Add m-esr52 at 52.6.0

1 files changed, 853 insertions, 0 deletions

diff --git a/dom/media/webaudio/AudioBufferSourceNode.cpp b/dom/media/webaudio/AudioBufferSourceNode.cpp
new file mode 100644
index 000000000..51b6bab4a
--- /dev/null
+++ b/dom/media/webaudio/AudioBufferSourceNode.cpp
@@ -0,0 +1,853 @@
+/* -*- 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, AudioNode, mBuffer, mPlaybackRate, mDetune)
+
+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 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(already_AddRefed<ThreadSharedFloatArrayBufferList> 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);
+    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(AudioBlock* aOutput,
+                           uint32_t aChannels,
+                           uintptr_t aOffsetWithinBlock,
+                           uint32_t aNumberOfFrames) {
+    for (uint32_t i = 0; i < aChannels; ++i) {
+      float* baseChannelData = aOutput->ChannelFloatsForWrite(i);
+      memcpy(baseChannelData + aOffsetWithinBlock,
+             mBuffer->GetData(i) + mBufferPosition,
+             aNumberOfFrames * sizeof(float));
+    }
+  }
+
+  // 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);
+
+      for (uint32_t i = 0; true; ) {
+        uint32_t inSamples = inputLimit;
+        const float* inputData = mBuffer->GetData(i) + mBufferPosition;
+
+        uint32_t outSamples = aAvailableInOutput;
+        float* outputData =
+          aOutput->ChannelFloatsForWrite(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 = 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 inputBufferAligned = true;
+    for (uint32_t i = 0; i < aChannels; ++i) {
+      if (!IS_ALIGNED16(mBuffer->GetData(i) + mBufferPosition)) {
+        inputBufferAligned = false;
+      }
+    }
+
+    if (numFrames == WEBAUDIO_BLOCK_SIZE && inputBufferAligned) {
+      MOZ_ASSERT(mBufferPosition < aBufferMax);
+      BorrowFromInputBuffer(aOutput, aChannels);
+    } else {
+      if (*aOffsetWithinBlock == 0) {
+        aOutput->AllocateChannels(aChannels);
+      }
+      MOZ_ASSERT(mBufferPosition < aBufferMax);
+      CopyFromInputBuffer(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 ? mBuffer->GetChannels() : 0;
+
+    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;
+  RefPtr<ThreadSharedFloatArrayBufferList> 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;
+  AudioNodeStream* mDestination;
+  AudioNodeStream* mSource;
+  AudioParamTimeline mPlaybackRateTimeline;
+  AudioParamTimeline mDetuneTimeline;
+  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(this, PLAYBACKRATE, 1.0f, "playbackRate"))
+  , mDetune(new AudioParam(this, DETUNE, 0.0f, "detune"))
+  , 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);
+}
+
+AudioBufferSourceNode::~AudioBufferSourceNode()
+{
+}
+
+void
+AudioBufferSourceNode::DestroyMediaStream()
+{
+  bool hadStream = mStream;
+  if (hadStream) {
+    mStream->RemoveMainThreadListener(this);
+  }
+  AudioNode::DestroyMediaStream();
+  if (hadStream && Context()) {
+    Context()->UnregisterAudioBufferSourceNode(this);
+  }
+}
+
+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 AudioBufferSourceNodeBinding::Wrap(aCx, this, aGivenProto);
+}
+
+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 = 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);
+  }
+}
+
+void
+AudioBufferSourceNode::SendBufferParameterToStream(JSContext* aCx)
+{
+  AudioNodeStream* ns = mStream;
+  if (!ns) {
+    return;
+  }
+
+  if (mBuffer) {
+    RefPtr<ThreadSharedFloatArrayBufferList> data =
+      mBuffer->GetThreadSharedChannelsForRate(aCx);
+    ns->SetBuffer(data.forget());
+
+    if (mStartCalled) {
+      SendOffsetAndDurationParametersToStream(ns);
+    }
+  } else {
+    ns->SetInt32Parameter(BUFFEREND, 0);
+    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();
+  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.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
+    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)
+      : 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;
+  };
+
+  NS_DispatchToMainThread(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