dom/media/webaudio/AudioBufferSourceNode.cpp
author Andreas Pehrson <apehrson@mozilla.com>
Fri, 23 Nov 2018 15:00:26 +0000
changeset 504258 befba547fb5850fd62d4e31784aa4b5198404500
parent 503941 3d997ec4174de4296a6bd30641d42309999703b0
child 509464 6d105dcaa3dfd622e0dbfda3d0405b69d08557c1
permissions -rw-r--r--
Bug 1423241 - Fix MediaStreamTrackListener::NotifyEnded. r=padenot Without this, NotifyEnded() happens before the track has been played out, at the time it's marked ended by its producer. This change will actually make us wait until the last chunk has been played out and then notify listeners. Differential Revision: https://phabricator.services.mozilla.com/D12269

/* -*- 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 "nsDebug.h"
#include "mozilla/dom/AudioBufferSourceNodeBinding.h"
#include "mozilla/dom/AudioParam.h"
#include "mozilla/FloatingPoint.h"
#include "nsContentUtils.h"
#include "nsMathUtils.h"
#include "AlignmentUtils.h"
#include "AudioNodeEngine.h"
#include "AudioNodeStream.h"
#include "AudioDestinationNode.h"
#include "AudioParamTimeline.h"
#include <limits>
#include <algorithm>

namespace mozilla {
namespace dom {

NS_IMPL_CYCLE_COLLECTION_INHERITED(AudioBufferSourceNode,
                                   AudioScheduledSourceNode, mBuffer,
                                   mPlaybackRate, mDetune)

NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(AudioBufferSourceNode)
NS_INTERFACE_MAP_END_INHERITING(AudioScheduledSourceNode)

NS_IMPL_ADDREF_INHERITED(AudioBufferSourceNode, AudioScheduledSourceNode)
NS_IMPL_RELEASE_INHERITED(AudioBufferSourceNode, AudioScheduledSourceNode)

/**
 * 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 final : public AudioNodeEngine {
 public:
  AudioBufferSourceNodeEngine(AudioNode* aNode,
                              AudioDestinationNode* aDestination)
      : AudioNodeEngine(aNode),
        mStart(0.0),
        mBeginProcessing(0),
        mStop(STREAM_TIME_MAX),
        mResampler(nullptr),
        mRemainingResamplerTail(0),
        mBufferEnd(0),
        mLoopStart(0),
        mLoopEnd(0),
        mBufferPosition(0),
        mBufferSampleRate(0),
        // mResamplerOutRate is initialized in UpdateResampler().
        mChannels(0),
        mDopplerShift(1.0f),
        mDestination(aDestination->Stream()),
        mPlaybackRateTimeline(1.0f),
        mDetuneTimeline(0.0f),
        mLoop(false) {}

  ~AudioBufferSourceNodeEngine() {
    if (mResampler) {
      speex_resampler_destroy(mResampler);
    }
  }

  void SetSourceStream(AudioNodeStream* aSource) { mSource = aSource; }

  void RecvTimelineEvent(uint32_t aIndex,
                         dom::AudioTimelineEvent& aEvent) override {
    MOZ_ASSERT(mDestination);
    WebAudioUtils::ConvertAudioTimelineEventToTicks(aEvent, mDestination);

    switch (aIndex) {
      case AudioBufferSourceNode::PLAYBACKRATE:
        mPlaybackRateTimeline.InsertEvent<int64_t>(aEvent);
        break;
      case AudioBufferSourceNode::DETUNE:
        mDetuneTimeline.InsertEvent<int64_t>(aEvent);
        break;
      default:
        NS_ERROR("Bad AudioBufferSourceNodeEngine TimelineParameter");
    }
  }
  void SetStreamTimeParameter(uint32_t aIndex, StreamTime aParam) override {
    switch (aIndex) {
      case AudioBufferSourceNode::STOP:
        mStop = aParam;
        break;
      default:
        NS_ERROR("Bad AudioBufferSourceNodeEngine StreamTimeParameter");
    }
  }
  void SetDoubleParameter(uint32_t aIndex, double aParam) override {
    switch (aIndex) {
      case AudioBufferSourceNode::START:
        MOZ_ASSERT(!mStart, "Another START?");
        mStart = aParam * mDestination->SampleRate();
        // Round to nearest
        mBeginProcessing = mStart + 0.5;
        break;
      case AudioBufferSourceNode::DOPPLERSHIFT:
        mDopplerShift = (aParam <= 0 || mozilla::IsNaN(aParam)) ? 1.0 : aParam;
        break;
      default:
        NS_ERROR("Bad AudioBufferSourceNodeEngine double parameter.");
    };
  }
  void SetInt32Parameter(uint32_t aIndex, int32_t aParam) override {
    switch (aIndex) {
      case AudioBufferSourceNode::SAMPLE_RATE:
        MOZ_ASSERT(aParam > 0);
        mBufferSampleRate = aParam;
        mSource->SetActive();
        break;
      case AudioBufferSourceNode::BUFFERSTART:
        MOZ_ASSERT(aParam >= 0);
        if (mBufferPosition == 0) {
          mBufferPosition = aParam;
        }
        break;
      case AudioBufferSourceNode::BUFFEREND:
        MOZ_ASSERT(aParam >= 0);
        mBufferEnd = aParam;
        break;
      case AudioBufferSourceNode::LOOP:
        mLoop = !!aParam;
        break;
      case AudioBufferSourceNode::LOOPSTART:
        MOZ_ASSERT(aParam >= 0);
        mLoopStart = aParam;
        break;
      case AudioBufferSourceNode::LOOPEND:
        MOZ_ASSERT(aParam >= 0);
        mLoopEnd = aParam;
        break;
      default:
        NS_ERROR("Bad AudioBufferSourceNodeEngine Int32Parameter");
    }
  }
  void SetBuffer(AudioChunk&& aBuffer) override { mBuffer = aBuffer; }

  bool BegunResampling() { return mBeginProcessing == -STREAM_TIME_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 (aChannels == 0 || (aOutRate == mBufferSampleRate && !mResampler)) {
      mResamplerOutRate = aOutRate;
      return;
    }

    if (!mResampler) {
      mChannels = aChannels;
      mResampler = speex_resampler_init(mChannels, mBufferSampleRate, aOutRate,
                                        SPEEX_RESAMPLER_QUALITY_MIN, nullptr);
    } else {
      if (mResamplerOutRate == aOutRate) {
        return;
      }
      if (speex_resampler_set_rate(mResampler, mBufferSampleRate, aOutRate) !=
          RESAMPLER_ERR_SUCCESS) {
        NS_ASSERTION(false, "speex_resampler_set_rate failed");
        return;
      }
    }

    mResamplerOutRate = 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(AudioBlock* aOutput, uint32_t aChannels) {
    aOutput->SetBuffer(mBuffer.mBuffer);
    aOutput->mChannelData.SetLength(aChannels);
    for (uint32_t i = 0; i < aChannels; ++i) {
      aOutput->mChannelData[i] =
          mBuffer.ChannelData<float>()[i] + mBufferPosition;
    }
    aOutput->mVolume = mBuffer.mVolume;
    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.
  template <typename T>
  void CopyFromInputBuffer(AudioBlock* aOutput, uint32_t aChannels,
                           uintptr_t aOffsetWithinBlock,
                           uint32_t aNumberOfFrames) {
    MOZ_ASSERT(mBuffer.mVolume == 1.0f);
    for (uint32_t i = 0; i < aChannels; ++i) {
      float* baseChannelData = aOutput->ChannelFloatsForWrite(i);
      ConvertAudioSamples(mBuffer.ChannelData<T>()[i] + mBufferPosition,
                          baseChannelData + aOffsetWithinBlock,
                          aNumberOfFrames);
    }
  }

  // Resamples input data to an output buffer, according to |mBufferSampleRate|
  // and the playbackRate/detune. 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(AudioBlock* aOutput,
                                         uint32_t aChannels,
                                         uint32_t* aOffsetWithinBlock,
                                         uint32_t aAvailableInOutput,
                                         StreamTime* aCurrentPosition,
                                         uint32_t aBufferMax) {
    if (*aOffsetWithinBlock == 0) {
      aOutput->AllocateChannels(aChannels);
    }
    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 = aAvailableInOutput * 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.
        int64_t skipFracNum = static_cast<int64_t>(inputLatency) * ratioDen;
        double leadTicks = mStart - *aCurrentPosition;
        if (leadTicks > 0.0) {
          // Round to nearest output subsample supported by the resampler at
          // these rates.
          int64_t leadSubsamples = leadTicks * ratioNum + 0.5;
          MOZ_ASSERT(leadSubsamples <= skipFracNum,
                     "mBeginProcessing is wrong?");
          skipFracNum -= leadSubsamples;
        }
        speex_resampler_set_skip_frac_num(
            resampler, std::min<int64_t>(skipFracNum, UINT32_MAX));

        mBeginProcessing = -STREAM_TIME_MAX;
      }
      inputLimit = std::min(inputLimit, availableInInputBuffer);

      MOZ_ASSERT(mBuffer.mVolume == 1.0f);
      for (uint32_t i = 0; true;) {
        uint32_t inSamples = inputLimit;

        uint32_t outSamples = aAvailableInOutput;
        float* outputData =
            aOutput->ChannelFloatsForWrite(i) + *aOffsetWithinBlock;

        if (mBuffer.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
          const float* inputData =
              mBuffer.ChannelData<float>()[i] + mBufferPosition;
          WebAudioUtils::SpeexResamplerProcess(
              resampler, i, inputData, &inSamples, outputData, &outSamples);
        } else {
          MOZ_ASSERT(mBuffer.mBufferFormat == AUDIO_FORMAT_S16);
          const int16_t* inputData =
              mBuffer.ChannelData<int16_t>()[i] + mBufferPosition;
          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 = aAvailableInOutput;
        float* outputData =
            aOutput->ChannelFloatsForWrite(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) {
          MOZ_ASSERT(inSamples <= mRemainingResamplerTail);
          mRemainingResamplerTail -= inSamples;
          *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(AudioBlock* aOutput, uint32_t aChannels,
                      uint32_t* aOffsetWithinBlock,
                      StreamTime* aCurrentPosition, StreamTime aMaxPos) {
    MOZ_ASSERT(*aCurrentPosition < aMaxPos);
    uint32_t numFrames = std::min<StreamTime>(
        WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock, aMaxPos - *aCurrentPosition);
    if (numFrames == WEBAUDIO_BLOCK_SIZE || !aChannels) {
      aOutput->SetNull(numFrames);
    } else {
      if (*aOffsetWithinBlock == 0) {
        aOutput->AllocateChannels(aChannels);
      }
      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(AudioBlock* aOutput, uint32_t aChannels,
                      uint32_t* aOffsetWithinBlock,
                      StreamTime* aCurrentPosition, uint32_t aBufferMax) {
    MOZ_ASSERT(*aCurrentPosition < mStop);
    uint32_t availableInOutput = std::min<StreamTime>(
        WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock, mStop - *aCurrentPosition);
    if (mResampler) {
      CopyFromInputBufferWithResampling(aOutput, aChannels, aOffsetWithinBlock,
                                        availableInOutput, aCurrentPosition,
                                        aBufferMax);
      return;
    }

    if (aChannels == 0) {
      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
      // There is no attempt here to limit advance so that mBufferPosition is
      // limited to aBufferMax.  The only observable affect of skipping the
      // check would be in the precise timing of the ended event if the loop
      // attribute is reset after playback has looped.
      *aOffsetWithinBlock += availableInOutput;
      *aCurrentPosition += availableInOutput;
      // Rounding at the start and end of the period means that fractional
      // increments essentially accumulate if outRate remains constant.  If
      // outRate is varying, then accumulation happens on average but not
      // precisely.
      TrackTicks start =
          *aCurrentPosition * mBufferSampleRate / mResamplerOutRate;
      TrackTicks end = (*aCurrentPosition + availableInOutput) *
                       mBufferSampleRate / mResamplerOutRate;
      mBufferPosition += end - start;
      return;
    }

    uint32_t numFrames =
        std::min(aBufferMax - mBufferPosition, availableInOutput);

    bool shouldBorrow = false;
    if (numFrames == WEBAUDIO_BLOCK_SIZE &&
        mBuffer.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
      shouldBorrow = true;
      for (uint32_t i = 0; i < aChannels; ++i) {
        if (!IS_ALIGNED16(mBuffer.ChannelData<float>()[i] + mBufferPosition)) {
          shouldBorrow = false;
          break;
        }
      }
    }
    MOZ_ASSERT(mBufferPosition < aBufferMax);
    if (shouldBorrow) {
      BorrowFromInputBuffer(aOutput, aChannels);
    } else {
      if (*aOffsetWithinBlock == 0) {
        aOutput->AllocateChannels(aChannels);
      }
      if (mBuffer.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
        CopyFromInputBuffer<float>(aOutput, aChannels, *aOffsetWithinBlock,
                                   numFrames);
      } else {
        MOZ_ASSERT(mBuffer.mBufferFormat == AUDIO_FORMAT_S16);
        CopyFromInputBuffer<int16_t>(aOutput, aChannels, *aOffsetWithinBlock,
                                     numFrames);
      }
    }
    *aOffsetWithinBlock += numFrames;
    *aCurrentPosition += numFrames;
    mBufferPosition += numFrames;
  }

  int32_t ComputeFinalOutSampleRate(float aPlaybackRate, float aDetune) {
    float computedPlaybackRate = aPlaybackRate * pow(2, aDetune / 1200.f);
    // 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() / (computedPlaybackRate * mDopplerShift));
    return rate ? rate : mBufferSampleRate;
  }

  void UpdateSampleRateIfNeeded(uint32_t aChannels,
                                StreamTime aStreamPosition) {
    float playbackRate;
    float detune;

    if (mPlaybackRateTimeline.HasSimpleValue()) {
      playbackRate = mPlaybackRateTimeline.GetValue();
    } else {
      playbackRate = mPlaybackRateTimeline.GetValueAtTime(aStreamPosition);
    }
    if (mDetuneTimeline.HasSimpleValue()) {
      detune = mDetuneTimeline.GetValue();
    } else {
      detune = mDetuneTimeline.GetValueAtTime(aStreamPosition);
    }
    if (playbackRate <= 0 || mozilla::IsNaN(playbackRate)) {
      playbackRate = 1.0f;
    }

    detune = std::min(std::max(-1200.f, detune), 1200.f);

    int32_t outRate = ComputeFinalOutSampleRate(playbackRate, detune);
    UpdateResampler(outRate, aChannels);
  }

  void ProcessBlock(AudioNodeStream* aStream, GraphTime aFrom,
                    const AudioBlock& aInput, AudioBlock* aOutput,
                    bool* aFinished) override {
    if (mBufferSampleRate == 0) {
      // start() has not yet been called or no buffer has yet been set
      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
      return;
    }

    StreamTime streamPosition = mDestination->GraphTimeToStreamTime(aFrom);
    uint32_t channels = mBuffer.ChannelCount();

    UpdateSampleRateIfNeeded(channels, streamPosition);

    uint32_t written = 0;
    while (written < WEBAUDIO_BLOCK_SIZE) {
      if (mStop != STREAM_TIME_MAX && streamPosition >= mStop) {
        FillWithZeroes(aOutput, channels, &written, &streamPosition,
                       STREAM_TIME_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(aOutput, channels, &written, &streamPosition, mLoopEnd);
      } else {
        if (mBufferPosition < mBufferEnd || mRemainingResamplerTail) {
          CopyFromBuffer(aOutput, channels, &written, &streamPosition,
                         mBufferEnd);
        } else {
          FillWithZeroes(aOutput, channels, &written, &streamPosition,
                         STREAM_TIME_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;
    }
  }

  bool IsActive() const override {
    // Whether buffer has been set and start() has been called.
    return mBufferSampleRate != 0;
  }

  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
    // Not owned:
    // - mBuffer - shared w/ AudioNode
    // - mPlaybackRateTimeline - shared w/ AudioNode
    // - mDetuneTimeline - 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;
  }

  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const 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 -STREAM_TIME_MAX
  // indicates that the resampler has begun processing.
  StreamTime mBeginProcessing;
  StreamTime mStop;
  AudioChunk mBuffer;
  SpeexResamplerState* mResampler;
  // mRemainingResamplerTail, like mBufferPosition, and
  // mBufferEnd, is measured in input buffer samples.
  uint32_t mRemainingResamplerTail;
  uint32_t mBufferEnd;
  uint32_t mLoopStart;
  uint32_t mLoopEnd;
  uint32_t mBufferPosition;
  int32_t mBufferSampleRate;
  int32_t mResamplerOutRate;
  uint32_t mChannels;
  float mDopplerShift;
  RefPtr<AudioNodeStream> mDestination;

  // mSource deletes the engine in its destructor.
  AudioNodeStream* MOZ_NON_OWNING_REF mSource;
  AudioParamTimeline mPlaybackRateTimeline;
  AudioParamTimeline mDetuneTimeline;
  bool mLoop;
};

AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* aContext)
    : AudioScheduledSourceNode(aContext, 2, ChannelCountMode::Max,
                               ChannelInterpretation::Speakers),
      mLoopStart(0.0),
      mLoopEnd(0.0)
      // mOffset and mDuration are initialized in Start().
      ,
      mPlaybackRate(new AudioParam(this, PLAYBACKRATE, "playbackRate", 1.0f)),
      mDetune(new AudioParam(this, DETUNE, "detune", 0.0f)),
      mLoop(false),
      mStartCalled(false) {
  AudioBufferSourceNodeEngine* engine =
      new AudioBufferSourceNodeEngine(this, aContext->Destination());
  mStream = AudioNodeStream::Create(aContext, engine,
                                    AudioNodeStream::NEED_MAIN_THREAD_FINISHED,
                                    aContext->Graph());
  engine->SetSourceStream(mStream);
  mStream->AddMainThreadListener(this);
}

/* static */ already_AddRefed<AudioBufferSourceNode>
AudioBufferSourceNode::Create(JSContext* aCx, AudioContext& aAudioContext,
                              const AudioBufferSourceOptions& aOptions,
                              ErrorResult& aRv) {
  if (aAudioContext.CheckClosed(aRv)) {
    return nullptr;
  }

  RefPtr<AudioBufferSourceNode> audioNode =
      new AudioBufferSourceNode(&aAudioContext);

  if (aOptions.mBuffer.WasPassed()) {
    MOZ_ASSERT(aCx);
    audioNode->SetBuffer(aCx, aOptions.mBuffer.Value());
  }

  audioNode->Detune()->SetValue(aOptions.mDetune);
  audioNode->SetLoop(aOptions.mLoop);
  audioNode->SetLoopEnd(aOptions.mLoopEnd);
  audioNode->SetLoopStart(aOptions.mLoopStart);
  audioNode->PlaybackRate()->SetValue(aOptions.mPlaybackRate);

  return audioNode.forget();
}
void AudioBufferSourceNode::DestroyMediaStream() {
  bool hadStream = mStream;
  if (hadStream) {
    mStream->RemoveMainThreadListener(this);
  }
  AudioNode::DestroyMediaStream();
}

size_t AudioBufferSourceNode::SizeOfExcludingThis(
    MallocSizeOf aMallocSizeOf) const {
  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);

  /* mBuffer can be shared and is accounted for separately. */

  amount += mPlaybackRate->SizeOfIncludingThis(aMallocSizeOf);
  amount += mDetune->SizeOfIncludingThis(aMallocSizeOf);
  return amount;
}

size_t AudioBufferSourceNode::SizeOfIncludingThis(
    MallocSizeOf aMallocSizeOf) const {
  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
}

JSObject* AudioBufferSourceNode::WrapObject(JSContext* aCx,
                                            JS::Handle<JSObject*> aGivenProto) {
  return AudioBufferSourceNode_Binding::Wrap(aCx, this, aGivenProto);
}

void AudioBufferSourceNode::Start(double aWhen, double aOffset,
                                  const Optional<double>& aDuration,
                                  ErrorResult& aRv) {
  if (!WebAudioUtils::IsTimeValid(aWhen)) {
    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>(
        NS_LITERAL_STRING("start time"));
    return;
  }
  if (aOffset < 0) {
    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>(NS_LITERAL_STRING("offset"));
    return;
  }
  if (aDuration.WasPassed() && !WebAudioUtils::IsTimeValid(aDuration.Value())) {
    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>(NS_LITERAL_STRING("duration"));
    return;
  }

  if (mStartCalled) {
    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
    return;
  }
  mStartCalled = true;

  AudioNodeStream* ns = mStream;
  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();

  WEB_AUDIO_API_LOG("%f: %s %u Start(%f, %g, %g)", Context()->CurrentTime(),
                    NodeType(), Id(), aWhen, aOffset, mDuration);

  // 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, aWhen);
  }

  Context()->NotifyScheduledSourceNodeStarted();
}

void AudioBufferSourceNode::Start(double aWhen, ErrorResult& aRv) {
  Start(aWhen, 0 /* offset */, Optional<double>(), aRv);
}

void AudioBufferSourceNode::SendBufferParameterToStream(JSContext* aCx) {
  AudioNodeStream* ns = mStream;
  if (!ns) {
    return;
  }

  if (mBuffer) {
    AudioChunk data = mBuffer->GetThreadSharedChannelsForRate(aCx);
    ns->SetBuffer(std::move(data));

    if (mStartCalled) {
      SendOffsetAndDurationParametersToStream(ns);
    }
  } else {
    ns->SetInt32Parameter(BUFFEREND, 0);
    ns->SetBuffer(AudioChunk());

    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();
  aStream->SetInt32Parameter(SAMPLE_RATE, rate);

  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()) {
    MOZ_ASSERT(mDuration >= 0.0);  // provided by Start()
    MOZ_ASSERT(rate >= 0.0f);      // provided by AudioBuffer::Create()
    static_assert(std::numeric_limits<double>::digits >=
                      std::numeric_limits<decltype(bufferEnd)>::digits,
                  "bufferEnd should be represented exactly by double");
    // + 0.5 rounds mDuration to nearest sample when assigned to bufferEnd.
    bufferEnd =
        std::min<double>(bufferEnd, offsetSamples + mDuration * rate + 0.5);
  }
  aStream->SetInt32Parameter(BUFFEREND, bufferEnd);

  MarkActive();
}

void AudioBufferSourceNode::Stop(double aWhen, ErrorResult& aRv) {
  if (!WebAudioUtils::IsTimeValid(aWhen)) {
    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>(NS_LITERAL_STRING("stop time"));
    return;
  }

  if (!mStartCalled) {
    aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
    return;
  }

  WEB_AUDIO_API_LOG("%f: %s %u Stop(%f)", Context()->CurrentTime(), NodeType(),
                    Id(), aWhen);

  AudioNodeStream* ns = mStream;
  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::NotifyMainThreadStreamFinished() {
  MOZ_ASSERT(mStream->IsFinished());

  class EndedEventDispatcher final : public Runnable {
   public:
    explicit EndedEventDispatcher(AudioBufferSourceNode* aNode)
        : mozilla::Runnable("EndedEventDispatcher"), mNode(aNode) {}
    NS_IMETHOD Run() override {
      // 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"));
      // Release stream resources.
      mNode->DestroyMediaStream();
      return NS_OK;
    }

   private:
    RefPtr<AudioBufferSourceNode> mNode;
  };

  Context()->Dispatch(do_AddRef(new EndedEventDispatcher(this)));

  // Drop the playing reference
  // Warning: The below line might delete this.
  MarkInactive();
}

void AudioBufferSourceNode::SendDopplerShiftToStream(double aDopplerShift) {
  MOZ_ASSERT(mStream, "Should have disconnected panner if no stream");
  SendDoubleParameterToStream(DOPPLERSHIFT, aDopplerShift);
}

void AudioBufferSourceNode::SendLoopParametersToStream() {
  if (!mStream) {
    return;
  }
  // 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 {
    SendInt32ParameterToStream(LOOP, 0);
  }
}

}  // namespace dom
}  // namespace mozilla