The Tor Browser: content/media/webaudio/AudioBufferSourceNode.cpp@b8a032363ba2 (annotated)

content/media/webaudio/AudioBufferSourceNode.cpp@b8a032363ba2 (annotated)

content/media/webaudio/AudioBufferSourceNode.cpp

Thu, 22 Jan 2015 13:21:57 +0100

author: Michael Schloh von Bennewitz <michael@schloh.com>
date: Thu, 22 Jan 2015 13:21:57 +0100
branch: TOR_BUG_9701
changeset 15: b8a032363ba2
permissions: -rw-r--r--

Incorporate requested changes from Mozilla in review:
https://bugzilla.mozilla.org/show_bug.cgi?id=1123480#c6

 /* -*- 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 "speex/speex_resampler.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 =
 * 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,
 ,
               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);
   }
 }
 }
 }

The Tor Browser / annotate

content/media/webaudio/AudioBufferSourceNode.cpp@b8a032363ba2 (annotated)

content/media/webaudio/AudioBufferSourceNode.cpp