputChannelCount = mNumberOfInputChannels;
break;
case ChannelCountMode::Clamped_max:
// Clamp the computed output channel count to mNumberOfInputChannels.
outputChannelCount = std::min(outputChannelCount, mNumberOfInputChannels);
break;
case ChannelCountMode::Max:
// Nothing to do here, just shut up the compiler warning.
break;
}
uint32_t inputChunkCount = inputChunks.Length();
if (inputChunkCount == 0 ||
(inputChunkCount == 1 && inputChunks[0]->mChannelData.Length() == 0)) {
aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
if (inputChunkCount == 1 &&
inputChunks[0]->mChannelData.Length() == outputChannelCount) {
aTmpChunk = *inputChunks[0];
return;
}
AllocateAudioBlock(outputChannelCount, &aTmpChunk);
float silenceChannel[WEBAUDIO_BLOCK_SIZE] = {0.f};
// The static storage here should be 1KB, so it's fine
nsAutoTArray<float, GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
for (uint32_t i = 0; i < inputChunkCount; ++i) {
AudioChunk* chunk = inputChunks[i];
nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channels;
channels.AppendElements(chunk->mChannelData);
if (channels.Length() < outputChannelCount) {
if (mChannelInterpretation == ChannelInterpretation::Speakers) {
AudioChannelsUpMix(&channels, outputChannelCount, nullptr);
NS_ASSERTION(outputChannelCount == channels.Length(),
"We called GetAudioChannelsSuperset to avoid this");
} else {
// Fill up the remaining channels by zeros
for (uint32_t j = channels.Length(); j < outputChannelCount; ++j) {
channels.AppendElement(silenceChannel);
}
}
} else if (channels.Length() > outputChannelCount) {
if (mChannelInterpretation == ChannelInterpretation::Speakers) {
nsAutoTArray<float*,GUESS_AUDIO_CHANNELS> outputChannels;
outputChannels.SetLength(outputChannelCount);
downmixBuffer.SetLength(outputChannelCount * WEBAUDIO_BLOCK_SIZE);
for (uint32_t j = 0; j < outputChannelCount; ++j) {
outputChannels[j] = &downmixBuffer[j * WEBAUDIO_BLOCK_SIZE];
}
AudioChannelsDownMix(channels, outputChannels.Elements(),
outputChannelCount, WEBAUDIO_BLOCK_SIZE);
channels.SetLength(outputChannelCount);
for (uint32_t j = 0; j < channels.Length(); ++j) {
channels[j] = outputChannels[j];
}
} else {
// Drop the remaining channels
channels.RemoveElementsAt(outputChannelCount,
channels.Length() - outputChannelCount);
}
}
for (uint32_t c = 0; c < channels.Length(); ++c) {
const float* inputData = static_cast<const float*>(channels[c]);
float* outputData = static_cast<float*>(const_cast<void*>(aTmpChunk.mChannelData[c]));
if (inputData) {
if (i == 0) {
AudioBlockCopyChannelWithScale(inputData, chunk->mVolume, outputData);
} else {
AudioBlockAddChannelWithScale(inputData, chunk->mVolume, outputData);
}
} else {
if (i == 0) {
memset(outputData, 0, WEBAUDIO_BLOCK_SIZE*sizeof(float));
}
}
}
}
}
void
AudioNodeExternalInputStream::ProcessInput(GraphTime aFrom, GraphTime aTo,
uint32_t aFlags)
{
// According to spec, number of outputs is always 1.
mLastChunks.SetLength(1);
// GC stuff can result in our input stream being destroyed before this stream.
// Handle that.
if (mInputs.IsEmpty()) {
mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
AdvanceOutputSegment();
return;
}
MOZ_ASSERT(mInputs.Length() == 1);
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) {
continue;
}
while (trackMapEntriesUsed.Length() <= trackMapIndex) {
trackMapEntriesUsed.AppendElement(false);
}
trackMapEntriesUsed[trackMapIndex] = true;
TrackMapEntry* trackMap = &mTrackMap[trackMapIndex];
AudioSegment segment;
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);
if (interval.mStart >= interval.mEnd)
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;
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;
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));
}
// Pad if we're looking past the end of the track
segment.AppendNullData(ticks - segment.GetDuration());
}
}
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);
}
}
uint32_t accumulateIndex = 0;
if (inputChannels) {
nsAutoTArray<float,GUESS_AUDIO_CHANNELS*WEBAUDIO_BLOCK_SIZE> downmixBuffer;
for (uint32_t i = 0; i < audioSegments.Length(); ++i) {
AudioChunk tmpChunk;
ConvertSegmentToAudioBlock(&audioSegments[i], &tmpChunk);
if (!tmpChunk.IsNull()) {
if (accumulateIndex == 0) {
AllocateAudioBlock(inputChannels, &mLastChunks[0]);
}
AccumulateInputChunk(accumulateIndex, tmpChunk, &mLastChunks[0], &downmixBuffer);
accumulateIndex++;
}
}
}
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();
}
