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gecko-dev/content/media/webaudio/AudioBufferSourceNode.cpp

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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/* vim:set ts=2 sw=2 sts=2 et cindent: */
/* 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 "AudioBufferSourceNode.h"
#include "mozilla/dom/AudioBufferSourceNodeBinding.h"
#include "mozilla/dom/AudioParam.h"
#include "nsMathUtils.h"
#include "AudioNodeEngine.h"
#include "AudioNodeStream.h"
#include "AudioDestinationNode.h"
#include "AudioParamTimeline.h"
#include <limits>
namespace mozilla {
namespace dom {
NS_IMPL_CYCLE_COLLECTION_CLASS(AudioBufferSourceNode)
NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(AudioBufferSourceNode)
NS_IMPL_CYCLE_COLLECTION_UNLINK(mBuffer)
NS_IMPL_CYCLE_COLLECTION_UNLINK(mPlaybackRate)
if (tmp->Context()) {
// AudioNode's Unlink implementation disconnects us from the graph
// too, but we need to do this right here to make sure that
// UnregisterAudioBufferSourceNode can properly untangle us from
// the possibly connected PannerNodes.
tmp->DisconnectFromGraph();
tmp->Context()->UnregisterAudioBufferSourceNode(tmp);
}
NS_IMPL_CYCLE_COLLECTION_UNLINK_END_INHERITED(AudioNode)
NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(AudioBufferSourceNode, AudioNode)
NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mBuffer)
NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mPlaybackRate)
NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(AudioBufferSourceNode)
NS_INTERFACE_MAP_END_INHERITING(AudioNode)
NS_IMPL_ADDREF_INHERITED(AudioBufferSourceNode, AudioNode)
NS_IMPL_RELEASE_INHERITED(AudioBufferSourceNode, AudioNode)
/**
* Media-thread playback engine for AudioBufferSourceNode.
* Nothing is played until a non-null buffer has been set (via
* AudioNodeStream::SetBuffer) and a non-zero mBufferEnd has been set (via
* AudioNodeStream::SetInt32Parameter).
*/
class AudioBufferSourceNodeEngine : public AudioNodeEngine
{
public:
explicit AudioBufferSourceNodeEngine(AudioNode* aNode,
AudioDestinationNode* aDestination) :
AudioNodeEngine(aNode),
mStart(0.0), mBeginProcessing(0),
mStop(TRACK_TICKS_MAX),
mResampler(nullptr), mRemainingResamplerTail(0),
mBufferEnd(0),
mLoopStart(0), mLoopEnd(0),
mBufferSampleRate(0), mBufferPosition(0), mChannels(0),
mDopplerShift(1.0f),
mDestination(static_cast<AudioNodeStream*>(aDestination->Stream())),
mPlaybackRateTimeline(1.0f), mLoop(false)
{}
~AudioBufferSourceNodeEngine()
{
if (mResampler) {
speex_resampler_destroy(mResampler);
}
}
void SetSourceStream(AudioNodeStream* aSource)
{
mSource = aSource;
}
virtual void SetTimelineParameter(uint32_t aIndex,
const dom::AudioParamTimeline& aValue,
TrackRate aSampleRate) MOZ_OVERRIDE
{
switch (aIndex) {
case AudioBufferSourceNode::PLAYBACKRATE:
mPlaybackRateTimeline = aValue;
WebAudioUtils::ConvertAudioParamToTicks(mPlaybackRateTimeline, mSource, mDestination);
break;
default:
NS_ERROR("Bad AudioBufferSourceNodeEngine TimelineParameter");
}
}
virtual void SetStreamTimeParameter(uint32_t aIndex, TrackTicks aParam)
{
switch (aIndex) {
case AudioBufferSourceNode::STOP: mStop = aParam; break;
default:
NS_ERROR("Bad AudioBufferSourceNodeEngine StreamTimeParameter");
}
}
virtual void SetDoubleParameter(uint32_t aIndex, double aParam)
{
switch (aIndex) {
case AudioBufferSourceNode::START:
MOZ_ASSERT(!mStart, "Another START?");
mStart = mSource->TimeFromDestinationTime(mDestination, aParam) *
mSource->SampleRate();
// Round to nearest
mBeginProcessing = mStart + 0.5;
break;
case AudioBufferSourceNode::DOPPLERSHIFT:
mDopplerShift = aParam > 0 && aParam == aParam ? aParam : 1.0;
break;
default:
NS_ERROR("Bad AudioBufferSourceNodeEngine double parameter.");
};
}
virtual void SetInt32Parameter(uint32_t aIndex, int32_t aParam)
{
switch (aIndex) {
case AudioBufferSourceNode::SAMPLE_RATE: mBufferSampleRate = aParam; break;
case AudioBufferSourceNode::BUFFERSTART:
if (mBufferPosition == 0) {
mBufferPosition = aParam;
}
break;
case AudioBufferSourceNode::BUFFEREND: mBufferEnd = aParam; break;
case AudioBufferSourceNode::LOOP: mLoop = !!aParam; break;
case AudioBufferSourceNode::LOOPSTART: mLoopStart = aParam; break;
case AudioBufferSourceNode::LOOPEND: mLoopEnd = aParam; break;
default:
NS_ERROR("Bad AudioBufferSourceNodeEngine Int32Parameter");
}
}
virtual void SetBuffer(already_AddRefed<ThreadSharedFloatArrayBufferList> aBuffer)
{
mBuffer = aBuffer;
}
bool BegunResampling()
{
return mBeginProcessing == -TRACK_TICKS_MAX;
}
void UpdateResampler(int32_t aOutRate, uint32_t aChannels)
{
if (mResampler &&
(aChannels != mChannels ||
// If the resampler has begun, then it will have moved
// mBufferPosition to after the samples it has read, but it hasn't
// output its buffered samples. Keep using the resampler, even if
// the rates now match, so that this latent segment is output.
(aOutRate == mBufferSampleRate && !BegunResampling()))) {
speex_resampler_destroy(mResampler);
mResampler = nullptr;
mRemainingResamplerTail = 0;
mBeginProcessing = mStart + 0.5;
}
if (aOutRate == mBufferSampleRate && !mResampler) {
return;
}
if (!mResampler) {
mChannels = aChannels;
mResampler = speex_resampler_init(mChannels, mBufferSampleRate, aOutRate,
SPEEX_RESAMPLER_QUALITY_DEFAULT,
nullptr);
} else {
uint32_t currentOutSampleRate, currentInSampleRate;
speex_resampler_get_rate(mResampler, &currentInSampleRate,
&currentOutSampleRate);
if (currentOutSampleRate == static_cast<uint32_t>(aOutRate)) {
return;
}
speex_resampler_set_rate(mResampler, currentInSampleRate, aOutRate);
}
if (!BegunResampling()) {
// Low pass filter effects from the resampler mean that samples before
// the start time are influenced by resampling the buffer. The input
// latency indicates half the filter width.
int64_t inputLatency = speex_resampler_get_input_latency(mResampler);
uint32_t ratioNum, ratioDen;
speex_resampler_get_ratio(mResampler, &ratioNum, &ratioDen);
// The output subsample resolution supported in aligning the resampler
// is ratioNum. First round the start time to the nearest subsample.
int64_t subsample = mStart * ratioNum + 0.5;
// Now include the leading effects of the filter, and round *up* to the
// next whole tick, because there is no effect on samples outside the
// filter width.
mBeginProcessing =
(subsample - inputLatency * ratioDen + ratioNum - 1) / ratioNum;
}
}
// Borrow a full buffer of size WEBAUDIO_BLOCK_SIZE from the source buffer
// at offset aSourceOffset. This avoids copying memory.
void BorrowFromInputBuffer(AudioChunk* aOutput,
uint32_t aChannels)
{
aOutput->mDuration = WEBAUDIO_BLOCK_SIZE;
aOutput->mBuffer = mBuffer;
aOutput->mChannelData.SetLength(aChannels);
for (uint32_t i = 0; i < aChannels; ++i) {
aOutput->mChannelData[i] = mBuffer->GetData(i) + mBufferPosition;
}
aOutput->mVolume = 1.0f;
aOutput->mBufferFormat = AUDIO_FORMAT_FLOAT32;
}
// Copy aNumberOfFrames frames from the source buffer at offset aSourceOffset
// and put it at offset aBufferOffset in the destination buffer.
void CopyFromInputBuffer(AudioChunk* aOutput,
uint32_t aChannels,
uintptr_t aOffsetWithinBlock,
uint32_t aNumberOfFrames) {
for (uint32_t i = 0; i < aChannels; ++i) {
float* baseChannelData = static_cast<float*>(const_cast<void*>(aOutput->mChannelData[i]));
memcpy(baseChannelData + aOffsetWithinBlock,
mBuffer->GetData(i) + mBufferPosition,
aNumberOfFrames * sizeof(float));
}
}
// Resamples input data to an output buffer, according to |mBufferSampleRate| and
// the playbackRate.
// The number of frames consumed/produced depends on the amount of space
// remaining in both the input and output buffer, and the playback rate (that
// is, the ratio between the output samplerate and the input samplerate).
void CopyFromInputBufferWithResampling(AudioNodeStream* aStream,
AudioChunk* aOutput,
uint32_t aChannels,
uint32_t* aOffsetWithinBlock,
TrackTicks* aCurrentPosition,
int32_t aBufferMax) {
// TODO: adjust for mStop (see bug 913854 comment 9).
uint32_t availableInOutputBuffer =
WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock;
SpeexResamplerState* resampler = mResampler;
MOZ_ASSERT(aChannels > 0);
if (mBufferPosition < aBufferMax) {
uint32_t availableInInputBuffer = aBufferMax - mBufferPosition;
uint32_t ratioNum, ratioDen;
speex_resampler_get_ratio(resampler, &ratioNum, &ratioDen);
// Limit the number of input samples copied and possibly
// format-converted for resampling by estimating how many will be used.
// This may be a little small if still filling the resampler with
// initial data, but we'll get called again and it will work out.
uint32_t inputLimit = availableInOutputBuffer * ratioNum / ratioDen + 10;
if (!BegunResampling()) {
// First time the resampler is used.
uint32_t inputLatency = speex_resampler_get_input_latency(resampler);
inputLimit += inputLatency;
// If starting after mStart, then play from the beginning of the
// buffer, but correct for input latency. If starting before mStart,
// then align the resampler so that the time corresponding to the
// first input sample is mStart.
uint32_t skipFracNum = inputLatency * ratioDen;
double leadTicks = mStart - *aCurrentPosition;
if (leadTicks > 0.0) {
// Round to nearest output subsample supported by the resampler at
// these rates.
skipFracNum -= leadTicks * ratioNum + 0.5;
MOZ_ASSERT(skipFracNum < INT32_MAX, "mBeginProcessing is wrong?");
}
speex_resampler_set_skip_frac_num(resampler, skipFracNum);
mBeginProcessing = -TRACK_TICKS_MAX;
}
inputLimit = std::min(inputLimit, availableInInputBuffer);
for (uint32_t i = 0; true; ) {
uint32_t inSamples = inputLimit;
const float* inputData = mBuffer->GetData(i) + mBufferPosition;
uint32_t outSamples = availableInOutputBuffer;
float* outputData =
static_cast<float*>(const_cast<void*>(aOutput->mChannelData[i])) +
*aOffsetWithinBlock;
WebAudioUtils::SpeexResamplerProcess(resampler, i,
inputData, &inSamples,
outputData, &outSamples);
if (++i == aChannels) {
mBufferPosition += inSamples;
MOZ_ASSERT(mBufferPosition <= mBufferEnd || mLoop);
*aOffsetWithinBlock += outSamples;
*aCurrentPosition += outSamples;
if (inSamples == availableInInputBuffer && !mLoop) {
// We'll feed in enough zeros to empty out the resampler's memory.
// This handles the output latency as well as capturing the low
// pass effects of the resample filter.
mRemainingResamplerTail =
2 * speex_resampler_get_input_latency(resampler) - 1;
}
return;
}
}
} else {
for (uint32_t i = 0; true; ) {
uint32_t inSamples = mRemainingResamplerTail;
uint32_t outSamples = availableInOutputBuffer;
float* outputData =
static_cast<float*>(const_cast<void*>(aOutput->mChannelData[i])) +
*aOffsetWithinBlock;
// AudioDataValue* for aIn selects the function that does not try to
// copy and format-convert input data.
WebAudioUtils::SpeexResamplerProcess(resampler, i,
static_cast<AudioDataValue*>(nullptr), &inSamples,
outputData, &outSamples);
if (++i == aChannels) {
mRemainingResamplerTail -= inSamples;
MOZ_ASSERT(mRemainingResamplerTail >= 0);
*aOffsetWithinBlock += outSamples;
*aCurrentPosition += outSamples;
break;
}
}
}
}
/**
* 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)
{
MOZ_ASSERT(*aCurrentPosition < aMaxPos);
uint32_t numFrames =
std::min<TrackTicks>(WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock,
aMaxPos - *aCurrentPosition);
if (numFrames == WEBAUDIO_BLOCK_SIZE) {
aOutput->SetNull(numFrames);
} else {
if (*aOffsetWithinBlock == 0) {
AllocateAudioBlock(aChannels, aOutput);
}
WriteZeroesToAudioBlock(aOutput, *aOffsetWithinBlock, numFrames);
}
*aOffsetWithinBlock += numFrames;
*aCurrentPosition += numFrames;
}
/**
* 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);
}
}
}
int32_t ComputeFinalOutSampleRate(float aPlaybackRate)
{
// Make sure the playback rate and the doppler shift are something
// our resampler can work with.
int32_t rate = WebAudioUtils::
TruncateFloatToInt<int32_t>(mSource->SampleRate() /
(aPlaybackRate * mDopplerShift));
return rate ? rate : mBufferSampleRate;
}
void UpdateSampleRateIfNeeded(uint32_t aChannels)
{
float playbackRate;
if (mPlaybackRateTimeline.HasSimpleValue()) {
playbackRate = mPlaybackRateTimeline.GetValue();
} else {
playbackRate = mPlaybackRateTimeline.GetValueAtTime(mSource->GetCurrentPosition());
}
if (playbackRate <= 0 || playbackRate != playbackRate) {
playbackRate = 1.0f;
}
int32_t outRate = ComputeFinalOutSampleRate(playbackRate);
UpdateResampler(outRate, aChannels);
}
virtual void ProcessBlock(AudioNodeStream* aStream,
const AudioChunk& aInput,
AudioChunk* aOutput,
bool* aFinished)
{
if (!mBuffer || !mBufferEnd) {
aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
uint32_t channels = mBuffer->GetChannels();
if (!channels) {
aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
return;
}
// WebKit treats the playbackRate as a k-rate parameter in their code,
// despite the spec saying that it should be an a-rate parameter. We treat
// it as k-rate. Spec bug: https://www.w3.org/Bugs/Public/show_bug.cgi?id=21592
UpdateSampleRateIfNeeded(channels);
uint32_t written = 0;
TrackTicks streamPosition = aStream->GetCurrentPosition();
while (written < WEBAUDIO_BLOCK_SIZE) {
if (mStop != TRACK_TICKS_MAX &&
streamPosition >= mStop) {
FillWithZeroes(aOutput, channels, &written, &streamPosition, TRACK_TICKS_MAX);
continue;
}
if (streamPosition < mBeginProcessing) {
FillWithZeroes(aOutput, channels, &written, &streamPosition,
mBeginProcessing);
continue;
}
if (mLoop) {
// mLoopEnd can become less than mBufferPosition when a LOOPEND engine
// parameter is received after "loopend" is changed on the node or a
// new buffer with lower samplerate is set.
if (mBufferPosition >= mLoopEnd) {
mBufferPosition = mLoopStart;
}
CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mLoopEnd);
} else {
if (mBufferPosition < mBufferEnd || mRemainingResamplerTail) {
CopyFromBuffer(aStream, aOutput, channels, &written, &streamPosition, mBufferEnd);
} else {
FillWithZeroes(aOutput, channels, &written, &streamPosition, TRACK_TICKS_MAX);
}
}
}
// We've finished if we've gone past mStop, or if we're past mDuration when
// looping is disabled.
if (streamPosition >= mStop ||
(!mLoop && mBufferPosition >= mBufferEnd && !mRemainingResamplerTail)) {
*aFinished = true;
}
}
virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const MOZ_OVERRIDE
{
// Not owned:
// - mBuffer - shared w/ AudioNode
// - mPlaybackRateTimeline - shared w/ AudioNode
size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
// NB: We need to modify speex if we want the full memory picture, internal
// fields that need measuring noted below.
// - mResampler->mem
// - mResampler->sinc_table
// - mResampler->last_sample
// - mResampler->magic_samples
// - mResampler->samp_frac_num
amount += aMallocSizeOf(mResampler);
return amount;
}
virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const MOZ_OVERRIDE
{
return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
}
double mStart; // including the fractional position between ticks
// Low pass filter effects from the resampler mean that samples before the
// start time are influenced by resampling the buffer. mBeginProcessing
// includes the extent of this filter. The special value of -TRACK_TICKS_MAX
// indicates that the resampler has begun processing.
TrackTicks mBeginProcessing;
TrackTicks mStop;
nsRefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
SpeexResamplerState* mResampler;
// mRemainingResamplerTail, like mBufferPosition, and
// mBufferEnd, is measured in input buffer samples.
int mRemainingResamplerTail;
int32_t mBufferEnd;
int32_t mLoopStart;
int32_t mLoopEnd;
int32_t mBufferSampleRate;
int32_t mBufferPosition;
uint32_t mChannels;
float mDopplerShift;
AudioNodeStream* mDestination;
AudioNodeStream* mSource;
AudioParamTimeline mPlaybackRateTimeline;
bool mLoop;
};
AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* aContext)
: AudioNode(aContext,
2,
ChannelCountMode::Max,
ChannelInterpretation::Speakers)
, mLoopStart(0.0)
, mLoopEnd(0.0)
// mOffset and mDuration are initialized in Start().
, mPlaybackRate(new AudioParam(MOZ_THIS_IN_INITIALIZER_LIST(),
SendPlaybackRateToStream, 1.0f))
, mLoop(false)
, mStartCalled(false)
, mStopped(false)
{
AudioBufferSourceNodeEngine* engine = new AudioBufferSourceNodeEngine(this, aContext->Destination());
mStream = aContext->Graph()->CreateAudioNodeStream(engine, MediaStreamGraph::SOURCE_STREAM);
engine->SetSourceStream(static_cast<AudioNodeStream*>(mStream.get()));
mStream->AddMainThreadListener(this);
}
AudioBufferSourceNode::~AudioBufferSourceNode()
{
if (Context()) {
Context()->UnregisterAudioBufferSourceNode(this);
}
}
size_t
AudioBufferSourceNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
{
size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
if (mBuffer) {
amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
}
amount += mPlaybackRate->SizeOfIncludingThis(aMallocSizeOf);
return amount;
}
size_t
AudioBufferSourceNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
{
return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
}
JSObject*
AudioBufferSourceNode::WrapObject(JSContext* aCx)
{
return AudioBufferSourceNodeBinding::Wrap(aCx, this);
}
void
AudioBufferSourceNode::Start(double aWhen, double aOffset,
const Optional<double>& aDuration, ErrorResult& aRv)
{
if (!WebAudioUtils::IsTimeValid(aWhen) ||
(aDuration.WasPassed() && !WebAudioUtils::IsTimeValid(aDuration.Value()))) {
aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
return;
}
if (mStartCalled) {
aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
return;
}
mStartCalled = true;
AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
if (!ns) {
// Nothing to play, or we're already dead for some reason
return;
}
// Remember our arguments so that we can use them when we get a new buffer.
mOffset = aOffset;
mDuration = aDuration.WasPassed() ? aDuration.Value()
: std::numeric_limits<double>::min();
// We can't send these parameters without a buffer because we don't know the
// buffer's sample rate or length.
if (mBuffer) {
SendOffsetAndDurationParametersToStream(ns);
}
// Don't set parameter unnecessarily
if (aWhen > 0.0) {
ns->SetDoubleParameter(START, mContext->DOMTimeToStreamTime(aWhen));
}
}
void
AudioBufferSourceNode::SendBufferParameterToStream(JSContext* aCx)
{
AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
MOZ_ASSERT(ns, "Why don't we have a stream here?");
if (mBuffer) {
float rate = mBuffer->SampleRate();
nsRefPtr<ThreadSharedFloatArrayBufferList> data =
mBuffer->GetThreadSharedChannelsForRate(aCx);
ns->SetBuffer(data.forget());
ns->SetInt32Parameter(SAMPLE_RATE, rate);
if (mStartCalled) {
SendOffsetAndDurationParametersToStream(ns);
}
} else {
ns->SetBuffer(nullptr);
MarkInactive();
}
}
void
AudioBufferSourceNode::SendOffsetAndDurationParametersToStream(AudioNodeStream* aStream)
{
NS_ASSERTION(mBuffer && mStartCalled,
"Only call this when we have a buffer and start() has been called");
float rate = mBuffer->SampleRate();
int32_t bufferEnd = mBuffer->Length();
int32_t offsetSamples = std::max(0, NS_lround(mOffset * rate));
// Don't set parameter unnecessarily
if (offsetSamples > 0) {
aStream->SetInt32Parameter(BUFFERSTART, offsetSamples);
}
if (mDuration != std::numeric_limits<double>::min()) {
bufferEnd = std::min(bufferEnd,
offsetSamples + NS_lround(mDuration * rate));
}
aStream->SetInt32Parameter(BUFFEREND, bufferEnd);
MarkActive();
}
void
AudioBufferSourceNode::Stop(double aWhen, ErrorResult& aRv)
{
if (!WebAudioUtils::IsTimeValid(aWhen)) {
aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
return;
}
if (!mStartCalled) {
aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
return;
}
AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
if (!ns || !Context()) {
// We've already stopped and had our stream shut down
return;
}
ns->SetStreamTimeParameter(STOP, Context(), std::max(0.0, aWhen));
}
void
AudioBufferSourceNode::NotifyMainThreadStateChanged()
{
if (mStream->IsFinished()) {
class EndedEventDispatcher : public nsRunnable
{
public:
explicit EndedEventDispatcher(AudioBufferSourceNode* aNode)
: mNode(aNode) {}
NS_IMETHODIMP Run()
{
// If it's not safe to run scripts right now, schedule this to run later
if (!nsContentUtils::IsSafeToRunScript()) {
nsContentUtils::AddScriptRunner(this);
return NS_OK;
}
mNode->DispatchTrustedEvent(NS_LITERAL_STRING("ended"));
return NS_OK;
}
private:
nsRefPtr<AudioBufferSourceNode> mNode;
};
if (!mStopped) {
// Only dispatch the ended event once
NS_DispatchToMainThread(new EndedEventDispatcher(this));
mStopped = true;
}
// Drop the playing reference
// Warning: The below line might delete this.
MarkInactive();
}
}
void
AudioBufferSourceNode::SendPlaybackRateToStream(AudioNode* aNode)
{
AudioBufferSourceNode* This = static_cast<AudioBufferSourceNode*>(aNode);
SendTimelineParameterToStream(This, PLAYBACKRATE, *This->mPlaybackRate);
}
void
AudioBufferSourceNode::SendDopplerShiftToStream(double aDopplerShift)
{
SendDoubleParameterToStream(DOPPLERSHIFT, aDopplerShift);
}
void
AudioBufferSourceNode::SendLoopParametersToStream()
{
// Don't compute and set the loop parameters unnecessarily
if (mLoop && mBuffer) {
float rate = mBuffer->SampleRate();
double length = (double(mBuffer->Length()) / mBuffer->SampleRate());
double actualLoopStart, actualLoopEnd;
if (mLoopStart >= 0.0 && mLoopEnd > 0.0 &&
mLoopStart < mLoopEnd) {
MOZ_ASSERT(mLoopStart != 0.0 || mLoopEnd != 0.0);
actualLoopStart = (mLoopStart > length) ? 0.0 : mLoopStart;
actualLoopEnd = std::min(mLoopEnd, length);
} else {
actualLoopStart = 0.0;
actualLoopEnd = length;
}
int32_t loopStartTicks = NS_lround(actualLoopStart * rate);
int32_t loopEndTicks = NS_lround(actualLoopEnd * rate);
if (loopStartTicks < loopEndTicks) {
SendInt32ParameterToStream(LOOPSTART, loopStartTicks);
SendInt32ParameterToStream(LOOPEND, loopEndTicks);
SendInt32ParameterToStream(LOOP, 1);
} else {
// Be explicit about looping not happening if the offsets make
// looping impossible.
SendInt32ParameterToStream(LOOP, 0);
}
} else if (!mLoop) {
SendInt32ParameterToStream(LOOP, 0);
}
}
}
}
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