gecko/content/media/AudioNodeStream.cpp

592 lines
19 KiB
C++

/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#include "AudioNodeStream.h"
#include "MediaStreamGraphImpl.h"
#include "AudioNodeEngine.h"
#include "ThreeDPoint.h"
#include "AudioChannelFormat.h"
#include "AudioParamTimeline.h"
#include "AudioContext.h"
using namespace mozilla::dom;
namespace mozilla {
/**
* An AudioNodeStream produces a single audio track with ID
* AUDIO_TRACK. This track has rate AudioContext::sIdealAudioRate
* for regular audio contexts, and the rate requested by the web content
* for offline audio contexts.
* Each chunk in the track is a single block of WEBAUDIO_BLOCK_SIZE samples.
* Note: This must be a different value than MEDIA_STREAM_DEST_TRACK_ID
*/
AudioNodeStream::~AudioNodeStream()
{
MOZ_COUNT_DTOR(AudioNodeStream);
}
size_t
AudioNodeStream::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
{
size_t amount = 0;
// Not reported:
// - mEngine
amount += ProcessedMediaStream::SizeOfExcludingThis(aMallocSizeOf);
amount += mLastChunks.SizeOfExcludingThis(aMallocSizeOf);
for (size_t i = 0; i < mLastChunks.Length(); i++) {
// NB: This is currently unshared only as there are instances of
// double reporting in DMD otherwise.
amount += mLastChunks[i].SizeOfExcludingThisIfUnshared(aMallocSizeOf);
}
return amount;
}
size_t
AudioNodeStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
{
return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
}
void
AudioNodeStream::SizeOfAudioNodesIncludingThis(MallocSizeOf aMallocSizeOf,
AudioNodeSizes& aUsage) const
{
// Explicitly separate out the stream memory.
aUsage.mStream = SizeOfIncludingThis(aMallocSizeOf);
if (mEngine) {
// This will fill out the rest of |aUsage|.
mEngine->SizeOfIncludingThis(aMallocSizeOf, aUsage);
}
}
void
AudioNodeStream::SetStreamTimeParameter(uint32_t aIndex, AudioContext* aContext,
double aStreamTime)
{
class Message : public ControlMessage {
public:
Message(AudioNodeStream* aStream, uint32_t aIndex, MediaStream* aRelativeToStream,
double aStreamTime)
: ControlMessage(aStream), mStreamTime(aStreamTime),
mRelativeToStream(aRelativeToStream), mIndex(aIndex) {}
virtual void Run()
{
static_cast<AudioNodeStream*>(mStream)->
SetStreamTimeParameterImpl(mIndex, mRelativeToStream, mStreamTime);
}
double mStreamTime;
MediaStream* mRelativeToStream;
uint32_t mIndex;
};
MOZ_ASSERT(this);
GraphImpl()->AppendMessage(new Message(this, aIndex,
aContext->DestinationStream(),
aContext->DOMTimeToStreamTime(aStreamTime)));
}
void
AudioNodeStream::SetStreamTimeParameterImpl(uint32_t aIndex, MediaStream* aRelativeToStream,
double aStreamTime)
{
TrackTicks ticks = TicksFromDestinationTime(aRelativeToStream, aStreamTime);
mEngine->SetStreamTimeParameter(aIndex, ticks);
}
void
AudioNodeStream::SetDoubleParameter(uint32_t aIndex, double aValue)
{
class Message : public ControlMessage {
public:
Message(AudioNodeStream* aStream, uint32_t aIndex, double aValue)
: ControlMessage(aStream), mValue(aValue), mIndex(aIndex) {}
virtual void Run()
{
static_cast<AudioNodeStream*>(mStream)->Engine()->
SetDoubleParameter(mIndex, mValue);
}
double mValue;
uint32_t mIndex;
};
MOZ_ASSERT(this);
GraphImpl()->AppendMessage(new Message(this, aIndex, aValue));
}
void
AudioNodeStream::SetInt32Parameter(uint32_t aIndex, int32_t aValue)
{
class Message : public ControlMessage {
public:
Message(AudioNodeStream* aStream, uint32_t aIndex, int32_t aValue)
: ControlMessage(aStream), mValue(aValue), mIndex(aIndex) {}
virtual void Run()
{
static_cast<AudioNodeStream*>(mStream)->Engine()->
SetInt32Parameter(mIndex, mValue);
}
int32_t mValue;
uint32_t mIndex;
};
MOZ_ASSERT(this);
GraphImpl()->AppendMessage(new Message(this, aIndex, aValue));
}
void
AudioNodeStream::SetTimelineParameter(uint32_t aIndex,
const AudioParamTimeline& aValue)
{
class Message : public ControlMessage {
public:
Message(AudioNodeStream* aStream, uint32_t aIndex,
const AudioParamTimeline& aValue)
: ControlMessage(aStream),
mValue(aValue),
mSampleRate(aStream->SampleRate()),
mIndex(aIndex) {}
virtual void Run()
{
static_cast<AudioNodeStream*>(mStream)->Engine()->
SetTimelineParameter(mIndex, mValue, mSampleRate);
}
AudioParamTimeline mValue;
TrackRate mSampleRate;
uint32_t mIndex;
};
GraphImpl()->AppendMessage(new Message(this, aIndex, aValue));
}
void
AudioNodeStream::SetThreeDPointParameter(uint32_t aIndex, const ThreeDPoint& aValue)
{
class Message : public ControlMessage {
public:
Message(AudioNodeStream* aStream, uint32_t aIndex, const ThreeDPoint& aValue)
: ControlMessage(aStream), mValue(aValue), mIndex(aIndex) {}
virtual void Run()
{
static_cast<AudioNodeStream*>(mStream)->Engine()->
SetThreeDPointParameter(mIndex, mValue);
}
ThreeDPoint mValue;
uint32_t mIndex;
};
MOZ_ASSERT(this);
GraphImpl()->AppendMessage(new Message(this, aIndex, aValue));
}
void
AudioNodeStream::SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList>&& aBuffer)
{
class Message : public ControlMessage {
public:
Message(AudioNodeStream* aStream,
already_AddRefed<ThreadSharedFloatArrayBufferList>& aBuffer)
: ControlMessage(aStream), mBuffer(aBuffer) {}
virtual void Run()
{
static_cast<AudioNodeStream*>(mStream)->Engine()->
SetBuffer(mBuffer.forget());
}
nsRefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
};
MOZ_ASSERT(this);
GraphImpl()->AppendMessage(new Message(this, aBuffer));
}
void
AudioNodeStream::SetRawArrayData(nsTArray<float>& aData)
{
class Message : public ControlMessage {
public:
Message(AudioNodeStream* aStream,
nsTArray<float>& aData)
: ControlMessage(aStream)
{
mData.SwapElements(aData);
}
virtual void Run()
{
static_cast<AudioNodeStream*>(mStream)->Engine()->SetRawArrayData(mData);
}
nsTArray<float> mData;
};
MOZ_ASSERT(this);
GraphImpl()->AppendMessage(new Message(this, aData));
}
void
AudioNodeStream::SetChannelMixingParameters(uint32_t aNumberOfChannels,
ChannelCountMode aChannelCountMode,
ChannelInterpretation aChannelInterpretation)
{
class Message : public ControlMessage {
public:
Message(AudioNodeStream* aStream,
uint32_t aNumberOfChannels,
ChannelCountMode aChannelCountMode,
ChannelInterpretation aChannelInterpretation)
: ControlMessage(aStream),
mNumberOfChannels(aNumberOfChannels),
mChannelCountMode(aChannelCountMode),
mChannelInterpretation(aChannelInterpretation)
{}
virtual void Run()
{
static_cast<AudioNodeStream*>(mStream)->
SetChannelMixingParametersImpl(mNumberOfChannels, mChannelCountMode,
mChannelInterpretation);
}
uint32_t mNumberOfChannels;
ChannelCountMode mChannelCountMode;
ChannelInterpretation mChannelInterpretation;
};
MOZ_ASSERT(this);
GraphImpl()->AppendMessage(new Message(this, aNumberOfChannels,
aChannelCountMode,
aChannelInterpretation));
}
void
AudioNodeStream::SetChannelMixingParametersImpl(uint32_t aNumberOfChannels,
ChannelCountMode aChannelCountMode,
ChannelInterpretation aChannelInterpretation)
{
// Make sure that we're not clobbering any significant bits by fitting these
// values in 16 bits.
MOZ_ASSERT(int(aChannelCountMode) < INT16_MAX);
MOZ_ASSERT(int(aChannelInterpretation) < INT16_MAX);
mNumberOfInputChannels = aNumberOfChannels;
mChannelCountMode = aChannelCountMode;
mChannelInterpretation = aChannelInterpretation;
}
uint32_t
AudioNodeStream::ComputedNumberOfChannels(uint32_t aInputChannelCount)
{
switch (mChannelCountMode) {
case ChannelCountMode::Explicit:
// Disregard the channel count we've calculated from inputs, and just use
// mNumberOfInputChannels.
return mNumberOfInputChannels;
case ChannelCountMode::Clamped_max:
// Clamp the computed output channel count to mNumberOfInputChannels.
return std::min(aInputChannelCount, mNumberOfInputChannels);
default:
case ChannelCountMode::Max:
// Nothing to do here, just shut up the compiler warning.
return aInputChannelCount;
}
}
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);
}
}
void
AudioNodeStream::AccumulateInputChunk(uint32_t aInputIndex, const AudioChunk& aChunk,
AudioChunk* aBlock,
nsTArray<float>* aDownmixBuffer)
{
nsAutoTArray<const void*,GUESS_AUDIO_CHANNELS> channels;
UpMixDownMixChunk(&aChunk, aBlock->mChannelData.Length(), channels, *aDownmixBuffer);
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*>(aBlock->mChannelData[c]));
if (inputData) {
if (aInputIndex == 0) {
AudioBlockCopyChannelWithScale(inputData, aChunk.mVolume, outputData);
} else {
AudioBlockAddChannelWithScale(inputData, aChunk.mVolume, outputData);
}
} else {
if (aInputIndex == 0) {
PodZero(outputData, WEBAUDIO_BLOCK_SIZE);
}
}
}
}
void
AudioNodeStream::UpMixDownMixChunk(const AudioChunk* aChunk,
uint32_t aOutputChannelCount,
nsTArray<const void*>& aOutputChannels,
nsTArray<float>& aDownmixBuffer)
{
static const float silenceChannel[WEBAUDIO_BLOCK_SIZE] = {0.f};
aOutputChannels.AppendElements(aChunk->mChannelData);
if (aOutputChannels.Length() < aOutputChannelCount) {
if (mChannelInterpretation == ChannelInterpretation::Speakers) {
AudioChannelsUpMix(&aOutputChannels, aOutputChannelCount, nullptr);
NS_ASSERTION(aOutputChannelCount == aOutputChannels.Length(),
"We called GetAudioChannelsSuperset to avoid this");
} else {
// Fill up the remaining aOutputChannels by zeros
for (uint32_t j = aOutputChannels.Length(); j < aOutputChannelCount; ++j) {
aOutputChannels.AppendElement(silenceChannel);
}
}
} else if (aOutputChannels.Length() > aOutputChannelCount) {
if (mChannelInterpretation == ChannelInterpretation::Speakers) {
nsAutoTArray<float*,GUESS_AUDIO_CHANNELS> outputChannels;
outputChannels.SetLength(aOutputChannelCount);
aDownmixBuffer.SetLength(aOutputChannelCount * WEBAUDIO_BLOCK_SIZE);
for (uint32_t j = 0; j < aOutputChannelCount; ++j) {
outputChannels[j] = &aDownmixBuffer[j * WEBAUDIO_BLOCK_SIZE];
}
AudioChannelsDownMix(aOutputChannels, outputChannels.Elements(),
aOutputChannelCount, WEBAUDIO_BLOCK_SIZE);
aOutputChannels.SetLength(aOutputChannelCount);
for (uint32_t j = 0; j < aOutputChannels.Length(); ++j) {
aOutputChannels[j] = outputChannels[j];
}
} else {
// Drop the remaining aOutputChannels
aOutputChannels.RemoveElementsAt(aOutputChannelCount,
aOutputChannels.Length() - aOutputChannelCount);
}
}
}
// The MediaStreamGraph guarantees that this is actually one block, for
// AudioNodeStreams.
void
AudioNodeStream::ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags)
{
EnsureTrack(AUDIO_TRACK, mSampleRate);
// No more tracks will be coming
mBuffer.AdvanceKnownTracksTime(STREAM_TIME_MAX);
uint16_t outputCount = std::max(uint16_t(1), mEngine->OutputCount());
mLastChunks.SetLength(outputCount);
// Consider this stream blocked if it has already finished output. Normally
// mBlocked would reflect this, but due to rounding errors our audio track may
// appear to extend slightly beyond aFrom, so we might not be blocked yet.
bool blocked = mFinished || mBlocked.GetAt(aFrom);
// If the stream has finished at this time, it will be blocked.
if (mMuted || blocked) {
for (uint16_t i = 0; i < outputCount; ++i) {
mLastChunks[i].SetNull(WEBAUDIO_BLOCK_SIZE);
}
} else {
// We need to generate at least one input
uint16_t maxInputs = std::max(uint16_t(1), mEngine->InputCount());
OutputChunks inputChunks;
inputChunks.SetLength(maxInputs);
for (uint16_t i = 0; i < maxInputs; ++i) {
ObtainInputBlock(inputChunks[i], i);
}
bool finished = false;
if (maxInputs <= 1 && mEngine->OutputCount() <= 1) {
mEngine->ProcessBlock(this, inputChunks[0], &mLastChunks[0], &finished);
} else {
mEngine->ProcessBlocksOnPorts(this, inputChunks, mLastChunks, &finished);
}
for (uint16_t i = 0; i < outputCount; ++i) {
NS_ASSERTION(mLastChunks[i].GetDuration() == WEBAUDIO_BLOCK_SIZE,
"Invalid WebAudio chunk size");
}
if (finished) {
mMarkAsFinishedAfterThisBlock = true;
}
if (mDisabledTrackIDs.Contains(static_cast<TrackID>(AUDIO_TRACK))) {
for (uint32_t i = 0; i < outputCount; ++i) {
mLastChunks[i].SetNull(WEBAUDIO_BLOCK_SIZE);
}
}
}
if (!blocked) {
// Don't output anything while blocked
AdvanceOutputSegment();
if (mMarkAsFinishedAfterThisBlock && (aFlags & ALLOW_FINISH)) {
// This stream was finished the last time that we looked at it, and all
// of the depending streams have finished their output as well, so now
// it's time to mark this stream as finished.
FinishOutput();
}
}
}
void
AudioNodeStream::AdvanceOutputSegment()
{
StreamBuffer::Track* track = EnsureTrack(AUDIO_TRACK, mSampleRate);
AudioSegment* segment = track->Get<AudioSegment>();
if (mKind == MediaStreamGraph::EXTERNAL_STREAM) {
segment->AppendAndConsumeChunk(&mLastChunks[0]);
} else {
segment->AppendNullData(mLastChunks[0].GetDuration());
}
for (uint32_t j = 0; j < mListeners.Length(); ++j) {
MediaStreamListener* l = mListeners[j];
AudioChunk copyChunk = mLastChunks[0];
AudioSegment tmpSegment;
tmpSegment.AppendAndConsumeChunk(&copyChunk);
l->NotifyQueuedTrackChanges(Graph(), AUDIO_TRACK,
mSampleRate, segment->GetDuration(), 0,
tmpSegment);
}
}
TrackTicks
AudioNodeStream::GetCurrentPosition()
{
return EnsureTrack(AUDIO_TRACK, mSampleRate)->Get<AudioSegment>()->GetDuration();
}
void
AudioNodeStream::FinishOutput()
{
if (IsFinishedOnGraphThread()) {
return;
}
StreamBuffer::Track* track = EnsureTrack(AUDIO_TRACK, mSampleRate);
track->SetEnded();
FinishOnGraphThread();
for (uint32_t j = 0; j < mListeners.Length(); ++j) {
MediaStreamListener* l = mListeners[j];
AudioSegment emptySegment;
l->NotifyQueuedTrackChanges(Graph(), AUDIO_TRACK,
mSampleRate,
track->GetSegment()->GetDuration(),
MediaStreamListener::TRACK_EVENT_ENDED, emptySegment);
}
}
double
AudioNodeStream::TimeFromDestinationTime(AudioNodeStream* aDestination,
double aSeconds)
{
MOZ_ASSERT(aDestination->SampleRate() == SampleRate());
double destinationSeconds = std::max(0.0, aSeconds);
StreamTime streamTime = SecondsToMediaTime(destinationSeconds);
// MediaTime does not have the resolution of double
double offset = destinationSeconds - MediaTimeToSeconds(streamTime);
GraphTime graphTime = aDestination->StreamTimeToGraphTime(streamTime);
StreamTime thisStreamTime = GraphTimeToStreamTimeOptimistic(graphTime);
double thisSeconds = MediaTimeToSeconds(thisStreamTime) + offset;
MOZ_ASSERT(thisSeconds >= 0.0);
return thisSeconds;
}
TrackTicks
AudioNodeStream::TicksFromDestinationTime(MediaStream* aDestination,
double aSeconds)
{
AudioNodeStream* destination = aDestination->AsAudioNodeStream();
MOZ_ASSERT(destination);
double thisSeconds = TimeFromDestinationTime(destination, aSeconds);
// Round to nearest
TrackTicks ticks = thisSeconds * SampleRate() + 0.5;
return ticks;
}
double
AudioNodeStream::DestinationTimeFromTicks(AudioNodeStream* aDestination,
TrackTicks aPosition)
{
MOZ_ASSERT(SampleRate() == aDestination->SampleRate());
StreamTime sourceTime = TicksToTimeRoundDown(SampleRate(), aPosition);
GraphTime graphTime = StreamTimeToGraphTime(sourceTime);
StreamTime destinationTime = aDestination->GraphTimeToStreamTimeOptimistic(graphTime);
return MediaTimeToSeconds(destinationTime);
}
}