Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

430 files changed, 49011 insertions, 0 deletions

diff --git a/dom/media/webaudio/AlignedTArray.h b/dom/media/webaudio/AlignedTArray.h
new file mode 100644
index 0000000000..28c0f094ef
--- /dev/null
+++ b/dom/media/webaudio/AlignedTArray.h
@@ -0,0 +1,115 @@
+/* -*- 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/. */
+
+#ifndef AlignedTArray_h__
+#define AlignedTArray_h__
+
+#include "mozilla/Alignment.h"
+#include "nsTArray.h"
+
+/**
+ * E: element type, must be a POD type.
+ * N: N bytes alignment for the first element, defaults to 32
+ * S: S bytes of inline storage
+ */
+template <typename E, int S, int N = 32>
+class AlignedAutoTArray : private AutoTArray<E, S + N> {
+  static_assert((N & (N - 1)) == 0, "N must be power of 2");
+  typedef AutoTArray<E, S + N> base_type;
+
+ public:
+  typedef E value_type;
+  typedef typename base_type::size_type size_type;
+  typedef typename base_type::index_type index_type;
+
+  AlignedAutoTArray() = default;
+  explicit AlignedAutoTArray(size_type capacity)
+      : base_type(capacity + sExtra) {}
+  value_type* Elements() { return getAligned(base_type::Elements()); }
+  const value_type* Elements() const {
+    return getAligned(base_type::Elements());
+  }
+  value_type& operator[](index_type i) { return Elements()[i]; }
+  const value_type& operator[](index_type i) const { return Elements()[i]; }
+
+  void SetLength(size_type newLen) { base_type::SetLength(newLen + sExtra); }
+
+  [[nodiscard]] bool SetLength(size_type newLen, const mozilla::fallible_t&) {
+    return base_type::SetLength(newLen + sExtra, mozilla::fallible);
+  }
+
+  size_type Length() const {
+    return base_type::Length() <= sExtra ? 0 : base_type::Length() - sExtra;
+  }
+
+  using base_type::ShallowSizeOfExcludingThis;
+  using base_type::ShallowSizeOfIncludingThis;
+
+ private:
+  AlignedAutoTArray(const AlignedAutoTArray& other) = delete;
+  void operator=(const AlignedAutoTArray& other) = delete;
+
+  static const size_type sPadding =
+      N <= MOZ_ALIGNOF(E) ? 0 : N - MOZ_ALIGNOF(E);
+  static const size_type sExtra = (sPadding + sizeof(E) - 1) / sizeof(E);
+
+  template <typename U>
+  static U* getAligned(U* p) {
+    return reinterpret_cast<U*>(((uintptr_t)p + N - 1) & ~(N - 1));
+  }
+};
+
+/**
+ * E: element type, must be a POD type.
+ * N: N bytes alignment for the first element, defaults to 32
+ */
+template <typename E, int N = 32>
+class AlignedTArray : private nsTArray_Impl<E, nsTArrayInfallibleAllocator> {
+  static_assert((N & (N - 1)) == 0, "N must be power of 2");
+  typedef nsTArray_Impl<E, nsTArrayInfallibleAllocator> base_type;
+
+ public:
+  typedef E value_type;
+  typedef typename base_type::size_type size_type;
+  typedef typename base_type::index_type index_type;
+
+  AlignedTArray() = default;
+  explicit AlignedTArray(size_type capacity) : base_type(capacity + sExtra) {}
+  value_type* Elements() { return getAligned(base_type::Elements()); }
+  const value_type* Elements() const {
+    return getAligned(base_type::Elements());
+  }
+  value_type& operator[](index_type i) { return Elements()[i]; }
+  const value_type& operator[](index_type i) const { return Elements()[i]; }
+
+  void SetLength(size_type newLen) { base_type::SetLength(newLen + sExtra); }
+
+  [[nodiscard]] bool SetLength(size_type newLen, const mozilla::fallible_t&) {
+    return base_type::SetLength(newLen + sExtra, mozilla::fallible);
+  }
+
+  size_type Length() const {
+    return base_type::Length() <= sExtra ? 0 : base_type::Length() - sExtra;
+  }
+
+  using base_type::ShallowSizeOfExcludingThis;
+  using base_type::ShallowSizeOfIncludingThis;
+
+ private:
+  AlignedTArray(const AlignedTArray& other) = delete;
+  void operator=(const AlignedTArray& other) = delete;
+
+  static const size_type sPadding =
+      N <= MOZ_ALIGNOF(E) ? 0 : N - MOZ_ALIGNOF(E);
+  static const size_type sExtra = (sPadding + sizeof(E) - 1) / sizeof(E);
+
+  template <typename U>
+  static U* getAligned(U* p) {
+    return reinterpret_cast<U*>(((uintptr_t)p + N - 1) & ~(N - 1));
+  }
+};
+
+#endif  // AlignedTArray_h__
diff --git a/dom/media/webaudio/AlignmentUtils.h b/dom/media/webaudio/AlignmentUtils.h
new file mode 100644
index 0000000000..c22dd4a58b
--- /dev/null
+++ b/dom/media/webaudio/AlignmentUtils.h
@@ -0,0 +1,29 @@
+/* -*- 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/. */
+
+#ifndef AlignmentUtils_h__
+#define AlignmentUtils_h__
+
+#define IS_ALIGNED16(ptr) ((((uintptr_t)ptr + 15) & ~0x0F) == (uintptr_t)ptr)
+
+#ifdef DEBUG
+#  define ASSERT_ALIGNED16(ptr)   \
+    MOZ_ASSERT(IS_ALIGNED16(ptr), \
+               #ptr " has to be aligned to a 16 byte boundary");
+#else
+#  define ASSERT_ALIGNED16(ptr)
+#endif
+
+#ifdef DEBUG
+#  define ASSERT_MULTIPLE16(v) \
+    MOZ_ASSERT(v % 16 == 0, #v " has to be a a multiple of 16");
+#else
+#  define ASSERT_MULTIPLE16(v)
+#endif
+
+#define ALIGNED16(ptr) (float*)(((uintptr_t)ptr + 15) & ~0x0F);
+
+#endif
diff --git a/dom/media/webaudio/AnalyserNode.cpp b/dom/media/webaudio/AnalyserNode.cpp
new file mode 100644
index 0000000000..d8b547477e
--- /dev/null
+++ b/dom/media/webaudio/AnalyserNode.cpp
@@ -0,0 +1,388 @@
+/* -*- 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 "mozilla/dom/AnalyserNode.h"
+#include "mozilla/dom/AnalyserNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/PodOperations.h"
+#include "nsMathUtils.h"
+#include "Tracing.h"
+
+namespace mozilla {
+
+static const uint32_t MAX_FFT_SIZE = 32768;
+static const size_t CHUNK_COUNT = MAX_FFT_SIZE >> WEBAUDIO_BLOCK_SIZE_BITS;
+static_assert(MAX_FFT_SIZE == CHUNK_COUNT * WEBAUDIO_BLOCK_SIZE,
+              "MAX_FFT_SIZE must be a multiple of WEBAUDIO_BLOCK_SIZE");
+static_assert((CHUNK_COUNT & (CHUNK_COUNT - 1)) == 0,
+              "CHUNK_COUNT must be power of 2 for remainder behavior");
+
+namespace dom {
+
+class AnalyserNodeEngine final : public AudioNodeEngine {
+  class TransferBuffer final : public Runnable {
+   public:
+    TransferBuffer(AudioNodeTrack* aTrack, const AudioChunk& aChunk)
+        : Runnable("dom::AnalyserNodeEngine::TransferBuffer"),
+          mTrack(aTrack),
+          mChunk(aChunk) {}
+
+    NS_IMETHOD Run() override {
+      RefPtr<AnalyserNode> node =
+          static_cast<AnalyserNode*>(mTrack->Engine()->NodeMainThread());
+      if (node) {
+        node->AppendChunk(mChunk);
+      }
+      return NS_OK;
+    }
+
+   private:
+    RefPtr<AudioNodeTrack> mTrack;
+    AudioChunk mChunk;
+  };
+
+ public:
+  explicit AnalyserNodeEngine(AnalyserNode* aNode) : AudioNodeEngine(aNode) {
+    MOZ_ASSERT(NS_IsMainThread());
+  }
+
+  virtual void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                            const AudioBlock& aInput, AudioBlock* aOutput,
+                            bool* aFinished) override {
+    TRACE("AnalyserNodeEngine::ProcessBlock");
+    *aOutput = aInput;
+
+    if (aInput.IsNull()) {
+      // If AnalyserNode::mChunks has only null chunks, then there is no need
+      // to send further null chunks.
+      if (mChunksToProcess == 0) {
+        return;
+      }
+
+      --mChunksToProcess;
+      if (mChunksToProcess == 0) {
+        aTrack->ScheduleCheckForInactive();
+      }
+
+    } else {
+      // This many null chunks will be required to empty AnalyserNode::mChunks.
+      mChunksToProcess = CHUNK_COUNT;
+    }
+
+    RefPtr<TransferBuffer> transfer =
+        new TransferBuffer(aTrack, aInput.AsAudioChunk());
+    AbstractThread::MainThread()->Dispatch(transfer.forget());
+  }
+
+  virtual bool IsActive() const override { return mChunksToProcess != 0; }
+
+  virtual size_t SizeOfIncludingThis(
+      MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  uint32_t mChunksToProcess = 0;
+};
+
+/* static */
+already_AddRefed<AnalyserNode> AnalyserNode::Create(
+    AudioContext& aAudioContext, const AnalyserOptions& aOptions,
+    ErrorResult& aRv) {
+  RefPtr<AnalyserNode> analyserNode = new AnalyserNode(&aAudioContext);
+
+  analyserNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  analyserNode->SetFftSize(aOptions.mFftSize, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  analyserNode->SetMinAndMaxDecibels(aOptions.mMinDecibels,
+                                     aOptions.mMaxDecibels, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  analyserNode->SetSmoothingTimeConstant(aOptions.mSmoothingTimeConstant, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  return analyserNode.forget();
+}
+
+AnalyserNode::AnalyserNode(AudioContext* aContext)
+    : AudioNode(aContext, 2, ChannelCountMode::Max,
+                ChannelInterpretation::Speakers),
+      mAnalysisBlock(2048),
+      mMinDecibels(-100.),
+      mMaxDecibels(-30.),
+      mSmoothingTimeConstant(.8) {
+  mTrack =
+      AudioNodeTrack::Create(aContext, new AnalyserNodeEngine(this),
+                             AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+
+  // Enough chunks must be recorded to handle the case of fftSize being
+  // increased to maximum immediately before getFloatTimeDomainData() is
+  // called, for example.
+  Unused << mChunks.SetLength(CHUNK_COUNT, fallible);
+
+  AllocateBuffer();
+}
+
+size_t AnalyserNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mAnalysisBlock.SizeOfExcludingThis(aMallocSizeOf);
+  amount += mChunks.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  amount += mOutputBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t AnalyserNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject* AnalyserNode::WrapObject(JSContext* aCx,
+                                   JS::Handle<JSObject*> aGivenProto) {
+  return AnalyserNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void AnalyserNode::SetFftSize(uint32_t aValue, ErrorResult& aRv) {
+  // Disallow values that are not a power of 2 and outside the [32,32768] range
+  if (aValue < 32 || aValue > MAX_FFT_SIZE || (aValue & (aValue - 1)) != 0) {
+    aRv.ThrowIndexSizeError(nsPrintfCString(
+        "FFT size %u is not a power of two in the range 32 to 32768", aValue));
+    return;
+  }
+  if (FftSize() != aValue) {
+    mAnalysisBlock.SetFFTSize(aValue);
+    AllocateBuffer();
+  }
+}
+
+void AnalyserNode::SetMinDecibels(double aValue, ErrorResult& aRv) {
+  if (aValue >= mMaxDecibels) {
+    aRv.ThrowIndexSizeError(nsPrintfCString(
+        "%g is not strictly smaller than current maxDecibels (%g)", aValue,
+        mMaxDecibels));
+    return;
+  }
+  mMinDecibels = aValue;
+}
+
+void AnalyserNode::SetMaxDecibels(double aValue, ErrorResult& aRv) {
+  if (aValue <= mMinDecibels) {
+    aRv.ThrowIndexSizeError(nsPrintfCString(
+        "%g is not strictly larger than current minDecibels (%g)", aValue,
+        mMinDecibels));
+    return;
+  }
+  mMaxDecibels = aValue;
+}
+
+void AnalyserNode::SetMinAndMaxDecibels(double aMinValue, double aMaxValue,
+                                        ErrorResult& aRv) {
+  if (aMinValue >= aMaxValue) {
+    aRv.ThrowIndexSizeError(nsPrintfCString(
+        "minDecibels value (%g) must be smaller than maxDecibels value (%g)",
+        aMinValue, aMaxValue));
+    return;
+  }
+  mMinDecibels = aMinValue;
+  mMaxDecibels = aMaxValue;
+}
+
+void AnalyserNode::SetSmoothingTimeConstant(double aValue, ErrorResult& aRv) {
+  if (aValue < 0 || aValue > 1) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("%g is not in the range [0, 1]", aValue));
+    return;
+  }
+  mSmoothingTimeConstant = aValue;
+}
+
+void AnalyserNode::GetFloatFrequencyData(const Float32Array& aArray) {
+  if (!FFTAnalysis()) {
+    // Might fail to allocate memory
+    return;
+  }
+
+  aArray.ProcessData([&](const Span<float>& aData, JS::AutoCheckCannotGC&&) {
+    size_t length = std::min(size_t(aData.Length()), mOutputBuffer.Length());
+
+    for (size_t i = 0; i < length; ++i) {
+      aData[i] = WebAudioUtils::ConvertLinearToDecibels(
+          mOutputBuffer[i], -std::numeric_limits<float>::infinity());
+    }
+  });
+}
+
+void AnalyserNode::GetByteFrequencyData(const Uint8Array& aArray) {
+  if (!FFTAnalysis()) {
+    // Might fail to allocate memory
+    return;
+  }
+
+  const double rangeScaleFactor = 1.0 / (mMaxDecibels - mMinDecibels);
+
+  aArray.ProcessData([&](const Span<uint8_t>& aData, JS::AutoCheckCannotGC&&) {
+    size_t length = std::min(size_t(aData.Length()), mOutputBuffer.Length());
+
+    for (size_t i = 0; i < length; ++i) {
+      const double decibels = WebAudioUtils::ConvertLinearToDecibels(
+          mOutputBuffer[i], mMinDecibels);
+      // scale down the value to the range of [0, UCHAR_MAX]
+      const double scaled = std::max(
+          0.0,
+          std::min(double(UCHAR_MAX),
+                   UCHAR_MAX * (decibels - mMinDecibels) * rangeScaleFactor));
+      aData[i] = static_cast<unsigned char>(scaled);
+    }
+  });
+}
+
+void AnalyserNode::GetFloatTimeDomainData(const Float32Array& aArray) {
+  aArray.ProcessData([&](const Span<float>& aData, JS::AutoCheckCannotGC&&) {
+    size_t length = std::min(aData.Length(), size_t(FftSize()));
+
+    GetTimeDomainData(aData.Elements(), length);
+  });
+}
+
+void AnalyserNode::GetByteTimeDomainData(const Uint8Array& aArray) {
+  aArray.ProcessData([&](const Span<uint8_t>& aData, JS::AutoCheckCannotGC&&) {
+    size_t length = std::min(aData.Length(), size_t(FftSize()));
+
+    AlignedTArray<float> tmpBuffer;
+    if (!tmpBuffer.SetLength(length, fallible)) {
+      return;
+    }
+
+    GetTimeDomainData(tmpBuffer.Elements(), length);
+
+    unsigned char* buffer = aData.Elements();
+    for (size_t i = 0; i < length; ++i) {
+      const float value = tmpBuffer[i];
+      // scale the value to the range of [0, UCHAR_MAX]
+      const float scaled =
+          std::max(0.0f, std::min(float(UCHAR_MAX), 128.0f * (value + 1.0f)));
+      buffer[i] = static_cast<unsigned char>(scaled);
+    }
+  });
+}
+
+bool AnalyserNode::FFTAnalysis() {
+  AlignedTArray<float> tmpBuffer;
+  size_t fftSize = FftSize();
+  if (!tmpBuffer.SetLength(fftSize, fallible)) {
+    return false;
+  }
+
+  float* inputBuffer = tmpBuffer.Elements();
+  GetTimeDomainData(inputBuffer, fftSize);
+  ApplyBlackmanWindow(inputBuffer, fftSize);
+  mAnalysisBlock.PerformFFT(inputBuffer);
+
+  // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT
+  // scaling factor).
+  const double magnitudeScale = 1.0 / fftSize;
+
+  for (uint32_t i = 0; i < mOutputBuffer.Length(); ++i) {
+    double scalarMagnitude =
+        fdlibm_hypot(mAnalysisBlock.RealData(i), mAnalysisBlock.ImagData(i)) *
+        magnitudeScale;
+    mOutputBuffer[i] = mSmoothingTimeConstant * mOutputBuffer[i] +
+                       (1.0 - mSmoothingTimeConstant) * scalarMagnitude;
+  }
+
+  return true;
+}
+
+void AnalyserNode::ApplyBlackmanWindow(float* aBuffer, uint32_t aSize) {
+  double alpha = 0.16;
+  double a0 = 0.5 * (1.0 - alpha);
+  double a1 = 0.5;
+  double a2 = 0.5 * alpha;
+
+  for (uint32_t i = 0; i < aSize; ++i) {
+    double x = double(i) / aSize;
+    double window =
+        a0 - a1 * fdlibm_cos(2 * M_PI * x) + a2 * fdlibm_cos(4 * M_PI * x);
+    aBuffer[i] *= window;
+  }
+}
+
+bool AnalyserNode::AllocateBuffer() {
+  bool result = true;
+  if (mOutputBuffer.Length() != FrequencyBinCount()) {
+    if (!mOutputBuffer.SetLength(FrequencyBinCount(), fallible)) {
+      return false;
+    }
+    memset(mOutputBuffer.Elements(), 0, sizeof(float) * FrequencyBinCount());
+  }
+  return result;
+}
+
+void AnalyserNode::AppendChunk(const AudioChunk& aChunk) {
+  if (mChunks.Length() == 0) {
+    return;
+  }
+
+  ++mCurrentChunk;
+  mChunks[mCurrentChunk & (CHUNK_COUNT - 1)] = aChunk;
+}
+
+// Reads into aData the oldest aLength samples of the fftSize most recent
+// samples.
+void AnalyserNode::GetTimeDomainData(float* aData, size_t aLength) {
+  size_t fftSize = FftSize();
+  MOZ_ASSERT(aLength <= fftSize);
+
+  if (mChunks.Length() == 0) {
+    PodZero(aData, aLength);
+    return;
+  }
+
+  size_t readChunk =
+      mCurrentChunk - ((fftSize - 1) >> WEBAUDIO_BLOCK_SIZE_BITS);
+  size_t readIndex = (0 - fftSize) & (WEBAUDIO_BLOCK_SIZE - 1);
+  MOZ_ASSERT(readIndex == 0 || readIndex + fftSize == WEBAUDIO_BLOCK_SIZE);
+
+  for (size_t writeIndex = 0; writeIndex < aLength;) {
+    const AudioChunk& chunk = mChunks[readChunk & (CHUNK_COUNT - 1)];
+    const size_t channelCount = chunk.ChannelCount();
+    size_t copyLength =
+        std::min<size_t>(aLength - writeIndex, WEBAUDIO_BLOCK_SIZE);
+    float* dataOut = &aData[writeIndex];
+
+    if (channelCount == 0) {
+      PodZero(dataOut, copyLength);
+    } else {
+      float scale = chunk.mVolume / channelCount;
+      {  // channel 0
+        auto channelData =
+            static_cast<const float*>(chunk.mChannelData[0]) + readIndex;
+        AudioBufferCopyWithScale(channelData, scale, dataOut, copyLength);
+      }
+      for (uint32_t i = 1; i < channelCount; ++i) {
+        auto channelData =
+            static_cast<const float*>(chunk.mChannelData[i]) + readIndex;
+        AudioBufferAddWithScale(channelData, scale, dataOut, copyLength);
+      }
+    }
+
+    readChunk++;
+    writeIndex += copyLength;
+  }
+}
+
+}  // namespace dom
+}  // namespace mozilla
diff --git a/dom/media/webaudio/AnalyserNode.h b/dom/media/webaudio/AnalyserNode.h
new file mode 100644
index 0000000000..bdec13f6b9
--- /dev/null
+++ b/dom/media/webaudio/AnalyserNode.h
@@ -0,0 +1,82 @@
+/* -*- 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/. */
+
+#ifndef AnalyserNode_h_
+#define AnalyserNode_h_
+
+#include "AudioNode.h"
+#include "FFTBlock.h"
+#include "AlignedTArray.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct AnalyserOptions;
+
+class AnalyserNode final : public AudioNode {
+ public:
+  static already_AddRefed<AnalyserNode> Create(AudioContext& aAudioContext,
+                                               const AnalyserOptions& aOptions,
+                                               ErrorResult& aRv);
+
+  NS_INLINE_DECL_REFCOUNTING_INHERITED(AnalyserNode, AudioNode)
+
+  virtual JSObject* WrapObject(JSContext* aCx,
+                               JS::Handle<JSObject*> aGivenProto) override;
+
+  static already_AddRefed<AnalyserNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const AnalyserOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  void GetFloatFrequencyData(const Float32Array& aArray);
+  void GetByteFrequencyData(const Uint8Array& aArray);
+  void GetFloatTimeDomainData(const Float32Array& aArray);
+  void GetByteTimeDomainData(const Uint8Array& aArray);
+  uint32_t FftSize() const { return mAnalysisBlock.FFTSize(); }
+  void SetFftSize(uint32_t aValue, ErrorResult& aRv);
+  uint32_t FrequencyBinCount() const { return FftSize() / 2; }
+  double MinDecibels() const { return mMinDecibels; }
+  void SetMinDecibels(double aValue, ErrorResult& aRv);
+  double MaxDecibels() const { return mMaxDecibels; }
+  void SetMaxDecibels(double aValue, ErrorResult& aRv);
+  double SmoothingTimeConstant() const { return mSmoothingTimeConstant; }
+  void SetSmoothingTimeConstant(double aValue, ErrorResult& aRv);
+
+  virtual const char* NodeType() const override { return "AnalyserNode"; }
+
+  virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+  void SetMinAndMaxDecibels(double aMinValue, double aMaxValue,
+                            ErrorResult& aRv);
+
+ private:
+  ~AnalyserNode() = default;
+
+  friend class AnalyserNodeEngine;
+  void AppendChunk(const AudioChunk& aChunk);
+  bool AllocateBuffer();
+  bool FFTAnalysis();
+  void ApplyBlackmanWindow(float* aBuffer, uint32_t aSize);
+  void GetTimeDomainData(float* aData, size_t aLength);
+
+ private:
+  explicit AnalyserNode(AudioContext* aContext);
+
+  FFTBlock mAnalysisBlock;
+  nsTArray<AudioChunk> mChunks;
+  double mMinDecibels;
+  double mMaxDecibels;
+  double mSmoothingTimeConstant;
+  size_t mCurrentChunk = 0;
+  AlignedTArray<float> mOutputBuffer;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/AudioBlock.cpp b/dom/media/webaudio/AudioBlock.cpp
new file mode 100644
index 0000000000..09bbd7f70a
--- /dev/null
+++ b/dom/media/webaudio/AudioBlock.cpp
@@ -0,0 +1,166 @@
+/* -*- 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 "AudioBlock.h"
+#include "AlignmentUtils.h"
+
+namespace mozilla {
+
+/**
+ * Heap-allocated buffer of channels of 128-sample float arrays, with
+ * threadsafe refcounting.  Typically you would allocate one of these, fill it
+ * in, and then treat it as immutable while it's shared.
+ *
+ * Downstream references are accounted specially so that the creator of the
+ * buffer can reuse and modify its contents next iteration if other references
+ * are all downstream temporary references held by AudioBlock.
+ *
+ * We guarantee 16 byte alignment of the channel data.
+ */
+class AudioBlockBuffer final : public ThreadSharedObject {
+ public:
+  virtual AudioBlockBuffer* AsAudioBlockBuffer() override { return this; };
+
+  uint32_t ChannelsAllocated() const { return mChannelsAllocated; }
+  float* ChannelData(uint32_t aChannel) const {
+    float* base =
+        reinterpret_cast<float*>(((uintptr_t)(this + 1) + 15) & ~0x0F);
+    ASSERT_ALIGNED16(base);
+    return base + aChannel * WEBAUDIO_BLOCK_SIZE;
+  }
+
+  static already_AddRefed<AudioBlockBuffer> Create(uint32_t aChannelCount) {
+    CheckedInt<size_t> size = WEBAUDIO_BLOCK_SIZE;
+    size *= aChannelCount;
+    size *= sizeof(float);
+    size += sizeof(AudioBlockBuffer);
+    size += 15;  // padding for alignment
+    if (!size.isValid()) {
+      MOZ_CRASH();
+    }
+
+    void* m = operator new(size.value());
+    RefPtr<AudioBlockBuffer> p = new (m) AudioBlockBuffer(aChannelCount);
+    NS_ASSERTION((reinterpret_cast<char*>(p.get() + 1) -
+                  reinterpret_cast<char*>(p.get())) %
+                         4 ==
+                     0,
+                 "AudioBlockBuffers should be at least 4-byte aligned");
+    return p.forget();
+  }
+
+  // Graph thread only.
+  void DownstreamRefAdded() { ++mDownstreamRefCount; }
+  void DownstreamRefRemoved() {
+    MOZ_ASSERT(mDownstreamRefCount > 0);
+    --mDownstreamRefCount;
+  }
+  // Whether this is shared by any owners that are not downstream.
+  // Called only from owners with a reference that is not a downstream
+  // reference.  Graph thread only.
+  bool HasLastingShares() const {
+    // mRefCnt is atomic and so reading its value is defined even when
+    // modifications may happen on other threads.  mDownstreamRefCount is
+    // not modified on any other thread.
+    //
+    // If all other references are downstream references (managed on this, the
+    // graph thread), then other threads are not using this buffer and cannot
+    // add further references.  This method can safely return false.  The
+    // buffer contents can be modified.
+    //
+    // If there are other references that are not downstream references, then
+    // this method will return true.  The buffer will be assumed to be still
+    // in use and so will not be reused.
+    nsrefcnt count = mRefCnt;
+    // This test is strictly less than because the caller has a reference
+    // that is not a downstream reference.
+    MOZ_ASSERT(mDownstreamRefCount < count);
+    return count != mDownstreamRefCount + 1;
+  }
+
+  virtual size_t SizeOfIncludingThis(
+      MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  explicit AudioBlockBuffer(uint32_t aChannelsAllocated)
+      : mChannelsAllocated(aChannelsAllocated) {}
+  ~AudioBlockBuffer() override { MOZ_ASSERT(mDownstreamRefCount == 0); }
+
+  nsAutoRefCnt mDownstreamRefCount;
+  const uint32_t mChannelsAllocated;
+};
+
+AudioBlock::~AudioBlock() { ClearDownstreamMark(); }
+
+void AudioBlock::SetBuffer(ThreadSharedObject* aNewBuffer) {
+  if (aNewBuffer == mBuffer) {
+    return;
+  }
+
+  ClearDownstreamMark();
+
+  mBuffer = aNewBuffer;
+
+  if (!aNewBuffer) {
+    return;
+  }
+
+  AudioBlockBuffer* buffer = aNewBuffer->AsAudioBlockBuffer();
+  if (buffer) {
+    buffer->DownstreamRefAdded();
+    mBufferIsDownstreamRef = true;
+  }
+}
+
+void AudioBlock::ClearDownstreamMark() {
+  if (mBufferIsDownstreamRef) {
+    mBuffer->AsAudioBlockBuffer()->DownstreamRefRemoved();
+    mBufferIsDownstreamRef = false;
+  }
+}
+
+bool AudioBlock::CanWrite() {
+  // If mBufferIsDownstreamRef is set then the buffer is not ours to use.
+  // It may be in use by another node which is not downstream.
+  return !mBufferIsDownstreamRef &&
+         !mBuffer->AsAudioBlockBuffer()->HasLastingShares();
+}
+
+void AudioBlock::AllocateChannels(uint32_t aChannelCount) {
+  MOZ_ASSERT(mDuration == WEBAUDIO_BLOCK_SIZE);
+
+  if (mBufferIsDownstreamRef) {
+    // This is not our buffer to re-use.
+    ClearDownstreamMark();
+  } else if (mBuffer) {
+    AudioBlockBuffer* buffer = mBuffer->AsAudioBlockBuffer();
+    if (buffer && !buffer->HasLastingShares() &&
+        buffer->ChannelsAllocated() >= aChannelCount) {
+      MOZ_ASSERT(mBufferFormat == AUDIO_FORMAT_FLOAT32);
+      // No need to allocate again.
+      uint32_t previousChannelCount = ChannelCount();
+      mChannelData.SetLength(aChannelCount);
+      for (uint32_t i = previousChannelCount; i < aChannelCount; ++i) {
+        mChannelData[i] = buffer->ChannelData(i);
+      }
+      mVolume = 1.0f;
+      return;
+    }
+  }
+
+  RefPtr<AudioBlockBuffer> buffer = AudioBlockBuffer::Create(aChannelCount);
+  mChannelData.SetLength(aChannelCount);
+  for (uint32_t i = 0; i < aChannelCount; ++i) {
+    mChannelData[i] = buffer->ChannelData(i);
+  }
+  mBuffer = std::move(buffer);
+  mVolume = 1.0f;
+  mBufferFormat = AUDIO_FORMAT_FLOAT32;
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webaudio/AudioBlock.h b/dom/media/webaudio/AudioBlock.h
new file mode 100644
index 0000000000..b5178c9d8a
--- /dev/null
+++ b/dom/media/webaudio/AudioBlock.h
@@ -0,0 +1,134 @@
+/* -*- 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/. */
+#ifndef MOZILLA_AUDIOBLOCK_H_
+#define MOZILLA_AUDIOBLOCK_H_
+
+#include "AudioSegment.h"
+
+namespace mozilla {
+
+/**
+ * An AudioChunk whose buffer contents need to be valid only for one
+ * processing block iteration, after which contents can be overwritten if the
+ * buffer has not been passed to longer term storage or to another thread,
+ * which may happen though AsAudioChunk() or AsMutableChunk().
+ *
+ * Use on graph thread only.
+ */
+class AudioBlock : private AudioChunk {
+ public:
+  AudioBlock() {
+    mDuration = WEBAUDIO_BLOCK_SIZE;
+    mBufferFormat = AUDIO_FORMAT_SILENCE;
+  }
+  // No effort is made in constructors to ensure that mBufferIsDownstreamRef
+  // is set because the block is expected to be a temporary and so the
+  // reference will be released before the next iteration.
+  // The custom copy constructor is required so as not to set
+  // mBufferIsDownstreamRef without notifying AudioBlockBuffer.
+  AudioBlock(const AudioBlock& aBlock) : AudioChunk(aBlock.AsAudioChunk()) {}
+  explicit AudioBlock(const AudioChunk& aChunk) : AudioChunk(aChunk) {
+    MOZ_ASSERT(aChunk.mDuration == WEBAUDIO_BLOCK_SIZE);
+  }
+  ~AudioBlock();
+
+  using AudioChunk::ChannelCount;
+  using AudioChunk::ChannelData;
+  using AudioChunk::GetDuration;
+  using AudioChunk::IsNull;
+  using AudioChunk::SizeOfExcludingThis;
+  using AudioChunk::SizeOfExcludingThisIfUnshared;
+  // mDuration is not exposed.  Use GetDuration().
+  // mBuffer is not exposed.  Use Get/SetBuffer().
+  using AudioChunk::mBufferFormat;
+  using AudioChunk::mChannelData;
+  using AudioChunk::mVolume;
+
+  const AudioChunk& AsAudioChunk() const { return *this; }
+  AudioChunk* AsMutableChunk() {
+    ClearDownstreamMark();
+    return this;
+  }
+
+  /**
+   * Allocates, if necessary, aChannelCount buffers of WEBAUDIO_BLOCK_SIZE float
+   * samples for writing.
+   */
+  void AllocateChannels(uint32_t aChannelCount);
+
+  /**
+   * ChannelFloatsForWrite() should only be used when the buffers have been
+   * created with AllocateChannels().
+   */
+  float* ChannelFloatsForWrite(size_t aChannel) {
+    MOZ_ASSERT(mBufferFormat == AUDIO_FORMAT_FLOAT32);
+    MOZ_ASSERT(CanWrite());
+    return static_cast<float*>(const_cast<void*>(mChannelData[aChannel]));
+  }
+
+  ThreadSharedObject* GetBuffer() const { return mBuffer; }
+  void SetBuffer(ThreadSharedObject* aNewBuffer);
+  void SetNull(TrackTime aDuration) {
+    MOZ_ASSERT(aDuration == WEBAUDIO_BLOCK_SIZE);
+    SetBuffer(nullptr);
+    mChannelData.Clear();
+    mVolume = 1.0f;
+    mBufferFormat = AUDIO_FORMAT_SILENCE;
+  }
+
+  AudioBlock& operator=(const AudioBlock& aBlock) {
+    // Instead of just copying, mBufferIsDownstreamRef must be first cleared
+    // if set.  It is set again for the new mBuffer if possible.  This happens
+    // in SetBuffer().
+    return *this = aBlock.AsAudioChunk();
+  }
+  AudioBlock& operator=(const AudioChunk& aChunk) {
+    MOZ_ASSERT(aChunk.mDuration == WEBAUDIO_BLOCK_SIZE);
+    SetBuffer(aChunk.mBuffer);
+    mChannelData = aChunk.mChannelData;
+    mVolume = aChunk.mVolume;
+    mBufferFormat = aChunk.mBufferFormat;
+    return *this;
+  }
+
+  bool IsMuted() const { return mVolume == 0.0f; }
+
+  bool IsSilentOrSubnormal() const {
+    if (!mBuffer) {
+      return true;
+    }
+
+    for (uint32_t i = 0, length = mChannelData.Length(); i < length; ++i) {
+      const float* channel = static_cast<const float*>(mChannelData[i]);
+      for (TrackTime frame = 0; frame < mDuration; ++frame) {
+        if (fabs(channel[frame]) >= FLT_MIN) {
+          return false;
+        }
+      }
+    }
+
+    return true;
+  }
+
+ private:
+  void ClearDownstreamMark();
+  bool CanWrite();
+
+  // mBufferIsDownstreamRef is set only when mBuffer references an
+  // AudioBlockBuffer created in a different AudioBlock.  That can happen when
+  // this AudioBlock is on a node downstream from the node which created the
+  // buffer.  When this is set, the AudioBlockBuffer is notified that this
+  // reference does not prevent the upstream node from re-using the buffer next
+  // iteration and modifying its contents.  The AudioBlockBuffer is also
+  // notified when mBuffer releases this reference.
+  bool mBufferIsDownstreamRef = false;
+};
+
+}  // namespace mozilla
+
+MOZ_DECLARE_RELOCATE_USING_MOVE_CONSTRUCTOR(mozilla::AudioBlock)
+
+#endif  // MOZILLA_AUDIOBLOCK_H_
diff --git a/dom/media/webaudio/AudioBuffer.cpp b/dom/media/webaudio/AudioBuffer.cpp
new file mode 100644
index 0000000000..8c2a36fe45
--- /dev/null
+++ b/dom/media/webaudio/AudioBuffer.cpp
@@ -0,0 +1,498 @@
+/* -*- 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 "AudioBuffer.h"
+#include "mozilla/dom/AudioBufferBinding.h"
+#include "jsfriendapi.h"
+#include "js/ArrayBuffer.h"             // JS::StealArrayBufferContents
+#include "js/experimental/TypedData.h"  // JS_NewFloat32Array, JS_GetFloat32ArrayData, JS_GetTypedArrayLength, JS_GetArrayBufferViewBuffer
+#include "mozilla/ErrorResult.h"
+#include "AudioSegment.h"
+#include "AudioChannelFormat.h"
+#include "mozilla/PodOperations.h"
+#include "mozilla/CheckedInt.h"
+#include "mozilla/HoldDropJSObjects.h"
+#include "mozilla/MemoryReporting.h"
+#include "AudioNodeEngine.h"
+#include "nsPrintfCString.h"
+#include "nsTHashSet.h"
+#include <numeric>
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(AudioBuffer)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(AudioBuffer)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mJSChannels)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
+  tmp->ClearJSChannels();
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(AudioBuffer)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_CYCLE_COLLECTION_TRACE_BEGIN(AudioBuffer)
+  NS_IMPL_CYCLE_COLLECTION_TRACE_PRESERVED_WRAPPER
+  for (uint32_t i = 0; i < tmp->mJSChannels.Length(); ++i) {
+    NS_IMPL_CYCLE_COLLECTION_TRACE_JS_MEMBER_CALLBACK(mJSChannels[i])
+  }
+NS_IMPL_CYCLE_COLLECTION_TRACE_END
+
+/**
+ * AudioBuffers can be shared between AudioContexts, so we need a separate
+ * mechanism to track their memory usage. This thread-safe class keeps track of
+ * all the AudioBuffers, and gets called back by the memory reporting system
+ * when a memory report is needed, reporting how much memory is used by the
+ * buffers backing AudioBuffer objects. */
+class AudioBufferMemoryTracker : public nsIMemoryReporter {
+  NS_DECL_THREADSAFE_ISUPPORTS
+  NS_DECL_NSIMEMORYREPORTER
+
+ private:
+  AudioBufferMemoryTracker();
+  virtual ~AudioBufferMemoryTracker();
+
+ public:
+  /* Those methods can be called on any thread. */
+  static void RegisterAudioBuffer(const AudioBuffer* aAudioBuffer);
+  static void UnregisterAudioBuffer(const AudioBuffer* aAudioBuffer);
+
+ private:
+  static AudioBufferMemoryTracker* GetInstance();
+  /* Those methods must be called with the lock held. */
+  void RegisterAudioBufferInternal(const AudioBuffer* aAudioBuffer);
+  /* Returns the number of buffers still present in the hash table. */
+  uint32_t UnregisterAudioBufferInternal(const AudioBuffer* aAudioBuffer);
+  void Init();
+
+  /* This protects all members of this class. */
+  static StaticMutex sMutex MOZ_UNANNOTATED;
+  static StaticRefPtr<AudioBufferMemoryTracker> sSingleton;
+  nsTHashSet<const AudioBuffer*> mBuffers;
+};
+
+StaticRefPtr<AudioBufferMemoryTracker> AudioBufferMemoryTracker::sSingleton;
+StaticMutex AudioBufferMemoryTracker::sMutex;
+
+NS_IMPL_ISUPPORTS(AudioBufferMemoryTracker, nsIMemoryReporter);
+
+AudioBufferMemoryTracker* AudioBufferMemoryTracker::GetInstance() {
+  sMutex.AssertCurrentThreadOwns();
+  if (!sSingleton) {
+    sSingleton = new AudioBufferMemoryTracker();
+    sSingleton->Init();
+  }
+  return sSingleton;
+}
+
+AudioBufferMemoryTracker::AudioBufferMemoryTracker() = default;
+
+void AudioBufferMemoryTracker::Init() { RegisterWeakMemoryReporter(this); }
+
+AudioBufferMemoryTracker::~AudioBufferMemoryTracker() {
+  UnregisterWeakMemoryReporter(this);
+}
+
+void AudioBufferMemoryTracker::RegisterAudioBuffer(
+    const AudioBuffer* aAudioBuffer) {
+  StaticMutexAutoLock lock(sMutex);
+  AudioBufferMemoryTracker* tracker = AudioBufferMemoryTracker::GetInstance();
+  tracker->RegisterAudioBufferInternal(aAudioBuffer);
+}
+
+void AudioBufferMemoryTracker::UnregisterAudioBuffer(
+    const AudioBuffer* aAudioBuffer) {
+  StaticMutexAutoLock lock(sMutex);
+  AudioBufferMemoryTracker* tracker = AudioBufferMemoryTracker::GetInstance();
+  uint32_t count;
+  count = tracker->UnregisterAudioBufferInternal(aAudioBuffer);
+  if (count == 0) {
+    sSingleton = nullptr;
+  }
+}
+
+void AudioBufferMemoryTracker::RegisterAudioBufferInternal(
+    const AudioBuffer* aAudioBuffer) {
+  sMutex.AssertCurrentThreadOwns();
+  mBuffers.Insert(aAudioBuffer);
+}
+
+uint32_t AudioBufferMemoryTracker::UnregisterAudioBufferInternal(
+    const AudioBuffer* aAudioBuffer) {
+  sMutex.AssertCurrentThreadOwns();
+  mBuffers.Remove(aAudioBuffer);
+  return mBuffers.Count();
+}
+
+MOZ_DEFINE_MALLOC_SIZE_OF(AudioBufferMemoryTrackerMallocSizeOf)
+
+NS_IMETHODIMP
+AudioBufferMemoryTracker::CollectReports(nsIHandleReportCallback* aHandleReport,
+                                         nsISupports* aData, bool) {
+  StaticMutexAutoLock lock(sMutex);
+  const size_t amount =
+      std::accumulate(mBuffers.cbegin(), mBuffers.cend(), size_t(0),
+                      [](size_t val, const AudioBuffer* buffer) {
+                        return val + buffer->SizeOfIncludingThis(
+                                         AudioBufferMemoryTrackerMallocSizeOf);
+                      });
+
+  MOZ_COLLECT_REPORT("explicit/webaudio/audiobuffer", KIND_HEAP, UNITS_BYTES,
+                     amount, "Memory used by AudioBuffer objects (Web Audio).");
+
+  return NS_OK;
+}
+
+AudioBuffer::AudioBuffer(nsPIDOMWindowInner* aWindow,
+                         uint32_t aNumberOfChannels, uint32_t aLength,
+                         float aSampleRate, ErrorResult& aRv)
+    : mOwnerWindow(do_GetWeakReference(aWindow)), mSampleRate(aSampleRate) {
+  // Note that a buffer with zero channels is permitted here for the sake of
+  // AudioProcessingEvent, where channel counts must match parameters passed
+  // to createScriptProcessor(), one of which may be zero.
+  if (aSampleRate < WebAudioUtils::MinSampleRate ||
+      aSampleRate > WebAudioUtils::MaxSampleRate) {
+    aRv.ThrowNotSupportedError(
+        nsPrintfCString("Sample rate (%g) is out of range", aSampleRate));
+    return;
+  }
+
+  if (aNumberOfChannels > WebAudioUtils::MaxChannelCount) {
+    aRv.ThrowNotSupportedError(nsPrintfCString(
+        "Number of channels (%u) is out of range", aNumberOfChannels));
+    return;
+  }
+
+  if (!aLength || aLength > INT32_MAX) {
+    aRv.ThrowNotSupportedError(
+        nsPrintfCString("Length (%u) is out of range", aLength));
+    return;
+  }
+
+  mSharedChannels.mDuration = aLength;
+  mJSChannels.SetLength(aNumberOfChannels);
+  mozilla::HoldJSObjects(this);
+  AudioBufferMemoryTracker::RegisterAudioBuffer(this);
+}
+
+AudioBuffer::~AudioBuffer() {
+  AudioBufferMemoryTracker::UnregisterAudioBuffer(this);
+  ClearJSChannels();
+  mozilla::DropJSObjects(this);
+}
+
+/* static */
+already_AddRefed<AudioBuffer> AudioBuffer::Constructor(
+    const GlobalObject& aGlobal, const AudioBufferOptions& aOptions,
+    ErrorResult& aRv) {
+  if (!aOptions.mNumberOfChannels) {
+    aRv.ThrowNotSupportedError("Must have nonzero number of channels");
+    return nullptr;
+  }
+
+  nsCOMPtr<nsPIDOMWindowInner> window =
+      do_QueryInterface(aGlobal.GetAsSupports());
+
+  return Create(window, aOptions.mNumberOfChannels, aOptions.mLength,
+                aOptions.mSampleRate, aRv);
+}
+
+void AudioBuffer::ClearJSChannels() { mJSChannels.Clear(); }
+
+void AudioBuffer::SetSharedChannels(
+    already_AddRefed<ThreadSharedFloatArrayBufferList> aBuffer) {
+  RefPtr<ThreadSharedFloatArrayBufferList> buffer = aBuffer;
+  uint32_t channelCount = buffer->GetChannels();
+  mSharedChannels.mChannelData.SetLength(channelCount);
+  for (uint32_t i = 0; i < channelCount; ++i) {
+    mSharedChannels.mChannelData[i] = buffer->GetData(i);
+  }
+  mSharedChannels.mBuffer = std::move(buffer);
+  mSharedChannels.mBufferFormat = AUDIO_FORMAT_FLOAT32;
+}
+
+/* static */
+already_AddRefed<AudioBuffer> AudioBuffer::Create(
+    nsPIDOMWindowInner* aWindow, uint32_t aNumberOfChannels, uint32_t aLength,
+    float aSampleRate,
+    already_AddRefed<ThreadSharedFloatArrayBufferList> aInitialContents,
+    ErrorResult& aRv) {
+  RefPtr<ThreadSharedFloatArrayBufferList> initialContents = aInitialContents;
+  RefPtr<AudioBuffer> buffer =
+      new AudioBuffer(aWindow, aNumberOfChannels, aLength, aSampleRate, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+
+  if (initialContents) {
+    MOZ_ASSERT(initialContents->GetChannels() == aNumberOfChannels);
+    buffer->SetSharedChannels(initialContents.forget());
+  }
+
+  return buffer.forget();
+}
+
+/* static */
+already_AddRefed<AudioBuffer> AudioBuffer::Create(
+    nsPIDOMWindowInner* aWindow, float aSampleRate,
+    AudioChunk&& aInitialContents) {
+  AudioChunk initialContents = aInitialContents;
+  ErrorResult rv;
+  RefPtr<AudioBuffer> buffer =
+      new AudioBuffer(aWindow, initialContents.ChannelCount(),
+                      initialContents.mDuration, aSampleRate, rv);
+  if (rv.Failed()) {
+    return nullptr;
+  }
+  buffer->mSharedChannels = std::move(aInitialContents);
+
+  return buffer.forget();
+}
+
+JSObject* AudioBuffer::WrapObject(JSContext* aCx,
+                                  JS::Handle<JSObject*> aGivenProto) {
+  return AudioBuffer_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+static void CopyChannelDataToFloat(const AudioChunk& aChunk, uint32_t aChannel,
+                                   uint32_t aSrcOffset, float* aOutput,
+                                   uint32_t aLength) {
+  MOZ_ASSERT(aChunk.mVolume == 1.0f);
+  if (aChunk.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
+    mozilla::PodCopy(
+        aOutput, aChunk.ChannelData<float>()[aChannel] + aSrcOffset, aLength);
+  } else {
+    MOZ_ASSERT(aChunk.mBufferFormat == AUDIO_FORMAT_S16);
+    ConvertAudioSamples(aChunk.ChannelData<int16_t>()[aChannel] + aSrcOffset,
+                        aOutput, aLength);
+  }
+}
+
+bool AudioBuffer::RestoreJSChannelData(JSContext* aJSContext) {
+  nsPIDOMWindowInner* global = GetParentObject();
+  if (!global || !global->AsGlobal()->HasJSGlobal()) {
+    return false;
+  }
+
+  JSAutoRealm ar(aJSContext, global->AsGlobal()->GetGlobalJSObject());
+
+  for (uint32_t i = 0; i < mJSChannels.Length(); ++i) {
+    if (mJSChannels[i]) {
+      // Already have data in JS array.
+      continue;
+    }
+
+    // The following code first zeroes the array and then copies our data
+    // into it. We could avoid this with additional JS APIs to construct
+    // an array (or ArrayBuffer) containing initial data.
+    JS::Rooted<JSObject*> array(aJSContext,
+                                JS_NewFloat32Array(aJSContext, Length()));
+    if (!array) {
+      return false;
+    }
+    if (!mSharedChannels.IsNull()) {
+      // "4. Attach ArrayBuffers containing copies of the data to the
+      // AudioBuffer, to be returned by the next call to getChannelData."
+      JS::AutoCheckCannotGC nogc;
+      bool isShared;
+      float* jsData = JS_GetFloat32ArrayData(array, &isShared, nogc);
+      MOZ_ASSERT(!isShared);  // Was created as unshared above
+      CopyChannelDataToFloat(mSharedChannels, i, 0, jsData, Length());
+    }
+    mJSChannels[i] = array;
+  }
+
+  mSharedChannels.SetNull(Length());
+
+  return true;
+}
+
+void AudioBuffer::CopyFromChannel(const Float32Array& aDestination,
+                                  uint32_t aChannelNumber,
+                                  uint32_t aBufferOffset, ErrorResult& aRv) {
+  if (aChannelNumber >= NumberOfChannels()) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("Channel number (%u) is out of range", aChannelNumber));
+    return;
+  }
+  uint32_t length = Length();
+  if (aBufferOffset >= length) {
+    return;
+  }
+  JS::AutoCheckCannotGC nogc;
+  MOZ_RELEASE_ASSERT(!JS_GetTypedArraySharedness(aDestination.Obj()));
+  auto calculateCount = [=](uint32_t aLength) -> uint32_t {
+    return std::min(length - aBufferOffset, aLength);
+  };
+
+  JSObject* channelArray = mJSChannels[aChannelNumber];
+  if (channelArray) {
+    if (JS_GetTypedArrayLength(channelArray) != length) {
+      // The array's buffer was detached.
+      return;
+    }
+    bool isShared = false;
+    const float* sourceData =
+        JS_GetFloat32ArrayData(channelArray, &isShared, nogc);
+    // The sourceData arrays should all have originated in
+    // RestoreJSChannelData, where they are created unshared.
+    MOZ_ASSERT(!isShared);
+    aDestination.ProcessData(
+        [&](const Span<float>& aData, JS::AutoCheckCannotGC&&) {
+          PodMove(aData.Elements(), sourceData + aBufferOffset,
+                  calculateCount(aData.Length()));
+        });
+    return;
+  }
+
+  if (!mSharedChannels.IsNull()) {
+    aDestination.ProcessData([&](const Span<float>& aData,
+                                 JS::AutoCheckCannotGC&&) {
+      CopyChannelDataToFloat(mSharedChannels, aChannelNumber, aBufferOffset,
+                             aData.Elements(), calculateCount(aData.Length()));
+    });
+    return;
+  }
+
+  aDestination.ProcessData(
+      [&](const Span<float>& aData, JS::AutoCheckCannotGC&&) {
+        PodZero(aData.Elements(), calculateCount(aData.Length()));
+      });
+}
+
+void AudioBuffer::CopyToChannel(JSContext* aJSContext,
+                                const Float32Array& aSource,
+                                uint32_t aChannelNumber, uint32_t aBufferOffset,
+                                ErrorResult& aRv) {
+  if (aChannelNumber >= NumberOfChannels()) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("Channel number (%u) is out of range", aChannelNumber));
+    return;
+  }
+
+  if (!RestoreJSChannelData(aJSContext)) {
+    aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+    return;
+  }
+
+  JS::AutoCheckCannotGC nogc;
+  JSObject* channelArray = mJSChannels[aChannelNumber];
+  // This may differ from Length() if the buffer has been detached.
+  uint32_t length = JS_GetTypedArrayLength(channelArray);
+  if (aBufferOffset >= length) {
+    return;
+  }
+
+  int64_t offset = aBufferOffset;
+  aSource.ProcessData([&](const Span<float>& aData, JS::AutoCheckCannotGC&&) {
+    MOZ_ASSERT_IF(std::numeric_limits<decltype(aData.Length())>::max() >
+                      std::numeric_limits<int64_t>::max(),
+                  aData.Length() <= std::numeric_limits<int64_t>::max());
+    int64_t srcLength = int64_t(aData.Length());
+    size_t count = std::max(int64_t(0), std::min(length - offset, srcLength));
+    bool isShared = false;
+    float* channelData = JS_GetFloat32ArrayData(channelArray, &isShared, nogc);
+    // The channelData arrays should all have originated in
+    // RestoreJSChannelData, where they are created unshared.
+    MOZ_ASSERT(!isShared);
+    PodMove(channelData + aBufferOffset, aData.Elements(), count);
+  });
+}
+
+void AudioBuffer::GetChannelData(JSContext* aJSContext, uint32_t aChannel,
+                                 JS::MutableHandle<JSObject*> aRetval,
+                                 ErrorResult& aRv) {
+  if (aChannel >= NumberOfChannels()) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("Channel number (%u) is out of range", aChannel));
+    return;
+  }
+
+  if (!RestoreJSChannelData(aJSContext)) {
+    aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+    return;
+  }
+
+  aRetval.set(mJSChannels[aChannel]);
+}
+
+already_AddRefed<ThreadSharedFloatArrayBufferList>
+AudioBuffer::StealJSArrayDataIntoSharedChannels(JSContext* aJSContext) {
+  nsPIDOMWindowInner* global = GetParentObject();
+  if (!global || !global->AsGlobal()->HasJSGlobal()) {
+    return nullptr;
+  }
+
+  JSAutoRealm ar(aJSContext, global->AsGlobal()->GetGlobalJSObject());
+
+  // "1. If any of the AudioBuffer's ArrayBuffer have been detached, abort
+  // these steps, and return a zero-length channel data buffers to the
+  // invoker."
+  for (uint32_t i = 0; i < mJSChannels.Length(); ++i) {
+    JSObject* channelArray = mJSChannels[i];
+    if (!channelArray || Length() != JS_GetTypedArrayLength(channelArray)) {
+      // Either empty buffer or one of the arrays' buffers was detached.
+      return nullptr;
+    }
+  }
+
+  // "2. Detach all ArrayBuffers for arrays previously returned by
+  // getChannelData on this AudioBuffer."
+  // "3. Retain the underlying data buffers from those ArrayBuffers and return
+  // references to them to the invoker."
+  RefPtr<ThreadSharedFloatArrayBufferList> result =
+      new ThreadSharedFloatArrayBufferList(mJSChannels.Length());
+  for (uint32_t i = 0; i < mJSChannels.Length(); ++i) {
+    JS::Rooted<JSObject*> arrayBufferView(aJSContext, mJSChannels[i]);
+    bool isSharedMemory;
+    JS::Rooted<JSObject*> arrayBuffer(
+        aJSContext, JS_GetArrayBufferViewBuffer(aJSContext, arrayBufferView,
+                                                &isSharedMemory));
+    // The channel data arrays should all have originated in
+    // RestoreJSChannelData, where they are created unshared.
+    MOZ_ASSERT(!isSharedMemory);
+    auto stolenData = arrayBuffer
+                          ? static_cast<float*>(JS::StealArrayBufferContents(
+                                aJSContext, arrayBuffer))
+                          : nullptr;
+    if (stolenData) {
+      result->SetData(i, stolenData, js_free, stolenData);
+    } else {
+      NS_ASSERTION(i == 0, "some channels lost when contents not acquired");
+      return nullptr;
+    }
+  }
+
+  for (uint32_t i = 0; i < mJSChannels.Length(); ++i) {
+    mJSChannels[i] = nullptr;
+  }
+
+  return result.forget();
+}
+
+const AudioChunk& AudioBuffer::GetThreadSharedChannelsForRate(
+    JSContext* aJSContext) {
+  if (mSharedChannels.IsNull()) {
+    // mDuration is set in constructor
+    RefPtr<ThreadSharedFloatArrayBufferList> buffer =
+        StealJSArrayDataIntoSharedChannels(aJSContext);
+
+    if (buffer) {
+      SetSharedChannels(buffer.forget());
+    }
+  }
+
+  return mSharedChannels;
+}
+
+size_t AudioBuffer::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = aMallocSizeOf(this);
+  amount += mJSChannels.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  amount += mSharedChannels.SizeOfExcludingThis(aMallocSizeOf, false);
+  return amount;
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioBuffer.h b/dom/media/webaudio/AudioBuffer.h
new file mode 100644
index 0000000000..e313c5c944
--- /dev/null
+++ b/dom/media/webaudio/AudioBuffer.h
@@ -0,0 +1,138 @@
+/* -*- 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/. */
+
+#ifndef AudioBuffer_h_
+#define AudioBuffer_h_
+
+#include "AudioSegment.h"
+#include "nsWrapperCache.h"
+#include "nsCycleCollectionParticipant.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/StaticPtr.h"
+#include "mozilla/StaticMutex.h"
+#include "nsTArray.h"
+#include "js/TypeDecls.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/dom/TypedArray.h"
+#include "nsPIDOMWindow.h"
+#include "nsIWeakReferenceUtils.h"
+
+namespace mozilla {
+
+class ErrorResult;
+class ThreadSharedFloatArrayBufferList;
+
+namespace dom {
+
+struct AudioBufferOptions;
+
+/**
+ * An AudioBuffer keeps its data either in the mJSChannels objects, which
+ * are Float32Arrays, or in mSharedChannels if the mJSChannels objects' buffers
+ * are detached.
+ */
+class AudioBuffer final : public nsWrapperCache {
+ public:
+  // If non-null, aInitialContents must have number of channels equal to
+  // aNumberOfChannels and their lengths must be at least aLength.
+  static already_AddRefed<AudioBuffer> Create(
+      nsPIDOMWindowInner* aWindow, uint32_t aNumberOfChannels, uint32_t aLength,
+      float aSampleRate,
+      already_AddRefed<ThreadSharedFloatArrayBufferList> aInitialContents,
+      ErrorResult& aRv);
+
+  static already_AddRefed<AudioBuffer> Create(nsPIDOMWindowInner* aWindow,
+                                              uint32_t aNumberOfChannels,
+                                              uint32_t aLength,
+                                              float aSampleRate,
+                                              ErrorResult& aRv) {
+    return Create(aWindow, aNumberOfChannels, aLength, aSampleRate, nullptr,
+                  aRv);
+  }
+
+  // Non-unit AudioChunk::mVolume is not supported
+  static already_AddRefed<AudioBuffer> Create(nsPIDOMWindowInner* aWindow,
+                                              float aSampleRate,
+                                              AudioChunk&& aInitialContents);
+
+  size_t SizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+  NS_INLINE_DECL_CYCLE_COLLECTING_NATIVE_REFCOUNTING(AudioBuffer)
+  NS_DECL_CYCLE_COLLECTION_SCRIPT_HOLDER_NATIVE_CLASS(AudioBuffer)
+
+  static already_AddRefed<AudioBuffer> Constructor(
+      const GlobalObject& aGlobal, const AudioBufferOptions& aOptions,
+      ErrorResult& aRv);
+
+  nsPIDOMWindowInner* GetParentObject() const {
+    nsCOMPtr<nsPIDOMWindowInner> parentObject = do_QueryReferent(mOwnerWindow);
+    return parentObject;
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  float SampleRate() const { return mSampleRate; }
+
+  uint32_t Length() const { return mSharedChannels.mDuration; }
+
+  double Duration() const {
+    return Length() / static_cast<double>(mSampleRate);
+  }
+
+  uint32_t NumberOfChannels() const { return mJSChannels.Length(); }
+
+  /**
+   * If mSharedChannels is non-null, copies its contents to
+   * new Float32Arrays in mJSChannels. Returns a Float32Array.
+   */
+  void GetChannelData(JSContext* aJSContext, uint32_t aChannel,
+                      JS::MutableHandle<JSObject*> aRetval, ErrorResult& aRv);
+
+  void CopyFromChannel(const Float32Array& aDestination,
+                       uint32_t aChannelNumber, uint32_t aBufferOffset,
+                       ErrorResult& aRv);
+  void CopyToChannel(JSContext* aJSContext, const Float32Array& aSource,
+                     uint32_t aChannelNumber, uint32_t aBufferOffset,
+                     ErrorResult& aRv);
+
+  /**
+   * Returns a reference to an AudioChunk containing the sample data.
+   * The AudioChunk can have a null buffer if there is no data.
+   */
+  const AudioChunk& GetThreadSharedChannelsForRate(JSContext* aContext);
+
+ protected:
+  AudioBuffer(nsPIDOMWindowInner* aWindow, uint32_t aNumberOfChannels,
+              uint32_t aLength, float aSampleRate, ErrorResult& aRv);
+  ~AudioBuffer();
+
+  void SetSharedChannels(
+      already_AddRefed<ThreadSharedFloatArrayBufferList> aBuffer);
+
+  bool RestoreJSChannelData(JSContext* aJSContext);
+
+  already_AddRefed<ThreadSharedFloatArrayBufferList>
+  StealJSArrayDataIntoSharedChannels(JSContext* aJSContext);
+
+  void ClearJSChannels();
+
+  // Float32Arrays
+  AutoTArray<JS::Heap<JSObject*>, 2> mJSChannels;
+  // mSharedChannels aggregates the data from mJSChannels. This is non-null
+  // if and only if the mJSChannels' buffers are detached, but its mDuration
+  // member keeps the buffer length regardless of whether the buffer is
+  // provided by mJSChannels or mSharedChannels.
+  AudioChunk mSharedChannels;
+
+  nsWeakPtr mOwnerWindow;
+  float mSampleRate;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/AudioBufferSourceNode.cpp b/dom/media/webaudio/AudioBufferSourceNode.cpp
new file mode 100644
index 0000000000..db34c49f48
--- /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 "AudioNodeTrack.h"
+#include "AudioDestinationNode.h"
+#include "AudioParamTimeline.h"
+#include <limits>
+#include <algorithm>
+#include "Tracing.h"
+
+namespace mozilla::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
+ * AudioNodeTrack::SetBuffer) and a non-zero mBufferSampleRate has been set
+ * (via AudioNodeTrack::SetInt32Parameter)
+ */
+class AudioBufferSourceNodeEngine final : public AudioNodeEngine {
+ public:
+  AudioBufferSourceNodeEngine(AudioNode* aNode,
+                              AudioDestinationNode* aDestination)
+      : AudioNodeEngine(aNode),
+        mStart(0.0),
+        mBeginProcessing(0),
+        mStop(TRACK_TIME_MAX),
+        mResampler(nullptr),
+        mRemainingResamplerTail(0),
+        mRemainingFrames(TRACK_TICKS_MAX),
+        mLoopStart(0),
+        mLoopEnd(0),
+        mBufferPosition(0),
+        mBufferSampleRate(0),
+        // mResamplerOutRate is initialized in UpdateResampler().
+        mChannels(0),
+        mDestination(aDestination->Track()),
+        mPlaybackRateTimeline(1.0f),
+        mDetuneTimeline(0.0f),
+        mLoop(false) {}
+
+  ~AudioBufferSourceNodeEngine() {
+    if (mResampler) {
+      speex_resampler_destroy(mResampler);
+    }
+  }
+
+  void SetSourceTrack(AudioNodeTrack* aSource) { mSource = aSource; }
+
+  void RecvTimelineEvent(uint32_t aIndex, AudioParamEvent& aEvent) override {
+    MOZ_ASSERT(mDestination);
+    aEvent.ConvertToTicks(mDestination);
+    mRecomputeOutRate = true;
+
+    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 SetTrackTimeParameter(uint32_t aIndex, TrackTime aParam) override {
+    switch (aIndex) {
+      case AudioBufferSourceNode::STOP:
+        mStop = aParam;
+        break;
+      default:
+        NS_ERROR("Bad AudioBufferSourceNodeEngine TrackTimeParameter");
+    }
+  }
+  void SetDoubleParameter(uint32_t aIndex, double aParam) override {
+    switch (aIndex) {
+      case AudioBufferSourceNode::START:
+        MOZ_ASSERT(!mStart, "Another START?");
+        mStart = aParam * mDestination->mSampleRate;
+        // Round to nearest
+        mBeginProcessing = llround(mStart);
+        break;
+      case AudioBufferSourceNode::DURATION:
+        MOZ_ASSERT(aParam >= 0);
+        mRemainingFrames = llround(aParam * mBufferSampleRate);
+        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);
+        MOZ_ASSERT(mRecomputeOutRate);
+        mBufferSampleRate = aParam;
+        mSource->SetActive();
+        break;
+      case AudioBufferSourceNode::BUFFERSTART:
+        MOZ_ASSERT(aParam >= 0);
+        if (mBufferPosition == 0) {
+          mBufferPosition = 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 == -TRACK_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 = llround(mStart);
+    }
+
+    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 = llround(mStart * ratioNum);
+      // 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,
+                                         TrackTime* 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 = llround(leadTicks * ratioNum);
+          MOZ_ASSERT(leadSubsamples <= skipFracNum,
+                     "mBeginProcessing is wrong?");
+          skipFracNum -= leadSubsamples;
+        }
+        speex_resampler_set_skip_frac_num(
+            resampler, std::min<int64_t>(skipFracNum, UINT32_MAX));
+
+        mBeginProcessing = -TRACK_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;
+          mRemainingFrames -= inSamples;
+          MOZ_ASSERT(mBufferPosition <= mBuffer.GetDuration());
+          MOZ_ASSERT(mRemainingFrames >= 0);
+          *aOffsetWithinBlock += outSamples;
+          *aCurrentPosition += outSamples;
+          if ((!mLoop && inSamples == availableInInputBuffer) ||
+              mRemainingFrames == 0) {
+            // 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, TrackTime* aCurrentPosition,
+                      TrackTime aMaxPos) {
+    MOZ_ASSERT(*aCurrentPosition < aMaxPos);
+    uint32_t numFrames = std::min<TrackTime>(
+        WEBAUDIO_BLOCK_SIZE - *aOffsetWithinBlock, aMaxPos - *aCurrentPosition);
+    if (numFrames == WEBAUDIO_BLOCK_SIZE || !aChannels) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+    } 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, TrackTime* aCurrentPosition,
+                      uint32_t aBufferMax) {
+    MOZ_ASSERT(*aCurrentPosition < mStop);
+    uint32_t availableInOutput = std::min<TrackTime>(
+        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;
+    mRemainingFrames -= numFrames;
+  }
+
+  int32_t ComputeFinalOutSampleRate(float aPlaybackRate, float aDetune) {
+    float computedPlaybackRate = aPlaybackRate * fdlibm_exp2f(aDetune / 1200.f);
+    // Make sure the playback rate is something our resampler can work with.
+    int32_t rate = WebAudioUtils::TruncateFloatToInt<int32_t>(
+        mSource->mSampleRate / computedPlaybackRate);
+    return rate ? rate : mBufferSampleRate;
+  }
+
+  void UpdateSampleRateIfNeeded(uint32_t aChannels, TrackTime aTrackPosition) {
+    bool simplePlaybackRate = mPlaybackRateTimeline.HasSimpleValue();
+    bool simpleDetune = mDetuneTimeline.HasSimpleValue();
+
+    if (simplePlaybackRate && simpleDetune && !mRecomputeOutRate) {
+      return;  // skipping the slow exp2f() for the detune
+    }
+    mRecomputeOutRate = false;
+
+    float playbackRate;
+    float detune;
+    if (simplePlaybackRate) {
+      playbackRate = mPlaybackRateTimeline.GetValue();
+    } else {
+      playbackRate =
+          mPlaybackRateTimeline.GetComplexValueAtTime(aTrackPosition);
+    }
+    if (simpleDetune) {
+      detune = mDetuneTimeline.GetValue();
+    } else {
+      detune = mDetuneTimeline.GetComplexValueAtTime(aTrackPosition);
+    }
+    if (playbackRate <= 0 || std::isnan(playbackRate)) {
+      playbackRate = 1.0f;
+    }
+
+    int32_t outRate = ComputeFinalOutSampleRate(playbackRate, detune);
+    UpdateResampler(outRate, aChannels);
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    TRACE("AudioBufferSourceNodeEngine::ProcessBlock");
+    if (mBufferSampleRate == 0) {
+      // start() has not yet been called or no buffer has yet been set
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      return;
+    }
+
+    TrackTime streamPosition = mDestination->GraphTimeToTrackTime(aFrom);
+    uint32_t channels = mBuffer.ChannelCount();
+
+    UpdateSampleRateIfNeeded(channels, streamPosition);
+
+    uint32_t written = 0;
+    while (true) {
+      if ((mStop != TRACK_TIME_MAX && streamPosition >= mStop) ||
+          (!mRemainingResamplerTail &&
+           ((mBufferPosition >= mBuffer.GetDuration() && !mLoop) ||
+            mRemainingFrames <= 0))) {
+        if (written != WEBAUDIO_BLOCK_SIZE) {
+          FillWithZeroes(aOutput, channels, &written, &streamPosition,
+                         TRACK_TIME_MAX);
+        }
+        *aFinished = true;
+        break;
+      }
+      if (written == WEBAUDIO_BLOCK_SIZE) {
+        break;
+      }
+      if (streamPosition < mBeginProcessing) {
+        FillWithZeroes(aOutput, channels, &written, &streamPosition,
+                       mBeginProcessing);
+        continue;
+      }
+
+      TrackTicks bufferLeft;
+      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;
+        }
+        bufferLeft =
+            std::min<TrackTicks>(mRemainingFrames, mLoopEnd - mBufferPosition);
+      } else {
+        bufferLeft =
+            std::min(mRemainingFrames, mBuffer.GetDuration() - mBufferPosition);
+      }
+
+      CopyFromBuffer(aOutput, channels, &written, &streamPosition,
+                     bufferLeft + mBufferPosition);
+    }
+  }
+
+  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 -TRACK_TIME_MAX
+  // indicates that the resampler has begun processing.
+  TrackTime mBeginProcessing;
+  TrackTime mStop;
+  AudioChunk mBuffer;
+  SpeexResamplerState* mResampler;
+  // mRemainingResamplerTail, like mBufferPosition
+  // is measured in input buffer samples.
+  uint32_t mRemainingResamplerTail;
+  TrackTicks mRemainingFrames;
+  uint32_t mLoopStart;
+  uint32_t mLoopEnd;
+  uint32_t mBufferPosition;
+  int32_t mBufferSampleRate;
+  int32_t mResamplerOutRate;
+  uint32_t mChannels;
+  RefPtr<AudioNodeTrack> mDestination;
+
+  // mSource deletes the engine in its destructor.
+  AudioNodeTrack* MOZ_NON_OWNING_REF mSource;
+  AudioParamTimeline mPlaybackRateTimeline;
+  AudioParamTimeline mDetuneTimeline;
+  bool mLoop;
+  bool mRecomputeOutRate = true;
+};
+
+AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* aContext)
+    : AudioScheduledSourceNode(aContext, 2, ChannelCountMode::Max,
+                               ChannelInterpretation::Speakers),
+      mLoopStart(0.0),
+      mLoopEnd(0.0),
+      // mOffset and mDuration are initialized in Start().
+      mLoop(false),
+      mStartCalled(false),
+      mBufferSet(false) {
+  mPlaybackRate = CreateAudioParam(PLAYBACKRATE, u"playbackRate"_ns, 1.0f);
+  mDetune = CreateAudioParam(DETUNE, u"detune"_ns, 0.0f);
+  AudioBufferSourceNodeEngine* engine =
+      new AudioBufferSourceNodeEngine(this, aContext->Destination());
+  mTrack = AudioNodeTrack::Create(aContext, engine,
+                                  AudioNodeTrack::NEED_MAIN_THREAD_ENDED,
+                                  aContext->Graph());
+  engine->SetSourceTrack(mTrack);
+  mTrack->AddMainThreadListener(this);
+}
+
+/* static */
+already_AddRefed<AudioBufferSourceNode> AudioBufferSourceNode::Create(
+    JSContext* aCx, AudioContext& aAudioContext,
+    const AudioBufferSourceOptions& aOptions) {
+  RefPtr<AudioBufferSourceNode> audioNode =
+      new AudioBufferSourceNode(&aAudioContext);
+
+  if (aOptions.mBuffer.WasPassed()) {
+    ErrorResult ignored;
+    MOZ_ASSERT(aCx);
+    audioNode->SetBuffer(aCx, aOptions.mBuffer.Value(), ignored);
+  }
+
+  audioNode->Detune()->SetInitialValue(aOptions.mDetune);
+  audioNode->SetLoop(aOptions.mLoop);
+  audioNode->SetLoopEnd(aOptions.mLoopEnd);
+  audioNode->SetLoopStart(aOptions.mLoopStart);
+  audioNode->PlaybackRate()->SetInitialValue(aOptions.mPlaybackRate);
+
+  return audioNode.forget();
+}
+void AudioBufferSourceNode::DestroyMediaTrack() {
+  bool hadTrack = mTrack;
+  if (hadTrack) {
+    mTrack->RemoveMainThreadListener(this);
+  }
+  AudioNode::DestroyMediaTrack();
+}
+
+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>("start time");
+    return;
+  }
+  if (aOffset < 0) {
+    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>("offset");
+    return;
+  }
+  if (aDuration.WasPassed() && !WebAudioUtils::IsTimeValid(aDuration.Value())) {
+    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>("duration");
+    return;
+  }
+
+  if (mStartCalled) {
+    aRv.ThrowInvalidStateError(
+        "Start has already been called on this AudioBufferSourceNode.");
+    return;
+  }
+  mStartCalled = true;
+
+  AudioNodeTrack* ns = mTrack;
+  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) {
+    SendOffsetAndDurationParametersToTrack(ns);
+  }
+
+  // Don't set parameter unnecessarily
+  if (aWhen > 0.0) {
+    ns->SetDoubleParameter(START, aWhen);
+  }
+
+  Context()->StartBlockedAudioContextIfAllowed();
+}
+
+void AudioBufferSourceNode::Start(double aWhen, ErrorResult& aRv) {
+  Start(aWhen, 0 /* offset */, Optional<double>(), aRv);
+}
+
+void AudioBufferSourceNode::SendBufferParameterToTrack(JSContext* aCx) {
+  AudioNodeTrack* ns = mTrack;
+  if (!ns) {
+    return;
+  }
+
+  if (mBuffer) {
+    AudioChunk data = mBuffer->GetThreadSharedChannelsForRate(aCx);
+    ns->SetBuffer(std::move(data));
+
+    if (mStartCalled) {
+      SendOffsetAndDurationParametersToTrack(ns);
+    }
+  } else {
+    ns->SetBuffer(AudioChunk());
+
+    MarkInactive();
+  }
+}
+
+void AudioBufferSourceNode::SendOffsetAndDurationParametersToTrack(
+    AudioNodeTrack* aTrack) {
+  NS_ASSERTION(
+      mBuffer && mStartCalled,
+      "Only call this when we have a buffer and start() has been called");
+
+  float rate = mBuffer->SampleRate();
+  aTrack->SetInt32Parameter(SAMPLE_RATE, rate);
+
+  int32_t offsetSamples = std::max(0, NS_lround(mOffset * rate));
+
+  // Don't set parameter unnecessarily
+  if (offsetSamples > 0) {
+    aTrack->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()
+    aTrack->SetDoubleParameter(DURATION, mDuration);
+  }
+  MarkActive();
+}
+
+void AudioBufferSourceNode::Stop(double aWhen, ErrorResult& aRv) {
+  if (!WebAudioUtils::IsTimeValid(aWhen)) {
+    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>("stop time");
+    return;
+  }
+
+  if (!mStartCalled) {
+    aRv.ThrowInvalidStateError(
+        "Start has not been called on this AudioBufferSourceNode.");
+    return;
+  }
+
+  WEB_AUDIO_API_LOG("%f: %s %u Stop(%f)", Context()->CurrentTime(), NodeType(),
+                    Id(), aWhen);
+
+  AudioNodeTrack* ns = mTrack;
+  if (!ns || !Context()) {
+    // We've already stopped and had our track shut down
+    return;
+  }
+
+  ns->SetTrackTimeParameter(STOP, Context(), std::max(0.0, aWhen));
+}
+
+void AudioBufferSourceNode::NotifyMainThreadTrackEnded() {
+  MOZ_ASSERT(mTrack->IsEnded());
+
+  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(u"ended"_ns);
+      // Release track resources.
+      mNode->DestroyMediaTrack();
+      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::SendLoopParametersToTrack() {
+  if (!mTrack) {
+    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) {
+      SendInt32ParameterToTrack(LOOPSTART, loopStartTicks);
+      SendInt32ParameterToTrack(LOOPEND, loopEndTicks);
+      SendInt32ParameterToTrack(LOOP, 1);
+    } else {
+      // Be explicit about looping not happening if the offsets make
+      // looping impossible.
+      SendInt32ParameterToTrack(LOOP, 0);
+    }
+  } else {
+    SendInt32ParameterToTrack(LOOP, 0);
+  }
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioBufferSourceNode.h b/dom/media/webaudio/AudioBufferSourceNode.h
new file mode 100644
index 0000000000..9a3861555c
--- /dev/null
+++ b/dom/media/webaudio/AudioBufferSourceNode.h
@@ -0,0 +1,129 @@
+/* -*- 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/. */
+
+#ifndef AudioBufferSourceNode_h_
+#define AudioBufferSourceNode_h_
+
+#include "AudioScheduledSourceNode.h"
+#include "AudioBuffer.h"
+
+namespace mozilla::dom {
+
+struct AudioBufferSourceOptions;
+class AudioParam;
+
+class AudioBufferSourceNode final : public AudioScheduledSourceNode,
+                                    public MainThreadMediaTrackListener {
+ public:
+  static already_AddRefed<AudioBufferSourceNode> Create(
+      JSContext* aCx, AudioContext& aAudioContext,
+      const AudioBufferSourceOptions& aOptions);
+
+  void DestroyMediaTrack() override;
+
+  uint16_t NumberOfInputs() const final { return 0; }
+  AudioBufferSourceNode* AsAudioBufferSourceNode() override { return this; }
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioBufferSourceNode,
+                                           AudioScheduledSourceNode)
+
+  static already_AddRefed<AudioBufferSourceNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const AudioBufferSourceOptions& aOptions) {
+    return Create(aGlobal.Context(), aAudioContext, aOptions);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  void Start(double aWhen, double aOffset, const Optional<double>& aDuration,
+             ErrorResult& aRv);
+
+  void Start(double aWhen, ErrorResult& aRv) override;
+  void Stop(double aWhen, ErrorResult& aRv) override;
+
+  AudioBuffer* GetBuffer(JSContext* aCx) const { return mBuffer; }
+  void SetBuffer(JSContext* aCx, AudioBuffer* aBuffer, ErrorResult& aRv) {
+    if (aBuffer && mBufferSet) {
+      aRv.ThrowInvalidStateError(
+          "Cannot set the buffer attribute of an AudioBufferSourceNode "
+          "with an AudioBuffer more than once");
+      return;
+    }
+    if (aBuffer) {
+      mBufferSet = true;
+    }
+    mBuffer = aBuffer;
+    SendBufferParameterToTrack(aCx);
+    SendLoopParametersToTrack();
+  }
+  AudioParam* PlaybackRate() const { return mPlaybackRate; }
+  AudioParam* Detune() const { return mDetune; }
+  bool Loop() const { return mLoop; }
+  void SetLoop(bool aLoop) {
+    mLoop = aLoop;
+    SendLoopParametersToTrack();
+  }
+  double LoopStart() const { return mLoopStart; }
+  void SetLoopStart(double aStart) {
+    mLoopStart = aStart;
+    SendLoopParametersToTrack();
+  }
+  double LoopEnd() const { return mLoopEnd; }
+  void SetLoopEnd(double aEnd) {
+    mLoopEnd = aEnd;
+    SendLoopParametersToTrack();
+  }
+  void NotifyMainThreadTrackEnded() override;
+
+  const char* NodeType() const override { return "AudioBufferSourceNode"; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ private:
+  explicit AudioBufferSourceNode(AudioContext* aContext);
+  ~AudioBufferSourceNode() = default;
+
+  friend class AudioBufferSourceNodeEngine;
+  // START is sent during Start().
+  // STOP is sent during Stop().
+  // BUFFERSTART and DURATION are sent when SetBuffer() and Start() have
+  // been called (along with sending the buffer).
+  enum EngineParameters {
+    SAMPLE_RATE,
+    START,
+    STOP,
+    // BUFFERSTART is the "offset" passed to start(), multiplied by
+    // buffer.sampleRate.
+    BUFFERSTART,
+    DURATION,
+    LOOP,
+    LOOPSTART,
+    LOOPEND,
+    PLAYBACKRATE,
+    DETUNE
+  };
+
+  void SendLoopParametersToTrack();
+  void SendBufferParameterToTrack(JSContext* aCx);
+  void SendOffsetAndDurationParametersToTrack(AudioNodeTrack* aTrack);
+
+  double mLoopStart;
+  double mLoopEnd;
+  double mOffset;
+  double mDuration;
+  RefPtr<AudioBuffer> mBuffer;
+  RefPtr<AudioParam> mPlaybackRate;
+  RefPtr<AudioParam> mDetune;
+  bool mLoop;
+  bool mStartCalled;
+  bool mBufferSet;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/AudioContext.cpp b/dom/media/webaudio/AudioContext.cpp
new file mode 100644
index 0000000000..2ef5fdd65b
--- /dev/null
+++ b/dom/media/webaudio/AudioContext.cpp
@@ -0,0 +1,1405 @@
+/* -*- 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 "AudioContext.h"
+
+#include "blink/PeriodicWave.h"
+
+#include "mozilla/ErrorResult.h"
+#include "mozilla/NotNull.h"
+#include "mozilla/OwningNonNull.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/Preferences.h"
+#include "mozilla/StaticPrefs_media.h"
+
+#include "mozilla/dom/AnalyserNode.h"
+#include "mozilla/dom/AnalyserNodeBinding.h"
+#include "mozilla/dom/AudioBufferSourceNodeBinding.h"
+#include "mozilla/dom/AudioContextBinding.h"
+#include "mozilla/dom/BaseAudioContextBinding.h"
+#include "mozilla/dom/BiquadFilterNodeBinding.h"
+#include "mozilla/dom/BrowsingContext.h"
+#include "mozilla/dom/CanonicalBrowsingContext.h"
+#include "mozilla/dom/ChannelMergerNodeBinding.h"
+#include "mozilla/dom/ChannelSplitterNodeBinding.h"
+#include "mozilla/dom/ContentChild.h"
+#include "mozilla/dom/ConvolverNodeBinding.h"
+#include "mozilla/dom/DelayNodeBinding.h"
+#include "mozilla/dom/DynamicsCompressorNodeBinding.h"
+#include "mozilla/dom/GainNodeBinding.h"
+#include "mozilla/dom/IIRFilterNodeBinding.h"
+#include "mozilla/dom/HTMLMediaElement.h"
+#include "mozilla/dom/MediaElementAudioSourceNodeBinding.h"
+#include "mozilla/dom/MediaStreamAudioSourceNodeBinding.h"
+#include "mozilla/dom/MediaStreamTrackAudioSourceNodeBinding.h"
+#include "mozilla/dom/OfflineAudioContextBinding.h"
+#include "mozilla/dom/OscillatorNodeBinding.h"
+#include "mozilla/dom/PannerNodeBinding.h"
+#include "mozilla/dom/PeriodicWaveBinding.h"
+#include "mozilla/dom/Performance.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/dom/StereoPannerNodeBinding.h"
+#include "mozilla/dom/WaveShaperNodeBinding.h"
+#include "mozilla/dom/Worklet.h"
+
+#include "AudioBuffer.h"
+#include "AudioBufferSourceNode.h"
+#include "AudioChannelService.h"
+#include "AudioDestinationNode.h"
+#include "AudioListener.h"
+#include "AudioNodeTrack.h"
+#include "AudioStream.h"
+#include "AudioWorkletImpl.h"
+#include "AutoplayPolicy.h"
+#include "BiquadFilterNode.h"
+#include "ChannelMergerNode.h"
+#include "ChannelSplitterNode.h"
+#include "ConstantSourceNode.h"
+#include "ConvolverNode.h"
+#include "DelayNode.h"
+#include "DynamicsCompressorNode.h"
+#include "GainNode.h"
+#include "IIRFilterNode.h"
+#include "js/ArrayBuffer.h"  // JS::StealArrayBufferContents
+#include "MediaElementAudioSourceNode.h"
+#include "MediaStreamAudioDestinationNode.h"
+#include "MediaStreamAudioSourceNode.h"
+#include "MediaTrackGraph.h"
+#include "MediaStreamTrackAudioSourceNode.h"
+#include "nsContentUtils.h"
+#include "nsIScriptError.h"
+#include "nsNetCID.h"
+#include "nsNetUtil.h"
+#include "nsPIDOMWindow.h"
+#include "nsPrintfCString.h"
+#include "nsRFPService.h"
+#include "OscillatorNode.h"
+#include "PannerNode.h"
+#include "PeriodicWave.h"
+#include "ScriptProcessorNode.h"
+#include "StereoPannerNode.h"
+#include "WaveShaperNode.h"
+#include "Tracing.h"
+
+extern mozilla::LazyLogModule gAutoplayPermissionLog;
+
+#define AUTOPLAY_LOG(msg, ...) \
+  MOZ_LOG(gAutoplayPermissionLog, LogLevel::Debug, (msg, ##__VA_ARGS__))
+
+namespace mozilla::dom {
+
+// 0 is a special value that MediaTracks use to denote they are not part of a
+// AudioContext.
+static dom::AudioContext::AudioContextId gAudioContextId = 1;
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(AudioContext)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(AudioContext)
+  // The destination node and AudioContext form a cycle and so the destination
+  // track will be destroyed.  mWorklet must be shut down before the track
+  // is destroyed.  Do this before clearing mWorklet.
+  tmp->ShutdownWorklet();
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mDestination)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mListener)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mWorklet)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mPromiseGripArray)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mPendingResumePromises)
+  if (tmp->mTracksAreSuspended || !tmp->mIsStarted) {
+    NS_IMPL_CYCLE_COLLECTION_UNLINK(mActiveNodes)
+  }
+  // mDecodeJobs owns the WebAudioDecodeJob objects whose lifetime is managed
+  // explicitly. mAllNodes is an array of weak pointers, ignore it here.
+  // mBasicWaveFormCache cannot participate in cycles, ignore it here.
+
+  // Remove weak reference on the global window as the context is not usable
+  // without mDestination.
+  tmp->DisconnectFromWindow();
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END_INHERITED(DOMEventTargetHelper)
+
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(AudioContext,
+                                                  DOMEventTargetHelper)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mDestination)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mListener)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mWorklet)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mPromiseGripArray)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mPendingResumePromises)
+  if (tmp->mTracksAreSuspended || !tmp->mIsStarted) {
+    NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mActiveNodes)
+  }
+  // mDecodeJobs owns the WebAudioDecodeJob objects whose lifetime is managed
+  // explicitly. mAllNodes is an array of weak pointers, ignore it here.
+  // mBasicWaveFormCache cannot participate in cycles, ignore it here.
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_ADDREF_INHERITED(AudioContext, DOMEventTargetHelper)
+NS_IMPL_RELEASE_INHERITED(AudioContext, DOMEventTargetHelper)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(AudioContext)
+  NS_INTERFACE_MAP_ENTRY(nsIMemoryReporter)
+NS_INTERFACE_MAP_END_INHERITING(DOMEventTargetHelper)
+
+static float GetSampleRateForAudioContext(bool aIsOffline, float aSampleRate,
+                                          bool aShouldResistFingerprinting) {
+  if (aIsOffline || aSampleRate != 0.0) {
+    return aSampleRate;
+  } else {
+    return static_cast<float>(
+        CubebUtils::PreferredSampleRate(aShouldResistFingerprinting));
+  }
+}
+
+AudioContext::AudioContext(nsPIDOMWindowInner* aWindow, bool aIsOffline,
+                           uint32_t aNumberOfChannels, uint32_t aLength,
+                           float aSampleRate)
+    : DOMEventTargetHelper(aWindow),
+      mId(gAudioContextId++),
+      mSampleRate(GetSampleRateForAudioContext(
+          aIsOffline, aSampleRate,
+          aWindow->AsGlobal()->ShouldResistFingerprinting(
+              RFPTarget::AudioSampleRate))),
+      mAudioContextState(AudioContextState::Suspended),
+      mNumberOfChannels(aNumberOfChannels),
+      mRTPCallerType(aWindow->AsGlobal()->GetRTPCallerType()),
+      mShouldResistFingerprinting(
+          aWindow->AsGlobal()->ShouldResistFingerprinting(
+              RFPTarget::AudioContext)),
+      mIsOffline(aIsOffline),
+      mIsStarted(!aIsOffline),
+      mIsShutDown(false),
+      mIsDisconnecting(false),
+      mCloseCalled(false),
+      // Realtime contexts start with suspended tracks until an
+      // AudioCallbackDriver is running.
+      mTracksAreSuspended(!aIsOffline),
+      mWasAllowedToStart(true),
+      mSuspendedByContent(false),
+      mSuspendedByChrome(aWindow->IsSuspended()),
+      mWasEverAllowedToStart(false),
+      mWasEverBlockedToStart(false),
+      mWouldBeAllowedToStart(true) {
+  bool mute = aWindow->AddAudioContext(this);
+
+  // Note: AudioDestinationNode needs an AudioContext that must already be
+  // bound to the window.
+  const bool allowedToStart = media::AutoplayPolicy::IsAllowedToPlay(*this);
+  mDestination =
+      new AudioDestinationNode(this, aIsOffline, aNumberOfChannels, aLength);
+  mDestination->Init();
+  // If an AudioContext is not allowed to start, we would postpone its state
+  // transition from `suspended` to `running` until sites explicitly call
+  // AudioContext.resume() or AudioScheduledSourceNode.start().
+  if (!allowedToStart) {
+    MOZ_ASSERT(!mIsOffline);
+    AUTOPLAY_LOG("AudioContext %p is not allowed to start", this);
+    ReportBlocked();
+  } else if (!mIsOffline) {
+    ResumeInternal();
+  }
+
+  // The context can't be muted until it has a destination.
+  if (mute) {
+    Mute();
+  }
+
+  UpdateAutoplayAssumptionStatus();
+
+  FFTBlock::MainThreadInit();
+}
+
+void AudioContext::StartBlockedAudioContextIfAllowed() {
+  MOZ_ASSERT(NS_IsMainThread());
+  MaybeUpdateAutoplayTelemetry();
+  // Only try to start AudioContext when AudioContext was not allowed to start.
+  if (mWasAllowedToStart) {
+    return;
+  }
+
+  const bool isAllowedToPlay = media::AutoplayPolicy::IsAllowedToPlay(*this);
+  AUTOPLAY_LOG("Trying to start AudioContext %p, IsAllowedToPlay=%d", this,
+               isAllowedToPlay);
+
+  // Only start the AudioContext if this resume() call was initiated by content,
+  // not if it was a result of the AudioContext starting after having been
+  // blocked because of the auto-play policy.
+  if (isAllowedToPlay && !mSuspendedByContent) {
+    ResumeInternal();
+  } else {
+    ReportBlocked();
+  }
+}
+
+void AudioContext::DisconnectFromWindow() {
+  MaybeClearPageAwakeRequest();
+  nsPIDOMWindowInner* window = GetOwner();
+  if (window) {
+    window->RemoveAudioContext(this);
+  }
+}
+
+AudioContext::~AudioContext() {
+  DisconnectFromWindow();
+  UnregisterWeakMemoryReporter(this);
+  MOZ_ASSERT(!mSetPageAwakeRequest, "forgot to revoke for page awake?");
+}
+
+JSObject* AudioContext::WrapObject(JSContext* aCx,
+                                   JS::Handle<JSObject*> aGivenProto) {
+  if (mIsOffline) {
+    return OfflineAudioContext_Binding::Wrap(aCx, this, aGivenProto);
+  } else {
+    return AudioContext_Binding::Wrap(aCx, this, aGivenProto);
+  }
+}
+
+static bool CheckFullyActive(nsPIDOMWindowInner* aWindow, ErrorResult& aRv) {
+  if (!aWindow->IsFullyActive()) {
+    aRv.ThrowInvalidStateError("The document is not fully active.");
+    return false;
+  }
+  return true;
+}
+
+/* static */
+already_AddRefed<AudioContext> AudioContext::Constructor(
+    const GlobalObject& aGlobal, const AudioContextOptions& aOptions,
+    ErrorResult& aRv) {
+  nsCOMPtr<nsPIDOMWindowInner> window =
+      do_QueryInterface(aGlobal.GetAsSupports());
+  if (!window) {
+    aRv.Throw(NS_ERROR_FAILURE);
+    return nullptr;
+  }
+  /**
+   * If the current settings object’s responsible document is NOT fully
+   * active, throw an InvalidStateError and abort these steps.
+   */
+  if (!CheckFullyActive(window, aRv)) {
+    return nullptr;
+  }
+
+  if (aOptions.mSampleRate.WasPassed() &&
+      (aOptions.mSampleRate.Value() < WebAudioUtils::MinSampleRate ||
+       aOptions.mSampleRate.Value() > WebAudioUtils::MaxSampleRate)) {
+    aRv.ThrowNotSupportedError(nsPrintfCString(
+        "Sample rate %g is not in the range [%u, %u]",
+        aOptions.mSampleRate.Value(), WebAudioUtils::MinSampleRate,
+        WebAudioUtils::MaxSampleRate));
+    return nullptr;
+  }
+  float sampleRate = aOptions.mSampleRate.WasPassed()
+                         ? aOptions.mSampleRate.Value()
+                         : MediaTrackGraph::REQUEST_DEFAULT_SAMPLE_RATE;
+
+  RefPtr<AudioContext> object =
+      new AudioContext(window, false, 2, 0, sampleRate);
+
+  RegisterWeakMemoryReporter(object);
+
+  return object.forget();
+}
+
+/* static */
+already_AddRefed<AudioContext> AudioContext::Constructor(
+    const GlobalObject& aGlobal, const OfflineAudioContextOptions& aOptions,
+    ErrorResult& aRv) {
+  return Constructor(aGlobal, aOptions.mNumberOfChannels, aOptions.mLength,
+                     aOptions.mSampleRate, aRv);
+}
+
+/* static */
+already_AddRefed<AudioContext> AudioContext::Constructor(
+    const GlobalObject& aGlobal, uint32_t aNumberOfChannels, uint32_t aLength,
+    float aSampleRate, ErrorResult& aRv) {
+  nsCOMPtr<nsPIDOMWindowInner> window =
+      do_QueryInterface(aGlobal.GetAsSupports());
+  if (!window) {
+    aRv.Throw(NS_ERROR_FAILURE);
+    return nullptr;
+  }
+  /**
+   * If the current settings object’s responsible document is NOT fully
+   * active, throw an InvalidStateError and abort these steps.
+   */
+  if (!CheckFullyActive(window, aRv)) {
+    return nullptr;
+  }
+
+  if (aNumberOfChannels == 0 ||
+      aNumberOfChannels > WebAudioUtils::MaxChannelCount) {
+    aRv.ThrowNotSupportedError(
+        nsPrintfCString("%u is not a valid channel count", aNumberOfChannels));
+    return nullptr;
+  }
+
+  if (aLength == 0) {
+    aRv.ThrowNotSupportedError("Length must be nonzero");
+    return nullptr;
+  }
+
+  if (aSampleRate < WebAudioUtils::MinSampleRate ||
+      aSampleRate > WebAudioUtils::MaxSampleRate) {
+    // The DOM binding protects us against infinity and NaN
+    aRv.ThrowNotSupportedError(nsPrintfCString(
+        "Sample rate %g is not in the range [%u, %u]", aSampleRate,
+        WebAudioUtils::MinSampleRate, WebAudioUtils::MaxSampleRate));
+    return nullptr;
+  }
+
+  RefPtr<AudioContext> object =
+      new AudioContext(window, true, aNumberOfChannels, aLength, aSampleRate);
+
+  RegisterWeakMemoryReporter(object);
+
+  return object.forget();
+}
+
+already_AddRefed<AudioBufferSourceNode> AudioContext::CreateBufferSource() {
+  return AudioBufferSourceNode::Create(nullptr, *this,
+                                       AudioBufferSourceOptions());
+}
+
+already_AddRefed<ConstantSourceNode> AudioContext::CreateConstantSource() {
+  RefPtr<ConstantSourceNode> constantSourceNode = new ConstantSourceNode(this);
+  return constantSourceNode.forget();
+}
+
+already_AddRefed<AudioBuffer> AudioContext::CreateBuffer(
+    uint32_t aNumberOfChannels, uint32_t aLength, float aSampleRate,
+    ErrorResult& aRv) {
+  if (!aNumberOfChannels) {
+    aRv.ThrowNotSupportedError("Number of channels must be nonzero");
+    return nullptr;
+  }
+
+  return AudioBuffer::Create(GetOwner(), aNumberOfChannels, aLength,
+                             aSampleRate, aRv);
+}
+
+namespace {
+
+bool IsValidBufferSize(uint32_t aBufferSize) {
+  switch (aBufferSize) {
+    case 0:  // let the implementation choose the buffer size
+    case 256:
+    case 512:
+    case 1024:
+    case 2048:
+    case 4096:
+    case 8192:
+    case 16384:
+      return true;
+    default:
+      return false;
+  }
+}
+
+}  // namespace
+
+already_AddRefed<MediaStreamAudioDestinationNode>
+AudioContext::CreateMediaStreamDestination(ErrorResult& aRv) {
+  return MediaStreamAudioDestinationNode::Create(*this, AudioNodeOptions(),
+                                                 aRv);
+}
+
+already_AddRefed<ScriptProcessorNode> AudioContext::CreateScriptProcessor(
+    uint32_t aBufferSize, uint32_t aNumberOfInputChannels,
+    uint32_t aNumberOfOutputChannels, ErrorResult& aRv) {
+  if (aNumberOfInputChannels == 0 && aNumberOfOutputChannels == 0) {
+    aRv.ThrowIndexSizeError(
+        "At least one of numberOfInputChannels and numberOfOutputChannels must "
+        "be nonzero");
+    return nullptr;
+  }
+
+  if (aNumberOfInputChannels > WebAudioUtils::MaxChannelCount) {
+    aRv.ThrowIndexSizeError(nsPrintfCString(
+        "%u is not a valid number of input channels", aNumberOfInputChannels));
+    return nullptr;
+  }
+
+  if (aNumberOfOutputChannels > WebAudioUtils::MaxChannelCount) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("%u is not a valid number of output channels",
+                        aNumberOfOutputChannels));
+    return nullptr;
+  }
+
+  if (!IsValidBufferSize(aBufferSize)) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("%u is not a valid bufferSize", aBufferSize));
+    return nullptr;
+  }
+
+  RefPtr<ScriptProcessorNode> scriptProcessor = new ScriptProcessorNode(
+      this, aBufferSize, aNumberOfInputChannels, aNumberOfOutputChannels);
+  return scriptProcessor.forget();
+}
+
+already_AddRefed<AnalyserNode> AudioContext::CreateAnalyser(ErrorResult& aRv) {
+  return AnalyserNode::Create(*this, AnalyserOptions(), aRv);
+}
+
+already_AddRefed<StereoPannerNode> AudioContext::CreateStereoPanner(
+    ErrorResult& aRv) {
+  return StereoPannerNode::Create(*this, StereoPannerOptions(), aRv);
+}
+
+already_AddRefed<MediaElementAudioSourceNode>
+AudioContext::CreateMediaElementSource(HTMLMediaElement& aMediaElement,
+                                       ErrorResult& aRv) {
+  MediaElementAudioSourceOptions options;
+  options.mMediaElement = aMediaElement;
+
+  return MediaElementAudioSourceNode::Create(*this, options, aRv);
+}
+
+already_AddRefed<MediaStreamAudioSourceNode>
+AudioContext::CreateMediaStreamSource(DOMMediaStream& aMediaStream,
+                                      ErrorResult& aRv) {
+  MediaStreamAudioSourceOptions options;
+  options.mMediaStream = aMediaStream;
+
+  return MediaStreamAudioSourceNode::Create(*this, options, aRv);
+}
+
+already_AddRefed<MediaStreamTrackAudioSourceNode>
+AudioContext::CreateMediaStreamTrackSource(MediaStreamTrack& aMediaStreamTrack,
+                                           ErrorResult& aRv) {
+  MediaStreamTrackAudioSourceOptions options;
+  options.mMediaStreamTrack = aMediaStreamTrack;
+
+  return MediaStreamTrackAudioSourceNode::Create(*this, options, aRv);
+}
+
+already_AddRefed<GainNode> AudioContext::CreateGain(ErrorResult& aRv) {
+  return GainNode::Create(*this, GainOptions(), aRv);
+}
+
+already_AddRefed<WaveShaperNode> AudioContext::CreateWaveShaper(
+    ErrorResult& aRv) {
+  return WaveShaperNode::Create(*this, WaveShaperOptions(), aRv);
+}
+
+already_AddRefed<DelayNode> AudioContext::CreateDelay(double aMaxDelayTime,
+                                                      ErrorResult& aRv) {
+  DelayOptions options;
+  options.mMaxDelayTime = aMaxDelayTime;
+  return DelayNode::Create(*this, options, aRv);
+}
+
+already_AddRefed<PannerNode> AudioContext::CreatePanner(ErrorResult& aRv) {
+  return PannerNode::Create(*this, PannerOptions(), aRv);
+}
+
+already_AddRefed<ConvolverNode> AudioContext::CreateConvolver(
+    ErrorResult& aRv) {
+  return ConvolverNode::Create(nullptr, *this, ConvolverOptions(), aRv);
+}
+
+already_AddRefed<ChannelSplitterNode> AudioContext::CreateChannelSplitter(
+    uint32_t aNumberOfOutputs, ErrorResult& aRv) {
+  ChannelSplitterOptions options;
+  options.mNumberOfOutputs = aNumberOfOutputs;
+  return ChannelSplitterNode::Create(*this, options, aRv);
+}
+
+already_AddRefed<ChannelMergerNode> AudioContext::CreateChannelMerger(
+    uint32_t aNumberOfInputs, ErrorResult& aRv) {
+  ChannelMergerOptions options;
+  options.mNumberOfInputs = aNumberOfInputs;
+  return ChannelMergerNode::Create(*this, options, aRv);
+}
+
+already_AddRefed<DynamicsCompressorNode> AudioContext::CreateDynamicsCompressor(
+    ErrorResult& aRv) {
+  return DynamicsCompressorNode::Create(*this, DynamicsCompressorOptions(),
+                                        aRv);
+}
+
+already_AddRefed<BiquadFilterNode> AudioContext::CreateBiquadFilter(
+    ErrorResult& aRv) {
+  return BiquadFilterNode::Create(*this, BiquadFilterOptions(), aRv);
+}
+
+already_AddRefed<IIRFilterNode> AudioContext::CreateIIRFilter(
+    const Sequence<double>& aFeedforward, const Sequence<double>& aFeedback,
+    mozilla::ErrorResult& aRv) {
+  IIRFilterOptions options;
+  options.mFeedforward = aFeedforward;
+  options.mFeedback = aFeedback;
+  return IIRFilterNode::Create(*this, options, aRv);
+}
+
+already_AddRefed<OscillatorNode> AudioContext::CreateOscillator(
+    ErrorResult& aRv) {
+  return OscillatorNode::Create(*this, OscillatorOptions(), aRv);
+}
+
+already_AddRefed<PeriodicWave> AudioContext::CreatePeriodicWave(
+    const Sequence<float>& aRealData, const Sequence<float>& aImagData,
+    const PeriodicWaveConstraints& aConstraints, ErrorResult& aRv) {
+  RefPtr<PeriodicWave> periodicWave = new PeriodicWave(
+      this, aRealData.Elements(), aRealData.Length(), aImagData.Elements(),
+      aImagData.Length(), aConstraints.mDisableNormalization, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+  return periodicWave.forget();
+}
+
+AudioListener* AudioContext::Listener() {
+  if (!mListener) {
+    mListener = new AudioListener(this);
+  }
+  return mListener;
+}
+
+double AudioContext::OutputLatency() {
+  if (mIsShutDown) {
+    return 0.0;
+  }
+  // When reduceFingerprinting is enabled, return a latency figure that is
+  // fixed, but plausible for the platform.
+  double latency_s = 0.0;
+  if (mShouldResistFingerprinting) {
+#ifdef XP_MACOSX
+    latency_s = 512. / mSampleRate;
+#elif MOZ_WIDGET_ANDROID
+    latency_s = 0.020;
+#elif XP_WIN
+    latency_s = 0.04;
+#else  // Catchall for other OSes, including Linux.
+    latency_s = 0.025;
+#endif
+  } else {
+    return Graph()->AudioOutputLatency();
+  }
+  return latency_s;
+}
+
+void AudioContext::GetOutputTimestamp(AudioTimestamp& aTimeStamp) {
+  if (!Destination()) {
+    aTimeStamp.mContextTime.Construct(0.0);
+    aTimeStamp.mPerformanceTime.Construct(0.0);
+    return;
+  }
+
+  // The currentTime currently being output is the currentTime minus the audio
+  // output latency. The resolution of CurrentTime() is already reduced.
+  aTimeStamp.mContextTime.Construct(
+      std::max(0.0, CurrentTime() - OutputLatency()));
+  nsPIDOMWindowInner* parent = GetParentObject();
+  Performance* perf = parent ? parent->GetPerformance() : nullptr;
+  if (perf) {
+    // perf->Now() already has reduced resolution here, no need to do it again.
+    aTimeStamp.mPerformanceTime.Construct(
+        std::max(0., perf->Now() - (OutputLatency() * 1000.)));
+  } else {
+    aTimeStamp.mPerformanceTime.Construct(0.0);
+  }
+}
+
+Worklet* AudioContext::GetAudioWorklet(ErrorResult& aRv) {
+  if (!mWorklet) {
+    mWorklet = AudioWorkletImpl::CreateWorklet(this, aRv);
+  }
+
+  return mWorklet;
+}
+bool AudioContext::IsRunning() const {
+  return mAudioContextState == AudioContextState::Running;
+}
+
+already_AddRefed<Promise> AudioContext::CreatePromise(ErrorResult& aRv) {
+  // Get the relevant global for the promise from the wrapper cache because
+  // DOMEventTargetHelper::GetOwner() returns null if the document is unloaded.
+  // We know the wrapper exists because it is being used for |this| from JS.
+  // See https://github.com/heycam/webidl/issues/932 for why the relevant
+  // global is used instead of the current global.
+  nsCOMPtr<nsIGlobalObject> global = xpc::NativeGlobal(GetWrapper());
+  RefPtr<Promise> promise = Promise::Create(global, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+  /**
+   * If this's relevant global object's associated Document is not fully
+   * active then return a promise rejected with "InvalidStateError"
+   * DOMException.
+   */
+  nsCOMPtr<nsPIDOMWindowInner> window = do_QueryInterface(global);
+  if (!window->IsFullyActive()) {
+    promise->MaybeRejectWithInvalidStateError(
+        "The document is not fully active.");
+  }
+  return promise.forget();
+}
+
+already_AddRefed<Promise> AudioContext::DecodeAudioData(
+    const ArrayBuffer& aBuffer,
+    const Optional<OwningNonNull<DecodeSuccessCallback>>& aSuccessCallback,
+    const Optional<OwningNonNull<DecodeErrorCallback>>& aFailureCallback,
+    ErrorResult& aRv) {
+  AutoJSAPI jsapi;
+  jsapi.Init();
+  JSContext* cx = jsapi.cx();
+
+  // CheckedUnwrapStatic is OK, since we know we have an ArrayBuffer.
+  JS::Rooted<JSObject*> obj(cx, js::CheckedUnwrapStatic(aBuffer.Obj()));
+  if (!obj) {
+    aRv.ThrowSecurityError("Can't get audio data from cross-origin object");
+    return nullptr;
+  }
+
+  RefPtr<Promise> promise = CreatePromise(aRv);
+  if (aRv.Failed() || promise->State() == Promise::PromiseState::Rejected) {
+    return promise.forget();
+  }
+
+  JSAutoRealm ar(cx, obj);
+
+  // Detach the array buffer
+  size_t length = JS::GetArrayBufferByteLength(obj);
+  uint8_t* data = static_cast<uint8_t*>(JS::StealArrayBufferContents(cx, obj));
+  if (!data) {
+    JS_ClearPendingException(cx);
+
+    // Throw if the buffer is detached
+    aRv.ThrowTypeError("Buffer argument can't be a detached buffer");
+    return nullptr;
+  }
+
+  // Sniff the content of the media.
+  // Failed type sniffing will be handled by AsyncDecodeWebAudio.
+  nsAutoCString contentType;
+  NS_SniffContent(NS_DATA_SNIFFER_CATEGORY, nullptr, data, length, contentType);
+
+  RefPtr<DecodeErrorCallback> failureCallback;
+  RefPtr<DecodeSuccessCallback> successCallback;
+  if (aFailureCallback.WasPassed()) {
+    failureCallback = &aFailureCallback.Value();
+  }
+  if (aSuccessCallback.WasPassed()) {
+    successCallback = &aSuccessCallback.Value();
+  }
+  UniquePtr<WebAudioDecodeJob> job(
+      new WebAudioDecodeJob(this, promise, successCallback, failureCallback));
+  AsyncDecodeWebAudio(contentType.get(), data, length, *job);
+  // Transfer the ownership to mDecodeJobs
+  mDecodeJobs.AppendElement(std::move(job));
+
+  return promise.forget();
+}
+
+void AudioContext::RemoveFromDecodeQueue(WebAudioDecodeJob* aDecodeJob) {
+  // Since UniquePtr doesn't provide an operator== which allows you to compare
+  // against raw pointers, we need to iterate manually.
+  for (uint32_t i = 0; i < mDecodeJobs.Length(); ++i) {
+    if (mDecodeJobs[i].get() == aDecodeJob) {
+      mDecodeJobs.RemoveElementAt(i);
+      break;
+    }
+  }
+}
+
+void AudioContext::RegisterActiveNode(AudioNode* aNode) {
+  if (!mCloseCalled) {
+    mActiveNodes.Insert(aNode);
+  }
+}
+
+void AudioContext::UnregisterActiveNode(AudioNode* aNode) {
+  mActiveNodes.Remove(aNode);
+}
+
+uint32_t AudioContext::MaxChannelCount() const {
+  if (mShouldResistFingerprinting) {
+    return 2;
+  }
+  return std::min<uint32_t>(
+      WebAudioUtils::MaxChannelCount,
+      mIsOffline ? mNumberOfChannels : CubebUtils::MaxNumberOfChannels());
+}
+
+uint32_t AudioContext::ActiveNodeCount() const { return mActiveNodes.Count(); }
+
+MediaTrackGraph* AudioContext::Graph() const {
+  return Destination()->Track()->Graph();
+}
+
+AudioNodeTrack* AudioContext::DestinationTrack() const {
+  if (Destination()) {
+    return Destination()->Track();
+  }
+  return nullptr;
+}
+
+void AudioContext::ShutdownWorklet() {
+  if (mWorklet) {
+    mWorklet->Impl()->NotifyWorkletFinished();
+  }
+}
+
+double AudioContext::CurrentTime() {
+  mozilla::MediaTrack* track = Destination()->Track();
+
+  double rawTime = track->TrackTimeToSeconds(track->GetCurrentTime());
+
+  // CurrentTime increments in intervals of 128/sampleRate. If the Timer
+  // Precision Reduction is smaller than this interval, the jittered time
+  // can always be reversed to the raw step of the interval. In that case
+  // we can simply return the un-reduced time; and avoid breaking tests.
+  // We have to convert each variable into a common magnitude, we choose ms.
+  if ((128 / mSampleRate) * 1000.0 >
+      nsRFPService::TimerResolution(mRTPCallerType) / 1000.0) {
+    return rawTime;
+  }
+
+  // The value of a MediaTrack's CurrentTime will always advance forward; it
+  // will never reset (even if one rewinds a video.) Therefore we can use a
+  // single Random Seed initialized at the same time as the object.
+  return nsRFPService::ReduceTimePrecisionAsSecs(
+      rawTime, GetRandomTimelineSeed(), mRTPCallerType);
+}
+
+nsISerialEventTarget* AudioContext::GetMainThread() const {
+  if (nsPIDOMWindowInner* window = GetParentObject()) {
+    return window->AsGlobal()->SerialEventTarget();
+  }
+  return GetCurrentSerialEventTarget();
+}
+
+void AudioContext::DisconnectFromOwner() {
+  mIsDisconnecting = true;
+  MaybeClearPageAwakeRequest();
+  OnWindowDestroy();
+  DOMEventTargetHelper::DisconnectFromOwner();
+}
+
+void AudioContext::OnWindowDestroy() {
+  // Avoid resend the Telemetry data.
+  if (!mIsShutDown) {
+    MaybeUpdateAutoplayTelemetryWhenShutdown();
+  }
+  mIsShutDown = true;
+
+  CloseInternal(nullptr, AudioContextOperationFlags::None);
+
+  // We don't want to touch promises if the global is going away soon.
+  if (!mIsDisconnecting) {
+    for (auto p : mPromiseGripArray) {
+      p->MaybeRejectWithInvalidStateError("Navigated away from page");
+    }
+
+    mPromiseGripArray.Clear();
+
+    for (const auto& p : mPendingResumePromises) {
+      p->MaybeRejectWithInvalidStateError("Navigated away from page");
+    }
+    mPendingResumePromises.Clear();
+  }
+
+  // On process shutdown, the MTG thread shuts down before the destination
+  // track is destroyed, but AudioWorklet needs to release objects on the MTG
+  // thread.  AudioContext::Shutdown() is invoked on processing the
+  // PBrowser::Destroy() message before xpcom shutdown begins.
+  ShutdownWorklet();
+
+  if (mDestination) {
+    // We can destroy the MediaTrackGraph at this point.
+    // Although there may be other clients using the graph, this graph is used
+    // only for clients in the same window and this window is going away.
+    // This will also interrupt any worklet script still running on the graph
+    // thread.
+    Graph()->ForceShutDown();
+    // AudioDestinationNodes on rendering offline contexts have a
+    // self-reference which needs removal.
+    if (mIsOffline) {
+      mDestination->OfflineShutdown();
+    }
+  }
+}
+
+/* This runnable allows to fire the "statechange" event */
+class OnStateChangeTask final : public Runnable {
+ public:
+  explicit OnStateChangeTask(AudioContext* aAudioContext)
+      : Runnable("dom::OnStateChangeTask"), mAudioContext(aAudioContext) {}
+
+  NS_IMETHODIMP
+  Run() override {
+    nsPIDOMWindowInner* parent = mAudioContext->GetParentObject();
+    if (!parent) {
+      return NS_ERROR_FAILURE;
+    }
+
+    Document* doc = parent->GetExtantDoc();
+    if (!doc) {
+      return NS_ERROR_FAILURE;
+    }
+
+    return nsContentUtils::DispatchTrustedEvent(
+        doc, mAudioContext, u"statechange"_ns, CanBubble::eNo, Cancelable::eNo);
+  }
+
+ private:
+  RefPtr<AudioContext> mAudioContext;
+};
+
+void AudioContext::Dispatch(already_AddRefed<nsIRunnable>&& aRunnable) {
+  MOZ_ASSERT(NS_IsMainThread());
+  // It can happen that this runnable took a long time to reach the main thread,
+  // and the global is not valid anymore.
+  if (GetParentObject()) {
+    AbstractThread::MainThread()->Dispatch(std::move(aRunnable));
+  } else {
+    RefPtr<nsIRunnable> runnable(aRunnable);
+    runnable = nullptr;
+  }
+}
+
+void AudioContext::OnStateChanged(void* aPromise, AudioContextState aNewState) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (mAudioContextState == AudioContextState::Closed) {
+    fprintf(stderr,
+            "Invalid transition: mAudioContextState: %d -> aNewState %d\n",
+            static_cast<int>(mAudioContextState), static_cast<int>(aNewState));
+    MOZ_ASSERT(false);
+  }
+
+  if (aPromise) {
+    Promise* promise = reinterpret_cast<Promise*>(aPromise);
+    // It is possible for the promise to have been removed from
+    // mPromiseGripArray if the cycle collector has severed our connections. DO
+    // NOT dereference the promise pointer in that case since it may point to
+    // already freed memory.
+    if (mPromiseGripArray.Contains(promise)) {
+      promise->MaybeResolveWithUndefined();
+      DebugOnly<bool> rv = mPromiseGripArray.RemoveElement(promise);
+      MOZ_ASSERT(rv, "Promise wasn't in the grip array?");
+    }
+  }
+
+  // Resolve all pending promises once the audio context has been allowed to
+  // start.
+  if (aNewState == AudioContextState::Running) {
+    for (const auto& p : mPendingResumePromises) {
+      p->MaybeResolveWithUndefined();
+    }
+    mPendingResumePromises.Clear();
+  }
+
+  if (mAudioContextState != aNewState) {
+    RefPtr<OnStateChangeTask> task = new OnStateChangeTask(this);
+    Dispatch(task.forget());
+  }
+
+  mAudioContextState = aNewState;
+  Destination()->NotifyAudioContextStateChanged();
+  MaybeUpdatePageAwakeRequest();
+}
+
+BrowsingContext* AudioContext::GetTopLevelBrowsingContext() {
+  nsCOMPtr<nsPIDOMWindowInner> window = GetParentObject();
+  if (!window) {
+    return nullptr;
+  }
+  BrowsingContext* bc = window->GetBrowsingContext();
+  if (!bc || bc->IsDiscarded()) {
+    return nullptr;
+  }
+  return bc->Top();
+}
+
+void AudioContext::MaybeUpdatePageAwakeRequest() {
+  // No need to keep page awake for offline context.
+  if (IsOffline()) {
+    return;
+  }
+
+  if (mIsShutDown) {
+    return;
+  }
+
+  if (IsRunning() && !mSetPageAwakeRequest) {
+    SetPageAwakeRequest(true);
+  } else if (!IsRunning() && mSetPageAwakeRequest) {
+    SetPageAwakeRequest(false);
+  }
+}
+
+void AudioContext::SetPageAwakeRequest(bool aShouldSet) {
+  mSetPageAwakeRequest = aShouldSet;
+  BrowsingContext* bc = GetTopLevelBrowsingContext();
+  if (!bc) {
+    return;
+  }
+  if (XRE_IsContentProcess()) {
+    ContentChild* contentChild = ContentChild::GetSingleton();
+    Unused << contentChild->SendAddOrRemovePageAwakeRequest(bc, aShouldSet);
+    return;
+  }
+  if (aShouldSet) {
+    bc->Canonical()->AddPageAwakeRequest();
+  } else {
+    bc->Canonical()->RemovePageAwakeRequest();
+  }
+}
+
+void AudioContext::MaybeClearPageAwakeRequest() {
+  if (mSetPageAwakeRequest) {
+    SetPageAwakeRequest(false);
+  }
+}
+
+nsTArray<RefPtr<mozilla::MediaTrack>> AudioContext::GetAllTracks() const {
+  nsTArray<RefPtr<mozilla::MediaTrack>> tracks;
+  for (AudioNode* node : mAllNodes) {
+    mozilla::MediaTrack* t = node->GetTrack();
+    if (t) {
+      tracks.AppendElement(t);
+    }
+    // Add the tracks of AudioParam.
+    const nsTArray<RefPtr<AudioParam>>& audioParams = node->GetAudioParams();
+    if (!audioParams.IsEmpty()) {
+      for (auto& param : audioParams) {
+        t = param->GetTrack();
+        if (t && !tracks.Contains(t)) {
+          tracks.AppendElement(t);
+        }
+      }
+    }
+  }
+  return tracks;
+}
+
+already_AddRefed<Promise> AudioContext::Suspend(ErrorResult& aRv) {
+  TRACE("AudioContext::Suspend");
+  RefPtr<Promise> promise = CreatePromise(aRv);
+  if (aRv.Failed() || promise->State() == Promise::PromiseState::Rejected) {
+    return promise.forget();
+  }
+  if (mIsOffline) {
+    // XXXbz This is not reachable, since we don't implement this
+    // method on OfflineAudioContext at all!
+    promise->MaybeRejectWithNotSupportedError(
+        "Can't suspend OfflineAudioContext yet");
+    return promise.forget();
+  }
+
+  if (mCloseCalled) {
+    promise->MaybeRejectWithInvalidStateError(
+        "Can't suspend if the control thread state is \"closed\"");
+    return promise.forget();
+  }
+
+  mSuspendedByContent = true;
+  mPromiseGripArray.AppendElement(promise);
+  SuspendInternal(promise, AudioContextOperationFlags::SendStateChange);
+  return promise.forget();
+}
+
+void AudioContext::SuspendFromChrome() {
+  if (mIsOffline || mIsShutDown) {
+    return;
+  }
+  MOZ_ASSERT(!mSuspendedByChrome);
+  mSuspendedByChrome = true;
+  SuspendInternal(nullptr, Preferences::GetBool("dom.audiocontext.testing")
+                               ? AudioContextOperationFlags::SendStateChange
+                               : AudioContextOperationFlags::None);
+}
+
+void AudioContext::SuspendInternal(void* aPromise,
+                                   AudioContextOperationFlags aFlags) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(!mIsOffline);
+  Destination()->Suspend();
+
+  nsTArray<RefPtr<mozilla::MediaTrack>> tracks;
+  // If mTracksAreSuspended is true then we already suspended all our tracks,
+  // so don't suspend them again (since suspend(); suspend(); resume(); should
+  // cancel both suspends). But we still need to do ApplyAudioContextOperation
+  // to ensure our new promise is resolved.
+  if (!mTracksAreSuspended) {
+    mTracksAreSuspended = true;
+    tracks = GetAllTracks();
+  }
+  auto promise = Graph()->ApplyAudioContextOperation(
+      DestinationTrack(), std::move(tracks), AudioContextOperation::Suspend);
+  if ((aFlags & AudioContextOperationFlags::SendStateChange)) {
+    promise->Then(
+        GetMainThread(), "AudioContext::OnStateChanged",
+        [self = RefPtr<AudioContext>(this),
+         aPromise](AudioContextState aNewState) {
+          self->OnStateChanged(aPromise, aNewState);
+        },
+        [] { MOZ_CRASH("Unexpected rejection"); });
+  }
+}
+
+void AudioContext::ResumeFromChrome() {
+  if (mIsOffline || mIsShutDown) {
+    return;
+  }
+  MOZ_ASSERT(mSuspendedByChrome);
+  mSuspendedByChrome = false;
+  if (!mWasAllowedToStart) {
+    return;
+  }
+  ResumeInternal();
+}
+
+already_AddRefed<Promise> AudioContext::Resume(ErrorResult& aRv) {
+  TRACE("AudioContext::Resume");
+  RefPtr<Promise> promise = CreatePromise(aRv);
+  if (aRv.Failed() || promise->State() == Promise::PromiseState::Rejected) {
+    return promise.forget();
+  }
+
+  if (mIsOffline) {
+    promise->MaybeRejectWithNotSupportedError(
+        "Can't resume OfflineAudioContext");
+    return promise.forget();
+  }
+
+  if (mCloseCalled) {
+    promise->MaybeRejectWithInvalidStateError(
+        "Can't resume if the control thread state is \"closed\"");
+    return promise.forget();
+  }
+
+  mSuspendedByContent = false;
+  mPendingResumePromises.AppendElement(promise);
+
+  const bool isAllowedToPlay = media::AutoplayPolicy::IsAllowedToPlay(*this);
+  AUTOPLAY_LOG("Trying to resume AudioContext %p, IsAllowedToPlay=%d", this,
+               isAllowedToPlay);
+  if (isAllowedToPlay) {
+    ResumeInternal();
+  } else {
+    ReportBlocked();
+  }
+
+  MaybeUpdateAutoplayTelemetry();
+
+  return promise.forget();
+}
+
+void AudioContext::ResumeInternal() {
+  MOZ_ASSERT(!mIsOffline);
+  AUTOPLAY_LOG("Allow to resume AudioContext %p", this);
+  mWasAllowedToStart = true;
+
+  if (mSuspendedByChrome || mSuspendedByContent || mCloseCalled) {
+    MOZ_ASSERT(mTracksAreSuspended);
+    return;
+  }
+
+  Destination()->Resume();
+
+  nsTArray<RefPtr<mozilla::MediaTrack>> tracks;
+  // If mTracksAreSuspended is false then we already resumed all our tracks,
+  // so don't resume them again (since suspend(); resume(); resume(); should
+  // be OK). But we still need to do ApplyAudioContextOperation
+  // to ensure our new promise is resolved.
+  if (mTracksAreSuspended) {
+    mTracksAreSuspended = false;
+    tracks = GetAllTracks();
+  }
+  // Check for statechange even when resumed from chrome because content may
+  // have called Resume() before chrome resumed the window.
+  Graph()
+      ->ApplyAudioContextOperation(DestinationTrack(), std::move(tracks),
+                                   AudioContextOperation::Resume)
+      ->Then(
+          GetMainThread(), "AudioContext::OnStateChanged",
+          [self = RefPtr<AudioContext>(this)](AudioContextState aNewState) {
+            self->OnStateChanged(nullptr, aNewState);
+          },
+          [] {});  // Promise may be rejected after graph shutdown.
+}
+
+void AudioContext::UpdateAutoplayAssumptionStatus() {
+  if (media::AutoplayPolicyTelemetryUtils::
+          WouldBeAllowedToPlayIfAutoplayDisabled(*this)) {
+    mWasEverAllowedToStart |= true;
+    mWouldBeAllowedToStart = true;
+  } else {
+    mWasEverBlockedToStart |= true;
+    mWouldBeAllowedToStart = false;
+  }
+}
+
+void AudioContext::MaybeUpdateAutoplayTelemetry() {
+  // Exclude offline AudioContext because it's always allowed to start.
+  if (mIsOffline) {
+    return;
+  }
+
+  if (media::AutoplayPolicyTelemetryUtils::
+          WouldBeAllowedToPlayIfAutoplayDisabled(*this) &&
+      !mWouldBeAllowedToStart) {
+    AccumulateCategorical(
+        mozilla::Telemetry::LABELS_WEB_AUDIO_AUTOPLAY::AllowedAfterBlocked);
+  }
+  UpdateAutoplayAssumptionStatus();
+}
+
+void AudioContext::MaybeUpdateAutoplayTelemetryWhenShutdown() {
+  // Exclude offline AudioContext because it's always allowed to start.
+  if (mIsOffline) {
+    return;
+  }
+
+  if (mWasEverAllowedToStart && !mWasEverBlockedToStart) {
+    AccumulateCategorical(
+        mozilla::Telemetry::LABELS_WEB_AUDIO_AUTOPLAY::NeverBlocked);
+  } else if (!mWasEverAllowedToStart && mWasEverBlockedToStart) {
+    AccumulateCategorical(
+        mozilla::Telemetry::LABELS_WEB_AUDIO_AUTOPLAY::NeverAllowed);
+  }
+}
+
+void AudioContext::ReportBlocked() {
+  ReportToConsole(nsIScriptError::warningFlag,
+                  "BlockAutoplayWebAudioStartError");
+  mWasAllowedToStart = false;
+
+  if (!StaticPrefs::media_autoplay_block_event_enabled()) {
+    return;
+  }
+
+  RefPtr<AudioContext> self = this;
+  RefPtr<nsIRunnable> r =
+      NS_NewRunnableFunction("AudioContext::AutoplayBlocked", [self]() {
+        nsPIDOMWindowInner* parent = self->GetParentObject();
+        if (!parent) {
+          return;
+        }
+
+        Document* doc = parent->GetExtantDoc();
+        if (!doc) {
+          return;
+        }
+
+        AUTOPLAY_LOG("Dispatch `blocked` event for AudioContext %p",
+                     self.get());
+        nsContentUtils::DispatchTrustedEvent(doc, self, u"blocked"_ns,
+                                             CanBubble::eNo, Cancelable::eNo);
+      });
+  Dispatch(r.forget());
+}
+
+already_AddRefed<Promise> AudioContext::Close(ErrorResult& aRv) {
+  TRACE("AudioContext::Close");
+  RefPtr<Promise> promise = CreatePromise(aRv);
+  if (aRv.Failed() || promise->State() == Promise::PromiseState::Rejected) {
+    return promise.forget();
+  }
+
+  if (mIsOffline) {
+    // XXXbz This is not reachable, since we don't implement this
+    // method on OfflineAudioContext at all!
+    promise->MaybeRejectWithNotSupportedError(
+        "Can't close OfflineAudioContext yet");
+    return promise.forget();
+  }
+
+  if (mCloseCalled) {
+    promise->MaybeRejectWithInvalidStateError(
+        "Can't close an AudioContext twice");
+    return promise.forget();
+  }
+
+  mPromiseGripArray.AppendElement(promise);
+
+  CloseInternal(promise, AudioContextOperationFlags::SendStateChange);
+
+  return promise.forget();
+}
+
+void AudioContext::OfflineClose() {
+  CloseInternal(nullptr, AudioContextOperationFlags::None);
+}
+
+void AudioContext::CloseInternal(void* aPromise,
+                                 AudioContextOperationFlags aFlags) {
+  // This can be called when freeing a document, and the tracks are dead at
+  // this point, so we need extra null-checks.
+  AudioNodeTrack* ds = DestinationTrack();
+  if (ds && !mIsOffline) {
+    Destination()->Close();
+
+    nsTArray<RefPtr<mozilla::MediaTrack>> tracks;
+    // If mTracksAreSuspended or mCloseCalled are true then we already suspended
+    // all our tracks, so don't suspend them again. But we still need to do
+    // ApplyAudioContextOperation to ensure our new promise is resolved.
+    if (!mTracksAreSuspended && !mCloseCalled) {
+      tracks = GetAllTracks();
+    }
+    auto promise = Graph()->ApplyAudioContextOperation(
+        ds, std::move(tracks), AudioContextOperation::Close);
+    if ((aFlags & AudioContextOperationFlags::SendStateChange)) {
+      promise->Then(
+          GetMainThread(), "AudioContext::OnStateChanged",
+          [self = RefPtr<AudioContext>(this),
+           aPromise](AudioContextState aNewState) {
+            self->OnStateChanged(aPromise, aNewState);
+          },
+          [] {});  // Promise may be rejected after graph shutdown.
+    }
+  }
+  mCloseCalled = true;
+  // Release references to active nodes.
+  // Active AudioNodes don't unregister in destructors, at which point the
+  // Node is already unregistered.
+  mActiveNodes.Clear();
+}
+
+void AudioContext::RegisterNode(AudioNode* aNode) {
+  MOZ_ASSERT(!mAllNodes.Contains(aNode));
+  mAllNodes.Insert(aNode);
+}
+
+void AudioContext::UnregisterNode(AudioNode* aNode) {
+  MOZ_ASSERT(mAllNodes.Contains(aNode));
+  mAllNodes.Remove(aNode);
+}
+
+already_AddRefed<Promise> AudioContext::StartRendering(ErrorResult& aRv) {
+  MOZ_ASSERT(mIsOffline, "This should only be called on OfflineAudioContext");
+  RefPtr<Promise> promise = CreatePromise(aRv);
+  if (aRv.Failed() || promise->State() == Promise::PromiseState::Rejected) {
+    return promise.forget();
+  }
+  if (mIsStarted) {
+    aRv.ThrowInvalidStateError("Rendering already started");
+    return nullptr;
+  }
+
+  mIsStarted = true;
+  mDestination->StartRendering(promise);
+
+  OnStateChanged(nullptr, AudioContextState::Running);
+
+  return promise.forget();
+}
+
+unsigned long AudioContext::Length() {
+  MOZ_ASSERT(mIsOffline);
+  return mDestination->Length();
+}
+
+void AudioContext::Mute() const {
+  MOZ_ASSERT(!mIsOffline);
+  if (mDestination) {
+    mDestination->Mute();
+  }
+}
+
+void AudioContext::Unmute() const {
+  MOZ_ASSERT(!mIsOffline);
+  if (mDestination) {
+    mDestination->Unmute();
+  }
+}
+
+void AudioContext::SetParamMapForWorkletName(
+    const nsAString& aName, AudioParamDescriptorMap* aParamMap) {
+  MOZ_ASSERT(!mWorkletParamDescriptors.Contains(aName));
+  Unused << mWorkletParamDescriptors.InsertOrUpdate(
+      aName, std::move(*aParamMap), fallible);
+}
+
+size_t AudioContext::SizeOfIncludingThis(
+    mozilla::MallocSizeOf aMallocSizeOf) const {
+  // AudioNodes are tracked separately because we do not want the AudioContext
+  // to track all of the AudioNodes it creates, so we wouldn't be able to
+  // traverse them from here.
+
+  size_t amount = aMallocSizeOf(this);
+  if (mListener) {
+    amount += mListener->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  amount += mDecodeJobs.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (uint32_t i = 0; i < mDecodeJobs.Length(); ++i) {
+    amount += mDecodeJobs[i]->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  amount += mActiveNodes.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+NS_IMETHODIMP
+AudioContext::CollectReports(nsIHandleReportCallback* aHandleReport,
+                             nsISupports* aData, bool aAnonymize) {
+  const nsLiteralCString nodeDescription(
+      "Memory used by AudioNode DOM objects (Web Audio).");
+  for (AudioNode* node : mAllNodes) {
+    int64_t amount = node->SizeOfIncludingThis(MallocSizeOf);
+    nsPrintfCString domNodePath("explicit/webaudio/audio-node/%s/dom-nodes",
+                                node->NodeType());
+    aHandleReport->Callback(""_ns, domNodePath, KIND_HEAP, UNITS_BYTES, amount,
+                            nodeDescription, aData);
+  }
+
+  int64_t amount = SizeOfIncludingThis(MallocSizeOf);
+  MOZ_COLLECT_REPORT("explicit/webaudio/audiocontext", KIND_HEAP, UNITS_BYTES,
+                     amount,
+                     "Memory used by AudioContext objects (Web Audio).");
+
+  return NS_OK;
+}
+
+BasicWaveFormCache* AudioContext::GetBasicWaveFormCache() {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (!mBasicWaveFormCache) {
+    mBasicWaveFormCache = new BasicWaveFormCache(SampleRate());
+  }
+  return mBasicWaveFormCache;
+}
+
+void AudioContext::ReportToConsole(uint32_t aErrorFlags,
+                                   const char* aMsg) const {
+  MOZ_ASSERT(aMsg);
+  Document* doc =
+      GetParentObject() ? GetParentObject()->GetExtantDoc() : nullptr;
+  nsContentUtils::ReportToConsole(aErrorFlags, "Media"_ns, doc,
+                                  nsContentUtils::eDOM_PROPERTIES, aMsg);
+}
+
+BasicWaveFormCache::BasicWaveFormCache(uint32_t aSampleRate)
+    : mSampleRate(aSampleRate) {
+  MOZ_ASSERT(NS_IsMainThread());
+}
+BasicWaveFormCache::~BasicWaveFormCache() = default;
+
+WebCore::PeriodicWave* BasicWaveFormCache::GetBasicWaveForm(
+    OscillatorType aType) {
+  MOZ_ASSERT(!NS_IsMainThread());
+  if (aType == OscillatorType::Sawtooth) {
+    if (!mSawtooth) {
+      mSawtooth = WebCore::PeriodicWave::createSawtooth(mSampleRate);
+    }
+    return mSawtooth;
+  }
+  if (aType == OscillatorType::Square) {
+    if (!mSquare) {
+      mSquare = WebCore::PeriodicWave::createSquare(mSampleRate);
+    }
+    return mSquare;
+  }
+  if (aType == OscillatorType::Triangle) {
+    if (!mTriangle) {
+      mTriangle = WebCore::PeriodicWave::createTriangle(mSampleRate);
+    }
+    return mTriangle;
+  }
+  MOZ_ASSERT(false, "Not reached");
+  return nullptr;
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioContext.h b/dom/media/webaudio/AudioContext.h
new file mode 100644
index 0000000000..81a64a18bf
--- /dev/null
+++ b/dom/media/webaudio/AudioContext.h
@@ -0,0 +1,479 @@
+/* -*- 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/. */
+
+#ifndef AudioContext_h_
+#define AudioContext_h_
+
+#include "X11UndefineNone.h"
+#include "AudioParamDescriptorMap.h"
+#include "mozilla/dom/OfflineAudioContextBinding.h"
+#include "mozilla/dom/AudioContextBinding.h"
+#include "MediaBufferDecoder.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/DOMEventTargetHelper.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/dom/TypedArray.h"
+#include "mozilla/RelativeTimeline.h"
+#include "mozilla/TypedEnumBits.h"
+#include "mozilla/UniquePtr.h"
+#include "nsCOMPtr.h"
+#include "nsCycleCollectionParticipant.h"
+#include "nsTHashMap.h"
+#include "nsHashKeys.h"
+#include "nsTHashSet.h"
+#include "js/TypeDecls.h"
+#include "nsIMemoryReporter.h"
+
+namespace WebCore {
+class PeriodicWave;
+}  // namespace WebCore
+
+class nsPIDOMWindowInner;
+
+namespace mozilla {
+
+class DOMMediaStream;
+class ErrorResult;
+class MediaTrack;
+class MediaTrackGraph;
+class AudioNodeTrack;
+
+namespace dom {
+
+enum class AudioContextState : uint8_t;
+class AnalyserNode;
+class AudioBuffer;
+class AudioBufferSourceNode;
+class AudioDestinationNode;
+class AudioListener;
+class AudioNode;
+class BiquadFilterNode;
+class BrowsingContext;
+class ChannelMergerNode;
+class ChannelSplitterNode;
+class ConstantSourceNode;
+class ConvolverNode;
+class DelayNode;
+class DynamicsCompressorNode;
+class GainNode;
+class GlobalObject;
+class HTMLMediaElement;
+class IIRFilterNode;
+class MediaElementAudioSourceNode;
+class MediaStreamAudioDestinationNode;
+class MediaStreamAudioSourceNode;
+class MediaStreamTrack;
+class MediaStreamTrackAudioSourceNode;
+class OscillatorNode;
+class PannerNode;
+class ScriptProcessorNode;
+class StereoPannerNode;
+class WaveShaperNode;
+class Worklet;
+class PeriodicWave;
+struct PeriodicWaveConstraints;
+class Promise;
+enum class OscillatorType : uint8_t;
+
+// This is addrefed by the OscillatorNodeEngine on the main thread
+// and then used from the MTG thread.
+// It can be released either from the graph thread or the main thread.
+class BasicWaveFormCache {
+ public:
+  explicit BasicWaveFormCache(uint32_t aSampleRate);
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(BasicWaveFormCache)
+  WebCore::PeriodicWave* GetBasicWaveForm(OscillatorType aType);
+
+ private:
+  ~BasicWaveFormCache();
+  RefPtr<WebCore::PeriodicWave> mSawtooth;
+  RefPtr<WebCore::PeriodicWave> mSquare;
+  RefPtr<WebCore::PeriodicWave> mTriangle;
+  uint32_t mSampleRate;
+};
+
+/* This runnable allows the MTG to notify the main thread when audio is actually
+ * flowing */
+class StateChangeTask final : public Runnable {
+ public:
+  /* This constructor should be used when this event is sent from the main
+   * thread. */
+  StateChangeTask(AudioContext* aAudioContext, void* aPromise,
+                  AudioContextState aNewState);
+
+  /* This constructor should be used when this event is sent from the audio
+   * thread. */
+  StateChangeTask(AudioNodeTrack* aTrack, void* aPromise,
+                  AudioContextState aNewState);
+
+  NS_IMETHOD Run() override;
+
+ private:
+  RefPtr<AudioContext> mAudioContext;
+  void* mPromise;
+  RefPtr<AudioNodeTrack> mAudioNodeTrack;
+  AudioContextState mNewState;
+};
+
+enum class AudioContextOperation : uint8_t { Suspend, Resume, Close };
+static const char* const kAudioContextOptionsStrings[] = {"Suspend", "Resume",
+                                                          "Close"};
+// When suspending or resuming an AudioContext, some operations have to notify
+// the main thread, so that the Promise is resolved, the state is modified, and
+// the statechanged event is sent. Some other operations don't go back to the
+// main thread, for example when the AudioContext is paused by something that is
+// not caused by the page itself: opening a debugger, breaking on a breakpoint,
+// reloading a document.
+enum class AudioContextOperationFlags { None, SendStateChange };
+MOZ_MAKE_ENUM_CLASS_BITWISE_OPERATORS(AudioContextOperationFlags);
+
+struct AudioContextOptions;
+
+class AudioContext final : public DOMEventTargetHelper,
+                           public nsIMemoryReporter,
+                           public RelativeTimeline {
+  AudioContext(nsPIDOMWindowInner* aParentWindow, bool aIsOffline,
+               uint32_t aNumberOfChannels = 0, uint32_t aLength = 0,
+               float aSampleRate = 0.0f);
+  ~AudioContext();
+
+ public:
+  typedef uint64_t AudioContextId;
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioContext, DOMEventTargetHelper)
+  MOZ_DEFINE_MALLOC_SIZE_OF(MallocSizeOf)
+
+  nsPIDOMWindowInner* GetParentObject() const { return GetOwner(); }
+
+  nsISerialEventTarget* GetMainThread() const;
+
+  virtual void DisconnectFromOwner() override;
+
+  void OnWindowDestroy();  // idempotent
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  using DOMEventTargetHelper::DispatchTrustedEvent;
+
+  // Constructor for regular AudioContext
+  static already_AddRefed<AudioContext> Constructor(
+      const GlobalObject& aGlobal, const AudioContextOptions& aOptions,
+      ErrorResult& aRv);
+
+  // Constructor for offline AudioContext with options object
+  static already_AddRefed<AudioContext> Constructor(
+      const GlobalObject& aGlobal, const OfflineAudioContextOptions& aOptions,
+      ErrorResult& aRv);
+
+  // Constructor for offline AudioContext
+  static already_AddRefed<AudioContext> Constructor(const GlobalObject& aGlobal,
+                                                    uint32_t aNumberOfChannels,
+                                                    uint32_t aLength,
+                                                    float aSampleRate,
+                                                    ErrorResult& aRv);
+
+  // AudioContext methods
+
+  AudioDestinationNode* Destination() const { return mDestination; }
+
+  float SampleRate() const { return mSampleRate; }
+
+  bool ShouldSuspendNewTrack() const {
+    return mTracksAreSuspended || mCloseCalled;
+  }
+  double CurrentTime();
+
+  AudioListener* Listener();
+
+  AudioContextState State() const { return mAudioContextState; }
+
+  double BaseLatency() const {
+    // Gecko does not do any buffering between rendering the audio and sending
+    // it to the audio subsystem.
+    return 0.0;
+  }
+
+  double OutputLatency();
+
+  void GetOutputTimestamp(AudioTimestamp& aTimeStamp);
+
+  Worklet* GetAudioWorklet(ErrorResult& aRv);
+
+  bool IsRunning() const;
+
+  // Called when an AudioScheduledSourceNode started or the source node starts,
+  // this method might resume the AudioContext if it was not allowed to start.
+  void StartBlockedAudioContextIfAllowed();
+
+  // Those three methods return a promise to content, that is resolved when an
+  // (possibly long) operation is completed on the MTG (and possibly other)
+  // thread(s). To avoid having to match the calls and asychronous result when
+  // the operation is completed, we keep a reference to the promises on the main
+  // thread, and then send the promises pointers down the MTG thread, as a void*
+  // (to make it very clear that the pointer is to merely be treated as an ID).
+  // When back on the main thread, we can resolve or reject the promise, by
+  // casting it back to a `Promise*` while asserting we're back on the main
+  // thread and removing the reference we added.
+  already_AddRefed<Promise> Suspend(ErrorResult& aRv);
+  already_AddRefed<Promise> Resume(ErrorResult& aRv);
+  already_AddRefed<Promise> Close(ErrorResult& aRv);
+  IMPL_EVENT_HANDLER(statechange)
+
+  // These two functions are similar with Suspend() and Resume(), the difference
+  // is they are designed for calling from chrome side, not content side. eg.
+  // calling from inner window, so we won't need to return promise for caller.
+  void SuspendFromChrome();
+  void ResumeFromChrome();
+  // Called on completion of offline rendering:
+  void OfflineClose();
+
+  already_AddRefed<AudioBufferSourceNode> CreateBufferSource();
+
+  already_AddRefed<ConstantSourceNode> CreateConstantSource();
+
+  already_AddRefed<AudioBuffer> CreateBuffer(uint32_t aNumberOfChannels,
+                                             uint32_t aLength,
+                                             float aSampleRate,
+                                             ErrorResult& aRv);
+
+  already_AddRefed<MediaStreamAudioDestinationNode>
+  CreateMediaStreamDestination(ErrorResult& aRv);
+
+  already_AddRefed<ScriptProcessorNode> CreateScriptProcessor(
+      uint32_t aBufferSize, uint32_t aNumberOfInputChannels,
+      uint32_t aNumberOfOutputChannels, ErrorResult& aRv);
+
+  already_AddRefed<StereoPannerNode> CreateStereoPanner(ErrorResult& aRv);
+
+  already_AddRefed<AnalyserNode> CreateAnalyser(ErrorResult& aRv);
+
+  already_AddRefed<GainNode> CreateGain(ErrorResult& aRv);
+
+  already_AddRefed<WaveShaperNode> CreateWaveShaper(ErrorResult& aRv);
+
+  already_AddRefed<MediaElementAudioSourceNode> CreateMediaElementSource(
+      HTMLMediaElement& aMediaElement, ErrorResult& aRv);
+  already_AddRefed<MediaStreamAudioSourceNode> CreateMediaStreamSource(
+      DOMMediaStream& aMediaStream, ErrorResult& aRv);
+  already_AddRefed<MediaStreamTrackAudioSourceNode>
+  CreateMediaStreamTrackSource(MediaStreamTrack& aMediaStreamTrack,
+                               ErrorResult& aRv);
+
+  already_AddRefed<DelayNode> CreateDelay(double aMaxDelayTime,
+                                          ErrorResult& aRv);
+
+  already_AddRefed<PannerNode> CreatePanner(ErrorResult& aRv);
+
+  already_AddRefed<ConvolverNode> CreateConvolver(ErrorResult& aRv);
+
+  already_AddRefed<ChannelSplitterNode> CreateChannelSplitter(
+      uint32_t aNumberOfOutputs, ErrorResult& aRv);
+
+  already_AddRefed<ChannelMergerNode> CreateChannelMerger(
+      uint32_t aNumberOfInputs, ErrorResult& aRv);
+
+  already_AddRefed<DynamicsCompressorNode> CreateDynamicsCompressor(
+      ErrorResult& aRv);
+
+  already_AddRefed<BiquadFilterNode> CreateBiquadFilter(ErrorResult& aRv);
+
+  already_AddRefed<IIRFilterNode> CreateIIRFilter(
+      const Sequence<double>& aFeedforward, const Sequence<double>& aFeedback,
+      mozilla::ErrorResult& aRv);
+
+  already_AddRefed<OscillatorNode> CreateOscillator(ErrorResult& aRv);
+
+  already_AddRefed<PeriodicWave> CreatePeriodicWave(
+      const Sequence<float>& aRealData, const Sequence<float>& aImagData,
+      const PeriodicWaveConstraints& aConstraints, ErrorResult& aRv);
+
+  already_AddRefed<Promise> DecodeAudioData(
+      const ArrayBuffer& aBuffer,
+      const Optional<OwningNonNull<DecodeSuccessCallback>>& aSuccessCallback,
+      const Optional<OwningNonNull<DecodeErrorCallback>>& aFailureCallback,
+      ErrorResult& aRv);
+
+  // OfflineAudioContext methods
+  already_AddRefed<Promise> StartRendering(ErrorResult& aRv);
+  IMPL_EVENT_HANDLER(complete)
+  unsigned long Length();
+
+  bool IsOffline() const { return mIsOffline; }
+
+  bool ShouldResistFingerprinting() const {
+    return mShouldResistFingerprinting;
+  }
+
+  MediaTrackGraph* Graph() const;
+  AudioNodeTrack* DestinationTrack() const;
+
+  // Nodes register here if they will produce sound even if they have silent
+  // or no input connections.  The AudioContext will keep registered nodes
+  // alive until the context is collected.  This takes care of "playing"
+  // references and "tail-time" references.
+  void RegisterActiveNode(AudioNode* aNode);
+  // Nodes unregister when they have finished producing sound for the
+  // foreseeable future.
+  // Do NOT call UnregisterActiveNode from an AudioNode destructor.
+  // If the destructor is called, then the Node has already been unregistered.
+  // The destructor may be called during hashtable enumeration, during which
+  // unregistering would not be safe.
+  void UnregisterActiveNode(AudioNode* aNode);
+
+  uint32_t MaxChannelCount() const;
+
+  uint32_t ActiveNodeCount() const;
+
+  void Mute() const;
+  void Unmute() const;
+
+  void RegisterNode(AudioNode* aNode);
+  void UnregisterNode(AudioNode* aNode);
+
+  void OnStateChanged(void* aPromise, AudioContextState aNewState);
+
+  BasicWaveFormCache* GetBasicWaveFormCache();
+
+  void ShutdownWorklet();
+  // Steals from |aParamMap|
+  void SetParamMapForWorkletName(const nsAString& aName,
+                                 AudioParamDescriptorMap* aParamMap);
+  const AudioParamDescriptorMap* GetParamMapForWorkletName(
+      const nsAString& aName) {
+    return mWorkletParamDescriptors.Lookup(aName).DataPtrOrNull();
+  }
+
+  void Dispatch(already_AddRefed<nsIRunnable>&& aRunnable);
+
+ private:
+  void DisconnectFromWindow();
+  already_AddRefed<Promise> CreatePromise(ErrorResult& aRv);
+  void RemoveFromDecodeQueue(WebAudioDecodeJob* aDecodeJob);
+  void ShutdownDecoder();
+
+  size_t SizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+  NS_DECL_NSIMEMORYREPORTER
+
+  friend struct ::mozilla::WebAudioDecodeJob;
+
+  nsTArray<RefPtr<mozilla::MediaTrack>> GetAllTracks() const;
+
+  void ResumeInternal();
+  void SuspendInternal(void* aPromise, AudioContextOperationFlags aFlags);
+  void CloseInternal(void* aPromise, AudioContextOperationFlags aFlags);
+
+  // Will report error message to console and dispatch testing event if needed
+  // when AudioContext is blocked by autoplay policy.
+  void ReportBlocked();
+
+  void ReportToConsole(uint32_t aErrorFlags, const char* aMsg) const;
+
+  // This function should be called everytime we decide whether allow to start
+  // audio context, it's used to update Telemetry related variables.
+  void UpdateAutoplayAssumptionStatus();
+
+  // These functions are used for updating Telemetry.
+  // - MaybeUpdateAutoplayTelemetry: update category 'AllowedAfterBlocked'
+  // - MaybeUpdateAutoplayTelemetryWhenShutdown: update category 'NeverBlocked'
+  //   and 'NeverAllowed', so we need to call it when shutdown AudioContext
+  void MaybeUpdateAutoplayTelemetry();
+  void MaybeUpdateAutoplayTelemetryWhenShutdown();
+
+  // If the pref `dom.suspend_inactive.enabled` is enabled, the dom window will
+  // be suspended when the window becomes inactive. In order to keep audio
+  // context running still, we will ask pages to keep awake in that situation.
+  void MaybeUpdatePageAwakeRequest();
+  void MaybeClearPageAwakeRequest();
+  void SetPageAwakeRequest(bool aShouldSet);
+
+  BrowsingContext* GetTopLevelBrowsingContext();
+
+ private:
+  // Each AudioContext has an id, that is passed down the MediaTracks that
+  // back the AudioNodes, so we can easily compute the set of all the
+  // MediaTracks for a given context, on the MediasTrackGraph side.
+  const AudioContextId mId;
+  // Note that it's important for mSampleRate to be initialized before
+  // mDestination, as mDestination's constructor needs to access it!
+  const float mSampleRate;
+  AudioContextState mAudioContextState;
+  RefPtr<AudioDestinationNode> mDestination;
+  RefPtr<AudioListener> mListener;
+  RefPtr<Worklet> mWorklet;
+  nsTArray<UniquePtr<WebAudioDecodeJob>> mDecodeJobs;
+  // This array is used to keep the suspend/close promises alive until
+  // they are resolved, so we can safely pass them accross threads.
+  nsTArray<RefPtr<Promise>> mPromiseGripArray;
+  // This array is used to onlly keep the resume promises alive until they are
+  // resolved, so we can safely pass them accross threads. If the audio context
+  // is not allowed to play, the promise would be pending in this array and be
+  // resolved until audio context has been allowed and user call resume() again.
+  nsTArray<RefPtr<Promise>> mPendingResumePromises;
+  // See RegisterActiveNode.  These will keep the AudioContext alive while it
+  // is rendering and the window remains alive.
+  nsTHashSet<RefPtr<AudioNode>> mActiveNodes;
+  // Raw (non-owning) references to all AudioNodes for this AudioContext.
+  nsTHashSet<AudioNode*> mAllNodes;
+  nsTHashMap<nsStringHashKey, AudioParamDescriptorMap> mWorkletParamDescriptors;
+  // Cache to avoid recomputing basic waveforms all the time.
+  RefPtr<BasicWaveFormCache> mBasicWaveFormCache;
+  // Number of channels passed in the OfflineAudioContext ctor.
+  uint32_t mNumberOfChannels;
+  const RTPCallerType mRTPCallerType;
+  const bool mShouldResistFingerprinting;
+  const bool mIsOffline;
+  // true iff realtime or startRendering() has been called.
+  bool mIsStarted;
+  bool mIsShutDown;
+  bool mIsDisconnecting;
+  // Close has been called; reject suspend and resume calls.
+  bool mCloseCalled;
+  // Whether the MediaTracks are suspended, due to one or more of
+  // !mWasAllowedToStart, mSuspendedByContent, or mSuspendedByChrome.
+  // false if offline.
+  bool mTracksAreSuspended;
+  // This flag stores the value of previous status of `allowed-to-start`.
+  // true if offline.
+  bool mWasAllowedToStart;
+  // Whether this AudioContext is suspended because the page called suspend().
+  // Unused if offline.
+  bool mSuspendedByContent;
+  // Whether this AudioContext is suspended because the Window is suspended.
+  // Unused if offline.
+  bool mSuspendedByChrome;
+
+  // These variables are used for telemetry, they're not reflect the actual
+  // status of AudioContext, they are based on the "assumption" of enabling
+  // blocking web audio. Because we want to record Telemetry no matter user
+  // enable blocking autoplay or not.
+  // - 'mWasEverAllowedToStart' would be true when AudioContext had ever been
+  //   allowed to start if we enable blocking web audio.
+  // - 'mWasEverBlockedToStart' would be true when AudioContext had ever been
+  //   blocked to start if we enable blocking web audio.
+  // - 'mWouldBeAllowedToStart' stores the value of previous status of
+  //   `allowed-to-start` if we enable blocking web audio.
+  bool mWasEverAllowedToStart;
+  bool mWasEverBlockedToStart;
+  bool mWouldBeAllowedToStart;
+
+  // Whether we have set the page awake reqeust when non-offline audio context
+  // is running. That will keep the audio context being able to continue running
+  // even if the window is inactive.
+  bool mSetPageAwakeRequest = false;
+};
+
+static const dom::AudioContext::AudioContextId NO_AUDIO_CONTEXT = 0;
+
+}  // namespace dom
+}  // namespace mozilla
+
+inline nsISupports* ToSupports(mozilla::dom::AudioContext* p) {
+  return NS_CYCLE_COLLECTION_CLASSNAME(mozilla::dom::AudioContext)::Upcast(p);
+}
+
+#endif
diff --git a/dom/media/webaudio/AudioDestinationNode.cpp b/dom/media/webaudio/AudioDestinationNode.cpp
new file mode 100644
index 0000000000..1a9c3c0f7b
--- /dev/null
+++ b/dom/media/webaudio/AudioDestinationNode.cpp
@@ -0,0 +1,675 @@
+/* -*- 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 "AudioDestinationNode.h"
+
+#include "AlignmentUtils.h"
+#include "AudibilityMonitor.h"
+#include "AudioChannelService.h"
+#include "AudioContext.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "CubebUtils.h"
+#include "MediaTrackGraph.h"
+#include "mozilla/StaticPrefs_dom.h"
+#include "mozilla/dom/AudioDestinationNodeBinding.h"
+#include "mozilla/dom/BaseAudioContextBinding.h"
+#include "mozilla/dom/OfflineAudioCompletionEvent.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/dom/ScriptSettings.h"
+#include "mozilla/dom/WakeLock.h"
+#include "mozilla/dom/power/PowerManagerService.h"
+#include "mozilla/Telemetry.h"
+#include "mozilla/TelemetryHistogramEnums.h"
+#include "nsContentUtils.h"
+#include "nsIInterfaceRequestorUtils.h"
+#include "nsIScriptObjectPrincipal.h"
+#include "nsServiceManagerUtils.h"
+#include "Tracing.h"
+
+extern mozilla::LazyLogModule gAudioChannelLog;
+
+#define AUDIO_CHANNEL_LOG(msg, ...) \
+  MOZ_LOG(gAudioChannelLog, LogLevel::Debug, (msg, ##__VA_ARGS__))
+
+namespace mozilla::dom {
+
+namespace {
+class OnCompleteTask final : public Runnable {
+ public:
+  OnCompleteTask(AudioContext* aAudioContext, AudioBuffer* aRenderedBuffer)
+      : Runnable("dom::OfflineDestinationNodeEngine::OnCompleteTask"),
+        mAudioContext(aAudioContext),
+        mRenderedBuffer(aRenderedBuffer) {}
+
+  NS_IMETHOD Run() override {
+    OfflineAudioCompletionEventInit param;
+    param.mRenderedBuffer = mRenderedBuffer;
+
+    RefPtr<OfflineAudioCompletionEvent> event =
+        OfflineAudioCompletionEvent::Constructor(mAudioContext, u"complete"_ns,
+                                                 param);
+    mAudioContext->DispatchTrustedEvent(event);
+
+    return NS_OK;
+  }
+
+ private:
+  RefPtr<AudioContext> mAudioContext;
+  RefPtr<AudioBuffer> mRenderedBuffer;
+};
+}  // anonymous namespace
+
+class OfflineDestinationNodeEngine final : public AudioNodeEngine {
+ public:
+  explicit OfflineDestinationNodeEngine(AudioDestinationNode* aNode)
+      : AudioNodeEngine(aNode),
+        mWriteIndex(0),
+        mNumberOfChannels(aNode->ChannelCount()),
+        mLength(aNode->Length()),
+        mSampleRate(aNode->Context()->SampleRate()),
+        mBufferAllocated(false) {}
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    TRACE("OfflineDestinationNodeEngine::ProcessBlock");
+    // Do this just for the sake of political correctness; this output
+    // will not go anywhere.
+    *aOutput = aInput;
+
+    // The output buffer is allocated lazily, on the rendering thread, when
+    // non-null input is received.
+    if (!mBufferAllocated && !aInput.IsNull()) {
+      // These allocations might fail if content provides a huge number of
+      // channels or size, but it's OK since we'll deal with the failure
+      // gracefully.
+      mBuffer = ThreadSharedFloatArrayBufferList::Create(mNumberOfChannels,
+                                                         mLength, fallible);
+      if (mBuffer && mWriteIndex) {
+        // Zero leading for any null chunks that were skipped.
+        for (uint32_t i = 0; i < mNumberOfChannels; ++i) {
+          float* channelData = mBuffer->GetDataForWrite(i);
+          PodZero(channelData, mWriteIndex);
+        }
+      }
+
+      mBufferAllocated = true;
+    }
+
+    // Skip copying if there is no buffer.
+    uint32_t outputChannelCount = mBuffer ? mNumberOfChannels : 0;
+
+    // Record our input buffer
+    MOZ_ASSERT(mWriteIndex < mLength, "How did this happen?");
+    const uint32_t duration =
+        std::min(WEBAUDIO_BLOCK_SIZE, mLength - mWriteIndex);
+    const uint32_t inputChannelCount = aInput.ChannelCount();
+    for (uint32_t i = 0; i < outputChannelCount; ++i) {
+      float* outputData = mBuffer->GetDataForWrite(i) + mWriteIndex;
+      if (aInput.IsNull() || i >= inputChannelCount) {
+        PodZero(outputData, duration);
+      } else {
+        const float* inputBuffer =
+            static_cast<const float*>(aInput.mChannelData[i]);
+        if (duration == WEBAUDIO_BLOCK_SIZE && IS_ALIGNED16(inputBuffer)) {
+          // Use the optimized version of the copy with scale operation
+          AudioBlockCopyChannelWithScale(inputBuffer, aInput.mVolume,
+                                         outputData);
+        } else {
+          if (aInput.mVolume == 1.0f) {
+            PodCopy(outputData, inputBuffer, duration);
+          } else {
+            for (uint32_t j = 0; j < duration; ++j) {
+              outputData[j] = aInput.mVolume * inputBuffer[j];
+            }
+          }
+        }
+      }
+    }
+    mWriteIndex += duration;
+
+    if (mWriteIndex >= mLength) {
+      NS_ASSERTION(mWriteIndex == mLength, "Overshot length");
+      // Go to finished state. When the graph's current time eventually reaches
+      // the end of the track, then the main thread will be notified and we'll
+      // shut down the AudioContext.
+      *aFinished = true;
+    }
+  }
+
+  bool IsActive() const override {
+    // Keep processing to track track time, which is used for all timelines
+    // associated with the same AudioContext.
+    return true;
+  }
+
+  already_AddRefed<AudioBuffer> CreateAudioBuffer(AudioContext* aContext) {
+    MOZ_ASSERT(NS_IsMainThread());
+    // Create the input buffer
+    ErrorResult rv;
+    RefPtr<AudioBuffer> renderedBuffer =
+        AudioBuffer::Create(aContext->GetOwner(), mNumberOfChannels, mLength,
+                            mSampleRate, mBuffer.forget(), rv);
+    if (rv.Failed()) {
+      rv.SuppressException();
+      return nullptr;
+    }
+
+    return renderedBuffer.forget();
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    if (mBuffer) {
+      amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
+    }
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  // The input to the destination node is recorded in mBuffer.
+  // When this buffer fills up with mLength frames, the buffered input is sent
+  // to the main thread in order to dispatch OfflineAudioCompletionEvent.
+  RefPtr<ThreadSharedFloatArrayBufferList> mBuffer;
+  // An index representing the next offset in mBuffer to be written to.
+  uint32_t mWriteIndex;
+  uint32_t mNumberOfChannels;
+  // How many frames the OfflineAudioContext intends to produce.
+  uint32_t mLength;
+  float mSampleRate;
+  bool mBufferAllocated;
+};
+
+class DestinationNodeEngine final : public AudioNodeEngine {
+ public:
+  explicit DestinationNodeEngine(AudioDestinationNode* aNode)
+      : AudioNodeEngine(aNode),
+        mSampleRate(CubebUtils::PreferredSampleRate(
+            aNode->Context()->ShouldResistFingerprinting())),
+        mVolume(1.0f),
+        mAudibilityMonitor(
+            mSampleRate,
+            StaticPrefs::dom_media_silence_duration_for_audibility()),
+        mSuspended(false),
+        mIsAudible(false) {
+    MOZ_ASSERT(aNode);
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    TRACE("DestinationNodeEngine::ProcessBlock");
+    *aOutput = aInput;
+    aOutput->mVolume *= mVolume;
+
+    if (mSuspended) {
+      return;
+    }
+
+    mAudibilityMonitor.Process(aInput);
+    bool isAudible =
+        mAudibilityMonitor.RecentlyAudible() && aOutput->mVolume > 0.0;
+    if (isAudible != mIsAudible) {
+      mIsAudible = isAudible;
+      RefPtr<AudioNodeTrack> track = aTrack;
+      auto r = [track, isAudible]() -> void {
+        MOZ_ASSERT(NS_IsMainThread());
+        RefPtr<AudioNode> node = track->Engine()->NodeMainThread();
+        if (node) {
+          RefPtr<AudioDestinationNode> destinationNode =
+              static_cast<AudioDestinationNode*>(node.get());
+          destinationNode->NotifyDataAudibleStateChanged(isAudible);
+        }
+      };
+
+      aTrack->Graph()->DispatchToMainThreadStableState(NS_NewRunnableFunction(
+          "dom::WebAudioAudibleStateChangedRunnable", r));
+    }
+  }
+
+  bool IsActive() const override {
+    // Keep processing to track track time, which is used for all timelines
+    // associated with the same AudioContext.  If there are no other engines
+    // for the AudioContext, then this could return false to suspend the
+    // track, but the track is blocked anyway through
+    // AudioDestinationNode::SetIsOnlyNodeForContext().
+    return true;
+  }
+
+  void SetDoubleParameter(uint32_t aIndex, double aParam) override {
+    if (aIndex == VOLUME) {
+      mVolume = static_cast<float>(aParam);
+    }
+  }
+
+  void SetInt32Parameter(uint32_t aIndex, int32_t aParam) override {
+    if (aIndex == SUSPENDED) {
+      mSuspended = !!aParam;
+      if (mSuspended) {
+        mIsAudible = false;
+      }
+    }
+  }
+
+  enum Parameters {
+    VOLUME,
+    SUSPENDED,
+  };
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  int mSampleRate;
+  float mVolume;
+  AudibilityMonitor mAudibilityMonitor;
+  bool mSuspended;
+  bool mIsAudible;
+};
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(AudioDestinationNode, AudioNode,
+                                   mAudioChannelAgent, mOfflineRenderingPromise)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(AudioDestinationNode)
+  NS_INTERFACE_MAP_ENTRY(nsIAudioChannelAgentCallback)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(AudioDestinationNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(AudioDestinationNode, AudioNode)
+
+const AudioNodeTrack::Flags kTrackFlags =
+    AudioNodeTrack::NEED_MAIN_THREAD_CURRENT_TIME |
+    AudioNodeTrack::NEED_MAIN_THREAD_ENDED | AudioNodeTrack::EXTERNAL_OUTPUT;
+
+AudioDestinationNode::AudioDestinationNode(AudioContext* aContext,
+                                           bool aIsOffline,
+                                           uint32_t aNumberOfChannels,
+                                           uint32_t aLength)
+    : AudioNode(aContext, aNumberOfChannels, ChannelCountMode::Explicit,
+                ChannelInterpretation::Speakers),
+      mFramesToProduce(aLength),
+      mIsOffline(aIsOffline),
+      mCreatedTime(TimeStamp::Now()) {
+  if (aIsOffline) {
+    // The track is created on demand to avoid creating a graph thread that
+    // may not be used.
+    return;
+  }
+
+  // GetParentObject can return nullptr here. This will end up creating another
+  // MediaTrackGraph
+  MediaTrackGraph* graph = MediaTrackGraph::GetInstance(
+      MediaTrackGraph::AUDIO_THREAD_DRIVER, aContext->GetParentObject(),
+      aContext->SampleRate(), MediaTrackGraph::DEFAULT_OUTPUT_DEVICE);
+  AudioNodeEngine* engine = new DestinationNodeEngine(this);
+
+  mTrack = AudioNodeTrack::Create(aContext, engine, kTrackFlags, graph);
+  mTrack->AddMainThreadListener(this);
+  // null key is fine: only one output per mTrack
+  mTrack->AddAudioOutput(nullptr, nullptr);
+}
+
+void AudioDestinationNode::Init() {
+  // The reason we don't do that in ctor is because we have to keep AudioContext
+  // holding a strong reference to the destination node first. If we don't do
+  // that, initializing the agent would cause an unexpected destroy of the
+  // destination node when destroying the local weak reference inside
+  // `InitWithWeakCallback()`.
+  if (!mIsOffline) {
+    CreateAndStartAudioChannelAgent();
+  }
+}
+
+void AudioDestinationNode::Close() {
+  DestroyAudioChannelAgentIfExists();
+  ReleaseAudioWakeLockIfExists();
+}
+
+void AudioDestinationNode::CreateAndStartAudioChannelAgent() {
+  MOZ_ASSERT(!mIsOffline);
+  MOZ_ASSERT(!mAudioChannelAgent);
+
+  AudioChannelAgent* agent = new AudioChannelAgent();
+  nsresult rv = agent->InitWithWeakCallback(GetOwner(), this);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    AUDIO_CHANNEL_LOG("Failed to init audio channel agent");
+    return;
+  }
+
+  AudibleState state =
+      IsAudible() ? AudibleState::eAudible : AudibleState::eNotAudible;
+  rv = agent->NotifyStartedPlaying(state);
+  if (NS_WARN_IF(NS_FAILED(rv))) {
+    AUDIO_CHANNEL_LOG("Failed to start audio channel agent");
+    return;
+  }
+
+  mAudioChannelAgent = agent;
+  mAudioChannelAgent->PullInitialUpdate();
+}
+
+AudioDestinationNode::~AudioDestinationNode() {
+  MOZ_ASSERT(!mAudioChannelAgent);
+  MOZ_ASSERT(!mWakeLock);
+  MOZ_ASSERT(!mCaptureTrackPort);
+}
+
+size_t AudioDestinationNode::SizeOfExcludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  // Might be useful in the future:
+  // - mAudioChannelAgent
+  return amount;
+}
+
+size_t AudioDestinationNode::SizeOfIncludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+AudioNodeTrack* AudioDestinationNode::Track() {
+  if (mTrack) {
+    return mTrack;
+  }
+
+  AudioContext* context = Context();
+  if (!context) {  // This node has been unlinked.
+    return nullptr;
+  }
+
+  MOZ_ASSERT(mIsOffline, "Realtime tracks are created in constructor");
+
+  // GetParentObject can return nullptr here when the document has been
+  // unlinked.
+  MediaTrackGraph* graph =
+      MediaTrackGraph::CreateNonRealtimeInstance(context->SampleRate());
+  AudioNodeEngine* engine = new OfflineDestinationNodeEngine(this);
+
+  mTrack = AudioNodeTrack::Create(context, engine, kTrackFlags, graph);
+  mTrack->AddMainThreadListener(this);
+
+  return mTrack;
+}
+
+void AudioDestinationNode::DestroyAudioChannelAgentIfExists() {
+  if (mAudioChannelAgent) {
+    mAudioChannelAgent->NotifyStoppedPlaying();
+    mAudioChannelAgent = nullptr;
+    if (IsCapturingAudio()) {
+      StopAudioCapturingTrack();
+    }
+  }
+}
+
+void AudioDestinationNode::DestroyMediaTrack() {
+  Close();
+  if (!mTrack) {
+    return;
+  }
+
+  Context()->ShutdownWorklet();
+
+  mTrack->RemoveMainThreadListener(this);
+  AudioNode::DestroyMediaTrack();
+}
+
+void AudioDestinationNode::NotifyMainThreadTrackEnded() {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(mTrack->IsEnded());
+
+  if (mIsOffline) {
+    AbstractThread::MainThread()->Dispatch(NewRunnableMethod(
+        "dom::AudioDestinationNode::FireOfflineCompletionEvent", this,
+        &AudioDestinationNode::FireOfflineCompletionEvent));
+  }
+}
+
+void AudioDestinationNode::FireOfflineCompletionEvent() {
+  AudioContext* context = Context();
+  context->OfflineClose();
+
+  OfflineDestinationNodeEngine* engine =
+      static_cast<OfflineDestinationNodeEngine*>(Track()->Engine());
+  RefPtr<AudioBuffer> renderedBuffer = engine->CreateAudioBuffer(context);
+  if (!renderedBuffer) {
+    return;
+  }
+  ResolvePromise(renderedBuffer);
+
+  context->Dispatch(do_AddRef(new OnCompleteTask(context, renderedBuffer)));
+
+  context->OnStateChanged(nullptr, AudioContextState::Closed);
+
+  mOfflineRenderingRef.Drop(this);
+}
+
+void AudioDestinationNode::ResolvePromise(AudioBuffer* aRenderedBuffer) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(mIsOffline);
+  mOfflineRenderingPromise->MaybeResolve(aRenderedBuffer);
+}
+
+uint32_t AudioDestinationNode::MaxChannelCount() const {
+  return Context()->MaxChannelCount();
+}
+
+void AudioDestinationNode::SetChannelCount(uint32_t aChannelCount,
+                                           ErrorResult& aRv) {
+  if (aChannelCount > MaxChannelCount()) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("%u is larger than maxChannelCount", aChannelCount));
+    return;
+  }
+
+  if (aChannelCount == ChannelCount()) {
+    return;
+  }
+
+  AudioNode::SetChannelCount(aChannelCount, aRv);
+}
+
+void AudioDestinationNode::Mute() {
+  MOZ_ASSERT(Context() && !Context()->IsOffline());
+  SendDoubleParameterToTrack(DestinationNodeEngine::VOLUME, 0.0f);
+}
+
+void AudioDestinationNode::Unmute() {
+  MOZ_ASSERT(Context() && !Context()->IsOffline());
+  SendDoubleParameterToTrack(DestinationNodeEngine::VOLUME, 1.0f);
+}
+
+void AudioDestinationNode::Suspend() {
+  SendInt32ParameterToTrack(DestinationNodeEngine::SUSPENDED, 1);
+}
+
+void AudioDestinationNode::Resume() {
+  SendInt32ParameterToTrack(DestinationNodeEngine::SUSPENDED, 0);
+}
+
+void AudioDestinationNode::NotifyAudioContextStateChanged() {
+  UpdateFinalAudibleStateIfNeeded(AudibleChangedReasons::ePauseStateChanged);
+}
+
+void AudioDestinationNode::OfflineShutdown() {
+  MOZ_ASSERT(Context() && Context()->IsOffline(),
+             "Should only be called on a valid OfflineAudioContext");
+
+  mOfflineRenderingRef.Drop(this);
+}
+
+JSObject* AudioDestinationNode::WrapObject(JSContext* aCx,
+                                           JS::Handle<JSObject*> aGivenProto) {
+  return AudioDestinationNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void AudioDestinationNode::StartRendering(Promise* aPromise) {
+  mOfflineRenderingPromise = aPromise;
+  mOfflineRenderingRef.Take(this);
+  Track()->Graph()->StartNonRealtimeProcessing(mFramesToProduce);
+}
+
+NS_IMETHODIMP
+AudioDestinationNode::WindowVolumeChanged(float aVolume, bool aMuted) {
+  MOZ_ASSERT(mAudioChannelAgent);
+  if (!mTrack) {
+    return NS_OK;
+  }
+
+  AUDIO_CHANNEL_LOG(
+      "AudioDestinationNode %p WindowVolumeChanged, "
+      "aVolume = %f, aMuted = %s\n",
+      this, aVolume, aMuted ? "true" : "false");
+
+  mAudioChannelVolume = aMuted ? 0.0f : aVolume;
+  mTrack->SetAudioOutputVolume(nullptr, mAudioChannelVolume);
+  UpdateFinalAudibleStateIfNeeded(AudibleChangedReasons::eVolumeChanged);
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+AudioDestinationNode::WindowSuspendChanged(nsSuspendedTypes aSuspend) {
+  MOZ_ASSERT(mAudioChannelAgent);
+  if (!mTrack) {
+    return NS_OK;
+  }
+
+  const bool shouldDisable = aSuspend == nsISuspendedTypes::SUSPENDED_BLOCK;
+  if (mAudioChannelDisabled == shouldDisable) {
+    return NS_OK;
+  }
+  mAudioChannelDisabled = shouldDisable;
+
+  AUDIO_CHANNEL_LOG(
+      "AudioDestinationNode %p WindowSuspendChanged, shouldDisable = %d\n",
+      this, mAudioChannelDisabled);
+
+  DisabledTrackMode disabledMode = mAudioChannelDisabled
+                                       ? DisabledTrackMode::SILENCE_BLACK
+                                       : DisabledTrackMode::ENABLED;
+  mTrack->SetDisabledTrackMode(disabledMode);
+  UpdateFinalAudibleStateIfNeeded(AudibleChangedReasons::ePauseStateChanged);
+  return NS_OK;
+}
+
+NS_IMETHODIMP
+AudioDestinationNode::WindowAudioCaptureChanged(bool aCapture) {
+  MOZ_ASSERT(mAudioChannelAgent);
+  if (!mTrack) {
+    return NS_OK;
+  }
+
+  nsCOMPtr<nsPIDOMWindowInner> ownerWindow = GetOwner();
+  if (!ownerWindow) {
+    return NS_OK;
+  }
+
+  if (aCapture == IsCapturingAudio()) {
+    return NS_OK;
+  }
+
+  if (aCapture) {
+    StartAudioCapturingTrack();
+  } else {
+    StopAudioCapturingTrack();
+  }
+
+  return NS_OK;
+}
+
+bool AudioDestinationNode::IsCapturingAudio() const {
+  return mCaptureTrackPort != nullptr;
+}
+
+void AudioDestinationNode::StartAudioCapturingTrack() {
+  MOZ_ASSERT(!IsCapturingAudio());
+  nsCOMPtr<nsPIDOMWindowInner> window = Context()->GetParentObject();
+  uint64_t id = window->WindowID();
+  mCaptureTrackPort = mTrack->Graph()->ConnectToCaptureTrack(id, mTrack);
+}
+
+void AudioDestinationNode::StopAudioCapturingTrack() {
+  MOZ_ASSERT(IsCapturingAudio());
+  mCaptureTrackPort->Destroy();
+  mCaptureTrackPort = nullptr;
+}
+
+void AudioDestinationNode::CreateAudioWakeLockIfNeeded() {
+  if (!mWakeLock && IsAudible()) {
+    RefPtr<power::PowerManagerService> pmService =
+        power::PowerManagerService::GetInstance();
+    NS_ENSURE_TRUE_VOID(pmService);
+
+    ErrorResult rv;
+    mWakeLock = pmService->NewWakeLock(u"audio-playing"_ns, GetOwner(), rv);
+  }
+}
+
+void AudioDestinationNode::ReleaseAudioWakeLockIfExists() {
+  if (mWakeLock) {
+    IgnoredErrorResult rv;
+    mWakeLock->Unlock(rv);
+    mWakeLock = nullptr;
+  }
+}
+
+void AudioDestinationNode::NotifyDataAudibleStateChanged(bool aAudible) {
+  MOZ_ASSERT(!mIsOffline);
+
+  AUDIO_CHANNEL_LOG(
+      "AudioDestinationNode %p NotifyDataAudibleStateChanged, audible=%d", this,
+      aAudible);
+
+  if (mDurationBeforeFirstTimeAudible.IsZero()) {
+    MOZ_ASSERT(aAudible);
+    mDurationBeforeFirstTimeAudible = TimeStamp::Now() - mCreatedTime;
+    Telemetry::Accumulate(Telemetry::WEB_AUDIO_BECOMES_AUDIBLE_TIME,
+                          mDurationBeforeFirstTimeAudible.ToSeconds());
+  }
+
+  mIsDataAudible = aAudible;
+  UpdateFinalAudibleStateIfNeeded(AudibleChangedReasons::eDataAudibleChanged);
+}
+
+void AudioDestinationNode::UpdateFinalAudibleStateIfNeeded(
+    AudibleChangedReasons aReason) {
+  // The audio context has been closed and we've destroyed the agent.
+  if (!mAudioChannelAgent) {
+    return;
+  }
+  const bool newAudibleState = IsAudible();
+  if (mFinalAudibleState == newAudibleState) {
+    return;
+  }
+  AUDIO_CHANNEL_LOG("AudioDestinationNode %p Final audible state=%d", this,
+                    newAudibleState);
+  mFinalAudibleState = newAudibleState;
+  AudibleState state =
+      mFinalAudibleState ? AudibleState::eAudible : AudibleState::eNotAudible;
+  mAudioChannelAgent->NotifyStartedAudible(state, aReason);
+  if (mFinalAudibleState) {
+    CreateAudioWakeLockIfNeeded();
+  } else {
+    ReleaseAudioWakeLockIfExists();
+  }
+}
+
+bool AudioDestinationNode::IsAudible() const {
+  // The desitionation node will be regarded as audible if all following
+  // conditions are true.
+  // (1) data audible state : both audio input and output are audible
+  // (2) window audible state : the tab isn't muted by tab sound indicator
+  // (3) audio context state : audio context should be running
+  return Context()->State() == AudioContextState::Running && mIsDataAudible &&
+         mAudioChannelVolume != 0.0;
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioDestinationNode.h b/dom/media/webaudio/AudioDestinationNode.h
new file mode 100644
index 0000000000..df4a9b16ec
--- /dev/null
+++ b/dom/media/webaudio/AudioDestinationNode.h
@@ -0,0 +1,133 @@
+/* -*- 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/. */
+
+#ifndef AudioDestinationNode_h_
+#define AudioDestinationNode_h_
+
+#include "AudioChannelService.h"
+#include "AudioNode.h"
+#include "AudioChannelAgent.h"
+#include "mozilla/TimeStamp.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+class WakeLock;
+
+class AudioDestinationNode final : public AudioNode,
+                                   public nsIAudioChannelAgentCallback,
+                                   public MainThreadMediaTrackListener {
+ public:
+  // This node type knows what MediaTrackGraph to use based on
+  // whether it's in offline mode.
+  AudioDestinationNode(AudioContext* aContext, bool aIsOffline,
+                       uint32_t aNumberOfChannels, uint32_t aLength);
+
+  void DestroyMediaTrack() override;
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioDestinationNode, AudioNode)
+  NS_DECL_NSIAUDIOCHANNELAGENTCALLBACK
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  uint16_t NumberOfOutputs() const final { return 0; }
+
+  uint32_t MaxChannelCount() const;
+  void SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv) override;
+
+  void Init();
+  void Close();
+
+  // Returns the track or null after unlink.
+  AudioNodeTrack* Track();
+
+  void Mute();
+  void Unmute();
+
+  void Suspend();
+  void Resume();
+
+  void StartRendering(Promise* aPromise);
+
+  void OfflineShutdown();
+
+  void NotifyMainThreadTrackEnded() override;
+  void FireOfflineCompletionEvent();
+
+  const char* NodeType() const override { return "AudioDestinationNode"; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+  void NotifyDataAudibleStateChanged(bool aAudible);
+  void ResolvePromise(AudioBuffer* aRenderedBuffer);
+
+  unsigned long Length() {
+    MOZ_ASSERT(mIsOffline);
+    return mFramesToProduce;
+  }
+
+  void NotifyAudioContextStateChanged();
+
+ protected:
+  virtual ~AudioDestinationNode();
+
+ private:
+  // This would be created for non-offline audio context in order to receive
+  // tab's mute/suspend/audio capture state change and update the audible state
+  // to the tab.
+  void CreateAndStartAudioChannelAgent();
+  void DestroyAudioChannelAgentIfExists();
+  RefPtr<AudioChannelAgent> mAudioChannelAgent;
+
+  // These members are related to audio capturing. We would start capturing
+  // audio if we're starting capturing audio from whole window, and MUST stop
+  // capturing explicitly when we don't need to capture audio any more, because
+  // we have to release the resource we allocated before.
+  bool IsCapturingAudio() const;
+  void StartAudioCapturingTrack();
+  void StopAudioCapturingTrack();
+  RefPtr<MediaInputPort> mCaptureTrackPort;
+
+  // These members are used to determine if the destination node is actual
+  // audible and `mFinalAudibleState` represents the final result.
+  using AudibleChangedReasons = AudioChannelService::AudibleChangedReasons;
+  using AudibleState = AudioChannelService::AudibleState;
+  void UpdateFinalAudibleStateIfNeeded(AudibleChangedReasons aReason);
+  bool IsAudible() const;
+  bool mFinalAudibleState = false;
+  bool mIsDataAudible = false;
+  float mAudioChannelVolume = 1.0;
+
+  // True if the audio channel disables the track for unvisited tab, and the
+  // track will be enabled again when the tab gets first visited, or a user
+  // presses the tab play icon.
+  bool mAudioChannelDisabled = false;
+
+  // When the destination node is audible, we would request a wakelock to
+  // prevent computer from sleeping in order to keep audio playing.
+  void CreateAudioWakeLockIfNeeded();
+  void ReleaseAudioWakeLockIfExists();
+  RefPtr<WakeLock> mWakeLock;
+
+  SelfReference<AudioDestinationNode> mOfflineRenderingRef;
+  uint32_t mFramesToProduce;
+
+  RefPtr<Promise> mOfflineRenderingPromise;
+
+  bool mIsOffline;
+
+  // These varaibles are used to know how long AudioContext would become audible
+  // since it was created.
+  TimeStamp mCreatedTime;
+  TimeDuration mDurationBeforeFirstTimeAudible;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/AudioEventTimeline.cpp b/dom/media/webaudio/AudioEventTimeline.cpp
new file mode 100644
index 0000000000..f960cadd62
--- /dev/null
+++ b/dom/media/webaudio/AudioEventTimeline.cpp
@@ -0,0 +1,513 @@
+/* -*- 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 "AudioEventTimeline.h"
+#include "AudioNodeTrack.h"
+
+#include "mozilla/ErrorResult.h"
+
+using mozilla::Span;
+
+// v1 and v0 are passed from float variables but converted to double for
+// double precision interpolation.
+static void FillLinearRamp(double aBufferStartTime, Span<float> aBuffer,
+                           double t0, double v0, double t1, double v1) {
+  double bufferStartDelta = aBufferStartTime - t0;
+  double gradient = (v1 - v0) / (t1 - t0);
+  for (size_t i = 0; i < aBuffer.Length(); ++i) {
+    double v = v0 + (bufferStartDelta + static_cast<double>(i)) * gradient;
+    aBuffer[i] = static_cast<float>(v);
+  }
+}
+
+static void FillExponentialRamp(double aBufferStartTime, Span<float> aBuffer,
+                                double t0, float v0, double t1, float v1) {
+  MOZ_ASSERT(aBuffer.Length() >= 1);
+  double fullRatio = static_cast<double>(v1) / v0;
+  if (v0 == 0.f || fullRatio < 0.0) {
+    std::fill_n(aBuffer.Elements(), aBuffer.Length(), v0);
+    return;
+  }
+
+  double tDelta = t1 - t0;
+  // Calculate the value for the first tick from the curve initial value.
+  // v(t) = v0 * (v1/v0)^((t-t0)/(t1-t0))
+  double exponent = (aBufferStartTime - t0) / tDelta;
+  // The power function can amplify rounding error in the exponent by
+  // ((t−t0)/(t1−t0)) ln (v1/v0).  The single precision exponent argument for
+  // powf() would be sufficient when max(v1/v0,v0/v1) <= e, where e is Euler's
+  // number, but fdlibm's single precision powf() is not expected to provide
+  // speed advantages over double precision pow().
+  double v = v0 * fdlibm_pow(fullRatio, exponent);
+  aBuffer[0] = static_cast<float>(v);
+  if (aBuffer.Length() == 1) {
+    return;
+  }
+
+  // Use the inter-tick ratio to calculate values at other ticks.
+  // v(t+1) = (v1/v0)^(1/(t1-t0)) * v(t)
+  // Double precision is used so that accumulation of rounding error is not
+  // significant.
+  double tickRatio = fdlibm_pow(fullRatio, 1.0 / tDelta);
+  for (size_t i = 1; i < aBuffer.Length(); ++i) {
+    v *= tickRatio;
+    aBuffer[i] = static_cast<float>(v);
+  }
+}
+
+template <typename TimeType, typename DurationType>
+static size_t LimitedCountForDuration(size_t aMax, DurationType aDuration);
+
+template <>
+size_t LimitedCountForDuration<double>(size_t aMax, double aDuration) {
+  // aDuration is in seconds, so tick arithmetic is inappropriate,
+  // and unnecessary.
+  // GetValuesAtTime() is not available, so at most one value is fetched.
+  MOZ_ASSERT(aMax <= 1);
+  return aMax;
+}
+template <>
+size_t LimitedCountForDuration<int64_t>(size_t aMax, int64_t aDuration) {
+  MOZ_ASSERT(aDuration >= 0);
+  // int64_t aDuration is in ticks.
+  // On 32-bit systems, aDuration may be larger than SIZE_MAX.
+  // Determine the larger with int64_t to avoid truncating before the
+  // comparison.
+  return static_cast<int64_t>(aMax) <= aDuration
+             ? aMax
+             : static_cast<size_t>(aDuration);
+}
+template <>
+size_t LimitedCountForDuration<int64_t>(size_t aMax, double aDuration) {
+  MOZ_ASSERT(aDuration >= 0);
+  // double aDuration is in ticks.
+  // AudioTimelineEvent::mDuration may be larger than INT64_MAX.
+  // On 32-bit systems, mDuration may be larger than SIZE_MAX.
+  // Determine the larger with double to avoid truncating before the
+  // comparison.
+  return static_cast<double>(aMax) <= aDuration
+             ? aMax
+             : static_cast<size_t>(aDuration);
+}
+
+static float* NewCurveCopy(Span<const float> aCurve) {
+  if (aCurve.Length() == 0) {
+    return nullptr;
+  }
+
+  float* curve = new float[aCurve.Length()];
+  mozilla::PodCopy(curve, aCurve.Elements(), aCurve.Length());
+  return curve;
+}
+
+namespace mozilla::dom {
+
+AudioTimelineEvent::AudioTimelineEvent(Type aType, double aTime, float aValue,
+                                       double aTimeConstant)
+    : mType(aType),
+      mValue(aValue),
+      mTimeConstant(aTimeConstant),
+      mPerTickRatio(std::numeric_limits<double>::quiet_NaN()),
+      mTime(aTime) {}
+
+AudioTimelineEvent::AudioTimelineEvent(Type aType,
+                                       const nsTArray<float>& aValues,
+                                       double aStartTime, double aDuration)
+    : mType(aType),
+      mCurveLength(aValues.Length()),
+      mCurve(NewCurveCopy(aValues)),
+      mDuration(aDuration),
+      mTime(aStartTime) {
+  MOZ_ASSERT(aType == AudioTimelineEvent::SetValueCurve);
+}
+
+// cppcoreguidelines-pro-type-member-init does not know PodCopy().
+// NOLINTNEXTLINE(cppcoreguidelines-pro-type-member-init)
+AudioTimelineEvent::AudioTimelineEvent(const AudioTimelineEvent& rhs)
+    : mType(rhs.mType) {
+  PodCopy(this, &rhs, 1);
+
+  if (rhs.mType == AudioTimelineEvent::SetValueCurve) {
+    mCurve = NewCurveCopy(Span(rhs.mCurve, rhs.mCurveLength));
+  }
+}
+
+AudioTimelineEvent::~AudioTimelineEvent() {
+  if (mType == AudioTimelineEvent::SetValueCurve) {
+    delete[] mCurve;
+  }
+}
+
+template <class TimeType>
+double AudioTimelineEvent::EndTime() const {
+  MOZ_ASSERT(mType != AudioTimelineEvent::SetTarget);
+  if (mType == AudioTimelineEvent::SetValueCurve) {
+    return Time<TimeType>() + mDuration;
+  }
+  return Time<TimeType>();
+};
+
+float AudioTimelineEvent::EndValue() const {
+  if (mType == AudioTimelineEvent::SetValueCurve) {
+    return mCurve[mCurveLength - 1];
+  }
+  return mValue;
+};
+
+void AudioTimelineEvent::ConvertToTicks(AudioNodeTrack* aDestination) {
+  mTime = aDestination->SecondsToNearestTrackTime(mTime.Get<double>());
+  switch (mType) {
+    case SetTarget:
+      mTimeConstant *= aDestination->mSampleRate;
+      // exp(-1/timeConstant) is usually very close to 1, but its effect
+      // depends on the difference from 1 and rounding errors would
+      // accumulate, so use double precision to retain precision in the
+      // difference.  Single precision expm1f() would be sufficient, but the
+      // arithmetic in AudioTimelineEvent::FillTargetApproach() is simpler
+      // with exp().
+      mPerTickRatio =
+          mTimeConstant == 0.0 ? 0.0 : fdlibm_exp(-1.0 / mTimeConstant);
+
+      break;
+    case SetValueCurve:
+      mDuration *= aDestination->mSampleRate;
+      break;
+    default:
+      break;
+  }
+}
+
+template <class TimeType>
+void AudioTimelineEvent::FillTargetApproach(TimeType aBufferStartTime,
+                                            Span<float> aBuffer,
+                                            double v0) const {
+  MOZ_ASSERT(mType == SetTarget);
+  MOZ_ASSERT(aBuffer.Length() >= 1);
+  double v1 = mValue;
+  double vDelta = v0 - v1;
+  if (vDelta == 0.0 || mTimeConstant == 0.0) {
+    std::fill_n(aBuffer.Elements(), aBuffer.Length(), mValue);
+    return;
+  }
+
+  // v(t) = v1 + vDelta(t) where vDelta(t) = (v0-v1) * e^(-(t-t0)/timeConstant).
+  // Calculate the value for the first element in the buffer using this
+  // formulation.
+  vDelta *= fdlibm_expf(-(aBufferStartTime - Time<TimeType>()) / mTimeConstant);
+  for (size_t i = 0; true;) {
+    aBuffer[i] = static_cast<float>(v1 + vDelta);
+    ++i;
+    if (i == aBuffer.Length()) {
+      return;
+    }
+    // For other buffer elements, use the pre-computed exp(-1/timeConstant)
+    // for the inter-tick ratio of the difference from the target.
+    // vDelta(t+1) = vDelta(t) * e^(-1/timeConstant)
+    vDelta *= mPerTickRatio;
+  }
+}
+
+static_assert(TRACK_TIME_MAX >> FloatingPoint<double>::kSignificandWidth == 0,
+              "double precision must be exact for integer tick counts");
+template <class TimeType>
+void AudioTimelineEvent::FillFromValueCurve(TimeType aBufferStartTime,
+                                            Span<float> aBuffer) const {
+  MOZ_ASSERT(mType == SetValueCurve);
+  double curveStartTime = Time<TimeType>();
+  MOZ_ASSERT(aBufferStartTime >= curveStartTime);
+  MOZ_ASSERT(aBufferStartTime - curveStartTime <= mDuration);
+  MOZ_ASSERT((std::is_same<TimeType, int64_t>::value) || aBuffer.Length() == 1);
+  MOZ_ASSERT((!std::is_same<TimeType, int64_t>::value) ||
+             aBufferStartTime - curveStartTime + aBuffer.Length() - 1 <=
+                 mDuration);
+  uint32_t stepCount = mCurveLength - 1;
+  double timeStep = mDuration / stepCount;
+
+  for (size_t fillStart = 0; fillStart < aBuffer.Length();) {
+    // Find the curve sample index, spec'd as `k`, corresponding to a time less
+    // than or equal to the first buffer element to be filled.
+    double stepPos =
+        (aBufferStartTime + fillStart - curveStartTime) / mDuration * stepCount;
+    // GetValuesAtTimeHelperInternal() calls this only when
+    // aBufferStartTime + fillStart - curveStartTime <= mDuration.
+    MOZ_ASSERT(stepPos >= 0 && stepPos <= UINT32_MAX - 1);
+    uint32_t currentNode = floor(stepPos);
+    if (currentNode >= stepCount) {
+      auto remaining = aBuffer.From(fillStart);
+      std::fill_n(remaining.Elements(), remaining.Length(), mCurve[stepCount]);
+      return;
+    }
+
+    // Linearly interpolate to fill the buffer elements for any ticks between
+    // curve samples k and k + 1 inclusive.
+    double tCurrent = curveStartTime + currentNode * timeStep;
+    uint32_t nextNode = currentNode + 1;
+    double tNext = curveStartTime + nextNode * timeStep;
+    // The first buffer index that cannot be filled with these curve samples
+    size_t fillEnd = LimitedCountForDuration<TimeType>(
+        aBuffer.Length(),
+        // This parameter is used only when time is in ticks:
+        // If tNext aligns exactly with a tick then fill to tNext, thus
+        // ensuring that fillStart is advanced even when timeStep is so small
+        // that tNext == tCurrent.
+        floor(tNext - aBufferStartTime) + 1.0);
+    TimeType fillStartTime =
+        aBufferStartTime + static_cast<TimeType>(fillStart);
+    FillLinearRamp(fillStartTime, aBuffer.FromTo(fillStart, fillEnd), tCurrent,
+                   mCurve[currentNode], tNext, mCurve[nextNode]);
+    fillStart = fillEnd;
+  }
+}
+
+template <class TimeType>
+float AudioEventTimeline::ComputeSetTargetStartValue(
+    const AudioTimelineEvent* aPreviousEvent, TimeType aTime) {
+  mSetTargetStartTime = aTime;
+  GetValuesAtTimeHelperInternal(aTime, Span(&mSetTargetStartValue, 1),
+                                aPreviousEvent, nullptr);
+  return mSetTargetStartValue;
+}
+
+template void AudioEventTimeline::CleanupEventsOlderThan(double);
+template void AudioEventTimeline::CleanupEventsOlderThan(int64_t);
+template <class TimeType>
+void AudioEventTimeline::CleanupEventsOlderThan(TimeType aTime) {
+  auto TimeOf =
+      [](const decltype(mEvents)::const_iterator& aEvent) -> TimeType {
+    return aEvent->Time<TimeType>();
+  };
+
+  if (mSimpleValue.isSome()) {
+    return;  // already only a single event
+  }
+
+  // Find first event to keep.  Keep one event prior to aTime.
+  auto begin = mEvents.cbegin();
+  auto end = mEvents.cend();
+  auto event = begin + 1;
+  for (; event < end && aTime > TimeOf(event); ++event) {
+  }
+  auto firstToKeep = event - 1;
+
+  if (firstToKeep->mType != AudioTimelineEvent::SetTarget) {
+    // The value is constant if there is a single remaining non-SetTarget event
+    // that has already passed.
+    if (end - firstToKeep == 1 && aTime >= firstToKeep->EndTime<TimeType>()) {
+      mSimpleValue.emplace(firstToKeep->EndValue());
+    }
+  } else {
+    // The firstToKeep event is a SetTarget.  Set its initial value if
+    // not already set.  First find the most recent event where the value at
+    // the end time of the event is known, either from the event or for
+    // SetTarget events because it has already been calculated.  This may not
+    // have been calculated if GetValuesAtTime() was not called for the start
+    // time of the SetTarget event.
+    for (event = firstToKeep;
+         event > begin && event->mType == AudioTimelineEvent::SetTarget &&
+         TimeOf(event) > mSetTargetStartTime.Get<TimeType>();
+         --event) {
+    }
+    // Compute SetTarget start times.
+    for (; event < firstToKeep; ++event) {
+      MOZ_ASSERT((event + 1)->mType == AudioTimelineEvent::SetTarget);
+      ComputeSetTargetStartValue(&*event, TimeOf(event + 1));
+    }
+  }
+  if (firstToKeep == begin) {
+    return;
+  }
+
+  mEvents.RemoveElementsRange(begin, firstToKeep);
+}
+
+// This method computes the AudioParam value at a given time based on the event
+// timeline
+template <class TimeType>
+void AudioEventTimeline::GetValuesAtTimeHelper(TimeType aTime, float* aBuffer,
+                                               const size_t aSize) {
+  MOZ_ASSERT(aBuffer);
+  MOZ_ASSERT(aSize);
+
+  auto TimeOf = [](const AudioTimelineEvent& aEvent) -> TimeType {
+    return aEvent.Time<TimeType>();
+  };
+
+  size_t eventIndex = 0;
+  const AudioTimelineEvent* previous = nullptr;
+
+  // Let's remove old events except the last one: we need it to calculate some
+  // curves.
+  CleanupEventsOlderThan(aTime);
+
+  for (size_t bufferIndex = 0; bufferIndex < aSize;) {
+    bool timeMatchesEventIndex = false;
+    const AudioTimelineEvent* next;
+    for (;; ++eventIndex) {
+      if (eventIndex >= mEvents.Length()) {
+        next = nullptr;
+        break;
+      }
+
+      next = &mEvents[eventIndex];
+      if (aTime < TimeOf(*next)) {
+        break;
+      }
+
+#ifdef DEBUG
+      MOZ_ASSERT(next->mType == AudioTimelineEvent::SetValueAtTime ||
+                 next->mType == AudioTimelineEvent::SetTarget ||
+                 next->mType == AudioTimelineEvent::LinearRamp ||
+                 next->mType == AudioTimelineEvent::ExponentialRamp ||
+                 next->mType == AudioTimelineEvent::SetValueCurve);
+#endif
+
+      if (TimesEqual(aTime, TimeOf(*next))) {
+        timeMatchesEventIndex = true;
+        aBuffer[bufferIndex] = GetValueAtTimeOfEvent<TimeType>(next, previous);
+        // Advance to next event, which may or may not have the same time.
+      }
+      previous = next;
+    }
+
+    if (timeMatchesEventIndex) {
+      // The time matches one of the events exactly.
+      MOZ_ASSERT(TimesEqual(aTime, TimeOf(mEvents[eventIndex - 1])));
+      ++bufferIndex;
+      ++aTime;
+    } else {
+      size_t count = aSize - bufferIndex;
+      if (next) {
+        count = LimitedCountForDuration<TimeType>(count, TimeOf(*next) - aTime);
+      }
+      GetValuesAtTimeHelperInternal(aTime, Span(aBuffer + bufferIndex, count),
+                                    previous, next);
+      bufferIndex += count;
+      aTime += static_cast<TimeType>(count);
+    }
+  }
+}
+template void AudioEventTimeline::GetValuesAtTimeHelper(double aTime,
+                                                        float* aBuffer,
+                                                        const size_t aSize);
+template void AudioEventTimeline::GetValuesAtTimeHelper(int64_t aTime,
+                                                        float* aBuffer,
+                                                        const size_t aSize);
+
+template <class TimeType>
+float AudioEventTimeline::GetValueAtTimeOfEvent(
+    const AudioTimelineEvent* aEvent, const AudioTimelineEvent* aPrevious) {
+  TimeType time = aEvent->Time<TimeType>();
+  switch (aEvent->mType) {
+    case AudioTimelineEvent::SetTarget:
+      // Start the curve, from the last value of the previous event.
+      return ComputeSetTargetStartValue(aPrevious, time);
+    case AudioTimelineEvent::SetValueCurve:
+      return aEvent->StartValue();
+    default:
+      // For other event types
+      return aEvent->NominalValue();
+  }
+}
+
+template <class TimeType>
+void AudioEventTimeline::GetValuesAtTimeHelperInternal(
+    TimeType aStartTime, Span<float> aBuffer,
+    const AudioTimelineEvent* aPrevious, const AudioTimelineEvent* aNext) {
+  MOZ_ASSERT(aBuffer.Length() >= 1);
+  MOZ_ASSERT((std::is_same<TimeType, int64_t>::value) || aBuffer.Length() == 1);
+
+  // If the requested time is before all of the existing events
+  if (!aPrevious) {
+    std::fill_n(aBuffer.Elements(), aBuffer.Length(), mDefaultValue);
+    return;
+  }
+
+  auto TimeOf = [](const AudioTimelineEvent* aEvent) -> TimeType {
+    return aEvent->Time<TimeType>();
+  };
+  auto EndTimeOf = [](const AudioTimelineEvent* aEvent) -> double {
+    return aEvent->EndTime<TimeType>();
+  };
+
+  // SetTarget nodes can be handled no matter what their next node is (if
+  // they have one)
+  if (aPrevious->mType == AudioTimelineEvent::SetTarget) {
+    aPrevious->FillTargetApproach(aStartTime, aBuffer, mSetTargetStartValue);
+    return;
+  }
+
+  // SetValueCurve events can be handled no matter what their next node is
+  // (if they have one), when aStartTime is in the curve region.
+  if (aPrevious->mType == AudioTimelineEvent::SetValueCurve) {
+    double remainingDuration =
+        TimeOf(aPrevious) - aStartTime + aPrevious->Duration();
+    if (remainingDuration >= 0.0) {
+      // aBuffer.Length() is 1 if remainingDuration is not in ticks.
+      size_t count = LimitedCountForDuration<TimeType>(
+          aBuffer.Length(),
+          // This parameter is used only when time is in ticks:
+          // Fill the last tick in the curve before possible ramps below.
+          floor(remainingDuration) + 1.0);
+      // GetValueAtTimeOfEvent() will set the value at the end of the curve if
+      // another event immediately follows.
+      MOZ_ASSERT(!aNext ||
+                 aStartTime + static_cast<TimeType>(count - 1) < TimeOf(aNext));
+      aPrevious->FillFromValueCurve(aStartTime,
+                                    Span(aBuffer.Elements(), count));
+      aBuffer = aBuffer.From(count);
+      if (aBuffer.Length() == 0) {
+        return;
+      }
+      aStartTime += static_cast<TimeType>(count);
+    }
+  }
+
+  // Handle the cases where our range ends up in a ramp event
+  if (aNext) {
+    switch (aNext->mType) {
+      case AudioTimelineEvent::LinearRamp:
+        FillLinearRamp(aStartTime, aBuffer, EndTimeOf(aPrevious),
+                       aPrevious->EndValue(), TimeOf(aNext),
+                       aNext->NominalValue());
+        return;
+      case AudioTimelineEvent::ExponentialRamp:
+        FillExponentialRamp(aStartTime, aBuffer, EndTimeOf(aPrevious),
+                            aPrevious->EndValue(), TimeOf(aNext),
+                            aNext->NominalValue());
+        return;
+      case AudioTimelineEvent::SetValueAtTime:
+      case AudioTimelineEvent::SetTarget:
+      case AudioTimelineEvent::SetValueCurve:
+        break;
+      case AudioTimelineEvent::SetValue:
+      case AudioTimelineEvent::Cancel:
+      case AudioTimelineEvent::Track:
+        MOZ_ASSERT(false, "Should have been handled earlier.");
+    }
+  }
+
+  // Now handle all other cases
+  switch (aPrevious->mType) {
+    case AudioTimelineEvent::SetValueAtTime:
+    case AudioTimelineEvent::LinearRamp:
+    case AudioTimelineEvent::ExponentialRamp:
+      break;
+    case AudioTimelineEvent::SetValueCurve:
+      MOZ_ASSERT(aStartTime - TimeOf(aPrevious) >= aPrevious->Duration());
+      break;
+    case AudioTimelineEvent::SetTarget:
+      MOZ_FALLTHROUGH_ASSERT("AudioTimelineEvent::SetTarget");
+    case AudioTimelineEvent::SetValue:
+    case AudioTimelineEvent::Cancel:
+    case AudioTimelineEvent::Track:
+      MOZ_ASSERT(false, "Should have been handled earlier.");
+  }
+  // If the next event type is neither linear or exponential ramp, the
+  // value is constant.
+  std::fill_n(aBuffer.Elements(), aBuffer.Length(), aPrevious->EndValue());
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioEventTimeline.h b/dom/media/webaudio/AudioEventTimeline.h
new file mode 100644
index 0000000000..9284fc5b6d
--- /dev/null
+++ b/dom/media/webaudio/AudioEventTimeline.h
@@ -0,0 +1,418 @@
+/* -*- 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/. */
+
+#ifndef AudioEventTimeline_h_
+#define AudioEventTimeline_h_
+
+#include <algorithm>
+#include "mozilla/Assertions.h"
+#include "mozilla/FloatingPoint.h"
+#include "mozilla/PodOperations.h"
+#include "mozilla/ErrorResult.h"
+
+#include "MainThreadUtils.h"
+#include "nsTArray.h"
+#include "math.h"
+#include "WebAudioUtils.h"
+
+// XXX Avoid including this here by moving function bodies to the cpp file
+#include "js/GCAPI.h"
+
+namespace mozilla {
+
+class AudioNodeTrack;
+
+namespace dom {
+
+struct AudioTimelineEvent {
+  enum Type : uint32_t {
+    SetValue,
+    SetValueAtTime,
+    LinearRamp,
+    ExponentialRamp,
+    SetTarget,
+    SetValueCurve,
+    Track,
+    Cancel
+  };
+
+  class TimeUnion {
+   public:
+    // double 0.0 is bit-identical to int64_t 0.
+    TimeUnion()
+        : mSeconds()
+#if DEBUG
+          ,
+          mIsInSeconds(true),
+          mIsInTicks(true)
+#endif
+    {
+    }
+    explicit TimeUnion(double aTime)
+        : mSeconds(aTime)
+#if DEBUG
+          ,
+          mIsInSeconds(true),
+          mIsInTicks(false)
+#endif
+    {
+    }
+    explicit TimeUnion(int64_t aTime)
+        : mTicks(aTime)
+#if DEBUG
+          ,
+          mIsInSeconds(false),
+          mIsInTicks(true)
+#endif
+    {
+    }
+
+    double operator=(double aTime) {
+#if DEBUG
+      mIsInSeconds = true;
+      mIsInTicks = true;
+#endif
+      return mSeconds = aTime;
+    }
+    int64_t operator=(int64_t aTime) {
+#if DEBUG
+      mIsInSeconds = true;
+      mIsInTicks = true;
+#endif
+      return mTicks = aTime;
+    }
+
+    template <class TimeType>
+    TimeType Get() const;
+
+   private:
+    union {
+      double mSeconds;
+      int64_t mTicks;
+    };
+#ifdef DEBUG
+    bool mIsInSeconds;
+    bool mIsInTicks;
+
+   public:
+    bool IsInTicks() const { return mIsInTicks; };
+#endif
+  };
+
+  AudioTimelineEvent(Type aType, double aTime, float aValue,
+                     double aTimeConstant = 0.0);
+  // For SetValueCurve
+  AudioTimelineEvent(Type aType, const nsTArray<float>& aValues,
+                     double aStartTime, double aDuration);
+  AudioTimelineEvent(const AudioTimelineEvent& rhs);
+  ~AudioTimelineEvent();
+
+  template <class TimeType>
+  TimeType Time() const {
+    return mTime.Get<TimeType>();
+  }
+  // If this event is a curve event, this returns the end time of the curve.
+  // Otherwise, this returns the time of the event.
+  template <class TimeType>
+  double EndTime() const;
+
+  float NominalValue() const {
+    MOZ_ASSERT(mType != SetValueCurve);
+    return mValue;
+  }
+  float StartValue() const {
+    MOZ_ASSERT(mType == SetValueCurve);
+    return mCurve[0];
+  }
+  // Value for an event, or for a ValueCurve event, this is the value of the
+  // last element of the curve.
+  float EndValue() const;
+
+  double TimeConstant() const {
+    MOZ_ASSERT(mType == SetTarget);
+    return mTimeConstant;
+  }
+  uint32_t CurveLength() const {
+    MOZ_ASSERT(mType == SetValueCurve);
+    return mCurveLength;
+  }
+  double Duration() const {
+    MOZ_ASSERT(mType == SetValueCurve);
+    return mDuration;
+  }
+  /**
+   * Converts an AudioTimelineEvent's floating point time members to tick
+   * values with respect to a destination AudioNodeTrack.
+   *
+   * This needs to be called for each AudioTimelineEvent that gets sent to an
+   * AudioNodeEngine, on the engine side where the AudioTimlineEvent is
+   * received.  This means that such engines need to be aware of their
+   * destination tracks as well.
+   */
+  void ConvertToTicks(AudioNodeTrack* aDestination);
+
+  template <class TimeType>
+  void FillTargetApproach(TimeType aBufferStartTime, Span<float> aBuffer,
+                          double v0) const;
+  template <class TimeType>
+  void FillFromValueCurve(TimeType aBufferStartTime, Span<float> aBuffer) const;
+
+  const Type mType;
+
+ private:
+  union {
+    float mValue;
+    uint32_t mCurveLength;  // for SetValueCurve
+  };
+  union {
+    double mTimeConstant;
+    // mCurve contains a buffer of SetValueCurve samples.  We sample the
+    // values in the buffer depending on how far along we are in time.
+    // If we're at time T and the event has started as time T0 and has a
+    // duration of D, we sample the buffer at floor(mCurveLength*(T-T0)/D)
+    // if T<T0+D, and just take the last sample in the buffer otherwise.
+    float* mCurve;
+  };
+  union {
+    // mPerTickRatio is used only with SetTarget and int64_t TimeType.
+    double mPerTickRatio;
+    double mDuration;  // for SetValueCurve
+  };
+
+  // This member is accessed using the `Time` method.
+  //
+  // The time for an event can either be in seconds or in ticks.
+  // Initially the time of the event is always in seconds.
+  // In order to convert it to ticks, call SetTimeInTicks.  Once this
+  // method has been called for an event, the time cannot be converted
+  // back to seconds.
+  TimeUnion mTime;
+};
+
+template <>
+inline double AudioTimelineEvent::TimeUnion::Get<double>() const {
+  MOZ_ASSERT(mIsInSeconds);
+  return mSeconds;
+}
+template <>
+inline int64_t AudioTimelineEvent::TimeUnion::Get<int64_t>() const {
+  MOZ_ASSERT(mIsInTicks);
+  return mTicks;
+}
+
+class AudioEventTimeline {
+ public:
+  explicit AudioEventTimeline(float aDefaultValue)
+      : mDefaultValue(aDefaultValue),
+        mSetTargetStartValue(aDefaultValue),
+        mSimpleValue(Some(aDefaultValue)) {}
+
+  bool ValidateEvent(const AudioTimelineEvent& aEvent, ErrorResult& aRv) const {
+    MOZ_ASSERT(NS_IsMainThread());
+
+    auto TimeOf = [](const AudioTimelineEvent& aEvent) -> double {
+      return aEvent.Time<double>();
+    };
+
+    // Validate the event itself
+    if (!WebAudioUtils::IsTimeValid(TimeOf(aEvent))) {
+      aRv.ThrowRangeError<MSG_INVALID_AUDIOPARAM_METHOD_START_TIME_ERROR>();
+      return false;
+    }
+
+    switch (aEvent.mType) {
+      case AudioTimelineEvent::SetValueCurve:
+        if (aEvent.CurveLength() < 2) {
+          aRv.ThrowInvalidStateError("Curve length must be at least 2");
+          return false;
+        }
+        if (aEvent.Duration() <= 0) {
+          aRv.ThrowRangeError(
+              "The curve duration for setValueCurveAtTime must be strictly "
+              "positive.");
+          return false;
+        }
+        MOZ_ASSERT(IsValid(aEvent.Duration()));
+        break;
+      case AudioTimelineEvent::SetTarget:
+        if (!WebAudioUtils::IsTimeValid(aEvent.TimeConstant())) {
+          aRv.ThrowRangeError(
+              "The exponential constant passed to setTargetAtTime must be "
+              "non-negative.");
+          return false;
+        }
+        [[fallthrough]];
+      default:
+        MOZ_ASSERT(IsValid(aEvent.NominalValue()));
+    }
+
+    // Make sure that new events don't fall within the duration of a
+    // curve event.
+    for (unsigned i = 0; i < mEvents.Length(); ++i) {
+      if (mEvents[i].mType == AudioTimelineEvent::SetValueCurve &&
+          TimeOf(mEvents[i]) <= TimeOf(aEvent) &&
+          TimeOf(mEvents[i]) + mEvents[i].Duration() > TimeOf(aEvent)) {
+        aRv.ThrowNotSupportedError("Can't add events during a curve event");
+        return false;
+      }
+    }
+
+    // Make sure that new curve events don't fall in a range which includes
+    // other events.
+    if (aEvent.mType == AudioTimelineEvent::SetValueCurve) {
+      for (unsigned i = 0; i < mEvents.Length(); ++i) {
+        if (TimeOf(aEvent) < TimeOf(mEvents[i]) &&
+            TimeOf(aEvent) + aEvent.Duration() > TimeOf(mEvents[i])) {
+          aRv.ThrowNotSupportedError(
+              "Can't add curve events that overlap other events");
+          return false;
+        }
+      }
+    }
+
+    // Make sure that invalid values are not used for exponential curves
+    if (aEvent.mType == AudioTimelineEvent::ExponentialRamp) {
+      if (aEvent.NominalValue() == 0.f) {
+        aRv.ThrowRangeError(
+            "The value passed to exponentialRampToValueAtTime must be "
+            "non-zero.");
+        return false;
+      }
+    }
+    return true;
+  }
+
+  template <typename TimeType>
+  void InsertEvent(const AudioTimelineEvent& aEvent) {
+    mSimpleValue.reset();
+    for (unsigned i = 0; i < mEvents.Length(); ++i) {
+      if (aEvent.Time<TimeType>() == mEvents[i].Time<TimeType>()) {
+        // If two events happen at the same time, have them in chronological
+        // order of insertion.
+        do {
+          ++i;
+        } while (i < mEvents.Length() &&
+                 aEvent.Time<TimeType>() == mEvents[i].Time<TimeType>());
+        mEvents.InsertElementAt(i, aEvent);
+        return;
+      }
+      // Otherwise, place the event right after the latest existing event
+      if (aEvent.Time<TimeType>() < mEvents[i].Time<TimeType>()) {
+        mEvents.InsertElementAt(i, aEvent);
+        return;
+      }
+    }
+
+    // If we couldn't find a place for the event, just append it to the list
+    mEvents.AppendElement(aEvent);
+  }
+
+  bool HasSimpleValue() const { return mSimpleValue.isSome(); }
+
+  float GetValue() const {
+    // This method should only be called if HasSimpleValue() returns true
+    MOZ_ASSERT(HasSimpleValue());
+    return mSimpleValue.value();
+  }
+
+  void SetValue(float aValue) {
+    // FIXME: bug 1308435
+    // A spec change means this should instead behave like setValueAtTime().
+
+    // Silently don't change anything if there are any events
+    if (mEvents.IsEmpty()) {
+      mSetTargetStartValue = mDefaultValue = aValue;
+      mSimpleValue = Some(aValue);
+    }
+  }
+
+  template <typename TimeType>
+  void CancelScheduledValues(TimeType aStartTime) {
+    for (unsigned i = 0; i < mEvents.Length(); ++i) {
+      if (mEvents[i].Time<TimeType>() >= aStartTime) {
+#ifdef DEBUG
+        // Sanity check: the array should be sorted, so all of the following
+        // events should have a time greater than aStartTime too.
+        for (unsigned j = i + 1; j < mEvents.Length(); ++j) {
+          MOZ_ASSERT(mEvents[j].Time<TimeType>() >= aStartTime);
+        }
+#endif
+        mEvents.TruncateLength(i);
+        break;
+      }
+    }
+    if (mEvents.IsEmpty()) {
+      mSimpleValue = Some(mDefaultValue);
+    }
+  }
+
+  static bool TimesEqual(int64_t aLhs, int64_t aRhs) { return aLhs == aRhs; }
+
+  // Since we are going to accumulate error by adding 0.01 multiple time in a
+  // loop, we want to fuzz the equality check in GetValueAtTime.
+  static bool TimesEqual(double aLhs, double aRhs) {
+    const float kEpsilon = 0.0000000001f;
+    return fabs(aLhs - aRhs) < kEpsilon;
+  }
+
+  template <class TimeType>
+  float GetValueAtTime(TimeType aTime) {
+    float result;
+    GetValuesAtTimeHelper(aTime, &result, 1);
+    return result;
+  }
+
+  void GetValuesAtTime(int64_t aTime, float* aBuffer, const size_t aSize) {
+    MOZ_ASSERT(aBuffer);
+    GetValuesAtTimeHelper(aTime, aBuffer, aSize);
+  }
+  void GetValuesAtTime(double aTime, float* aBuffer,
+                       const size_t aSize) = delete;
+
+  // Return the number of events scheduled
+  uint32_t GetEventCount() const { return mEvents.Length(); }
+
+  template <class TimeType>
+  void CleanupEventsOlderThan(TimeType aTime);
+
+ private:
+  template <class TimeType>
+  void GetValuesAtTimeHelper(TimeType aTime, float* aBuffer,
+                             const size_t aSize);
+
+  template <class TimeType>
+  float GetValueAtTimeOfEvent(const AudioTimelineEvent* aEvent,
+                              const AudioTimelineEvent* aPrevious);
+
+  template <class TimeType>
+  void GetValuesAtTimeHelperInternal(TimeType aStartTime, Span<float> aBuffer,
+                                     const AudioTimelineEvent* aPrevious,
+                                     const AudioTimelineEvent* aNext);
+
+  static bool IsValid(double value) { return std::isfinite(value); }
+
+  template <class TimeType>
+  float ComputeSetTargetStartValue(const AudioTimelineEvent* aPreviousEvent,
+                                   TimeType aTime);
+
+  // This is a sorted array of the events in the timeline.  Queries of this
+  // data structure should probably be more frequent than modifications to it,
+  // and that is the reason why we're using a simple array as the data
+  // structure. We can optimize this in the future if the performance of the
+  // array ends up being a bottleneck.
+  nsTArray<AudioTimelineEvent> mEvents;
+  float mDefaultValue;
+  // This is the value of this AudioParam at the end of the previous
+  // event for SetTarget curves.
+  float mSetTargetStartValue;
+  AudioTimelineEvent::TimeUnion mSetTargetStartTime;
+  Maybe<float> mSimpleValue;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/AudioListener.cpp b/dom/media/webaudio/AudioListener.cpp
new file mode 100644
index 0000000000..b8c4e9bc65
--- /dev/null
+++ b/dom/media/webaudio/AudioListener.cpp
@@ -0,0 +1,121 @@
+/* -*- 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 "AudioContext.h"
+#include "AudioListener.h"
+#include "MediaTrackGraph.h"
+#include "Tracing.h"
+#include "mozilla/dom/AudioListenerBinding.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE(AudioListener, mContext)
+
+AudioListenerEngine::AudioListenerEngine()
+    : mFrontVector(0., 0., -1.), mRightVector(1., 0., 0.) {}
+
+void AudioListenerEngine::RecvListenerEngineEvent(
+    AudioListenerEngine::AudioListenerParameter aParameter,
+    const ThreeDPoint& aValue) {
+  switch (aParameter) {
+    case AudioListenerParameter::POSITION:
+      mPosition = aValue;
+      break;
+    case AudioListenerParameter::FRONT:
+      mFrontVector = aValue;
+      break;
+    case AudioListenerParameter::RIGHT:
+      mRightVector = aValue;
+      break;
+    default:
+      MOZ_CRASH("Not handled");
+  }
+}
+
+const ThreeDPoint& AudioListenerEngine::Position() const { return mPosition; }
+const ThreeDPoint& AudioListenerEngine::FrontVector() const {
+  return mFrontVector;
+}
+const ThreeDPoint& AudioListenerEngine::RightVector() const {
+  return mRightVector;
+}
+
+AudioListener::AudioListener(AudioContext* aContext)
+    : mContext(aContext),
+      mEngine(new AudioListenerEngine()),
+
+      mFrontVector(0., 0., -1.),
+      mRightVector(1., 0., 0.) {
+  MOZ_ASSERT(aContext);
+}
+
+JSObject* AudioListener::WrapObject(JSContext* aCx,
+                                    JS::Handle<JSObject*> aGivenProto) {
+  return AudioListener_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void AudioListener::SetOrientation(double aX, double aY, double aZ, double aXUp,
+                                   double aYUp, double aZUp) {
+  ThreeDPoint front(aX, aY, aZ);
+  // The panning effect and the azimuth and elevation calculation in the Web
+  // Audio spec becomes undefined with linearly dependent vectors, so keep
+  // existing state in these situations.
+  if (front.IsZero()) {
+    return;
+  }
+  // Normalize before using CrossProduct() to avoid overflow.
+  front.Normalize();
+  ThreeDPoint up(aXUp, aYUp, aZUp);
+  if (up.IsZero()) {
+    return;
+  }
+  up.Normalize();
+  ThreeDPoint right = front.CrossProduct(up);
+  if (right.IsZero()) {
+    return;
+  }
+  right.Normalize();
+
+  if (!mFrontVector.FuzzyEqual(front)) {
+    mFrontVector = front;
+    SendListenerEngineEvent(AudioListenerEngine::AudioListenerParameter::FRONT,
+                            mFrontVector);
+  }
+  if (!mRightVector.FuzzyEqual(right)) {
+    mRightVector = right;
+    SendListenerEngineEvent(AudioListenerEngine::AudioListenerParameter::RIGHT,
+                            mRightVector);
+  }
+}
+
+void AudioListener::SetPosition(double aX, double aY, double aZ) {
+  if (WebAudioUtils::FuzzyEqual(mPosition.x, aX) &&
+      WebAudioUtils::FuzzyEqual(mPosition.y, aY) &&
+      WebAudioUtils::FuzzyEqual(mPosition.z, aZ)) {
+    return;
+  }
+  mPosition.x = aX;
+  mPosition.y = aY;
+  mPosition.z = aZ;
+  SendListenerEngineEvent(AudioListenerEngine::AudioListenerParameter::POSITION,
+                          mPosition);
+}
+
+void AudioListener::SendListenerEngineEvent(
+    AudioListenerEngine::AudioListenerParameter aParameter,
+    const ThreeDPoint& aValue) {
+  mContext->DestinationTrack()->QueueControlMessageWithNoShutdown(
+      [engine = RefPtr(Engine()), aParameter, aValue] {
+        TRACE("AudioListener::RecvListenerEngineEvent");
+        engine->RecvListenerEngineEvent(aParameter, aValue);
+      });
+}
+
+size_t AudioListener::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioListener.h b/dom/media/webaudio/AudioListener.h
new file mode 100644
index 0000000000..7d7fad3bfb
--- /dev/null
+++ b/dom/media/webaudio/AudioListener.h
@@ -0,0 +1,85 @@
+/* -*- 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/. */
+
+#ifndef AudioListener_h_
+#define AudioListener_h_
+
+#include "nsWrapperCache.h"
+#include "nsCycleCollectionParticipant.h"
+#include "mozilla/Attributes.h"
+#include "ThreeDPoint.h"
+#include "AudioContext.h"
+#include "PannerNode.h"
+#include "WebAudioUtils.h"
+#include "js/TypeDecls.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace mozilla::dom {
+
+class AudioListenerEngine final {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(AudioListenerEngine)
+
+  AudioListenerEngine();
+
+  enum class AudioListenerParameter {
+    POSITION,
+    FRONT,  // unit length
+    RIGHT   // unit length, orthogonal to FRONT
+  };
+  void RecvListenerEngineEvent(
+      AudioListenerEngine::AudioListenerParameter aParameter,
+      const ThreeDPoint& aValue);
+  const ThreeDPoint& Position() const;
+  const ThreeDPoint& FrontVector() const;
+  const ThreeDPoint& RightVector() const;
+
+ private:
+  ~AudioListenerEngine() = default;
+
+  ThreeDPoint mPosition;
+  ThreeDPoint mFrontVector;
+  ThreeDPoint mRightVector;
+};
+
+class AudioListener final : public nsWrapperCache {
+ public:
+  explicit AudioListener(AudioContext* aContext);
+
+  NS_INLINE_DECL_CYCLE_COLLECTING_NATIVE_REFCOUNTING(AudioListener)
+  NS_DECL_CYCLE_COLLECTION_NATIVE_WRAPPERCACHE_CLASS(AudioListener)
+
+  size_t SizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+  AudioContext* GetParentObject() const { return mContext; }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  void SetPosition(double aX, double aY, double aZ);
+  void SetOrientation(double aX, double aY, double aZ, double aXUp, double aYUp,
+                      double aZUp);
+
+  AudioListenerEngine* Engine() { return mEngine.get(); }
+
+ private:
+  void SendListenerEngineEvent(
+      AudioListenerEngine::AudioListenerParameter aParameter,
+      const ThreeDPoint& aValue);
+
+  ~AudioListener() = default;
+
+ private:
+  RefPtr<AudioContext> mContext;
+  RefPtr<AudioListenerEngine> mEngine;
+  ThreeDPoint mPosition;
+  ThreeDPoint mFrontVector;
+  ThreeDPoint mRightVector;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/AudioNode.cpp b/dom/media/webaudio/AudioNode.cpp
new file mode 100644
index 0000000000..f5c56d0da6
--- /dev/null
+++ b/dom/media/webaudio/AudioNode.cpp
@@ -0,0 +1,608 @@
+/* -*- 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 "AudioNode.h"
+#include "mozilla/ErrorResult.h"
+#include "AudioNodeTrack.h"
+#include "AudioNodeEngine.h"
+#include "mozilla/dom/AudioParam.h"
+#include "mozilla/Services.h"
+#include "nsIObserverService.h"
+
+namespace mozilla::dom {
+
+static const uint32_t INVALID_PORT = 0xffffffff;
+static uint32_t gId = 0;
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(AudioNode)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(AudioNode, DOMEventTargetHelper)
+  tmp->DisconnectFromGraph();
+  if (tmp->mContext) {
+    tmp->mContext->UnregisterNode(tmp);
+  }
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mContext)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mParams)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mOutputNodes)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mOutputParams)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_WEAK_REFERENCE
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(AudioNode,
+                                                  DOMEventTargetHelper)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mContext)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mParams)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mOutputNodes)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mOutputParams)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_ADDREF_INHERITED(AudioNode, DOMEventTargetHelper)
+NS_IMPL_RELEASE_INHERITED(AudioNode, DOMEventTargetHelper)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(AudioNode)
+  NS_INTERFACE_MAP_ENTRY(nsISupportsWeakReference)
+NS_INTERFACE_MAP_END_INHERITING(DOMEventTargetHelper)
+
+AudioNode::AudioNode(AudioContext* aContext, uint32_t aChannelCount,
+                     ChannelCountMode aChannelCountMode,
+                     ChannelInterpretation aChannelInterpretation)
+    : DOMEventTargetHelper(aContext->GetParentObject()),
+      mContext(aContext),
+      mChannelCount(aChannelCount),
+      mChannelCountMode(aChannelCountMode),
+      mChannelInterpretation(aChannelInterpretation),
+      mId(gId++),
+      mPassThrough(false) {
+  MOZ_ASSERT(aContext);
+  aContext->RegisterNode(this);
+}
+
+AudioNode::~AudioNode() {
+  MOZ_ASSERT(mInputNodes.IsEmpty());
+  MOZ_ASSERT(mOutputNodes.IsEmpty());
+  MOZ_ASSERT(mOutputParams.IsEmpty());
+  MOZ_ASSERT(!mTrack,
+             "The webaudio-node-demise notification must have been sent");
+  if (mContext) {
+    mContext->UnregisterNode(this);
+  }
+}
+
+void AudioNode::Initialize(const AudioNodeOptions& aOptions, ErrorResult& aRv) {
+  if (aOptions.mChannelCount.WasPassed()) {
+    SetChannelCount(aOptions.mChannelCount.Value(), aRv);
+    if (NS_WARN_IF(aRv.Failed())) {
+      return;
+    }
+  }
+
+  if (aOptions.mChannelCountMode.WasPassed()) {
+    SetChannelCountModeValue(aOptions.mChannelCountMode.Value(), aRv);
+    if (NS_WARN_IF(aRv.Failed())) {
+      return;
+    }
+  }
+
+  if (aOptions.mChannelInterpretation.WasPassed()) {
+    SetChannelInterpretationValue(aOptions.mChannelInterpretation.Value(), aRv);
+    if (NS_WARN_IF(aRv.Failed())) {
+      return;
+    }
+  }
+}
+
+size_t AudioNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  // Not owned:
+  // - mContext
+  // - mTrack
+  size_t amount = 0;
+
+  amount += mInputNodes.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < mInputNodes.Length(); i++) {
+    amount += mInputNodes[i].SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  // Just measure the array. The entire audio node graph is measured via the
+  // MediaTrackGraph's tracks, so we don't want to double-count the elements.
+  amount += mOutputNodes.ShallowSizeOfExcludingThis(aMallocSizeOf);
+
+  amount += mOutputParams.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < mOutputParams.Length(); i++) {
+    amount += mOutputParams[i]->SizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  return amount;
+}
+
+size_t AudioNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+template <class InputNode>
+static size_t FindIndexOfNode(const nsTArray<InputNode>& aInputNodes,
+                              const AudioNode* aNode) {
+  for (size_t i = 0; i < aInputNodes.Length(); ++i) {
+    if (aInputNodes[i].mInputNode == aNode) {
+      return i;
+    }
+  }
+  return nsTArray<InputNode>::NoIndex;
+}
+
+template <class InputNode>
+static size_t FindIndexOfNodeWithPorts(const nsTArray<InputNode>& aInputNodes,
+                                       const AudioNode* aNode,
+                                       uint32_t aInputPort,
+                                       uint32_t aOutputPort) {
+  for (size_t i = 0; i < aInputNodes.Length(); ++i) {
+    if (aInputNodes[i].mInputNode == aNode &&
+        aInputNodes[i].mInputPort == aInputPort &&
+        aInputNodes[i].mOutputPort == aOutputPort) {
+      return i;
+    }
+  }
+  return nsTArray<InputNode>::NoIndex;
+}
+
+void AudioNode::DisconnectFromGraph() {
+  MOZ_ASSERT(mRefCnt.get() > mInputNodes.Length(),
+             "Caller should be holding a reference");
+
+  // The idea here is that we remove connections one by one, and at each step
+  // the graph is in a valid state.
+
+  // Disconnect inputs. We don't need them anymore.
+  while (!mInputNodes.IsEmpty()) {
+    InputNode inputNode = mInputNodes.PopLastElement();
+    inputNode.mInputNode->mOutputNodes.RemoveElement(this);
+  }
+
+  while (!mOutputNodes.IsEmpty()) {
+    RefPtr<AudioNode> output = mOutputNodes.PopLastElement();
+    size_t inputIndex = FindIndexOfNode(output->mInputNodes, this);
+    // It doesn't matter which one we remove, since we're going to remove all
+    // entries for this node anyway.
+    output->mInputNodes.RemoveElementAt(inputIndex);
+    // This effects of this connection will remain.
+    output->NotifyHasPhantomInput();
+  }
+
+  while (!mOutputParams.IsEmpty()) {
+    RefPtr<AudioParam> output = mOutputParams.PopLastElement();
+    size_t inputIndex = FindIndexOfNode(output->InputNodes(), this);
+    // It doesn't matter which one we remove, since we're going to remove all
+    // entries for this node anyway.
+    output->RemoveInputNode(inputIndex);
+  }
+
+  DestroyMediaTrack();
+}
+
+AudioNode* AudioNode::Connect(AudioNode& aDestination, uint32_t aOutput,
+                              uint32_t aInput, ErrorResult& aRv) {
+  if (aOutput >= NumberOfOutputs()) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("Output index %u is out of bounds", aOutput));
+    return nullptr;
+  }
+
+  if (aInput >= aDestination.NumberOfInputs()) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("Input index %u is out of bounds", aInput));
+    return nullptr;
+  }
+
+  if (Context() != aDestination.Context()) {
+    aRv.ThrowInvalidAccessError(
+        "Can't connect nodes from different AudioContexts");
+    return nullptr;
+  }
+
+  if (FindIndexOfNodeWithPorts(aDestination.mInputNodes, this, aInput,
+                               aOutput) !=
+      nsTArray<AudioNode::InputNode>::NoIndex) {
+    // connection already exists.
+    return &aDestination;
+  }
+
+  WEB_AUDIO_API_LOG("%f: %s %u Connect() to %s %u", Context()->CurrentTime(),
+                    NodeType(), Id(), aDestination.NodeType(),
+                    aDestination.Id());
+
+  // The MediaTrackGraph will handle cycle detection. We don't need to do it
+  // here.
+
+  mOutputNodes.AppendElement(&aDestination);
+  InputNode* input = aDestination.mInputNodes.AppendElement();
+  input->mInputNode = this;
+  input->mInputPort = aInput;
+  input->mOutputPort = aOutput;
+  AudioNodeTrack* destinationTrack = aDestination.mTrack;
+  if (mTrack && destinationTrack) {
+    // Connect tracks in the MediaTrackGraph
+    MOZ_ASSERT(aInput <= UINT16_MAX, "Unexpected large input port number");
+    MOZ_ASSERT(aOutput <= UINT16_MAX, "Unexpected large output port number");
+    input->mTrackPort = destinationTrack->AllocateInputPort(
+        mTrack, static_cast<uint16_t>(aInput), static_cast<uint16_t>(aOutput));
+  }
+  aDestination.NotifyInputsChanged();
+
+  return &aDestination;
+}
+
+void AudioNode::Connect(AudioParam& aDestination, uint32_t aOutput,
+                        ErrorResult& aRv) {
+  if (aOutput >= NumberOfOutputs()) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("Output index %u is out of bounds", aOutput));
+    return;
+  }
+
+  if (Context() != aDestination.GetParentObject()) {
+    aRv.ThrowInvalidAccessError(
+        "Can't connect a node to an AudioParam from a different AudioContext");
+    return;
+  }
+
+  if (FindIndexOfNodeWithPorts(aDestination.InputNodes(), this, INVALID_PORT,
+                               aOutput) !=
+      nsTArray<AudioNode::InputNode>::NoIndex) {
+    // connection already exists.
+    return;
+  }
+
+  mOutputParams.AppendElement(&aDestination);
+  InputNode* input = aDestination.AppendInputNode();
+  input->mInputNode = this;
+  input->mInputPort = INVALID_PORT;
+  input->mOutputPort = aOutput;
+
+  mozilla::MediaTrack* track = aDestination.Track();
+  MOZ_ASSERT(track->AsProcessedTrack());
+  ProcessedMediaTrack* ps = static_cast<ProcessedMediaTrack*>(track);
+  if (mTrack) {
+    // Setup our track as an input to the AudioParam's track
+    MOZ_ASSERT(aOutput <= UINT16_MAX, "Unexpected large output port number");
+    input->mTrackPort =
+        ps->AllocateInputPort(mTrack, 0, static_cast<uint16_t>(aOutput));
+  }
+}
+
+void AudioNode::SendDoubleParameterToTrack(uint32_t aIndex, double aValue) {
+  MOZ_ASSERT(mTrack, "How come we don't have a track here?");
+  mTrack->SetDoubleParameter(aIndex, aValue);
+}
+
+void AudioNode::SendInt32ParameterToTrack(uint32_t aIndex, int32_t aValue) {
+  MOZ_ASSERT(mTrack, "How come we don't have a track here?");
+  mTrack->SetInt32Parameter(aIndex, aValue);
+}
+
+void AudioNode::SendChannelMixingParametersToTrack() {
+  if (mTrack) {
+    mTrack->SetChannelMixingParameters(mChannelCount, mChannelCountMode,
+                                       mChannelInterpretation);
+  }
+}
+
+template <>
+bool AudioNode::DisconnectFromOutputIfConnected<AudioNode>(
+    uint32_t aOutputNodeIndex, uint32_t aInputIndex) {
+  WEB_AUDIO_API_LOG("%f: %s %u Disconnect()", Context()->CurrentTime(),
+                    NodeType(), Id());
+
+  AudioNode* destination = mOutputNodes[aOutputNodeIndex];
+
+  MOZ_ASSERT(aOutputNodeIndex < mOutputNodes.Length());
+  MOZ_ASSERT(aInputIndex < destination->InputNodes().Length());
+
+  // An upstream node may be starting to play on the graph thread, and the
+  // engine for a downstream node may be sending a PlayingRefChangeHandler
+  // ADDREF message to this (main) thread.  Wait for a round trip before
+  // releasing nodes, to give engines receiving sound now time to keep their
+  // nodes alive.
+  class RunnableRelease final : public Runnable {
+   public:
+    explicit RunnableRelease(already_AddRefed<AudioNode> aNode)
+        : mozilla::Runnable("RunnableRelease"), mNode(aNode) {}
+
+    NS_IMETHOD Run() override {
+      mNode = nullptr;
+      return NS_OK;
+    }
+
+   private:
+    RefPtr<AudioNode> mNode;
+  };
+
+  InputNode& input = destination->mInputNodes[aInputIndex];
+  if (input.mInputNode != this) {
+    return false;
+  }
+
+  // Remove one instance of 'dest' from mOutputNodes. There could be
+  // others, and it's not correct to remove them all since some of them
+  // could be for different output ports.
+  RefPtr<AudioNode> output = std::move(mOutputNodes[aOutputNodeIndex]);
+  mOutputNodes.RemoveElementAt(aOutputNodeIndex);
+  // Destroying the InputNode here sends a message to the graph thread
+  // to disconnect the tracks, which should be sent before the
+  // RunAfterPendingUpdates() call below.
+  destination->mInputNodes.RemoveElementAt(aInputIndex);
+  output->NotifyInputsChanged();
+  if (mTrack) {
+    nsCOMPtr<nsIRunnable> runnable = new RunnableRelease(output.forget());
+    mTrack->RunAfterPendingUpdates(runnable.forget());
+  }
+  return true;
+}
+
+template <>
+bool AudioNode::DisconnectFromOutputIfConnected<AudioParam>(
+    uint32_t aOutputParamIndex, uint32_t aInputIndex) {
+  MOZ_ASSERT(aOutputParamIndex < mOutputParams.Length());
+
+  AudioParam* destination = mOutputParams[aOutputParamIndex];
+
+  MOZ_ASSERT(aInputIndex < destination->InputNodes().Length());
+
+  const InputNode& input = destination->InputNodes()[aInputIndex];
+  if (input.mInputNode != this) {
+    return false;
+  }
+  destination->RemoveInputNode(aInputIndex);
+  // Remove one instance of 'dest' from mOutputParams. There could be
+  // others, and it's not correct to remove them all since some of them
+  // could be for different output ports.
+  mOutputParams.RemoveElementAt(aOutputParamIndex);
+  return true;
+}
+
+template <>
+const nsTArray<AudioNode::InputNode>&
+AudioNode::InputsForDestination<AudioNode>(uint32_t aOutputNodeIndex) const {
+  return mOutputNodes[aOutputNodeIndex]->InputNodes();
+}
+
+template <>
+const nsTArray<AudioNode::InputNode>&
+AudioNode::InputsForDestination<AudioParam>(uint32_t aOutputNodeIndex) const {
+  return mOutputParams[aOutputNodeIndex]->InputNodes();
+}
+
+template <typename DestinationType, typename Predicate>
+bool AudioNode::DisconnectMatchingDestinationInputs(uint32_t aDestinationIndex,
+                                                    Predicate aPredicate) {
+  bool wasConnected = false;
+  uint32_t inputCount =
+      InputsForDestination<DestinationType>(aDestinationIndex).Length();
+
+  for (int32_t inputIndex = inputCount - 1; inputIndex >= 0; --inputIndex) {
+    const InputNode& input =
+        InputsForDestination<DestinationType>(aDestinationIndex)[inputIndex];
+    if (aPredicate(input)) {
+      if (DisconnectFromOutputIfConnected<DestinationType>(aDestinationIndex,
+                                                           inputIndex)) {
+        wasConnected = true;
+        break;
+      }
+    }
+  }
+  return wasConnected;
+}
+
+void AudioNode::Disconnect(ErrorResult& aRv) {
+  for (int32_t outputIndex = mOutputNodes.Length() - 1; outputIndex >= 0;
+       --outputIndex) {
+    DisconnectMatchingDestinationInputs<AudioNode>(
+        outputIndex, [](const InputNode&) { return true; });
+  }
+
+  for (int32_t outputIndex = mOutputParams.Length() - 1; outputIndex >= 0;
+       --outputIndex) {
+    DisconnectMatchingDestinationInputs<AudioParam>(
+        outputIndex, [](const InputNode&) { return true; });
+  }
+}
+
+void AudioNode::Disconnect(uint32_t aOutput, ErrorResult& aRv) {
+  if (aOutput >= NumberOfOutputs()) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("Output index %u is out of bounds", aOutput));
+    return;
+  }
+
+  for (int32_t outputIndex = mOutputNodes.Length() - 1; outputIndex >= 0;
+       --outputIndex) {
+    DisconnectMatchingDestinationInputs<AudioNode>(
+        outputIndex, [aOutput](const InputNode& aInputNode) {
+          return aInputNode.mOutputPort == aOutput;
+        });
+  }
+
+  for (int32_t outputIndex = mOutputParams.Length() - 1; outputIndex >= 0;
+       --outputIndex) {
+    DisconnectMatchingDestinationInputs<AudioParam>(
+        outputIndex, [aOutput](const InputNode& aInputNode) {
+          return aInputNode.mOutputPort == aOutput;
+        });
+  }
+}
+
+void AudioNode::Disconnect(AudioNode& aDestination, ErrorResult& aRv) {
+  bool wasConnected = false;
+
+  for (int32_t outputIndex = mOutputNodes.Length() - 1; outputIndex >= 0;
+       --outputIndex) {
+    if (mOutputNodes[outputIndex] != &aDestination) {
+      continue;
+    }
+    wasConnected |= DisconnectMatchingDestinationInputs<AudioNode>(
+        outputIndex, [](const InputNode&) { return true; });
+  }
+
+  if (!wasConnected) {
+    aRv.ThrowInvalidAccessError(
+        "Trying to disconnect from a node we're not connected to");
+    return;
+  }
+}
+
+void AudioNode::Disconnect(AudioNode& aDestination, uint32_t aOutput,
+                           ErrorResult& aRv) {
+  if (aOutput >= NumberOfOutputs()) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("Output index %u is out of bounds", aOutput));
+    return;
+  }
+
+  bool wasConnected = false;
+
+  for (int32_t outputIndex = mOutputNodes.Length() - 1; outputIndex >= 0;
+       --outputIndex) {
+    if (mOutputNodes[outputIndex] != &aDestination) {
+      continue;
+    }
+    wasConnected |= DisconnectMatchingDestinationInputs<AudioNode>(
+        outputIndex, [aOutput](const InputNode& aInputNode) {
+          return aInputNode.mOutputPort == aOutput;
+        });
+  }
+
+  if (!wasConnected) {
+    aRv.ThrowInvalidAccessError(
+        "Trying to disconnect from a node we're not connected to");
+    return;
+  }
+}
+
+void AudioNode::Disconnect(AudioNode& aDestination, uint32_t aOutput,
+                           uint32_t aInput, ErrorResult& aRv) {
+  if (aOutput >= NumberOfOutputs()) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("Output index %u is out of bounds", aOutput));
+    return;
+  }
+
+  if (aInput >= aDestination.NumberOfInputs()) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("Input index %u is out of bounds", aInput));
+    return;
+  }
+
+  bool wasConnected = false;
+
+  for (int32_t outputIndex = mOutputNodes.Length() - 1; outputIndex >= 0;
+       --outputIndex) {
+    if (mOutputNodes[outputIndex] != &aDestination) {
+      continue;
+    }
+    wasConnected |= DisconnectMatchingDestinationInputs<AudioNode>(
+        outputIndex, [aOutput, aInput](const InputNode& aInputNode) {
+          return aInputNode.mOutputPort == aOutput &&
+                 aInputNode.mInputPort == aInput;
+        });
+  }
+
+  if (!wasConnected) {
+    aRv.ThrowInvalidAccessError(
+        "Trying to disconnect from a node we're not connected to");
+    return;
+  }
+}
+
+void AudioNode::Disconnect(AudioParam& aDestination, ErrorResult& aRv) {
+  bool wasConnected = false;
+
+  for (int32_t outputIndex = mOutputParams.Length() - 1; outputIndex >= 0;
+       --outputIndex) {
+    if (mOutputParams[outputIndex] != &aDestination) {
+      continue;
+    }
+    wasConnected |= DisconnectMatchingDestinationInputs<AudioParam>(
+        outputIndex, [](const InputNode&) { return true; });
+  }
+
+  if (!wasConnected) {
+    aRv.ThrowInvalidAccessError(
+        "Trying to disconnect from an AudioParam we're not connected to");
+    return;
+  }
+}
+
+void AudioNode::Disconnect(AudioParam& aDestination, uint32_t aOutput,
+                           ErrorResult& aRv) {
+  if (aOutput >= NumberOfOutputs()) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("Output index %u is out of bounds", aOutput));
+    return;
+  }
+
+  bool wasConnected = false;
+
+  for (int32_t outputIndex = mOutputParams.Length() - 1; outputIndex >= 0;
+       --outputIndex) {
+    if (mOutputParams[outputIndex] != &aDestination) {
+      continue;
+    }
+    wasConnected |= DisconnectMatchingDestinationInputs<AudioParam>(
+        outputIndex, [aOutput](const InputNode& aInputNode) {
+          return aInputNode.mOutputPort == aOutput;
+        });
+  }
+
+  if (!wasConnected) {
+    aRv.ThrowInvalidAccessError(
+        "Trying to disconnect from an AudioParam we're not connected to");
+    return;
+  }
+}
+
+void AudioNode::DestroyMediaTrack() {
+  if (mTrack) {
+    // Remove the node pointer on the engine.
+    AudioNodeTrack* ns = mTrack;
+    MOZ_ASSERT(ns, "How come we don't have a track here?");
+    MOZ_ASSERT(ns->Engine()->NodeMainThread() == this,
+               "Invalid node reference");
+    ns->Engine()->ClearNode();
+
+    mTrack->Destroy();
+    mTrack = nullptr;
+
+    nsCOMPtr<nsIObserverService> obs = services::GetObserverService();
+    if (obs) {
+      nsAutoString id;
+      id.AppendPrintf("%u", mId);
+      obs->NotifyObservers(nullptr, "webaudio-node-demise", id.get());
+    }
+  }
+}
+
+void AudioNode::RemoveOutputParam(AudioParam* aParam) {
+  mOutputParams.RemoveElement(aParam);
+}
+
+bool AudioNode::PassThrough() const {
+  MOZ_ASSERT(NumberOfInputs() <= 1 && NumberOfOutputs() == 1);
+  return mPassThrough;
+}
+
+void AudioNode::SetPassThrough(bool aPassThrough) {
+  MOZ_ASSERT(NumberOfInputs() <= 1 && NumberOfOutputs() == 1);
+  mPassThrough = aPassThrough;
+  if (mTrack) {
+    mTrack->SetPassThrough(mPassThrough);
+  }
+}
+
+AudioParam* AudioNode::CreateAudioParam(uint32_t aIndex, const nsAString& aName,
+                                        float aDefaultValue, float aMinValue,
+                                        float aMaxValue) {
+  return *mParams.AppendElement(
+      new AudioParam(this, aIndex, aName, aDefaultValue, aMinValue, aMaxValue));
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioNode.h b/dom/media/webaudio/AudioNode.h
new file mode 100644
index 0000000000..561749fd61
--- /dev/null
+++ b/dom/media/webaudio/AudioNode.h
@@ -0,0 +1,290 @@
+/* -*- 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/. */
+
+#ifndef AudioNode_h_
+#define AudioNode_h_
+
+#include "mozilla/DOMEventTargetHelper.h"
+#include "mozilla/dom/AudioNodeBinding.h"
+#include "nsCycleCollectionParticipant.h"
+#include "nsTArray.h"
+#include "AudioContext.h"
+#include "MediaTrackGraph.h"
+#include "WebAudioUtils.h"
+#include "mozilla/ErrorResult.h"
+#include "mozilla/MemoryReporting.h"
+#include "nsPrintfCString.h"
+#include "nsWeakReference.h"
+#include "SelfRef.h"
+
+namespace mozilla {
+
+class AbstractThread;
+
+namespace dom {
+
+class AudioContext;
+class AudioBufferSourceNode;
+class AudioParam;
+class AudioParamTimeline;
+struct ThreeDPoint;
+
+/**
+ * The DOM object representing a Web Audio AudioNode.
+ *
+ * Each AudioNode has a MediaTrack representing the actual
+ * real-time processing and output of this AudioNode.
+ *
+ * We track the incoming and outgoing connections to other AudioNodes.
+ * Outgoing connections have strong ownership.  Also, AudioNodes that will
+ * produce sound on their output even when they have silent or no input ask
+ * the AudioContext to keep playing or tail-time references to keep them alive
+ * until the context is finished.
+ *
+ * Explicit disconnections will only remove references from output nodes after
+ * the graph is notified and the main thread receives a reply.  Similarly,
+ * nodes with playing or tail-time references release these references only
+ * after receiving notification from their engine on the graph thread that
+ * playing has stopped.  Engines notifying the main thread that they have
+ * finished do so strictly *after* producing and returning their last block.
+ * In this way, an engine that receives non-null input knows that the input
+ * comes from nodes that are still alive and will keep their output nodes
+ * alive for at least as long as it takes to process messages from the graph
+ * thread.  i.e. the engine receiving non-null input knows that its node is
+ * still alive, and will still be alive when it receives a message from the
+ * engine.
+ */
+class AudioNode : public DOMEventTargetHelper, public nsSupportsWeakReference {
+ protected:
+  // You can only use refcounting to delete this object
+  virtual ~AudioNode();
+
+ public:
+  AudioNode(AudioContext* aContext, uint32_t aChannelCount,
+            ChannelCountMode aChannelCountMode,
+            ChannelInterpretation aChannelInterpretation);
+
+  // This should be idempotent (safe to call multiple times).
+  virtual void DestroyMediaTrack();
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioNode, DOMEventTargetHelper)
+
+  virtual AudioBufferSourceNode* AsAudioBufferSourceNode() { return nullptr; }
+
+  AudioContext* GetParentObject() const { return mContext; }
+
+  AudioContext* Context() const { return mContext; }
+
+  virtual AudioNode* Connect(AudioNode& aDestination, uint32_t aOutput,
+                             uint32_t aInput, ErrorResult& aRv);
+
+  virtual void Connect(AudioParam& aDestination, uint32_t aOutput,
+                       ErrorResult& aRv);
+
+  virtual void Disconnect(ErrorResult& aRv);
+  virtual void Disconnect(uint32_t aOutput, ErrorResult& aRv);
+  virtual void Disconnect(AudioNode& aDestination, ErrorResult& aRv);
+  virtual void Disconnect(AudioNode& aDestination, uint32_t aOutput,
+                          ErrorResult& aRv);
+  virtual void Disconnect(AudioNode& aDestination, uint32_t aOutput,
+                          uint32_t aInput, ErrorResult& aRv);
+  virtual void Disconnect(AudioParam& aDestination, ErrorResult& aRv);
+  virtual void Disconnect(AudioParam& aDestination, uint32_t aOutput,
+                          ErrorResult& aRv);
+
+  // Called after input nodes have been explicitly added or removed through
+  // the Connect() or Disconnect() methods.
+  virtual void NotifyInputsChanged() {}
+  // Indicate that the node should continue indefinitely to behave as if an
+  // input is connected, even though there is no longer a corresponding entry
+  // in mInputNodes.  Called after an input node has been removed because it
+  // is being garbage collected.
+  virtual void NotifyHasPhantomInput() {}
+
+  // The following two virtual methods must be implemented by each node type
+  // to provide their number of input and output ports. These numbers are
+  // constant for the lifetime of the node. Both default to 1.
+  virtual uint16_t NumberOfInputs() const { return 1; }
+  virtual uint16_t NumberOfOutputs() const { return 1; }
+
+  uint32_t Id() const { return mId; }
+
+  bool PassThrough() const;
+  void SetPassThrough(bool aPassThrough);
+
+  uint32_t ChannelCount() const { return mChannelCount; }
+  virtual void SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv) {
+    if (aChannelCount == 0 || aChannelCount > WebAudioUtils::MaxChannelCount) {
+      aRv.ThrowNotSupportedError(
+          nsPrintfCString("Channel count (%u) must be in the range [1, max "
+                          "supported channel count]",
+                          aChannelCount));
+      return;
+    }
+    mChannelCount = aChannelCount;
+    SendChannelMixingParametersToTrack();
+  }
+  ChannelCountMode ChannelCountModeValue() const { return mChannelCountMode; }
+  virtual void SetChannelCountModeValue(ChannelCountMode aMode,
+                                        ErrorResult& aRv) {
+    mChannelCountMode = aMode;
+    SendChannelMixingParametersToTrack();
+  }
+  ChannelInterpretation ChannelInterpretationValue() const {
+    return mChannelInterpretation;
+  }
+  virtual void SetChannelInterpretationValue(ChannelInterpretation aMode,
+                                             ErrorResult& aRv) {
+    mChannelInterpretation = aMode;
+    SendChannelMixingParametersToTrack();
+  }
+
+  struct InputNode final {
+    InputNode() = default;
+    InputNode(const InputNode&) = delete;
+    InputNode(InputNode&&) = default;
+
+    ~InputNode() {
+      if (mTrackPort) {
+        mTrackPort->Destroy();
+      }
+    }
+
+    size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+      size_t amount = 0;
+      if (mTrackPort) {
+        amount += mTrackPort->SizeOfIncludingThis(aMallocSizeOf);
+      }
+
+      return amount;
+    }
+
+    // The InputNode is destroyed when mInputNode is disconnected.
+    AudioNode* MOZ_NON_OWNING_REF mInputNode;
+    RefPtr<MediaInputPort> mTrackPort;
+    // The index of the input port this node feeds into.
+    // This is not used for connections to AudioParams.
+    uint32_t mInputPort;
+    // The index of the output port this node comes out of.
+    uint32_t mOutputPort;
+  };
+
+  // Returns the track, if any.
+  AudioNodeTrack* GetTrack() const { return mTrack; }
+
+  const nsTArray<InputNode>& InputNodes() const { return mInputNodes; }
+  const nsTArray<RefPtr<AudioNode>>& OutputNodes() const {
+    return mOutputNodes;
+  }
+  const nsTArray<RefPtr<AudioParam>>& OutputParams() const {
+    return mOutputParams;
+  }
+
+  template <typename T>
+  const nsTArray<InputNode>& InputsForDestination(uint32_t aOutputIndex) const;
+
+  void RemoveOutputParam(AudioParam* aParam);
+
+  // MarkActive() asks the context to keep the AudioNode alive until the
+  // context is finished.  This takes care of "playing" references and
+  // "tail-time" references.
+  void MarkActive() { Context()->RegisterActiveNode(this); }
+  // Active nodes call MarkInactive() when they have finished producing sound
+  // for the foreseeable future.
+  // Do not call MarkInactive from a node destructor.  If the destructor is
+  // called, then the node is already inactive.
+  // MarkInactive() may delete |this|.
+  void MarkInactive() { Context()->UnregisterActiveNode(this); }
+
+  virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const;
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const;
+
+  // Returns a string from constant static storage identifying the dom node
+  // type.
+  virtual const char* NodeType() const = 0;
+
+  const nsTArray<RefPtr<AudioParam>>& GetAudioParams() const { return mParams; }
+
+ private:
+  // Given:
+  //
+  // - a DestinationType, that can be an AudioNode or an AudioParam ;
+  // - a Predicate, a function that takes an InputNode& and returns a bool ;
+  //
+  // This method iterates on the InputNodes() of the node at the index
+  // aDestinationIndex, and calls `DisconnectFromOutputIfConnected` with this
+  // input node, if aPredicate returns true.
+  template <typename DestinationType, typename Predicate>
+  bool DisconnectMatchingDestinationInputs(uint32_t aDestinationIndex,
+                                           Predicate aPredicate);
+
+  virtual void LastRelease() override {
+    // We are about to be deleted, disconnect the object from the graph before
+    // the derived type is destroyed.
+    DisconnectFromGraph();
+  }
+  // Callers must hold a reference to 'this'.
+  void DisconnectFromGraph();
+
+  template <typename DestinationType>
+  bool DisconnectFromOutputIfConnected(uint32_t aOutputIndex,
+                                       uint32_t aInputIndex);
+
+ protected:
+  // Helper for the Constructors for nodes.
+  void Initialize(const AudioNodeOptions& aOptions, ErrorResult& aRv);
+
+  // Helpers for sending different value types to tracks
+  void SendDoubleParameterToTrack(uint32_t aIndex, double aValue);
+  void SendInt32ParameterToTrack(uint32_t aIndex, int32_t aValue);
+  void SendChannelMixingParametersToTrack();
+
+ private:
+  RefPtr<AudioContext> mContext;
+
+ protected:
+  // Set in the constructor of all nodes except offline AudioDestinationNode.
+  // Must not become null until finished.
+  RefPtr<AudioNodeTrack> mTrack;
+
+  // The reference pointing out all audio params which belong to this node.
+  nsTArray<RefPtr<AudioParam>> mParams;
+  // Use this function to create an AudioParam, so as to automatically add
+  // the new AudioParam to `mParams`.
+  AudioParam* CreateAudioParam(
+      uint32_t aIndex, const nsAString& aName, float aDefaultValue,
+      float aMinValue = std::numeric_limits<float>::lowest(),
+      float aMaxValue = std::numeric_limits<float>::max());
+
+ private:
+  // For every InputNode, there is a corresponding entry in mOutputNodes of the
+  // InputNode's mInputNode.
+  nsTArray<InputNode> mInputNodes;
+  // For every mOutputNode entry, there is a corresponding entry in mInputNodes
+  // of the mOutputNode entry. We won't necessarily be able to identify the
+  // exact matching entry, since mOutputNodes doesn't include the port
+  // identifiers and the same node could be connected on multiple ports.
+  nsTArray<RefPtr<AudioNode>> mOutputNodes;
+  // For every mOutputParams entry, there is a corresponding entry in
+  // AudioParam::mInputNodes of the mOutputParams entry. We won't necessarily be
+  // able to identify the exact matching entry, since mOutputParams doesn't
+  // include the port identifiers and the same node could be connected on
+  // multiple ports.
+  nsTArray<RefPtr<AudioParam>> mOutputParams;
+  uint32_t mChannelCount;
+  ChannelCountMode mChannelCountMode;
+  ChannelInterpretation mChannelInterpretation;
+  const uint32_t mId;
+  // Whether the node just passes through its input.  This is a devtools API
+  // that only works for some node types.
+  bool mPassThrough;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/AudioNodeEngine.cpp b/dom/media/webaudio/AudioNodeEngine.cpp
new file mode 100644
index 0000000000..3fba688b8c
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngine.cpp
@@ -0,0 +1,437 @@
+/* -*- 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 "AudioNodeEngine.h"
+
+#include "mozilla/AbstractThread.h"
+#ifdef USE_NEON
+#  include "mozilla/arm.h"
+#  include "AudioNodeEngineGeneric.h"
+#endif
+#ifdef USE_SSE2
+#  include "mozilla/SSE.h"
+#  include "AudioNodeEngineGeneric.h"
+#endif
+#if defined(USE_SSE42) && defined(USE_FMA3)
+#  include "mozilla/SSE.h"
+#  include "AudioNodeEngineGeneric.h"
+#endif
+#include "AudioBlock.h"
+#include "Tracing.h"
+
+namespace mozilla {
+
+already_AddRefed<ThreadSharedFloatArrayBufferList>
+ThreadSharedFloatArrayBufferList::Create(uint32_t aChannelCount, size_t aLength,
+                                         const mozilla::fallible_t&) {
+  RefPtr<ThreadSharedFloatArrayBufferList> buffer =
+      new ThreadSharedFloatArrayBufferList(aChannelCount);
+
+  for (uint32_t i = 0; i < aChannelCount; ++i) {
+    float* channelData = js_pod_malloc<float>(aLength);
+    if (!channelData) {
+      return nullptr;
+    }
+
+    buffer->SetData(i, channelData, js_free, channelData);
+  }
+
+  return buffer.forget();
+}
+
+void WriteZeroesToAudioBlock(AudioBlock* aChunk, uint32_t aStart,
+                             uint32_t aLength) {
+  MOZ_ASSERT(aStart + aLength <= WEBAUDIO_BLOCK_SIZE);
+  MOZ_ASSERT(!aChunk->IsNull(), "You should pass a non-null chunk");
+  if (aLength == 0) {
+    return;
+  }
+
+  for (uint32_t i = 0; i < aChunk->ChannelCount(); ++i) {
+    PodZero(aChunk->ChannelFloatsForWrite(i) + aStart, aLength);
+  }
+}
+
+void AudioBufferCopyWithScale(const float* aInput, float aScale, float* aOutput,
+                              uint32_t aSize) {
+  if (aScale == 1.0f) {
+    PodCopy(aOutput, aInput, aSize);
+  } else {
+    for (uint32_t i = 0; i < aSize; ++i) {
+      aOutput[i] = aInput[i] * aScale;
+    }
+  }
+}
+
+void AudioBufferAddWithScale(const float* aInput, float aScale, float* aOutput,
+                             uint32_t aSize) {
+#ifdef USE_NEON
+  if (mozilla::supports_neon()) {
+    Engine<xsimd::neon>::AudioBufferAddWithScale(aInput, aScale, aOutput,
+                                                 aSize);
+    return;
+  }
+#endif
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse2()) {
+#  if defined(USE_SSE42) && defined(USE_FMA3)
+    if (mozilla::supports_fma3() && mozilla::supports_sse4_2()) {
+      Engine<xsimd::fma3<xsimd::sse4_2>>::AudioBufferAddWithScale(
+          aInput, aScale, aOutput, aSize);
+    } else
+#  endif
+    {
+      Engine<xsimd::sse2>::AudioBufferAddWithScale(aInput, aScale, aOutput,
+                                                   aSize);
+    }
+    return;
+  }
+#endif
+
+  if (aScale == 1.0f) {
+    for (uint32_t i = 0; i < aSize; ++i) {
+      aOutput[i] += aInput[i];
+    }
+  } else {
+    for (uint32_t i = 0; i < aSize; ++i) {
+      aOutput[i] += aInput[i] * aScale;
+    }
+  }
+}
+
+void AudioBlockAddChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                                   float aScale,
+                                   float aOutput[WEBAUDIO_BLOCK_SIZE]) {
+  AudioBufferAddWithScale(aInput, aScale, aOutput, WEBAUDIO_BLOCK_SIZE);
+}
+
+void AudioBlockCopyChannelWithScale(const float* aInput, float aScale,
+                                    float* aOutput) {
+  if (aScale == 1.0f) {
+    memcpy(aOutput, aInput, WEBAUDIO_BLOCK_SIZE * sizeof(float));
+  } else {
+#ifdef USE_NEON
+    if (mozilla::supports_neon()) {
+      Engine<xsimd::neon>::AudioBlockCopyChannelWithScale(aInput, aScale,
+                                                          aOutput);
+      return;
+    }
+#endif
+
+#ifdef USE_SSE2
+    if (mozilla::supports_sse2()) {
+      Engine<xsimd::sse2>::AudioBlockCopyChannelWithScale(aInput, aScale,
+                                                          aOutput);
+      return;
+    }
+#endif
+
+    for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+      aOutput[i] = aInput[i] * aScale;
+    }
+  }
+}
+
+void BufferComplexMultiply(const float* aInput, const float* aScale,
+                           float* aOutput, uint32_t aSize) {
+#ifdef USE_NEON
+  if (mozilla::supports_neon()) {
+    Engine<xsimd::neon>::BufferComplexMultiply(aInput, aScale, aOutput, aSize);
+    return;
+  }
+#endif
+#ifdef USE_SSE2
+  if (mozilla::supports_sse()) {
+#  if defined(USE_SSE42) && defined(USE_FMA3)
+    if (mozilla::supports_fma3() && mozilla::supports_sse4_2()) {
+      Engine<xsimd::fma3<xsimd::sse4_2>>::BufferComplexMultiply(aInput, aScale,
+                                                                aOutput, aSize);
+    } else
+#  endif
+    {
+      Engine<xsimd::sse2>::BufferComplexMultiply(aInput, aScale, aOutput,
+                                                 aSize);
+    }
+    return;
+  }
+#endif
+
+  for (uint32_t i = 0; i < aSize * 2; i += 2) {
+    float real1 = aInput[i];
+    float imag1 = aInput[i + 1];
+    float real2 = aScale[i];
+    float imag2 = aScale[i + 1];
+    float realResult = real1 * real2 - imag1 * imag2;
+    float imagResult = real1 * imag2 + imag1 * real2;
+    aOutput[i] = realResult;
+    aOutput[i + 1] = imagResult;
+  }
+}
+
+float AudioBufferPeakValue(const float* aInput, uint32_t aSize) {
+  float max = 0.0f;
+  for (uint32_t i = 0; i < aSize; i++) {
+    float mag = fabs(aInput[i]);
+    if (mag > max) {
+      max = mag;
+    }
+  }
+  return max;
+}
+
+void AudioBlockCopyChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                                    const float aScale[WEBAUDIO_BLOCK_SIZE],
+                                    float aOutput[WEBAUDIO_BLOCK_SIZE]) {
+#ifdef USE_NEON
+  if (mozilla::supports_neon()) {
+    Engine<xsimd::neon>::AudioBlockCopyChannelWithScale(aInput, aScale,
+                                                        aOutput);
+    return;
+  }
+#endif
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse2()) {
+    Engine<xsimd::sse2>::AudioBlockCopyChannelWithScale(aInput, aScale,
+                                                        aOutput);
+    return;
+  }
+#endif
+
+  for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+    aOutput[i] = aInput[i] * aScale[i];
+  }
+}
+
+void AudioBlockInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE], float aScale) {
+  AudioBufferInPlaceScale(aBlock, aScale, WEBAUDIO_BLOCK_SIZE);
+}
+
+void AudioBlockInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE],
+                            float aScale[WEBAUDIO_BLOCK_SIZE]) {
+  AudioBufferInPlaceScale(aBlock, aScale, WEBAUDIO_BLOCK_SIZE);
+}
+
+void AudioBufferInPlaceScale(float* aBlock, float aScale, uint32_t aSize) {
+  if (aScale == 1.0f) {
+    return;
+  }
+#ifdef USE_NEON
+  if (mozilla::supports_neon()) {
+    Engine<xsimd::neon>::AudioBufferInPlaceScale(aBlock, aScale, aSize);
+    return;
+  }
+#endif
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse2()) {
+    Engine<xsimd::sse2>::AudioBufferInPlaceScale(aBlock, aScale, aSize);
+    return;
+  }
+#endif
+
+  for (uint32_t i = 0; i < aSize; ++i) {
+    *aBlock++ *= aScale;
+  }
+}
+
+void AudioBufferInPlaceScale(float* aBlock, float* aScale, uint32_t aSize) {
+#ifdef USE_NEON
+  if (mozilla::supports_neon()) {
+    Engine<xsimd::neon>::AudioBufferInPlaceScale(aBlock, aScale, aSize);
+    return;
+  }
+#endif
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse2()) {
+    Engine<xsimd::sse2>::AudioBufferInPlaceScale(aBlock, aScale, aSize);
+    return;
+  }
+#endif
+
+  for (uint32_t i = 0; i < aSize; ++i) {
+    *aBlock++ *= *aScale++;
+  }
+}
+
+void AudioBlockPanMonoToStereo(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                               float aGainL[WEBAUDIO_BLOCK_SIZE],
+                               float aGainR[WEBAUDIO_BLOCK_SIZE],
+                               float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                               float aOutputR[WEBAUDIO_BLOCK_SIZE]) {
+  AudioBlockCopyChannelWithScale(aInput, aGainL, aOutputL);
+  AudioBlockCopyChannelWithScale(aInput, aGainR, aOutputR);
+}
+
+void AudioBlockPanMonoToStereo(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                               float aGainL, float aGainR,
+                               float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                               float aOutputR[WEBAUDIO_BLOCK_SIZE]) {
+  AudioBlockCopyChannelWithScale(aInput, aGainL, aOutputL);
+  AudioBlockCopyChannelWithScale(aInput, aGainR, aOutputR);
+}
+
+void AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                                 const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                                 float aGainL, float aGainR, bool aIsOnTheLeft,
+                                 float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                                 float aOutputR[WEBAUDIO_BLOCK_SIZE]) {
+#ifdef USE_NEON
+  if (mozilla::supports_neon()) {
+    Engine<xsimd::neon>::AudioBlockPanStereoToStereo(
+        aInputL, aInputR, aGainL, aGainR, aIsOnTheLeft, aOutputL, aOutputR);
+    return;
+  }
+#endif
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse2()) {
+#  if defined(USE_SSE42) && defined(USE_FMA3)
+    if (mozilla::supports_fma3() && mozilla::supports_sse4_2()) {
+      Engine<xsimd::fma3<xsimd::sse4_2>>::AudioBlockPanStereoToStereo(
+          aInputL, aInputR, aGainL, aGainR, aIsOnTheLeft, aOutputL, aOutputR);
+    } else
+#  endif
+    {
+      Engine<xsimd::sse2>::AudioBlockPanStereoToStereo(
+          aInputL, aInputR, aGainL, aGainR, aIsOnTheLeft, aOutputL, aOutputR);
+    }
+    return;
+  }
+#endif
+
+  uint32_t i;
+
+  if (aIsOnTheLeft) {
+    for (i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+      aOutputL[i] = aInputL[i] + aInputR[i] * aGainL;
+      aOutputR[i] = aInputR[i] * aGainR;
+    }
+  } else {
+    for (i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+      aOutputL[i] = aInputL[i] * aGainL;
+      aOutputR[i] = aInputR[i] + aInputL[i] * aGainR;
+    }
+  }
+}
+
+void AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                                 const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                                 const float aGainL[WEBAUDIO_BLOCK_SIZE],
+                                 const float aGainR[WEBAUDIO_BLOCK_SIZE],
+                                 const bool aIsOnTheLeft[WEBAUDIO_BLOCK_SIZE],
+                                 float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                                 float aOutputR[WEBAUDIO_BLOCK_SIZE]) {
+#ifdef USE_NEON
+  if (mozilla::supports_neon()) {
+    Engine<xsimd::neon>::AudioBlockPanStereoToStereo(
+        aInputL, aInputR, aGainL, aGainR, aIsOnTheLeft, aOutputL, aOutputR);
+    return;
+  }
+#endif
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse2()) {
+#  if defined(USE_SSE42) && defined(USE_FMA3)
+    if (mozilla::supports_fma3() && mozilla::supports_sse4_2()) {
+      Engine<xsimd::fma3<xsimd::sse2>>::AudioBlockPanStereoToStereo(
+          aInputL, aInputR, aGainL, aGainR, aIsOnTheLeft, aOutputL, aOutputR);
+    } else
+#  endif
+    {
+      Engine<xsimd::sse2>::AudioBlockPanStereoToStereo(
+          aInputL, aInputR, aGainL, aGainR, aIsOnTheLeft, aOutputL, aOutputR);
+    }
+    return;
+  }
+#endif
+
+  uint32_t i;
+  for (i = 0; i < WEBAUDIO_BLOCK_SIZE; i++) {
+    if (aIsOnTheLeft[i]) {
+      aOutputL[i] = aInputL[i] + aInputR[i] * aGainL[i];
+      aOutputR[i] = aInputR[i] * aGainR[i];
+    } else {
+      aOutputL[i] = aInputL[i] * aGainL[i];
+      aOutputR[i] = aInputR[i] + aInputL[i] * aGainR[i];
+    }
+  }
+}
+
+float AudioBufferSumOfSquares(const float* aInput, uint32_t aLength) {
+#ifdef USE_NEON
+  if (mozilla::supports_neon()) {
+    return Engine<xsimd::neon>::AudioBufferSumOfSquares(aInput, aLength);
+  }
+#endif
+
+#ifdef USE_SSE2
+  if (mozilla::supports_sse()) {
+#  if defined(USE_SSE42) && defined(USE_FMA3)
+    if (mozilla::supports_fma3() && mozilla::supports_sse4_2()) {
+      return Engine<xsimd::fma3<xsimd::sse4_2>>::AudioBufferSumOfSquares(
+          aInput, aLength);
+    } else
+#  endif
+    {
+      return Engine<xsimd::sse2>::AudioBufferSumOfSquares(aInput, aLength);
+    }
+  }
+#endif
+
+  float sum = 0.f;
+  while (aLength--) {
+    sum += *aInput * *aInput;
+    ++aInput;
+  }
+  return sum;
+}
+
+void NaNToZeroInPlace(float* aSamples, size_t aCount) {
+#ifdef USE_SSE2
+  if (mozilla::supports_sse2()) {
+    Engine<xsimd::sse2>::NaNToZeroInPlace(aSamples, aCount);
+    return;
+  }
+#endif
+  for (size_t i = 0; i < aCount; i++) {
+    if (aSamples[i] != aSamples[i]) {
+      aSamples[i] = 0.0;
+    }
+  }
+}
+
+AudioNodeEngine::AudioNodeEngine(dom::AudioNode* aNode)
+    : mNode(aNode),
+      mNodeType(aNode ? aNode->NodeType() : nullptr),
+      mInputCount(aNode ? aNode->NumberOfInputs() : 1),
+      mOutputCount(aNode ? aNode->NumberOfOutputs() : 0) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_COUNT_CTOR(AudioNodeEngine);
+}
+
+void AudioNodeEngine::ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                                   const AudioBlock& aInput,
+                                   AudioBlock* aOutput, bool* aFinished) {
+  MOZ_ASSERT(mInputCount <= 1 && mOutputCount <= 1);
+  TRACE("AudioNodeEngine::ProcessBlock");
+  *aOutput = aInput;
+}
+
+void AudioNodeEngine::ProcessBlocksOnPorts(AudioNodeTrack* aTrack,
+                                           GraphTime aFrom,
+                                           Span<const AudioBlock> aInput,
+                                           Span<AudioBlock> aOutput,
+                                           bool* aFinished) {
+  MOZ_ASSERT(mInputCount > 1 || mOutputCount > 1);
+  TRACE("AudioNodeEngine::ProcessBlocksOnPorts");
+  // Only produce one output port, and drop all other input ports.
+  aOutput[0] = aInput[0];
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webaudio/AudioNodeEngine.h b/dom/media/webaudio/AudioNodeEngine.h
new file mode 100644
index 0000000000..7a30e1b4cc
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngine.h
@@ -0,0 +1,389 @@
+/* -*- 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/. */
+#ifndef MOZILLA_AUDIONODEENGINE_H_
+#define MOZILLA_AUDIONODEENGINE_H_
+
+#include "AudioSegment.h"
+#include "mozilla/dom/AudioNode.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/Mutex.h"
+
+namespace WebCore {
+class Reverb;
+}  // namespace WebCore
+
+namespace mozilla {
+
+namespace dom {
+struct ThreeDPoint;
+class AudioParamTimeline;
+class DelayNodeEngine;
+struct AudioParamEvent;
+}  // namespace dom
+
+class AbstractThread;
+class AudioBlock;
+class AudioNodeTrack;
+
+/**
+ * This class holds onto a set of immutable channel buffers. The storage
+ * for the buffers must be malloced, but the buffer pointers and the malloc
+ * pointers can be different (e.g. if the buffers are contained inside
+ * some malloced object).
+ */
+class ThreadSharedFloatArrayBufferList final : public ThreadSharedObject {
+ public:
+  /**
+   * Construct with null channel data pointers.
+   */
+  explicit ThreadSharedFloatArrayBufferList(uint32_t aCount) {
+    mContents.SetLength(aCount);
+  }
+  /**
+   * Create with buffers suitable for transfer to
+   * JS::NewArrayBufferWithContents().  The buffer contents are uninitialized
+   * and so should be set using GetDataForWrite().
+   */
+  static already_AddRefed<ThreadSharedFloatArrayBufferList> Create(
+      uint32_t aChannelCount, size_t aLength, const mozilla::fallible_t&);
+
+  ThreadSharedFloatArrayBufferList* AsThreadSharedFloatArrayBufferList()
+      override {
+    return this;
+  };
+
+  struct Storage final {
+    Storage() : mDataToFree(nullptr), mFree(nullptr), mSampleData(nullptr) {}
+    ~Storage() {
+      if (mFree) {
+        mFree(mDataToFree);
+      } else {
+        MOZ_ASSERT(!mDataToFree);
+      }
+    }
+    size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+      // NB: mSampleData might not be owned, if it is it just points to
+      //     mDataToFree.
+      return aMallocSizeOf(mDataToFree);
+    }
+    void* mDataToFree;
+    void (*mFree)(void*);
+    float* mSampleData;
+  };
+
+  /**
+   * This can be called on any thread.
+   */
+  uint32_t GetChannels() const { return mContents.Length(); }
+  /**
+   * This can be called on any thread.
+   */
+  const float* GetData(uint32_t aIndex) const {
+    return mContents[aIndex].mSampleData;
+  }
+  /**
+   * This can be called on any thread, but only when the calling thread is the
+   * only owner.
+   */
+  float* GetDataForWrite(uint32_t aIndex) {
+    MOZ_ASSERT(!IsShared());
+    return mContents[aIndex].mSampleData;
+  }
+
+  /**
+   * Call this only during initialization, before the object is handed to
+   * any other thread.
+   */
+  void SetData(uint32_t aIndex, void* aDataToFree, void (*aFreeFunc)(void*),
+               float* aData) {
+    Storage* s = &mContents[aIndex];
+    if (s->mFree) {
+      s->mFree(s->mDataToFree);
+    } else {
+      MOZ_ASSERT(!s->mDataToFree);
+    }
+
+    s->mDataToFree = aDataToFree;
+    s->mFree = aFreeFunc;
+    s->mSampleData = aData;
+  }
+
+  /**
+   * Put this object into an error state where there are no channels.
+   */
+  void Clear() { mContents.Clear(); }
+
+  size_t SizeOfExcludingThis(
+      mozilla::MallocSizeOf aMallocSizeOf) const override {
+    size_t amount = ThreadSharedObject::SizeOfExcludingThis(aMallocSizeOf);
+    amount += mContents.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    for (size_t i = 0; i < mContents.Length(); i++) {
+      amount += mContents[i].SizeOfExcludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  AutoTArray<Storage, 2> mContents;
+};
+
+/**
+ * aChunk must have been allocated by AllocateAudioBlock.
+ */
+void WriteZeroesToAudioBlock(AudioBlock* aChunk, uint32_t aStart,
+                             uint32_t aLength);
+
+/**
+ * Copy with scale. aScale == 1.0f should be optimized.
+ */
+void AudioBufferCopyWithScale(const float* aInput, float aScale, float* aOutput,
+                              uint32_t aSize);
+
+/**
+ * Pointwise multiply-add operation. aScale == 1.0f should be optimized.
+ */
+void AudioBufferAddWithScale(const float* aInput, float aScale, float* aOutput,
+                             uint32_t aSize);
+
+/**
+ * Pointwise multiply-add operation. aScale == 1.0f should be optimized.
+ */
+void AudioBlockAddChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                                   float aScale,
+                                   float aOutput[WEBAUDIO_BLOCK_SIZE]);
+
+/**
+ * Pointwise copy-scaled operation. aScale == 1.0f should be optimized.
+ *
+ * Buffer size is implicitly assumed to be WEBAUDIO_BLOCK_SIZE.
+ */
+void AudioBlockCopyChannelWithScale(const float* aInput, float aScale,
+                                    float* aOutput);
+
+/**
+ * Vector copy-scaled operation.
+ */
+void AudioBlockCopyChannelWithScale(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                                    const float aScale[WEBAUDIO_BLOCK_SIZE],
+                                    float aOutput[WEBAUDIO_BLOCK_SIZE]);
+
+/**
+ * Vector complex multiplication on arbitrary sized buffers.
+ */
+void BufferComplexMultiply(const float* aInput, const float* aScale,
+                           float* aOutput, uint32_t aSize);
+
+/**
+ * Vector maximum element magnitude ( max(abs(aInput)) ).
+ */
+float AudioBufferPeakValue(const float* aInput, uint32_t aSize);
+
+/**
+ * In place gain. aScale == 1.0f should be optimized.
+ */
+void AudioBlockInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE], float aScale);
+
+/**
+ * In place gain. aScale == 1.0f should be optimized.
+ */
+void AudioBufferInPlaceScale(float* aBlock, float aScale, uint32_t aSize);
+
+/**
+ * a-rate in place gain.
+ */
+void AudioBlockInPlaceScale(float aBlock[WEBAUDIO_BLOCK_SIZE],
+                            float aScale[WEBAUDIO_BLOCK_SIZE]);
+/**
+ * a-rate in place gain.
+ */
+void AudioBufferInPlaceScale(float* aBlock, float* aScale, uint32_t aSize);
+
+/**
+ * Upmix a mono input to a stereo output, scaling the two output channels by two
+ * different gain value.
+ * This algorithm is specified in the WebAudio spec.
+ */
+void AudioBlockPanMonoToStereo(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                               float aGainL, float aGainR,
+                               float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                               float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+
+void AudioBlockPanMonoToStereo(const float aInput[WEBAUDIO_BLOCK_SIZE],
+                               float aGainL[WEBAUDIO_BLOCK_SIZE],
+                               float aGainR[WEBAUDIO_BLOCK_SIZE],
+                               float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                               float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+/**
+ * Pan a stereo source according to right and left gain, and the position
+ * (whether the listener is on the left of the source or not).
+ * This algorithm is specified in the WebAudio spec.
+ */
+void AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                                 const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                                 float aGainL, float aGainR, bool aIsOnTheLeft,
+                                 float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                                 float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+void AudioBlockPanStereoToStereo(const float aInputL[WEBAUDIO_BLOCK_SIZE],
+                                 const float aInputR[WEBAUDIO_BLOCK_SIZE],
+                                 const float aGainL[WEBAUDIO_BLOCK_SIZE],
+                                 const float aGainR[WEBAUDIO_BLOCK_SIZE],
+                                 const bool aIsOnTheLeft[WEBAUDIO_BLOCK_SIZE],
+                                 float aOutputL[WEBAUDIO_BLOCK_SIZE],
+                                 float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+
+/**
+ * Replace NaN by zeros in aSamples.
+ */
+void NaNToZeroInPlace(float* aSamples, size_t aCount);
+
+/**
+ * Return the sum of squares of all of the samples in the input.
+ */
+float AudioBufferSumOfSquares(const float* aInput, uint32_t aLength);
+
+/**
+ * All methods of this class and its subclasses are called on the
+ * MediaTrackGraph thread.
+ */
+class AudioNodeEngine {
+ public:
+  // This should be compatible with AudioNodeTrack::OutputChunks.
+  typedef AutoTArray<AudioBlock, 1> OutputChunks;
+
+  explicit AudioNodeEngine(dom::AudioNode* aNode);
+
+  virtual ~AudioNodeEngine() {
+    MOZ_ASSERT(!mNode, "The node reference must be already cleared");
+    MOZ_COUNT_DTOR(AudioNodeEngine);
+  }
+
+  virtual dom::DelayNodeEngine* AsDelayNodeEngine() { return nullptr; }
+
+  virtual void SetTrackTimeParameter(uint32_t aIndex, TrackTime aParam) {
+    NS_ERROR("Invalid SetTrackTimeParameter index");
+  }
+  virtual void SetDoubleParameter(uint32_t aIndex, double aParam) {
+    NS_ERROR("Invalid SetDoubleParameter index");
+  }
+  virtual void SetInt32Parameter(uint32_t aIndex, int32_t aParam) {
+    NS_ERROR("Invalid SetInt32Parameter index");
+  }
+  virtual void RecvTimelineEvent(uint32_t aIndex,
+                                 dom::AudioParamEvent& aValue) {
+    NS_ERROR("Invalid RecvTimelineEvent index");
+  }
+  virtual void SetBuffer(AudioChunk&& aBuffer) {
+    NS_ERROR("SetBuffer called on engine that doesn't support it");
+  }
+  // This consumes the contents of aData.  aData will be emptied after this
+  // returns.
+  virtual void SetRawArrayData(nsTArray<float>&& aData) {
+    NS_ERROR("SetRawArrayData called on an engine that doesn't support it");
+  }
+
+  virtual void SetReverb(WebCore::Reverb* aBuffer,
+                         uint32_t aImpulseChannelCount) {
+    NS_ERROR("SetReverb called on engine that doesn't support it");
+  }
+
+  /**
+   * Produce the next block of audio samples, given input samples aInput
+   * (the mixed data for input 0).
+   * aInput is guaranteed to have float sample format (if it has samples at all)
+   * and to have been resampled to the sampling rate for the track, and to have
+   * exactly WEBAUDIO_BLOCK_SIZE samples.
+   * *aFinished is set to false by the caller. The callee must not set this to
+   * true unless silent output is produced. If set to true, we'll finish the
+   * track, consider this input inactive on any downstream nodes, and not
+   * call this again.
+   */
+  virtual void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                            const AudioBlock& aInput, AudioBlock* aOutput,
+                            bool* aFinished);
+  /**
+   * Produce the next block of audio samples, before input is provided.
+   * ProcessBlock() will be called later, and it then should not change
+   * aOutput.  This is used only for DelayNodeEngine in a feedback loop.
+   */
+  virtual void ProduceBlockBeforeInput(AudioNodeTrack* aTrack, GraphTime aFrom,
+                                       AudioBlock* aOutput) {
+    MOZ_ASSERT_UNREACHABLE("ProduceBlockBeforeInput called on wrong engine");
+  }
+
+  /**
+   * Produce the next block of audio samples, given input samples in the aInput
+   * array.  There is one input sample per port in aInput, in order.
+   * This is the multi-input/output version of ProcessBlock.  Only one kind
+   * of ProcessBlock is called on each node.  ProcessBlocksOnPorts() is called
+   * instead of ProcessBlock() if either the number of inputs or the number of
+   * outputs is greater than 1.
+   *
+   * The numbers of AudioBlocks in aInput and aOutput are always guaranteed to
+   * match the numbers of inputs and outputs for the node.
+   */
+  virtual void ProcessBlocksOnPorts(AudioNodeTrack* aTrack, GraphTime aFrom,
+                                    Span<const AudioBlock> aInput,
+                                    Span<AudioBlock> aOutput, bool* aFinished);
+
+  // IsActive() returns true if the engine needs to continue processing an
+  // unfinished track even when it has silent or no input connections.  This
+  // includes tail-times and when sources have been scheduled to start.  If
+  // returning false, then the track can be suspended.
+  virtual bool IsActive() const { return false; }
+
+  // Called on graph thread when the engine will not be used again.
+  virtual void OnGraphThreadDone() {}
+
+  bool HasNode() const {
+    MOZ_ASSERT(NS_IsMainThread());
+    return !!mNode;
+  }
+
+  dom::AudioNode* NodeMainThread() const {
+    MOZ_ASSERT(NS_IsMainThread());
+    return mNode;
+  }
+
+  void ClearNode() {
+    MOZ_ASSERT(NS_IsMainThread());
+    MOZ_ASSERT(mNode != nullptr);
+    mNode = nullptr;
+  }
+
+  uint16_t InputCount() const { return mInputCount; }
+  uint16_t OutputCount() const { return mOutputCount; }
+
+  virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+    // NB: |mNode| is tracked separately so it is excluded here.
+    return 0;
+  }
+
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  void SizeOfIncludingThis(MallocSizeOf aMallocSizeOf,
+                           AudioNodeSizes& aUsage) const {
+    aUsage.mEngine = SizeOfIncludingThis(aMallocSizeOf);
+    aUsage.mNodeType = mNodeType;
+  }
+
+ private:
+  // This is cleared from AudioNode::DestroyMediaTrack()
+  dom::AudioNode* MOZ_NON_OWNING_REF mNode;  // main thread only
+  const char* const mNodeType;
+  const uint16_t mInputCount;
+  const uint16_t mOutputCount;
+};
+
+}  // namespace mozilla
+
+#endif /* MOZILLA_AUDIONODEENGINE_H_ */
diff --git a/dom/media/webaudio/AudioNodeEngineGeneric.h b/dom/media/webaudio/AudioNodeEngineGeneric.h
new file mode 100644
index 0000000000..de436ebc8d
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngineGeneric.h
@@ -0,0 +1,58 @@
+/* -*- mode: c++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* this source code form is subject to the terms of the mozilla public
+ * license, v. 2.0. if a copy of the mpl was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_AUDIONODEENGINEGENERIC_H_
+#define MOZILLA_AUDIONODEENGINEGENERIC_H_
+
+#include "AudioNodeEngine.h"
+
+#include "xsimd/xsimd.hpp"
+
+namespace mozilla {
+
+template <class Arch>
+struct Engine {
+  static void AudioBufferAddWithScale(const float* aInput, float aScale,
+                                      float* aOutput, uint32_t aSize);
+
+  static void AudioBlockCopyChannelWithScale(const float* aInput, float aScale,
+                                             float* aOutput);
+
+  static void AudioBlockCopyChannelWithScale(
+      const float aInput[WEBAUDIO_BLOCK_SIZE],
+      const float aScale[WEBAUDIO_BLOCK_SIZE],
+      float aOutput[WEBAUDIO_BLOCK_SIZE]);
+
+  static void AudioBufferInPlaceScale(float* aBlock, float aScale,
+                                      uint32_t aSize);
+
+  static void AudioBufferInPlaceScale(float* aBlock, float* aScale,
+                                      uint32_t aSize);
+
+  static void AudioBlockPanStereoToStereo(
+      const float aInputL[WEBAUDIO_BLOCK_SIZE],
+      const float aInputR[WEBAUDIO_BLOCK_SIZE], float aGainL, float aGainR,
+      bool aIsOnTheLeft, float aOutputL[WEBAUDIO_BLOCK_SIZE],
+      float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+
+  static void BufferComplexMultiply(const float* aInput, const float* aScale,
+                                    float* aOutput, uint32_t aSize);
+
+  static float AudioBufferSumOfSquares(const float* aInput, uint32_t aLength);
+
+  static void NaNToZeroInPlace(float* aSamples, size_t aCount);
+
+  static void AudioBlockPanStereoToStereo(
+      const float aInputL[WEBAUDIO_BLOCK_SIZE],
+      const float aInputR[WEBAUDIO_BLOCK_SIZE],
+      const float aGainL[WEBAUDIO_BLOCK_SIZE],
+      const float aGainR[WEBAUDIO_BLOCK_SIZE],
+      const bool aIsOnTheLeft[WEBAUDIO_BLOCK_SIZE],
+      float aOutputL[WEBAUDIO_BLOCK_SIZE], float aOutputR[WEBAUDIO_BLOCK_SIZE]);
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/AudioNodeEngineGenericImpl.h b/dom/media/webaudio/AudioNodeEngineGenericImpl.h
new file mode 100644
index 0000000000..b7fabc7a0b
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngineGenericImpl.h
@@ -0,0 +1,341 @@
+/* -*- mode: c++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* this source code form is subject to the terms of the mozilla public
+ * license, v. 2.0. if a copy of the mpl was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_AUDIONODEENGINEGENERICIMPL_H_
+#define MOZILLA_AUDIONODEENGINEGENERICIMPL_H_
+
+#include "AudioNodeEngineGeneric.h"
+#include "AlignmentUtils.h"
+
+#if defined(__GNUC__) && __GNUC__ > 7
+#  define MOZ_PRAGMA(tokens) _Pragma(#tokens)
+#  define MOZ_UNROLL(factor) MOZ_PRAGMA(GCC unroll factor)
+#elif defined(__INTEL_COMPILER) || (defined(__clang__) && __clang_major__ > 3)
+#  define MOZ_PRAGMA(tokens) _Pragma(#tokens)
+#  define MOZ_UNROLL(factor) MOZ_PRAGMA(unroll factor)
+#else
+#  define MOZ_UNROLL(_)
+#endif
+
+namespace mozilla {
+
+template <class Arch>
+static bool is_aligned(const void* ptr) {
+  return (reinterpret_cast<uintptr_t>(ptr) &
+          ~(static_cast<uintptr_t>(Arch::alignment()) - 1)) ==
+         reinterpret_cast<uintptr_t>(ptr);
+};
+
+template <class Arch>
+void Engine<Arch>::AudioBufferAddWithScale(const float* aInput, float aScale,
+                                           float* aOutput, uint32_t aSize) {
+  if constexpr (Arch::requires_alignment()) {
+    if (aScale == 1.0f) {
+      while (!is_aligned<Arch>(aInput) || !is_aligned<Arch>(aOutput)) {
+        if (!aSize) return;
+        *aOutput += *aInput;
+        ++aOutput;
+        ++aInput;
+        --aSize;
+      }
+    } else {
+      while (!is_aligned<Arch>(aInput) || !is_aligned<Arch>(aOutput)) {
+        if (!aSize) return;
+        *aOutput += *aInput * aScale;
+        ++aOutput;
+        ++aInput;
+        --aSize;
+      }
+    }
+  }
+  MOZ_ASSERT(is_aligned<Arch>(aInput), "aInput is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aOutput), "aOutput is aligned");
+
+  xsimd::batch<float, Arch> vgain(aScale);
+
+  uint32_t aVSize = aSize & ~(xsimd::batch<float, Arch>::size - 1);
+  MOZ_UNROLL(4)
+  for (unsigned i = 0; i < aVSize; i += xsimd::batch<float, Arch>::size) {
+    auto vin1 = xsimd::batch<float, Arch>::load_aligned(&aInput[i]);
+    auto vin2 = xsimd::batch<float, Arch>::load_aligned(&aOutput[i]);
+    auto vout = xsimd::fma(vin1, vgain, vin2);
+    vout.store_aligned(&aOutput[i]);
+  }
+
+  for (unsigned i = aVSize; i < aSize; ++i) {
+    aOutput[i] += aInput[i] * aScale;
+  }
+}
+
+template <class Arch>
+void Engine<Arch>::AudioBlockCopyChannelWithScale(const float* aInput,
+                                                  float aScale,
+                                                  float* aOutput) {
+  MOZ_ASSERT(is_aligned<Arch>(aInput), "aInput is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aOutput), "aOutput is aligned");
+
+  MOZ_ASSERT((WEBAUDIO_BLOCK_SIZE % xsimd::batch<float, Arch>::size == 0),
+             "requires tail processing");
+
+  xsimd::batch<float, Arch> vgain = (aScale);
+
+  MOZ_UNROLL(4)
+  for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE;
+       i += xsimd::batch<float, Arch>::size) {
+    auto vin = xsimd::batch<float, Arch>::load_aligned(&aInput[i]);
+    auto vout = vin * vgain;
+    vout.store_aligned(&aOutput[i]);
+  }
+};
+
+template <class Arch>
+void Engine<Arch>::AudioBlockCopyChannelWithScale(
+    const float aInput[WEBAUDIO_BLOCK_SIZE],
+    const float aScale[WEBAUDIO_BLOCK_SIZE],
+    float aOutput[WEBAUDIO_BLOCK_SIZE]) {
+  MOZ_ASSERT(is_aligned<Arch>(aInput), "aInput is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aOutput), "aOutput is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aScale), "aScale is aligned");
+
+  MOZ_ASSERT((WEBAUDIO_BLOCK_SIZE % xsimd::batch<float, Arch>::size == 0),
+             "requires tail processing");
+
+  MOZ_UNROLL(4)
+  for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE;
+       i += xsimd::batch<float, Arch>::size) {
+    auto vscaled = xsimd::batch<float, Arch>::load_aligned(&aScale[i]);
+    auto vin = xsimd::batch<float, Arch>::load_aligned(&aInput[i]);
+    auto vout = vin * vscaled;
+    vout.store_aligned(&aOutput[i]);
+  }
+};
+
+template <class Arch>
+void Engine<Arch>::AudioBufferInPlaceScale(float* aBlock, float aScale,
+                                           uint32_t aSize) {
+  MOZ_ASSERT(is_aligned<Arch>(aBlock), "aBlock is aligned");
+
+  xsimd::batch<float, Arch> vgain(aScale);
+
+  uint32_t aVSize = aSize & ~(xsimd::batch<float, Arch>::size - 1);
+  MOZ_UNROLL(4)
+  for (unsigned i = 0; i < aVSize; i += xsimd::batch<float, Arch>::size) {
+    auto vin = xsimd::batch<float, Arch>::load_aligned(&aBlock[i]);
+    auto vout = vin * vgain;
+    vout.store_aligned(&aBlock[i]);
+  }
+  for (unsigned i = aVSize; i < aSize; ++i) aBlock[i] *= aScale;
+};
+
+template <class Arch>
+void Engine<Arch>::AudioBufferInPlaceScale(float* aBlock, float* aScale,
+                                           uint32_t aSize) {
+  MOZ_ASSERT(is_aligned<Arch>(aBlock), "aBlock is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aScale), "aScale is aligned");
+
+  uint32_t aVSize = aSize & ~(xsimd::batch<float, Arch>::size - 1);
+  MOZ_UNROLL(4)
+  for (unsigned i = 0; i < aVSize; i += xsimd::batch<float, Arch>::size) {
+    auto vin = xsimd::batch<float, Arch>::load_aligned(&aBlock[i]);
+    auto vgain = xsimd::batch<float, Arch>::load_aligned(&aScale[i]);
+    auto vout = vin * vgain;
+    vout.store_aligned(&aBlock[i]);
+  }
+  for (uint32_t i = aVSize; i < aSize; ++i) {
+    *aBlock++ *= *aScale++;
+  }
+};
+
+template <class Arch>
+void Engine<Arch>::AudioBlockPanStereoToStereo(
+    const float aInputL[WEBAUDIO_BLOCK_SIZE],
+    const float aInputR[WEBAUDIO_BLOCK_SIZE], float aGainL, float aGainR,
+    bool aIsOnTheLeft, float aOutputL[WEBAUDIO_BLOCK_SIZE],
+    float aOutputR[WEBAUDIO_BLOCK_SIZE]) {
+  MOZ_ASSERT(is_aligned<Arch>(aInputL), "aInputL is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aInputR), "aInputR is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aOutputL), "aOutputL is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aOutputR), "aOutputR is aligned");
+
+  MOZ_ASSERT((WEBAUDIO_BLOCK_SIZE % xsimd::batch<float, Arch>::size == 0),
+             "requires tail processing");
+
+  xsimd::batch<float, Arch> vgainl(aGainL);
+  xsimd::batch<float, Arch> vgainr(aGainR);
+
+  if (aIsOnTheLeft) {
+    MOZ_UNROLL(2)
+    for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE;
+         i += xsimd::batch<float, Arch>::size) {
+      auto vinl = xsimd::batch<float, Arch>::load_aligned(&aInputL[i]);
+      auto vinr = xsimd::batch<float, Arch>::load_aligned(&aInputR[i]);
+
+      /* left channel : aOutputL  = aInputL + aInputR * gainL */
+      auto vout = xsimd::fma(vinr, vgainl, vinl);
+      vout.store_aligned(&aOutputL[i]);
+
+      /* right channel : aOutputR = aInputR * gainR */
+      auto vscaled = vinr * vgainr;
+      vscaled.store_aligned(&aOutputR[i]);
+    }
+  } else {
+    MOZ_UNROLL(2)
+    for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE;
+         i += xsimd::batch<float, Arch>::size) {
+      auto vinl = xsimd::batch<float, Arch>::load_aligned(&aInputL[i]);
+      auto vinr = xsimd::batch<float, Arch>::load_aligned(&aInputR[i]);
+
+      /* left channel : aInputL * gainL */
+      auto vscaled = vinl * vgainl;
+      vscaled.store_aligned(&aOutputL[i]);
+
+      /* right channel: aOutputR = aInputR + aInputL * gainR */
+      auto vout = xsimd::fma(vinl, vgainr, vinr);
+      vout.store_aligned(&aOutputR[i]);
+    }
+  }
+};
+
+template <class Arch>
+void Engine<Arch>::BufferComplexMultiply(const float* aInput,
+                                         const float* aScale, float* aOutput,
+                                         uint32_t aSize) {
+  MOZ_ASSERT(is_aligned<Arch>(aInput), "aInput is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aOutput), "aOutput is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aScale), "aScale is aligned");
+  MOZ_ASSERT((aSize % xsimd::batch<float, Arch>::size == 0),
+             "requires tail processing");
+
+  MOZ_UNROLL(2)
+  for (unsigned i = 0; i < aSize * 2;
+       i += 2 * xsimd::batch<std::complex<float>, Arch>::size) {
+    auto in1 = xsimd::batch<std::complex<float>, Arch>::load_aligned(
+        reinterpret_cast<const std::complex<float>*>(&aInput[i]));
+    auto in2 = xsimd::batch<std::complex<float>, Arch>::load_aligned(
+        reinterpret_cast<const std::complex<float>*>(&aScale[i]));
+    auto out = in1 * in2;
+    out.store_aligned(reinterpret_cast<std::complex<float>*>(&aOutput[i]));
+  }
+};
+
+template <class Arch>
+float Engine<Arch>::AudioBufferSumOfSquares(const float* aInput,
+                                            uint32_t aLength) {
+  float sum = 0.f;
+
+  if constexpr (Arch::requires_alignment()) {
+    while (!is_aligned<Arch>(aInput)) {
+      if (!aLength) {
+        return sum;
+      }
+      sum += *aInput * *aInput;
+      ++aInput;
+      --aLength;
+    }
+  }
+
+  MOZ_ASSERT(is_aligned<Arch>(aInput), "aInput is aligned");
+
+  constexpr uint32_t unroll_factor = 4;
+  xsimd::batch<float, Arch> accs[unroll_factor] = {0.f, 0.f, 0.f, 0.f};
+
+  uint32_t vLength =
+      aLength & ~(unroll_factor * xsimd::batch<float, Arch>::size - 1);
+
+  for (uint32_t i = 0; i < vLength;
+       i += unroll_factor * xsimd::batch<float, Arch>::size) {
+    MOZ_UNROLL(4)
+    for (uint32_t j = 0; j < unroll_factor; ++j) {
+      auto in = xsimd::batch<float, Arch>::load_aligned(
+          &aInput[i + xsimd::batch<float, Arch>::size * j]);
+      accs[j] = xsimd::fma(in, in, accs[j]);
+    }
+  }
+
+  sum += reduce_add((accs[0] + accs[1]) + (accs[2] + accs[3]));
+  for (uint32_t i = vLength; i < aLength; ++i) sum += aInput[i] * aInput[i];
+  return sum;
+};
+
+template <class Arch>
+void Engine<Arch>::NaNToZeroInPlace(float* aSamples, size_t aCount) {
+  if constexpr (Arch::requires_alignment()) {
+    while (!is_aligned<Arch>(aSamples)) {
+      if (!aCount) {
+        return;
+      }
+      if (*aSamples != *aSamples) {
+        *aSamples = 0.0;
+      }
+      ++aSamples;
+      --aCount;
+    }
+  }
+
+  MOZ_ASSERT(is_aligned<Arch>(aSamples), "aSamples is aligned");
+
+  uint32_t vCount = aCount & ~(xsimd::batch<float, Arch>::size - 1);
+
+  MOZ_UNROLL(4)
+  for (uint32_t i = 0; i < vCount; i += xsimd::batch<float, Arch>::size) {
+    auto vin = xsimd::batch<float, Arch>::load_aligned(&aSamples[i]);
+    auto vout =
+        xsimd::select(xsimd::isnan(vin), xsimd::batch<float, Arch>(0.f), vin);
+    vout.store_aligned(&aSamples[i]);
+  }
+
+  for (uint32_t i = vCount; i < aCount; i++) {
+    if (aSamples[i] != aSamples[i]) {
+      aSamples[i] = 0.0;
+    }
+  }
+};
+
+template <class Arch>
+void Engine<Arch>::AudioBlockPanStereoToStereo(
+    const float aInputL[WEBAUDIO_BLOCK_SIZE],
+    const float aInputR[WEBAUDIO_BLOCK_SIZE],
+    const float aGainL[WEBAUDIO_BLOCK_SIZE],
+    const float aGainR[WEBAUDIO_BLOCK_SIZE],
+    const bool aIsOnTheLeft[WEBAUDIO_BLOCK_SIZE],
+    float aOutputL[WEBAUDIO_BLOCK_SIZE], float aOutputR[WEBAUDIO_BLOCK_SIZE]) {
+  MOZ_ASSERT(is_aligned<Arch>(aInputL), "aInputL is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aInputR), "aInputR is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aGainL), "aGainL is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aGainR), "aGainR is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aIsOnTheLeft), "aIsOnTheLeft is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aOutputL), "aOutputL is aligned");
+  MOZ_ASSERT(is_aligned<Arch>(aOutputR), "aOutputR is aligned");
+
+  MOZ_ASSERT((WEBAUDIO_BLOCK_SIZE % xsimd::batch<float, Arch>::size == 0),
+             "requires tail processing");
+
+  MOZ_UNROLL(2)
+  for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE;
+       i += xsimd::batch<float, Arch>::size) {
+    auto mask = xsimd::batch_bool<float, Arch>::load_aligned(&aIsOnTheLeft[i]);
+
+    auto inputL = xsimd::batch<float, Arch>::load_aligned(&aInputL[i]);
+    auto inputR = xsimd::batch<float, Arch>::load_aligned(&aInputR[i]);
+    auto gainL = xsimd::batch<float, Arch>::load_aligned(&aGainL[i]);
+    auto gainR = xsimd::batch<float, Arch>::load_aligned(&aGainR[i]);
+
+    auto outL_true = xsimd::fma(inputR, gainL, inputL);
+    auto outR_true = inputR * gainR;
+
+    auto outL_false = inputL * gainL;
+    auto outR_false = xsimd::fma(inputL, gainR, inputR);
+
+    auto outL = xsimd::select(mask, outL_true, outL_false);
+    auto outR = xsimd::select(mask, outR_true, outR_false);
+
+    outL.store_aligned(&aOutputL[i]);
+    outR.store_aligned(&aOutputR[i]);
+  }
+}
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/AudioNodeEngineNEON.cpp b/dom/media/webaudio/AudioNodeEngineNEON.cpp
new file mode 100644
index 0000000000..93f5c85285
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngineNEON.cpp
@@ -0,0 +1,9 @@
+/* -*- mode: c++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* this source code form is subject to the terms of the mozilla public
+ * license, v. 2.0. if a copy of the mpl was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioNodeEngineGenericImpl.h"
+namespace mozilla {
+template struct Engine<xsimd::neon>;
+}  // namespace mozilla
diff --git a/dom/media/webaudio/AudioNodeEngineSSE2.cpp b/dom/media/webaudio/AudioNodeEngineSSE2.cpp
new file mode 100644
index 0000000000..558cd866fd
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngineSSE2.cpp
@@ -0,0 +1,10 @@
+/* -*- mode: c++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* this source code form is subject to the terms of the mozilla public
+ * license, v. 2.0. if a copy of the mpl was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioNodeEngineGenericImpl.h"
+
+namespace mozilla {
+template struct Engine<xsimd::sse2>;
+}  // namespace mozilla
diff --git a/dom/media/webaudio/AudioNodeEngineSSE4_2_FMA3.cpp b/dom/media/webaudio/AudioNodeEngineSSE4_2_FMA3.cpp
new file mode 100644
index 0000000000..ce0134ee4b
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeEngineSSE4_2_FMA3.cpp
@@ -0,0 +1,10 @@
+/* -*- mode: c++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* this source code form is subject to the terms of the mozilla public
+ * license, v. 2.0. if a copy of the mpl was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioNodeEngineGenericImpl.h"
+
+namespace mozilla {
+template struct Engine<xsimd::fma3<xsimd::sse4_2>>;
+}  // namespace mozilla
diff --git a/dom/media/webaudio/AudioNodeExternalInputTrack.cpp b/dom/media/webaudio/AudioNodeExternalInputTrack.cpp
new file mode 100644
index 0000000000..8867a2dbf2
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeExternalInputTrack.cpp
@@ -0,0 +1,225 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AlignedTArray.h"
+#include "AlignmentUtils.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeExternalInputTrack.h"
+#include "AudioChannelFormat.h"
+#include "mozilla/dom/MediaStreamAudioSourceNode.h"
+
+using namespace mozilla::dom;
+
+namespace mozilla {
+
+AudioNodeExternalInputTrack::AudioNodeExternalInputTrack(
+    AudioNodeEngine* aEngine, TrackRate aSampleRate)
+    : AudioNodeTrack(aEngine, NO_TRACK_FLAGS, aSampleRate) {
+  MOZ_COUNT_CTOR(AudioNodeExternalInputTrack);
+}
+
+AudioNodeExternalInputTrack::~AudioNodeExternalInputTrack() {
+  MOZ_COUNT_DTOR(AudioNodeExternalInputTrack);
+}
+
+/* static */
+already_AddRefed<AudioNodeExternalInputTrack>
+AudioNodeExternalInputTrack::Create(MediaTrackGraph* aGraph,
+                                    AudioNodeEngine* aEngine) {
+  AudioContext* ctx = aEngine->NodeMainThread()->Context();
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(aGraph == ctx->Graph());
+
+  RefPtr<AudioNodeExternalInputTrack> track =
+      new AudioNodeExternalInputTrack(aEngine, aGraph->GraphRate());
+  track->mSuspendedCount += ctx->ShouldSuspendNewTrack();
+  aGraph->AddTrack(track);
+  return track.forget();
+}
+
+/**
+ * Copies the data in aInput to aOffsetInBlock within aBlock.
+ * aBlock must have been allocated with AllocateInputBlock and have a channel
+ * count that's a superset of the channels in aInput.
+ */
+template <typename T>
+static void CopyChunkToBlock(AudioChunk& aInput, AudioBlock* aBlock,
+                             uint32_t aOffsetInBlock) {
+  uint32_t blockChannels = aBlock->ChannelCount();
+  AutoTArray<const T*, 2> channels;
+  if (aInput.IsNull()) {
+    channels.SetLength(blockChannels);
+    PodZero(channels.Elements(), blockChannels);
+  } else {
+    Span inputChannels = aInput.ChannelData<T>();
+    channels.SetLength(inputChannels.Length());
+    PodCopy(channels.Elements(), inputChannels.Elements(), channels.Length());
+    if (channels.Length() != blockChannels) {
+      // We only need to upmix here because aBlock's channel count has been
+      // chosen to be a superset of the channel count of every chunk.
+      AudioChannelsUpMix(&channels, blockChannels, static_cast<T*>(nullptr));
+    }
+  }
+
+  for (uint32_t c = 0; c < blockChannels; ++c) {
+    float* outputData = aBlock->ChannelFloatsForWrite(c) + aOffsetInBlock;
+    if (channels[c]) {
+      ConvertAudioSamplesWithScale(channels[c], outputData,
+                                   aInput.GetDuration(), aInput.mVolume);
+    } else {
+      PodZero(outputData, aInput.GetDuration());
+    }
+  }
+}
+
+/**
+ * Converts the data in aSegment to a single chunk aBlock. aSegment must have
+ * duration WEBAUDIO_BLOCK_SIZE. aFallbackChannelCount is a superset of the
+ * channels in every chunk of aSegment. aBlock must be float format or null.
+ */
+static void ConvertSegmentToAudioBlock(AudioSegment* aSegment,
+                                       AudioBlock* aBlock,
+                                       int32_t aFallbackChannelCount) {
+  NS_ASSERTION(aSegment->GetDuration() == WEBAUDIO_BLOCK_SIZE,
+               "Bad segment duration");
+
+  {
+    AudioSegment::ChunkIterator ci(*aSegment);
+    NS_ASSERTION(!ci.IsEnded(), "Should be at least one chunk!");
+    if (ci->GetDuration() == WEBAUDIO_BLOCK_SIZE &&
+        (ci->IsNull() || ci->mBufferFormat == AUDIO_FORMAT_FLOAT32)) {
+      bool aligned = true;
+      for (size_t i = 0; i < ci->mChannelData.Length(); ++i) {
+        if (!IS_ALIGNED16(ci->mChannelData[i])) {
+          aligned = false;
+          break;
+        }
+      }
+
+      // Return this chunk directly to avoid copying data.
+      if (aligned) {
+        *aBlock = *ci;
+        return;
+      }
+    }
+  }
+
+  aBlock->AllocateChannels(aFallbackChannelCount);
+
+  uint32_t duration = 0;
+  for (AudioSegment::ChunkIterator ci(*aSegment); !ci.IsEnded(); ci.Next()) {
+    switch (ci->mBufferFormat) {
+      case AUDIO_FORMAT_S16: {
+        CopyChunkToBlock<int16_t>(*ci, aBlock, duration);
+        break;
+      }
+      case AUDIO_FORMAT_FLOAT32: {
+        CopyChunkToBlock<float>(*ci, aBlock, duration);
+        break;
+      }
+      case AUDIO_FORMAT_SILENCE: {
+        // The actual type of the sample does not matter here, but we still need
+        // to send some audio to the graph.
+        CopyChunkToBlock<float>(*ci, aBlock, duration);
+        break;
+      }
+    }
+    duration += ci->GetDuration();
+  }
+}
+
+void AudioNodeExternalInputTrack::ProcessInput(GraphTime aFrom, GraphTime aTo,
+                                               uint32_t aFlags) {
+  // According to spec, number of outputs is always 1.
+  MOZ_ASSERT(mLastChunks.Length() == 1);
+
+  // GC stuff can result in our input track being destroyed before this track.
+  // Handle that.
+  if (!IsEnabled() || mInputs.IsEmpty() || mPassThrough) {
+    mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  MOZ_ASSERT(mInputs.Length() == 1);
+
+  MediaTrack* source = mInputs[0]->GetSource();
+  AutoTArray<AudioSegment, 1> audioSegments;
+  uint32_t inputChannels = 0;
+
+  MOZ_ASSERT(source->GetData()->GetType() == MediaSegment::AUDIO,
+             "AudioNodeExternalInputTrack shouldn't have a video input");
+
+  const AudioSegment& inputSegment =
+      *mInputs[0]->GetSource()->GetData<AudioSegment>();
+  if (!inputSegment.IsNull()) {
+    AudioSegment& segment = *audioSegments.AppendElement();
+    GraphTime next;
+    for (GraphTime t = aFrom; t < aTo; t = next) {
+      MediaInputPort::InputInterval interval =
+          MediaInputPort::GetNextInputInterval(mInputs[0], t);
+      interval.mEnd = std::min(interval.mEnd, aTo);
+      if (interval.mStart >= interval.mEnd) {
+        break;
+      }
+      next = interval.mEnd;
+
+      // We know this track does not block during the processing interval ---
+      // we're not finished, we don't underrun, and we're not suspended.
+      TrackTime outputStart = GraphTimeToTrackTime(interval.mStart);
+      TrackTime outputEnd = GraphTimeToTrackTime(interval.mEnd);
+      TrackTime ticks = outputEnd - outputStart;
+
+      if (interval.mInputIsBlocked) {
+        segment.AppendNullData(ticks);
+      } else {
+        // The input track is not blocked in this interval, so no need to call
+        // GraphTimeToTrackTimeWithBlocking.
+        TrackTime inputStart =
+            std::min(inputSegment.GetDuration(),
+                     source->GraphTimeToTrackTime(interval.mStart));
+        TrackTime inputEnd =
+            std::min(inputSegment.GetDuration(),
+                     source->GraphTimeToTrackTime(interval.mEnd));
+
+        segment.AppendSlice(inputSegment, inputStart, inputEnd);
+        // Pad if we're looking past the end of the track
+        segment.AppendNullData(ticks - (inputEnd - inputStart));
+      }
+    }
+
+    for (AudioSegment::ChunkIterator iter(segment); !iter.IsEnded();
+         iter.Next()) {
+      inputChannels =
+          GetAudioChannelsSuperset(inputChannels, iter->ChannelCount());
+    }
+  }
+
+  uint32_t accumulateIndex = 0;
+  if (inputChannels) {
+    DownmixBufferType downmixBuffer;
+    ASSERT_ALIGNED16(downmixBuffer.Elements());
+    for (auto& audioSegment : audioSegments) {
+      AudioBlock tmpChunk;
+      ConvertSegmentToAudioBlock(&audioSegment, &tmpChunk, inputChannels);
+      if (!tmpChunk.IsNull()) {
+        if (accumulateIndex == 0) {
+          mLastChunks[0].AllocateChannels(inputChannels);
+        }
+        AccumulateInputChunk(accumulateIndex, tmpChunk, &mLastChunks[0],
+                             &downmixBuffer);
+        accumulateIndex++;
+      }
+    }
+  }
+  if (accumulateIndex == 0) {
+    mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
+  }
+}
+
+bool AudioNodeExternalInputTrack::IsEnabled() {
+  return ((MediaStreamAudioSourceNodeEngine*)Engine())->IsEnabled();
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webaudio/AudioNodeExternalInputTrack.h b/dom/media/webaudio/AudioNodeExternalInputTrack.h
new file mode 100644
index 0000000000..f273fee696
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeExternalInputTrack.h
@@ -0,0 +1,46 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_AUDIONODEEXTERNALINPUTTRACK_H_
+#define MOZILLA_AUDIONODEEXTERNALINPUTTRACK_H_
+
+#include "MediaTrackGraph.h"
+#include "AudioNodeTrack.h"
+#include "mozilla/Atomics.h"
+
+namespace mozilla {
+
+class AbstractThread;
+
+/**
+ * This is a MediaTrack implementation that acts for a Web Audio node but
+ * unlike other AudioNodeTracks, supports any kind of MediaTrack as an
+ * input --- handling any number of audio tracks and handling blocking of
+ * the input MediaTrack.
+ */
+class AudioNodeExternalInputTrack final : public AudioNodeTrack {
+ public:
+  static already_AddRefed<AudioNodeExternalInputTrack> Create(
+      MediaTrackGraph* aGraph, AudioNodeEngine* aEngine);
+
+ protected:
+  AudioNodeExternalInputTrack(AudioNodeEngine* aEngine, TrackRate aSampleRate);
+  ~AudioNodeExternalInputTrack();
+
+ public:
+  void ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags) override;
+
+ private:
+  /**
+   * Determines if this is enabled or not.  Disabled nodes produce silence.
+   * This node becomes disabled if the document principal does not subsume the
+   * DOMMediaStream principal.
+   */
+  bool IsEnabled();
+};
+
+}  // namespace mozilla
+
+#endif /* MOZILLA_AUDIONODEEXTERNALINPUTTRACK_H_ */
diff --git a/dom/media/webaudio/AudioNodeTrack.cpp b/dom/media/webaudio/AudioNodeTrack.cpp
new file mode 100644
index 0000000000..ea411a16ac
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeTrack.cpp
@@ -0,0 +1,594 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#include "AudioNodeTrack.h"
+
+#include "MediaTrackGraph.h"
+#include "MediaTrackListener.h"
+#include "AudioNodeEngine.h"
+#include "ThreeDPoint.h"
+#include "Tracing.h"
+#include "AudioChannelFormat.h"
+#include "AudioParamTimeline.h"
+#include "AudioContext.h"
+#include "nsMathUtils.h"
+#include "AlignmentUtils.h"
+#include "blink/Reverb.h"
+
+using namespace mozilla::dom;
+
+namespace mozilla {
+
+/**
+ * An AudioNodeTrack produces a single audio track with ID
+ * AUDIO_TRACK. This track has rate AudioContext::sIdealAudioRate
+ * for regular audio contexts, and the rate requested by the web content
+ * for offline audio contexts.
+ * Each chunk in the track is a single block of WEBAUDIO_BLOCK_SIZE samples.
+ * Note: This must be a different value than MEDIA_STREAM_DEST_TRACK_ID
+ */
+
+AudioNodeTrack::AudioNodeTrack(AudioNodeEngine* aEngine, Flags aFlags,
+                               TrackRate aSampleRate)
+    : ProcessedMediaTrack(
+          aSampleRate, MediaSegment::AUDIO,
+          (aFlags & EXTERNAL_OUTPUT) ? new AudioSegment() : nullptr),
+      mEngine(aEngine),
+      mFlags(aFlags),
+      mNumberOfInputChannels(2),
+      mIsActive(aEngine->IsActive()),
+      mMarkAsEndedAfterThisBlock(false),
+      mAudioParamTrack(false),
+      mPassThrough(false) {
+  MOZ_ASSERT(NS_IsMainThread());
+  mSuspendedCount = !(mIsActive || mFlags & EXTERNAL_OUTPUT);
+  mChannelCountMode = ChannelCountMode::Max;
+  mChannelInterpretation = ChannelInterpretation::Speakers;
+  mLastChunks.SetLength(std::max(uint16_t(1), mEngine->OutputCount()));
+  MOZ_COUNT_CTOR(AudioNodeTrack);
+}
+
+AudioNodeTrack::~AudioNodeTrack() {
+  MOZ_ASSERT(mActiveInputCount == 0);
+  MOZ_COUNT_DTOR(AudioNodeTrack);
+}
+
+void AudioNodeTrack::OnGraphThreadDone() { mEngine->OnGraphThreadDone(); }
+
+void AudioNodeTrack::DestroyImpl() {
+  // These are graph thread objects, so clean up on graph thread.
+  mInputChunks.Clear();
+  mLastChunks.Clear();
+
+  ProcessedMediaTrack::DestroyImpl();
+}
+
+/* static */
+already_AddRefed<AudioNodeTrack> AudioNodeTrack::Create(
+    AudioContext* aCtx, AudioNodeEngine* aEngine, Flags aFlags,
+    MediaTrackGraph* aGraph) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_RELEASE_ASSERT(aGraph);
+
+  // MediaRecorders use an AudioNodeTrack, but no AudioNode
+  AudioNode* node = aEngine->NodeMainThread();
+
+  RefPtr<AudioNodeTrack> track =
+      new AudioNodeTrack(aEngine, aFlags, aGraph->GraphRate());
+  if (node) {
+    track->SetChannelMixingParametersImpl(node->ChannelCount(),
+                                          node->ChannelCountModeValue(),
+                                          node->ChannelInterpretationValue());
+  }
+  // All realtime tracks are initially suspended.
+  // ApplyAudioContextOperation() is used to start tracks so that a new track
+  // will not be started before the existing tracks, which may be awaiting an
+  // AudioCallbackDriver to resume.
+  bool isRealtime = !aCtx->IsOffline();
+  track->mSuspendedCount += isRealtime;
+  aGraph->AddTrack(track);
+  if (isRealtime && !aCtx->ShouldSuspendNewTrack()) {
+    nsTArray<RefPtr<mozilla::MediaTrack>> tracks;
+    tracks.AppendElement(track);
+    aGraph->ApplyAudioContextOperation(aCtx->DestinationTrack(),
+                                       std::move(tracks),
+                                       AudioContextOperation::Resume);
+  }
+  return track.forget();
+}
+
+size_t AudioNodeTrack::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = 0;
+
+  // Not reported:
+  // - mEngine
+
+  amount += ProcessedMediaTrack::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mLastChunks.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < mLastChunks.Length(); i++) {
+    // NB: This is currently unshared only as there are instances of
+    //     double reporting in DMD otherwise.
+    amount += mLastChunks[i].SizeOfExcludingThisIfUnshared(aMallocSizeOf);
+  }
+
+  return amount;
+}
+
+size_t AudioNodeTrack::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+void AudioNodeTrack::SizeOfAudioNodesIncludingThis(
+    MallocSizeOf aMallocSizeOf, AudioNodeSizes& aUsage) const {
+  // Explicitly separate out the track memory.
+  aUsage.mTrack = SizeOfIncludingThis(aMallocSizeOf);
+
+  if (mEngine) {
+    // This will fill out the rest of |aUsage|.
+    mEngine->SizeOfIncludingThis(aMallocSizeOf, aUsage);
+  }
+}
+
+void AudioNodeTrack::SetTrackTimeParameter(uint32_t aIndex,
+                                           AudioContext* aContext,
+                                           double aTrackTime) {
+  QueueControlMessageWithNoShutdown(
+      [self = RefPtr{this}, this, aIndex,
+       relativeToTrack = RefPtr{aContext->DestinationTrack()}, aTrackTime] {
+        TRACE("AudioNodeTrack::SetTrackTimeParameterImpl");
+        SetTrackTimeParameterImpl(aIndex, relativeToTrack, aTrackTime);
+      });
+}
+
+void AudioNodeTrack::SetTrackTimeParameterImpl(uint32_t aIndex,
+                                               MediaTrack* aRelativeToTrack,
+                                               double aTrackTime) {
+  TrackTime ticks = aRelativeToTrack->SecondsToNearestTrackTime(aTrackTime);
+  mEngine->SetTrackTimeParameter(aIndex, ticks);
+}
+
+void AudioNodeTrack::SetDoubleParameter(uint32_t aIndex, double aValue) {
+  QueueControlMessageWithNoShutdown(
+      [self = RefPtr{this}, this, aIndex, aValue] {
+        TRACE("AudioNodeTrack::SetDoubleParameter");
+        Engine()->SetDoubleParameter(aIndex, aValue);
+      });
+}
+
+void AudioNodeTrack::SetInt32Parameter(uint32_t aIndex, int32_t aValue) {
+  QueueControlMessageWithNoShutdown(
+      [self = RefPtr{this}, this, aIndex, aValue] {
+        TRACE("AudioNodeTrack::SetInt32Parameter");
+        Engine()->SetInt32Parameter(aIndex, aValue);
+      });
+}
+
+void AudioNodeTrack::SendTimelineEvent(uint32_t aIndex,
+                                       const AudioParamEvent& aEvent) {
+  QueueControlMessageWithNoShutdown(
+      [self = RefPtr{this}, this, aIndex, event = aEvent]() mutable {
+        TRACE("AudioNodeTrack::RecvTimelineEvent");
+        Engine()->RecvTimelineEvent(aIndex, event);
+      });
+}
+
+void AudioNodeTrack::SetBuffer(AudioChunk&& aBuffer) {
+  QueueControlMessageWithNoShutdown(
+      [self = RefPtr{this}, this, buffer = std::move(aBuffer)]() mutable {
+        TRACE("AudioNodeTrack::SetBuffer");
+        Engine()->SetBuffer(std::move(buffer));
+      });
+}
+
+void AudioNodeTrack::SetReverb(WebCore::Reverb* aReverb,
+                               uint32_t aImpulseChannelCount) {
+  QueueControlMessageWithNoShutdown([self = RefPtr{this}, this,
+                                     reverb = WrapUnique(aReverb),
+                                     aImpulseChannelCount]() mutable {
+    TRACE("AudioNodeTrack::SetReverb");
+    Engine()->SetReverb(reverb.release(), aImpulseChannelCount);
+  });
+}
+
+void AudioNodeTrack::SetRawArrayData(nsTArray<float>&& aData) {
+  QueueControlMessageWithNoShutdown(
+      [self = RefPtr{this}, this, data = std::move(aData)]() mutable {
+        TRACE("AudioNodeTrack::SetRawArrayData");
+        Engine()->SetRawArrayData(std::move(data));
+      });
+}
+
+void AudioNodeTrack::SetChannelMixingParameters(
+    uint32_t aNumberOfChannels, ChannelCountMode aChannelCountMode,
+    ChannelInterpretation aChannelInterpretation) {
+  QueueControlMessageWithNoShutdown([self = RefPtr{this}, this,
+                                     aNumberOfChannels, aChannelCountMode,
+                                     aChannelInterpretation] {
+    TRACE("AudioNodeTrack::SetChannelMixingParameters");
+    SetChannelMixingParametersImpl(aNumberOfChannels, aChannelCountMode,
+                                   aChannelInterpretation);
+  });
+}
+
+void AudioNodeTrack::SetPassThrough(bool aPassThrough) {
+  QueueControlMessageWithNoShutdown([self = RefPtr{this}, this, aPassThrough] {
+    TRACE("AudioNodeTrack::SetPassThrough");
+    mPassThrough = aPassThrough;
+  });
+}
+
+void AudioNodeTrack::SendRunnable(already_AddRefed<nsIRunnable> aRunnable) {
+  QueueControlMessageWithNoShutdown([runnable = nsCOMPtr{aRunnable}] {
+    TRACE("AudioNodeTrack::SendRunnable");
+    runnable->Run();
+  });
+}
+
+void AudioNodeTrack::SetChannelMixingParametersImpl(
+    uint32_t aNumberOfChannels, ChannelCountMode aChannelCountMode,
+    ChannelInterpretation aChannelInterpretation) {
+  mNumberOfInputChannels = aNumberOfChannels;
+  mChannelCountMode = aChannelCountMode;
+  mChannelInterpretation = aChannelInterpretation;
+}
+
+uint32_t AudioNodeTrack::ComputedNumberOfChannels(uint32_t aInputChannelCount) {
+  switch (mChannelCountMode) {
+    case ChannelCountMode::Explicit:
+      // Disregard the channel count we've calculated from inputs, and just use
+      // mNumberOfInputChannels.
+      return mNumberOfInputChannels;
+    case ChannelCountMode::Clamped_max:
+      // Clamp the computed output channel count to mNumberOfInputChannels.
+      return std::min(aInputChannelCount, mNumberOfInputChannels);
+    default:
+    case ChannelCountMode::Max:
+      // Nothing to do here, just shut up the compiler warning.
+      return aInputChannelCount;
+  }
+}
+
+uint32_t AudioNodeTrack::NumberOfChannels() const {
+  AssertOnGraphThread();
+
+  return mNumberOfInputChannels;
+}
+
+void AudioNodeTrack::AdvanceAndResume(TrackTime aAdvance) {
+  mMainThreadCurrentTime += aAdvance;
+  QueueControlMessageWithNoShutdown([self = RefPtr{this}, this, aAdvance] {
+    TRACE("AudioNodeTrack::AdvanceAndResumeMessage");
+    mStartTime -= aAdvance;
+    mSegment->AppendNullData(aAdvance);
+    DecrementSuspendCount();
+  });
+}
+
+void AudioNodeTrack::ObtainInputBlock(AudioBlock& aTmpChunk,
+                                      uint32_t aPortIndex) {
+  uint32_t inputCount = mInputs.Length();
+  uint32_t outputChannelCount = 1;
+  AutoTArray<const AudioBlock*, 250> inputChunks;
+  for (uint32_t i = 0; i < inputCount; ++i) {
+    if (aPortIndex != mInputs[i]->InputNumber()) {
+      // This input is connected to a different port
+      continue;
+    }
+    MediaTrack* t = mInputs[i]->GetSource();
+    AudioNodeTrack* a = static_cast<AudioNodeTrack*>(t);
+    MOZ_ASSERT(a == t->AsAudioNodeTrack());
+    if (a->IsAudioParamTrack()) {
+      continue;
+    }
+
+    const AudioBlock* chunk = &a->mLastChunks[mInputs[i]->OutputNumber()];
+    MOZ_ASSERT(chunk);
+    if (chunk->IsNull() || chunk->mChannelData.IsEmpty()) {
+      continue;
+    }
+
+    inputChunks.AppendElement(chunk);
+    outputChannelCount =
+        GetAudioChannelsSuperset(outputChannelCount, chunk->ChannelCount());
+  }
+
+  outputChannelCount = ComputedNumberOfChannels(outputChannelCount);
+
+  uint32_t inputChunkCount = inputChunks.Length();
+  if (inputChunkCount == 0 ||
+      (inputChunkCount == 1 && inputChunks[0]->ChannelCount() == 0)) {
+    aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  if (inputChunkCount == 1 &&
+      inputChunks[0]->ChannelCount() == outputChannelCount) {
+    aTmpChunk = *inputChunks[0];
+    return;
+  }
+
+  if (outputChannelCount == 0) {
+    aTmpChunk.SetNull(WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  aTmpChunk.AllocateChannels(outputChannelCount);
+  DownmixBufferType downmixBuffer;
+  ASSERT_ALIGNED16(downmixBuffer.Elements());
+
+  for (uint32_t i = 0; i < inputChunkCount; ++i) {
+    AccumulateInputChunk(i, *inputChunks[i], &aTmpChunk, &downmixBuffer);
+  }
+}
+
+void AudioNodeTrack::AccumulateInputChunk(uint32_t aInputIndex,
+                                          const AudioBlock& aChunk,
+                                          AudioBlock* aBlock,
+                                          DownmixBufferType* aDownmixBuffer) {
+  AutoTArray<const float*, GUESS_AUDIO_CHANNELS> channels;
+  UpMixDownMixChunk(&aChunk, aBlock->ChannelCount(), channels, *aDownmixBuffer);
+
+  for (uint32_t c = 0; c < channels.Length(); ++c) {
+    const float* inputData = static_cast<const float*>(channels[c]);
+    float* outputData = aBlock->ChannelFloatsForWrite(c);
+    if (inputData) {
+      if (aInputIndex == 0) {
+        AudioBlockCopyChannelWithScale(inputData, aChunk.mVolume, outputData);
+      } else {
+        AudioBlockAddChannelWithScale(inputData, aChunk.mVolume, outputData);
+      }
+    } else {
+      if (aInputIndex == 0) {
+        PodZero(outputData, WEBAUDIO_BLOCK_SIZE);
+      }
+    }
+  }
+}
+
+void AudioNodeTrack::UpMixDownMixChunk(const AudioBlock* aChunk,
+                                       uint32_t aOutputChannelCount,
+                                       nsTArray<const float*>& aOutputChannels,
+                                       DownmixBufferType& aDownmixBuffer) {
+  for (uint32_t i = 0; i < aChunk->ChannelCount(); i++) {
+    aOutputChannels.AppendElement(
+        static_cast<const float*>(aChunk->mChannelData[i]));
+  }
+  if (aOutputChannels.Length() < aOutputChannelCount) {
+    if (mChannelInterpretation == ChannelInterpretation::Speakers) {
+      AudioChannelsUpMix<float>(&aOutputChannels, aOutputChannelCount, nullptr);
+      NS_ASSERTION(aOutputChannelCount == aOutputChannels.Length(),
+                   "We called GetAudioChannelsSuperset to avoid this");
+    } else {
+      // Fill up the remaining aOutputChannels by zeros
+      for (uint32_t j = aOutputChannels.Length(); j < aOutputChannelCount;
+           ++j) {
+        aOutputChannels.AppendElement(nullptr);
+      }
+    }
+  } else if (aOutputChannels.Length() > aOutputChannelCount) {
+    if (mChannelInterpretation == ChannelInterpretation::Speakers) {
+      AutoTArray<float*, GUESS_AUDIO_CHANNELS> outputChannels;
+      outputChannels.SetLength(aOutputChannelCount);
+      aDownmixBuffer.SetLength(aOutputChannelCount * WEBAUDIO_BLOCK_SIZE);
+      for (uint32_t j = 0; j < aOutputChannelCount; ++j) {
+        outputChannels[j] = &aDownmixBuffer[j * WEBAUDIO_BLOCK_SIZE];
+      }
+
+      AudioChannelsDownMix<float, float>(aOutputChannels, outputChannels,
+                                         WEBAUDIO_BLOCK_SIZE);
+
+      aOutputChannels.SetLength(aOutputChannelCount);
+      for (uint32_t j = 0; j < aOutputChannels.Length(); ++j) {
+        aOutputChannels[j] = outputChannels[j];
+      }
+    } else {
+      // Drop the remaining aOutputChannels
+      aOutputChannels.RemoveLastElements(aOutputChannels.Length() -
+                                         aOutputChannelCount);
+    }
+  }
+}
+
+// The MediaTrackGraph guarantees that this is actually one block, for
+// AudioNodeTracks.
+void AudioNodeTrack::ProcessInput(GraphTime aFrom, GraphTime aTo,
+                                  uint32_t aFlags) {
+  uint16_t outputCount = mLastChunks.Length();
+  MOZ_ASSERT(outputCount == std::max(uint16_t(1), mEngine->OutputCount()));
+
+  if (!mIsActive) {
+    // mLastChunks are already null.
+#ifdef DEBUG
+    for (const auto& chunk : mLastChunks) {
+      MOZ_ASSERT(chunk.IsNull());
+    }
+#endif
+  } else if (InMutedCycle()) {
+    mInputChunks.Clear();
+    for (uint16_t i = 0; i < outputCount; ++i) {
+      mLastChunks[i].SetNull(WEBAUDIO_BLOCK_SIZE);
+    }
+  } else {
+    // We need to generate at least one input
+    uint16_t maxInputs = std::max(uint16_t(1), mEngine->InputCount());
+    mInputChunks.SetLength(maxInputs);
+    for (uint16_t i = 0; i < maxInputs; ++i) {
+      ObtainInputBlock(mInputChunks[i], i);
+    }
+    bool finished = false;
+    if (mPassThrough) {
+      MOZ_ASSERT(outputCount == 1,
+                 "For now, we only support nodes that have one output port");
+      mLastChunks[0] = mInputChunks[0];
+    } else {
+      if (maxInputs <= 1 && outputCount <= 1) {
+        mEngine->ProcessBlock(this, aFrom, mInputChunks[0], &mLastChunks[0],
+                              &finished);
+      } else {
+        mEngine->ProcessBlocksOnPorts(
+            this, aFrom, Span(mInputChunks.Elements(), mEngine->InputCount()),
+            Span(mLastChunks.Elements(), mEngine->OutputCount()), &finished);
+      }
+    }
+    for (uint16_t i = 0; i < outputCount; ++i) {
+      NS_ASSERTION(mLastChunks[i].GetDuration() == WEBAUDIO_BLOCK_SIZE,
+                   "Invalid WebAudio chunk size");
+    }
+    if (finished) {
+      mMarkAsEndedAfterThisBlock = true;
+      if (mIsActive) {
+        ScheduleCheckForInactive();
+      }
+    }
+
+    if (mDisabledMode != DisabledTrackMode::ENABLED) {
+      for (uint32_t i = 0; i < outputCount; ++i) {
+        mLastChunks[i].SetNull(WEBAUDIO_BLOCK_SIZE);
+      }
+    }
+  }
+
+  if (!mEnded) {
+    // Don't output anything while finished
+    if (mFlags & EXTERNAL_OUTPUT) {
+      AdvanceOutputSegment();
+    }
+    if (mMarkAsEndedAfterThisBlock && (aFlags & ALLOW_END)) {
+      // This track was ended the last time that we looked at it, and all
+      // of the depending tracks have ended their output as well, so now
+      // it's time to mark this track as ended.
+      mEnded = true;
+    }
+  }
+}
+
+void AudioNodeTrack::ProduceOutputBeforeInput(GraphTime aFrom) {
+  MOZ_ASSERT(mEngine->AsDelayNodeEngine());
+  MOZ_ASSERT(mEngine->OutputCount() == 1,
+             "DelayNodeEngine output count should be 1");
+  MOZ_ASSERT(!InMutedCycle(), "DelayNodes should break cycles");
+  MOZ_ASSERT(mLastChunks.Length() == 1);
+
+  if (!mIsActive) {
+    mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
+  } else {
+    mEngine->ProduceBlockBeforeInput(this, aFrom, &mLastChunks[0]);
+    NS_ASSERTION(mLastChunks[0].GetDuration() == WEBAUDIO_BLOCK_SIZE,
+                 "Invalid WebAudio chunk size");
+    if (mDisabledMode != DisabledTrackMode::ENABLED) {
+      mLastChunks[0].SetNull(WEBAUDIO_BLOCK_SIZE);
+    }
+  }
+}
+
+void AudioNodeTrack::AdvanceOutputSegment() {
+  AudioSegment* segment = GetData<AudioSegment>();
+
+  AudioChunk copyChunk = *mLastChunks[0].AsMutableChunk();
+  AudioSegment tmpSegment;
+  tmpSegment.AppendAndConsumeChunk(std::move(copyChunk));
+
+  for (const auto& l : mTrackListeners) {
+    // Notify MediaTrackListeners.
+    l->NotifyQueuedChanges(Graph(), segment->GetDuration(), tmpSegment);
+  }
+
+  if (mLastChunks[0].IsNull()) {
+    segment->AppendNullData(tmpSegment.GetDuration());
+  } else {
+    segment->AppendFrom(&tmpSegment);
+  }
+}
+
+void AudioNodeTrack::AddInput(MediaInputPort* aPort) {
+  ProcessedMediaTrack::AddInput(aPort);
+  AudioNodeTrack* ns = aPort->GetSource()->AsAudioNodeTrack();
+  // Tracks that are not AudioNodeTracks are considered active.
+  if (!ns || (ns->mIsActive && !ns->IsAudioParamTrack())) {
+    IncrementActiveInputCount();
+  }
+}
+void AudioNodeTrack::RemoveInput(MediaInputPort* aPort) {
+  ProcessedMediaTrack::RemoveInput(aPort);
+  AudioNodeTrack* ns = aPort->GetSource()->AsAudioNodeTrack();
+  // Tracks that are not AudioNodeTracks are considered active.
+  if (!ns || (ns->mIsActive && !ns->IsAudioParamTrack())) {
+    DecrementActiveInputCount();
+  }
+}
+
+void AudioNodeTrack::SetActive() {
+  if (mIsActive || mMarkAsEndedAfterThisBlock) {
+    return;
+  }
+
+  mIsActive = true;
+  if (!(mFlags & EXTERNAL_OUTPUT)) {
+    DecrementSuspendCount();
+  }
+  if (IsAudioParamTrack()) {
+    // Consumers merely influence track order.
+    // They do not read from the track.
+    return;
+  }
+
+  for (const auto& consumer : mConsumers) {
+    AudioNodeTrack* ns = consumer->GetDestination()->AsAudioNodeTrack();
+    if (ns) {
+      ns->IncrementActiveInputCount();
+    }
+  }
+}
+
+void AudioNodeTrack::ScheduleCheckForInactive() {
+  if (mActiveInputCount > 0 && !mMarkAsEndedAfterThisBlock) {
+    return;
+  }
+
+  RunAfterProcessing([self = RefPtr{this}, this] {
+    TRACE("AudioNodeTrack::CheckForInactive");
+    CheckForInactive();
+  });
+}
+
+void AudioNodeTrack::CheckForInactive() {
+  if (((mActiveInputCount > 0 || mEngine->IsActive()) &&
+       !mMarkAsEndedAfterThisBlock) ||
+      !mIsActive) {
+    return;
+  }
+
+  mIsActive = false;
+  mInputChunks.Clear();  // not required for foreseeable future
+  for (auto& chunk : mLastChunks) {
+    chunk.SetNull(WEBAUDIO_BLOCK_SIZE);
+  }
+  if (!(mFlags & EXTERNAL_OUTPUT)) {
+    IncrementSuspendCount();
+  }
+  if (IsAudioParamTrack()) {
+    return;
+  }
+
+  for (const auto& consumer : mConsumers) {
+    AudioNodeTrack* ns = consumer->GetDestination()->AsAudioNodeTrack();
+    if (ns) {
+      ns->DecrementActiveInputCount();
+    }
+  }
+}
+
+void AudioNodeTrack::IncrementActiveInputCount() {
+  ++mActiveInputCount;
+  SetActive();
+}
+
+void AudioNodeTrack::DecrementActiveInputCount() {
+  MOZ_ASSERT(mActiveInputCount > 0);
+  --mActiveInputCount;
+  CheckForInactive();
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webaudio/AudioNodeTrack.h b/dom/media/webaudio/AudioNodeTrack.h
new file mode 100644
index 0000000000..505ea96718
--- /dev/null
+++ b/dom/media/webaudio/AudioNodeTrack.h
@@ -0,0 +1,233 @@
+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this file,
+ * You can obtain one at http://mozilla.org/MPL/2.0/. */
+
+#ifndef MOZILLA_AUDIONODETRACK_H_
+#define MOZILLA_AUDIONODETRACK_H_
+
+#include "MediaTrackGraph.h"
+#include "mozilla/dom/AudioNodeBinding.h"
+#include "AlignedTArray.h"
+#include "AudioBlock.h"
+#include "AudioSegment.h"
+
+namespace WebCore {
+class Reverb;
+}  // namespace WebCore
+
+namespace mozilla {
+
+namespace dom {
+struct ThreeDPoint;
+struct AudioParamEvent;
+class AudioContext;
+}  // namespace dom
+
+class AbstractThread;
+class ThreadSharedFloatArrayBufferList;
+class AudioNodeEngine;
+
+typedef AlignedAutoTArray<float, GUESS_AUDIO_CHANNELS * WEBAUDIO_BLOCK_SIZE, 16>
+    DownmixBufferType;
+
+/**
+ * An AudioNodeTrack produces one audio track with ID AUDIO_TRACK.
+ * The start time of the AudioTrack is aligned to the start time of the
+ * AudioContext's destination node track, plus some multiple of BLOCK_SIZE
+ * samples.
+ *
+ * An AudioNodeTrack has an AudioNodeEngine plugged into it that does the
+ * actual audio processing. AudioNodeTrack contains the glue code that
+ * integrates audio processing with the MediaTrackGraph.
+ */
+class AudioNodeTrack : public ProcessedMediaTrack {
+  typedef dom::ChannelCountMode ChannelCountMode;
+  typedef dom::ChannelInterpretation ChannelInterpretation;
+
+ public:
+  typedef mozilla::dom::AudioContext AudioContext;
+
+  enum { AUDIO_TRACK = 1 };
+
+  typedef AutoTArray<AudioBlock, 1> OutputChunks;
+
+  // Flags re main thread updates and track output.
+  typedef unsigned Flags;
+  enum : Flags {
+    NO_TRACK_FLAGS = 0U,
+    NEED_MAIN_THREAD_ENDED = 1U << 0,
+    NEED_MAIN_THREAD_CURRENT_TIME = 1U << 1,
+    // Internal AudioNodeTracks can only pass their output to another
+    // AudioNode, whereas external AudioNodeTracks can pass their output
+    // to other ProcessedMediaTracks or hardware audio output.
+    EXTERNAL_OUTPUT = 1U << 2,
+  };
+  /**
+   * Create a track that will process audio for an AudioNode.
+   * Takes ownership of aEngine.
+   * aGraph is required and equals the graph of aCtx in most cases. An exception
+   * is AudioDestinationNode where the context's graph hasn't been set up yet.
+   */
+  static already_AddRefed<AudioNodeTrack> Create(AudioContext* aCtx,
+                                                 AudioNodeEngine* aEngine,
+                                                 Flags aKind,
+                                                 MediaTrackGraph* aGraph);
+
+ protected:
+  /**
+   * Transfers ownership of aEngine to the new AudioNodeTrack.
+   */
+  AudioNodeTrack(AudioNodeEngine* aEngine, Flags aFlags, TrackRate aSampleRate);
+
+  ~AudioNodeTrack();
+
+ public:
+  // Control API
+  /**
+   * Sets a parameter that's a time relative to some track's played time.
+   * This time is converted to a time relative to this track when it's set.
+   */
+  void SetTrackTimeParameter(uint32_t aIndex, AudioContext* aContext,
+                             double aTrackTime);
+  void SetDoubleParameter(uint32_t aIndex, double aValue);
+  void SetInt32Parameter(uint32_t aIndex, int32_t aValue);
+  void SetThreeDPointParameter(uint32_t aIndex, const dom::ThreeDPoint& aValue);
+  void SetBuffer(AudioChunk&& aBuffer);
+  void SetReverb(WebCore::Reverb* aReverb, uint32_t aImpulseChannelCount);
+  // This sends a single event to the timeline on the MTG thread side.
+  void SendTimelineEvent(uint32_t aIndex, const dom::AudioParamEvent& aEvent);
+  // This consumes the contents of aData.  aData will be emptied after this
+  // returns.
+  void SetRawArrayData(nsTArray<float>&& aData);
+  void SetChannelMixingParameters(uint32_t aNumberOfChannels,
+                                  ChannelCountMode aChannelCountMoe,
+                                  ChannelInterpretation aChannelInterpretation);
+  void SetPassThrough(bool aPassThrough);
+  void SendRunnable(already_AddRefed<nsIRunnable> aRunnable);
+  ChannelInterpretation GetChannelInterpretation() {
+    return mChannelInterpretation;
+  }
+
+  void SetAudioParamHelperTrack() {
+    MOZ_ASSERT(!mAudioParamTrack, "Can only do this once");
+    mAudioParamTrack = true;
+  }
+  // The value for channelCount on an AudioNode, but on the audio thread side.
+  uint32_t NumberOfChannels() const override;
+
+  /*
+   * Resume track after updating its concept of current time by aAdvance.
+   * Main thread.  Used only from AudioDestinationNode when resuming a track
+   * suspended to save running the MediaTrackGraph when there are no other
+   * nodes in the AudioContext.
+   */
+  void AdvanceAndResume(TrackTime aAdvance);
+
+  AudioNodeTrack* AsAudioNodeTrack() override { return this; }
+  void AddInput(MediaInputPort* aPort) override;
+  void RemoveInput(MediaInputPort* aPort) override;
+
+  // Graph thread only
+  void SetTrackTimeParameterImpl(uint32_t aIndex, MediaTrack* aRelativeToTrack,
+                                 double aTrackTime);
+  void SetChannelMixingParametersImpl(
+      uint32_t aNumberOfChannels, ChannelCountMode aChannelCountMoe,
+      ChannelInterpretation aChannelInterpretation);
+  void ProcessInput(GraphTime aFrom, GraphTime aTo, uint32_t aFlags) override;
+  /**
+   * Produce the next block of output, before input is provided.
+   * ProcessInput() will be called later, and it then should not change
+   * the output.  This is used only for DelayNodeEngine in a feedback loop.
+   */
+  void ProduceOutputBeforeInput(GraphTime aFrom);
+  bool IsAudioParamTrack() const { return mAudioParamTrack; }
+
+  const OutputChunks& LastChunks() const { return mLastChunks; }
+  bool MainThreadNeedsUpdates() const override {
+    return ((mFlags & NEED_MAIN_THREAD_ENDED) && mEnded) ||
+           (mFlags & NEED_MAIN_THREAD_CURRENT_TIME);
+  }
+
+  // Any thread
+  AudioNodeEngine* Engine() { return mEngine.get(); }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+  void SizeOfAudioNodesIncludingThis(MallocSizeOf aMallocSizeOf,
+                                     AudioNodeSizes& aUsage) const;
+
+  /*
+   * SetActive() is called when either an active input is added or the engine
+   * for a source node transitions from inactive to active.  This is not
+   * called from engines for processing nodes because they only become active
+   * when there are active input tracks, in which case this track is already
+   * active.
+   */
+  void SetActive();
+  /*
+   * ScheduleCheckForInactive() is called during track processing when the
+   * engine transitions from active to inactive, or the track finishes.  It
+   * schedules a call to CheckForInactive() after track processing.
+   */
+  void ScheduleCheckForInactive();
+
+ protected:
+  void OnGraphThreadDone() override;
+  void DestroyImpl() override;
+
+  /*
+   * CheckForInactive() is called when the engine transitions from active to
+   * inactive, or an active input is removed, or the track finishes.  If the
+   * track is now inactive, then mInputChunks will be cleared and mLastChunks
+   * will be set to null.  ProcessBlock() will not be called on the engine
+   * again until SetActive() is called.
+   */
+  void CheckForInactive();
+
+  void AdvanceOutputSegment();
+  void FinishOutput();
+  void AccumulateInputChunk(uint32_t aInputIndex, const AudioBlock& aChunk,
+                            AudioBlock* aBlock,
+                            DownmixBufferType* aDownmixBuffer);
+  void UpMixDownMixChunk(const AudioBlock* aChunk, uint32_t aOutputChannelCount,
+                         nsTArray<const float*>& aOutputChannels,
+                         DownmixBufferType& aDownmixBuffer);
+
+  uint32_t ComputedNumberOfChannels(uint32_t aInputChannelCount);
+  void ObtainInputBlock(AudioBlock& aTmpChunk, uint32_t aPortIndex);
+  void IncrementActiveInputCount();
+  void DecrementActiveInputCount();
+
+  // The engine that will generate output for this node.
+  const UniquePtr<AudioNodeEngine> mEngine;
+  // The mixed input blocks are kept from iteration to iteration to avoid
+  // reallocating channel data arrays and any buffers for mixing.
+  OutputChunks mInputChunks;
+  // The last block produced by this node.
+  OutputChunks mLastChunks;
+  // Whether this is an internal or external track
+  const Flags mFlags;
+  // The number of input tracks that may provide non-silent input.
+  uint32_t mActiveInputCount = 0;
+  // The number of input channels that this track requires. 0 means don't care.
+  uint32_t mNumberOfInputChannels;
+  // The mixing modes
+  ChannelCountMode mChannelCountMode;
+  ChannelInterpretation mChannelInterpretation;
+  // Tracks are considered active if the track has not finished and either
+  // the engine is active or there are active input tracks.
+  bool mIsActive;
+  // Whether the track should be marked as ended as soon
+  // as the current time range has been computed block by block.
+  bool mMarkAsEndedAfterThisBlock;
+  // Whether the track is an AudioParamHelper track.
+  bool mAudioParamTrack;
+  // Whether the track just passes its input through.
+  bool mPassThrough;
+};
+
+}  // namespace mozilla
+
+#endif /* MOZILLA_AUDIONODETRACK_H_ */
diff --git a/dom/media/webaudio/AudioParam.cpp b/dom/media/webaudio/AudioParam.cpp
new file mode 100644
index 0000000000..58d3b225c5
--- /dev/null
+++ b/dom/media/webaudio/AudioParam.cpp
@@ -0,0 +1,170 @@
+/* -*- 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 "AudioParam.h"
+#include "mozilla/dom/AudioParamBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "AudioContext.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE_CLASS(AudioParam)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(AudioParam)
+  tmp->DisconnectFromGraphAndDestroyTrack();
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mNode)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN(AudioParam)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mNode)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_CYCLE_COLLECTING_NATIVE_ADDREF(AudioParam)
+NS_IMPL_CYCLE_COLLECTING_NATIVE_RELEASE(AudioParam)
+
+AudioParam::AudioParam(AudioNode* aNode, uint32_t aIndex,
+                       const nsAString& aName, float aDefaultValue,
+                       float aMinValue, float aMaxValue)
+    : AudioParamTimeline(aDefaultValue),
+      mNode(aNode),
+      mName(aName),
+      mIndex(aIndex),
+      mDefaultValue(aDefaultValue),
+      mMinValue(aMinValue),
+      mMaxValue(aMaxValue) {}
+
+AudioParam::~AudioParam() { DisconnectFromGraphAndDestroyTrack(); }
+
+JSObject* AudioParam::WrapObject(JSContext* aCx,
+                                 JS::Handle<JSObject*> aGivenProto) {
+  return AudioParam_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void AudioParam::DisconnectFromGraphAndDestroyTrack() {
+  MOZ_ASSERT(mRefCnt.get() > mInputNodes.Length(),
+             "Caller should be holding a reference or have called "
+             "mRefCnt.stabilizeForDeletion()");
+
+  while (!mInputNodes.IsEmpty()) {
+    RefPtr<AudioNode> input = mInputNodes.PopLastElement().mInputNode;
+    input->RemoveOutputParam(this);
+  }
+
+  if (mNodeTrackPort) {
+    mNodeTrackPort->Destroy();
+    mNodeTrackPort = nullptr;
+  }
+
+  if (mTrack) {
+    mTrack->Destroy();
+    mTrack = nullptr;
+  }
+}
+
+mozilla::MediaTrack* AudioParam::GetTrack() const { return mTrack; }
+
+mozilla::MediaTrack* AudioParam::Track() {
+  if (mTrack) {
+    return mTrack;
+  }
+
+  AudioNodeEngine* engine = new AudioNodeEngine(nullptr);
+  mTrack = AudioNodeTrack::Create(mNode->Context(), engine,
+                                  AudioNodeTrack::NO_TRACK_FLAGS,
+                                  mNode->Context()->Graph());
+
+  // Force the input to have only one channel, and make it down-mix using
+  // the speaker rules if needed.
+  mTrack->SetChannelMixingParametersImpl(1, ChannelCountMode::Explicit,
+                                         ChannelInterpretation::Speakers);
+  // Mark as an AudioParam helper track
+  mTrack->SetAudioParamHelperTrack();
+
+  // Setup the AudioParam's track as an input to the owner AudioNode's track
+  AudioNodeTrack* nodeTrack = mNode->GetTrack();
+  if (nodeTrack) {
+    mNodeTrackPort = nodeTrack->AllocateInputPort(mTrack);
+  }
+
+  // Send the track to the timeline on the MTG side.
+  AudioParamEvent event(mTrack);
+  SendEventToEngine(event);
+
+  return mTrack;
+}
+
+static const char* ToString(AudioTimelineEvent::Type aType) {
+  switch (aType) {
+    case AudioTimelineEvent::SetValue:
+      return "SetValue";
+    case AudioTimelineEvent::SetValueAtTime:
+      return "SetValueAtTime";
+    case AudioTimelineEvent::LinearRamp:
+      return "LinearRamp";
+    case AudioTimelineEvent::ExponentialRamp:
+      return "ExponentialRamp";
+    case AudioTimelineEvent::SetTarget:
+      return "SetTarget";
+    case AudioTimelineEvent::SetValueCurve:
+      return "SetValueCurve";
+    case AudioTimelineEvent::Track:
+      return "Track";
+    case AudioTimelineEvent::Cancel:
+      return "Cancel";
+    default:
+      return "unknown AudioTimelineEvent";
+  }
+}
+
+void AudioParam::SendEventToEngine(const AudioParamEvent& aEvent) {
+  WEB_AUDIO_API_LOG(
+      "%f: %s for %u %s %s=%g time=%f %s=%g", GetParentObject()->CurrentTime(),
+      NS_ConvertUTF16toUTF8(mName).get(), ParentNodeId(),
+      ToString(aEvent.mType),
+      aEvent.mType == AudioTimelineEvent::SetValueCurve ? "length" : "value",
+      aEvent.mType == AudioTimelineEvent::SetValueCurve
+          ? static_cast<double>(aEvent.CurveLength())
+          : static_cast<double>(aEvent.NominalValue()),
+      aEvent.Time<double>(),
+      aEvent.mType == AudioTimelineEvent::SetValueCurve ? "duration"
+                                                        : "constant",
+      aEvent.mType == AudioTimelineEvent::SetValueCurve
+          ? aEvent.Duration()
+          : aEvent.TimeConstant());
+
+  AudioNodeTrack* track = mNode->GetTrack();
+  if (track) {
+    track->SendTimelineEvent(mIndex, aEvent);
+  }
+}
+
+void AudioParam::CleanupOldEvents() {
+  MOZ_ASSERT(NS_IsMainThread());
+  double currentTime = mNode->Context()->CurrentTime();
+
+  CleanupEventsOlderThan(currentTime);
+}
+
+float AudioParamTimeline::AudioNodeInputValue(size_t aCounter) const {
+  MOZ_ASSERT(mTrack);
+
+  // If we have a chunk produced by the AudioNode inputs to the AudioParam,
+  // get its value now.  We use aCounter to tell us which frame of the last
+  // AudioChunk to look at.
+  float audioNodeInputValue = 0.0f;
+  const AudioBlock& lastAudioNodeChunk = mTrack->LastChunks()[0];
+  if (!lastAudioNodeChunk.IsNull()) {
+    MOZ_ASSERT(lastAudioNodeChunk.GetDuration() == WEBAUDIO_BLOCK_SIZE);
+    audioNodeInputValue =
+        static_cast<const float*>(lastAudioNodeChunk.mChannelData[0])[aCounter];
+    audioNodeInputValue *= lastAudioNodeChunk.mVolume;
+  }
+
+  return audioNodeInputValue;
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioParam.h b/dom/media/webaudio/AudioParam.h
new file mode 100644
index 0000000000..2365857a5b
--- /dev/null
+++ b/dom/media/webaudio/AudioParam.h
@@ -0,0 +1,239 @@
+/* -*- 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/. */
+
+#ifndef AudioParam_h_
+#define AudioParam_h_
+
+#include "AudioParamTimeline.h"
+#include "nsWrapperCache.h"
+#include "nsCycleCollectionParticipant.h"
+#include "AudioNode.h"
+#include "mozilla/dom/TypedArray.h"
+#include "WebAudioUtils.h"
+#include "js/TypeDecls.h"
+
+namespace mozilla::dom {
+
+class AudioParam final : public nsWrapperCache, public AudioParamTimeline {
+  virtual ~AudioParam();
+
+ public:
+  AudioParam(AudioNode* aNode, uint32_t aIndex, const nsAString& aName,
+             float aDefaultValue,
+             float aMinValue = std::numeric_limits<float>::lowest(),
+             float aMaxValue = std::numeric_limits<float>::max());
+
+  NS_IMETHOD_(MozExternalRefCountType) AddRef(void);
+  NS_IMETHOD_(MozExternalRefCountType) Release(void);
+  NS_DECL_CYCLE_COLLECTION_NATIVE_WRAPPERCACHE_CLASS(AudioParam)
+
+  AudioContext* GetParentObject() const { return mNode->Context(); }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  float Value() {
+    return AudioParamTimeline::GetValueAtTime<double>(
+        GetParentObject()->CurrentTime());
+  }
+
+  // We override SetValueCurveAtTime to convert the Float32Array to the wrapper
+  // object.
+  AudioParam* SetValueCurveAtTime(const nsTArray<float>& aValues,
+                                  double aStartTime, double aDuration,
+                                  ErrorResult& aRv) {
+    if (!WebAudioUtils::IsTimeValid(aStartTime)) {
+      aRv.ThrowRangeError<MSG_INVALID_AUDIOPARAM_METHOD_START_TIME_ERROR>();
+      return this;
+    }
+    aStartTime = std::max(aStartTime, GetParentObject()->CurrentTime());
+    AudioParamEvent event(AudioTimelineEvent::SetValueCurve, aValues,
+                          aStartTime, aDuration);
+    ValidateAndInsertEvent(event, aRv);
+    return this;
+  }
+
+  // Intended for use in AudioNode creation, when the setter should not throw.
+  void SetInitialValue(float aValue) {
+    MOZ_ASSERT(HasSimpleValue(), "Existing events unexpected");
+    AudioParamEvent event(AudioTimelineEvent::SetValue, 0.0f, aValue);
+
+    DebugOnly<ErrorResult> rv;
+    MOZ_ASSERT(ValidateEvent(event, rv), "This event should be valid");
+
+    AudioParamTimeline::SetValue(aValue);
+
+    SendEventToEngine(event);
+  }
+
+  void SetValue(float aValue, ErrorResult& aRv) {
+    AudioParamEvent event(AudioTimelineEvent::SetValue, 0.0f, aValue);
+
+    if (!ValidateEvent(event, aRv)) {
+      return;
+    }
+
+    AudioParamTimeline::SetValue(aValue);
+
+    SendEventToEngine(event);
+  }
+
+  AudioParam* SetValueAtTime(float aValue, double aStartTime,
+                             ErrorResult& aRv) {
+    if (!WebAudioUtils::IsTimeValid(aStartTime)) {
+      aRv.ThrowRangeError<MSG_INVALID_AUDIOPARAM_METHOD_START_TIME_ERROR>();
+      return this;
+    }
+    aStartTime = std::max(aStartTime, GetParentObject()->CurrentTime());
+    AudioParamEvent event(AudioTimelineEvent::SetValueAtTime, aStartTime,
+                          aValue);
+    ValidateAndInsertEvent(event, aRv);
+    return this;
+  }
+
+  AudioParam* LinearRampToValueAtTime(float aValue, double aEndTime,
+                                      ErrorResult& aRv) {
+    if (!WebAudioUtils::IsTimeValid(aEndTime)) {
+      aRv.ThrowRangeError<MSG_INVALID_AUDIOPARAM_METHOD_END_TIME_ERROR>();
+      return this;
+    }
+    aEndTime = std::max(aEndTime, GetParentObject()->CurrentTime());
+    AudioParamEvent event(AudioTimelineEvent::LinearRamp, aEndTime, aValue);
+    ValidateAndInsertEvent(event, aRv);
+    return this;
+  }
+
+  AudioParam* ExponentialRampToValueAtTime(float aValue, double aEndTime,
+                                           ErrorResult& aRv) {
+    if (!WebAudioUtils::IsTimeValid(aEndTime)) {
+      aRv.ThrowRangeError<MSG_INVALID_AUDIOPARAM_METHOD_END_TIME_ERROR>();
+      return this;
+    }
+    aEndTime = std::max(aEndTime, GetParentObject()->CurrentTime());
+    AudioParamEvent event(AudioTimelineEvent::ExponentialRamp, aEndTime,
+                          aValue);
+    ValidateAndInsertEvent(event, aRv);
+    return this;
+  }
+
+  AudioParam* SetTargetAtTime(float aTarget, double aStartTime,
+                              double aTimeConstant, ErrorResult& aRv) {
+    if (!WebAudioUtils::IsTimeValid(aStartTime) ||
+        !WebAudioUtils::IsTimeValid(aTimeConstant)) {
+      aRv.ThrowRangeError<MSG_INVALID_AUDIOPARAM_METHOD_START_TIME_ERROR>();
+      return this;
+    }
+    aStartTime = std::max(aStartTime, GetParentObject()->CurrentTime());
+    AudioParamEvent event(AudioTimelineEvent::SetTarget, aStartTime, aTarget,
+                          aTimeConstant);
+    ValidateAndInsertEvent(event, aRv);
+    return this;
+  }
+
+  AudioParam* CancelScheduledValues(double aStartTime, ErrorResult& aRv) {
+    if (!WebAudioUtils::IsTimeValid(aStartTime)) {
+      aRv.ThrowRangeError<MSG_INVALID_AUDIOPARAM_METHOD_START_TIME_ERROR>();
+      return this;
+    }
+
+    aStartTime = std::max(aStartTime, GetParentObject()->CurrentTime());
+
+    // Remove some events on the main thread copy.
+    AudioEventTimeline::CancelScheduledValues(aStartTime);
+
+    AudioParamEvent event(AudioTimelineEvent::Cancel, aStartTime, 0.0f);
+
+    SendEventToEngine(event);
+
+    return this;
+  }
+
+  uint32_t ParentNodeId() { return mNode->Id(); }
+
+  void GetName(nsAString& aName) { aName.Assign(mName); }
+
+  float DefaultValue() const { return mDefaultValue; }
+
+  float MinValue() const { return mMinValue; }
+
+  float MaxValue() const { return mMaxValue; }
+
+  bool IsTrackSuspended() const {
+    return mTrack ? mTrack->IsSuspended() : false;
+  }
+
+  const nsTArray<AudioNode::InputNode>& InputNodes() const {
+    return mInputNodes;
+  }
+
+  void RemoveInputNode(uint32_t aIndex) { mInputNodes.RemoveElementAt(aIndex); }
+
+  AudioNode::InputNode* AppendInputNode() {
+    return mInputNodes.AppendElement();
+  }
+
+  // May create the track if it doesn't exist
+  mozilla::MediaTrack* Track();
+
+  // Return nullptr if track doesn't exist.
+  mozilla::MediaTrack* GetTrack() const;
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    size_t amount = AudioParamTimeline::SizeOfExcludingThis(aMallocSizeOf);
+    // Not owned:
+    // - mNode
+
+    // Just count the array, actual nodes are counted in mNode.
+    amount += mInputNodes.ShallowSizeOfExcludingThis(aMallocSizeOf);
+
+    if (mNodeTrackPort) {
+      amount += mNodeTrackPort->SizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  void ValidateAndInsertEvent(const AudioParamEvent& aEvent, ErrorResult& aRv) {
+    if (!ValidateEvent(aEvent, aRv)) {
+      return;
+    }
+
+    AudioEventTimeline::InsertEvent<double>(aEvent);
+
+    SendEventToEngine(aEvent);
+
+    CleanupOldEvents();
+  }
+
+  void CleanupOldEvents();
+
+  void SendEventToEngine(const AudioParamEvent& aEvent);
+
+  void DisconnectFromGraphAndDestroyTrack();
+
+  nsCycleCollectingAutoRefCnt mRefCnt;
+  NS_DECL_OWNINGTHREAD
+  RefPtr<AudioNode> mNode;
+  // For every InputNode, there is a corresponding entry in mOutputParams of the
+  // InputNode's mInputNode.
+  nsTArray<AudioNode::InputNode> mInputNodes;
+  const nsString mName;
+  // The input port used to connect the AudioParam's track to its node's track
+  RefPtr<MediaInputPort> mNodeTrackPort;
+  const uint32_t mIndex;
+  const float mDefaultValue;
+  const float mMinValue;
+  const float mMaxValue;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/AudioParamDescriptorMap.h b/dom/media/webaudio/AudioParamDescriptorMap.h
new file mode 100644
index 0000000000..978e8bc90c
--- /dev/null
+++ b/dom/media/webaudio/AudioParamDescriptorMap.h
@@ -0,0 +1,22 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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 https://mozilla.org/MPL/2.0/. */
+
+#ifndef mozilla_dom_AudioParamDescriptor_h
+#define mozilla_dom_AudioParamDescriptor_h
+
+#include "mozilla/dom/AudioParamDescriptorBinding.h"
+#include "nsTArray.h"
+
+namespace mozilla::dom {
+
+// Note: we call this "map" to match the spec, but we store audio param
+// descriptors in an array, because we need ordered access, and don't need the
+// map functionalities.
+typedef nsTArray<AudioParamDescriptor> AudioParamDescriptorMap;
+
+}  // namespace mozilla::dom
+
+#endif  // mozilla_dom_AudioParamDescriptor_h
diff --git a/dom/media/webaudio/AudioParamMap.cpp b/dom/media/webaudio/AudioParamMap.cpp
new file mode 100644
index 0000000000..2d4f936361
--- /dev/null
+++ b/dom/media/webaudio/AudioParamMap.cpp
@@ -0,0 +1,21 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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 https://mozilla.org/MPL/2.0/. */
+
+#include "AudioParamMap.h"
+#include "mozilla/dom/AudioParamMapBinding.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE(AudioParamMap, mParent)
+
+AudioParamMap::AudioParamMap(AudioWorkletNode* aParent) : mParent(aParent) {}
+
+JSObject* AudioParamMap::WrapObject(JSContext* aCx,
+                                    JS::Handle<JSObject*> aGivenProto) {
+  return AudioParamMap_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioParamMap.h b/dom/media/webaudio/AudioParamMap.h
new file mode 100644
index 0000000000..c7bb0bdbea
--- /dev/null
+++ b/dom/media/webaudio/AudioParamMap.h
@@ -0,0 +1,34 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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 https://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioParamMap_h_
+#define AudioParamMap_h_
+
+#include "nsWrapperCache.h"
+#include "AudioWorkletNode.h"
+
+namespace mozilla::dom {
+
+class AudioParamMap final : public nsWrapperCache {
+ public:
+  NS_INLINE_DECL_CYCLE_COLLECTING_NATIVE_REFCOUNTING(AudioParamMap)
+  NS_DECL_CYCLE_COLLECTION_NATIVE_WRAPPERCACHE_CLASS(AudioParamMap)
+
+  explicit AudioParamMap(AudioWorkletNode* aParent);
+
+  AudioWorkletNode* GetParentObject() const { return mParent; }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+ private:
+  ~AudioParamMap() = default;
+  RefPtr<AudioWorkletNode> mParent;
+};
+
+}  // namespace mozilla::dom
+
+#endif  // AudioParamMap_h_
diff --git a/dom/media/webaudio/AudioParamTimeline.h b/dom/media/webaudio/AudioParamTimeline.h
new file mode 100644
index 0000000000..bd0ba99ffe
--- /dev/null
+++ b/dom/media/webaudio/AudioParamTimeline.h
@@ -0,0 +1,144 @@
+/* -*- 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/. */
+
+#ifndef AudioParamTimeline_h_
+#define AudioParamTimeline_h_
+
+#include "AudioEventTimeline.h"
+#include "AudioNodeTrack.h"
+#include "AudioSegment.h"
+
+namespace mozilla::dom {
+
+struct AudioParamEvent final : public AudioTimelineEvent {
+  AudioParamEvent(Type aType, double aTime, float aValue,
+                  double aTimeConstant = 0.0)
+      : AudioTimelineEvent(aType, aTime, aValue, aTimeConstant) {}
+  AudioParamEvent(Type aType, const nsTArray<float>& aValues, double aStartTime,
+                  double aDuration)
+      : AudioTimelineEvent(aType, aValues, aStartTime, aDuration) {}
+  explicit AudioParamEvent(AudioNodeTrack* aTrack)
+      : AudioTimelineEvent(Track, 0.0, 0.f), mTrack(aTrack) {}
+
+  RefPtr<AudioNodeTrack> mTrack;
+};
+
+// This helper class is used to represent the part of the AudioParam
+// class that gets sent to AudioNodeEngine instances.  In addition to
+// AudioEventTimeline methods, it holds a pointer to an optional
+// AudioNodeTrack which represents the AudioNode inputs to the AudioParam.
+// This AudioNodeTrack is managed by the AudioParam subclass on the main
+// thread, and can only be obtained from the AudioNodeEngine instances
+// consuming this class.
+class AudioParamTimeline : public AudioEventTimeline {
+  typedef AudioEventTimeline BaseClass;
+
+ public:
+  explicit AudioParamTimeline(float aDefaultValue) : BaseClass(aDefaultValue) {}
+
+  AudioNodeTrack* Track() const { return mTrack; }
+
+  bool HasSimpleValue() const {
+    return BaseClass::HasSimpleValue() &&
+           (!mTrack || mTrack->LastChunks()[0].IsNull());
+  }
+
+  template <class TimeType>
+  float GetValueAtTime(TimeType aTime);
+
+  // Prefer this method over GetValueAtTime() only if HasSimpleValue() is
+  // known false.
+  float GetComplexValueAtTime(int64_t aTime);
+  float GetComplexValueAtTime(double aTime) = delete;
+
+  template <typename TimeType>
+  void InsertEvent(const AudioParamEvent& aEvent) {
+    if (aEvent.mType == AudioTimelineEvent::Cancel) {
+      CancelScheduledValues(aEvent.Time<TimeType>());
+      return;
+    }
+    if (aEvent.mType == AudioTimelineEvent::Track) {
+      mTrack = aEvent.mTrack;
+      return;
+    }
+    if (aEvent.mType == AudioTimelineEvent::SetValue) {
+      AudioEventTimeline::SetValue(aEvent.NominalValue());
+      return;
+    }
+    AudioEventTimeline::InsertEvent<TimeType>(aEvent);
+  }
+
+  // Get the values of the AudioParam at time aTime + (0 to aSize).
+  // aBuffer must have the correct aSize.
+  // aSize here is an offset to aTime if we try to get the value in ticks,
+  // otherwise it should always be zero.  aSize is meant to be used when
+  // getting the value of an a-rate AudioParam for each tick inside an
+  // AudioNodeEngine implementation.
+  void GetValuesAtTime(int64_t aTime, float* aBuffer, const size_t aSize);
+  void GetValuesAtTime(double aTime, float* aBuffer,
+                       const size_t aSize) = delete;
+
+  virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+    return mTrack ? mTrack->SizeOfIncludingThis(aMallocSizeOf) : 0;
+  }
+
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  float AudioNodeInputValue(size_t aCounter) const;
+
+ protected:
+  // This is created lazily when needed.
+  RefPtr<AudioNodeTrack> mTrack;
+};
+
+template <>
+inline float AudioParamTimeline::GetValueAtTime(double aTime) {
+  // Getting an AudioParam value on an AudioNode does not consider input from
+  // other AudioNodes, which is managed only on the graph thread.
+  return BaseClass::GetValueAtTime(aTime);
+}
+
+template <>
+inline float AudioParamTimeline::GetValueAtTime(int64_t aTime) {
+  // Use GetValuesAtTime() for a-rate parameters.
+  MOZ_ASSERT(aTime % WEBAUDIO_BLOCK_SIZE == 0);
+  if (HasSimpleValue()) {
+    return GetValue();
+  }
+  return GetComplexValueAtTime(aTime);
+}
+
+inline float AudioParamTimeline::GetComplexValueAtTime(int64_t aTime) {
+  MOZ_ASSERT(aTime % WEBAUDIO_BLOCK_SIZE == 0);
+
+  // Mix the value of the AudioParam itself with that of the AudioNode inputs.
+  return BaseClass::GetValueAtTime(aTime) +
+         (mTrack ? AudioNodeInputValue(0) : 0.0f);
+}
+
+inline void AudioParamTimeline::GetValuesAtTime(int64_t aTime, float* aBuffer,
+                                                const size_t aSize) {
+  MOZ_ASSERT(aBuffer);
+  MOZ_ASSERT(aSize <= WEBAUDIO_BLOCK_SIZE);
+  MOZ_ASSERT(aSize == 1 || !HasSimpleValue());
+
+  // Mix the value of the AudioParam itself with that of the AudioNode inputs.
+  BaseClass::GetValuesAtTime(aTime, aBuffer, aSize);
+  if (mTrack) {
+    uint32_t blockOffset = aTime % WEBAUDIO_BLOCK_SIZE;
+    MOZ_ASSERT(blockOffset + aSize <= WEBAUDIO_BLOCK_SIZE);
+    for (size_t i = 0; i < aSize; ++i) {
+      aBuffer[i] += AudioNodeInputValue(blockOffset + i);
+    }
+  }
+}
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/AudioProcessingEvent.cpp b/dom/media/webaudio/AudioProcessingEvent.cpp
new file mode 100644
index 0000000000..7d9987db07
--- /dev/null
+++ b/dom/media/webaudio/AudioProcessingEvent.cpp
@@ -0,0 +1,44 @@
+/* -*- 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 "AudioProcessingEvent.h"
+#include "mozilla/dom/AudioProcessingEventBinding.h"
+#include "mozilla/dom/ScriptSettings.h"
+#include "AudioContext.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(AudioProcessingEvent, Event, mInputBuffer,
+                                   mOutputBuffer, mNode)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(AudioProcessingEvent)
+NS_INTERFACE_MAP_END_INHERITING(Event)
+
+NS_IMPL_ADDREF_INHERITED(AudioProcessingEvent, Event)
+NS_IMPL_RELEASE_INHERITED(AudioProcessingEvent, Event)
+
+AudioProcessingEvent::AudioProcessingEvent(ScriptProcessorNode* aOwner,
+                                           nsPresContext* aPresContext,
+                                           WidgetEvent* aEvent)
+    : Event(aOwner, aPresContext, aEvent), mPlaybackTime(0.0), mNode(aOwner) {}
+
+AudioProcessingEvent::~AudioProcessingEvent() = default;
+
+JSObject* AudioProcessingEvent::WrapObjectInternal(
+    JSContext* aCx, JS::Handle<JSObject*> aGivenProto) {
+  return AudioProcessingEvent_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+already_AddRefed<AudioBuffer> AudioProcessingEvent::LazilyCreateBuffer(
+    uint32_t aNumberOfChannels, ErrorResult& aRv) {
+  RefPtr<AudioBuffer> buffer = AudioBuffer::Create(
+      mNode->Context()->GetOwner(), aNumberOfChannels, mNode->BufferSize(),
+      mNode->Context()->SampleRate(), aRv);
+  MOZ_ASSERT(buffer || aRv.ErrorCodeIs(NS_ERROR_OUT_OF_MEMORY));
+  return buffer.forget();
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioProcessingEvent.h b/dom/media/webaudio/AudioProcessingEvent.h
new file mode 100644
index 0000000000..3b8b9cf008
--- /dev/null
+++ b/dom/media/webaudio/AudioProcessingEvent.h
@@ -0,0 +1,71 @@
+/* -*- 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/. */
+
+#ifndef AudioProcessingEvent_h_
+#define AudioProcessingEvent_h_
+
+#include "AudioBuffer.h"
+#include "ScriptProcessorNode.h"
+#include "mozilla/dom/Event.h"
+
+namespace mozilla::dom {
+
+class AudioProcessingEvent final : public Event {
+ public:
+  AudioProcessingEvent(ScriptProcessorNode* aOwner, nsPresContext* aPresContext,
+                       WidgetEvent* aEvent);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioProcessingEvent, Event)
+
+  JSObject* WrapObjectInternal(JSContext* aCx,
+                               JS::Handle<JSObject*> aGivenProto) override;
+
+  using Event::InitEvent;
+  void InitEvent(AudioBuffer* aInputBuffer, uint32_t aNumberOfInputChannels,
+                 double aPlaybackTime) {
+    InitEvent(u"audioprocess"_ns, CanBubble::eNo, Cancelable::eNo);
+    mInputBuffer = aInputBuffer;
+    mNumberOfInputChannels = aNumberOfInputChannels;
+    mPlaybackTime = aPlaybackTime;
+  }
+
+  double PlaybackTime() const { return mPlaybackTime; }
+
+  AudioBuffer* GetInputBuffer(ErrorResult& aRv) {
+    if (!mInputBuffer) {
+      mInputBuffer = LazilyCreateBuffer(mNumberOfInputChannels, aRv);
+    }
+    return mInputBuffer;
+  }
+
+  AudioBuffer* GetOutputBuffer(ErrorResult& aRv) {
+    if (!mOutputBuffer) {
+      mOutputBuffer = LazilyCreateBuffer(mNode->NumberOfOutputChannels(), aRv);
+    }
+    return mOutputBuffer;
+  }
+
+  bool HasOutputBuffer() const { return !!mOutputBuffer; }
+
+ protected:
+  virtual ~AudioProcessingEvent();
+
+ private:
+  already_AddRefed<AudioBuffer> LazilyCreateBuffer(uint32_t aNumberOfChannels,
+                                                   ErrorResult& rv);
+
+ private:
+  double mPlaybackTime;
+  RefPtr<AudioBuffer> mInputBuffer;
+  RefPtr<AudioBuffer> mOutputBuffer;
+  RefPtr<ScriptProcessorNode> mNode;
+  uint32_t mNumberOfInputChannels;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/AudioScheduledSourceNode.cpp b/dom/media/webaudio/AudioScheduledSourceNode.cpp
new file mode 100644
index 0000000000..c6609def51
--- /dev/null
+++ b/dom/media/webaudio/AudioScheduledSourceNode.cpp
@@ -0,0 +1,17 @@
+/* -*- 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 "AudioScheduledSourceNode.h"
+
+namespace mozilla::dom {
+
+AudioScheduledSourceNode::AudioScheduledSourceNode(
+    AudioContext* aContext, uint32_t aChannelCount,
+    ChannelCountMode aChannelCountMode,
+    ChannelInterpretation aChannelInterpretation)
+    : AudioNode(aContext, aChannelCount, aChannelCountMode,
+                aChannelInterpretation) {}
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioScheduledSourceNode.h b/dom/media/webaudio/AudioScheduledSourceNode.h
new file mode 100644
index 0000000000..0102d4a6e8
--- /dev/null
+++ b/dom/media/webaudio/AudioScheduledSourceNode.h
@@ -0,0 +1,33 @@
+/* -*- 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/. */
+
+#ifndef AudioScheduledSourceNode_h_
+#define AudioScheduledSourceNode_h_
+
+#include "AudioNode.h"
+#include "mozilla/dom/AudioScheduledSourceNodeBinding.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+
+class AudioScheduledSourceNode : public AudioNode {
+ public:
+  virtual void Start(double aWhen, ErrorResult& aRv) = 0;
+  virtual void Stop(double aWhen, ErrorResult& aRv) = 0;
+
+  IMPL_EVENT_HANDLER(ended)
+
+ protected:
+  AudioScheduledSourceNode(AudioContext* aContext, uint32_t aChannelCount,
+                           ChannelCountMode aChannelCountMode,
+                           ChannelInterpretation aChannelInterpretation);
+  virtual ~AudioScheduledSourceNode() = default;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/AudioWorkletGlobalScope.cpp b/dom/media/webaudio/AudioWorkletGlobalScope.cpp
new file mode 100644
index 0000000000..18b0b4fa75
--- /dev/null
+++ b/dom/media/webaudio/AudioWorkletGlobalScope.cpp
@@ -0,0 +1,358 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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 "AudioWorkletGlobalScope.h"
+
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "AudioWorkletImpl.h"
+#include "jsapi.h"
+#include "js/ForOfIterator.h"
+#include "js/PropertyAndElement.h"  // JS_GetProperty
+#include "mozilla/BasePrincipal.h"
+#include "mozilla/dom/AudioWorkletGlobalScopeBinding.h"
+#include "mozilla/dom/AudioWorkletProcessor.h"
+#include "mozilla/dom/BindingCallContext.h"
+#include "mozilla/dom/MessagePort.h"
+#include "mozilla/dom/StructuredCloneHolder.h"
+#include "mozilla/dom/AudioParamDescriptorBinding.h"
+#include "nsPrintfCString.h"
+#include "nsTHashSet.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(AudioWorkletGlobalScope, WorkletGlobalScope,
+                                   mNameToProcessorMap);
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(AudioWorkletGlobalScope)
+NS_INTERFACE_MAP_END_INHERITING(WorkletGlobalScope)
+
+NS_IMPL_ADDREF_INHERITED(AudioWorkletGlobalScope, WorkletGlobalScope)
+NS_IMPL_RELEASE_INHERITED(AudioWorkletGlobalScope, WorkletGlobalScope)
+
+AudioWorkletGlobalScope::AudioWorkletGlobalScope(AudioWorkletImpl* aImpl)
+    : WorkletGlobalScope(aImpl) {}
+
+AudioWorkletImpl* AudioWorkletGlobalScope::Impl() const {
+  return static_cast<AudioWorkletImpl*>(mImpl.get());
+}
+
+bool AudioWorkletGlobalScope::WrapGlobalObject(
+    JSContext* aCx, JS::MutableHandle<JSObject*> aReflector) {
+  // |this| is being exposed to JS and content script will soon be running.
+  // The graph needs a handle on the JSContext so it can interrupt JS.
+  Impl()->DestinationTrack()->Graph()->NotifyJSContext(aCx);
+
+  JS::RealmOptions options = CreateRealmOptions();
+  return AudioWorkletGlobalScope_Binding::Wrap(
+      aCx, this, this, options, BasePrincipal::Cast(mImpl->Principal()),
+      aReflector);
+}
+
+void AudioWorkletGlobalScope::RegisterProcessor(
+    JSContext* aCx, const nsAString& aName,
+    AudioWorkletProcessorConstructor& aProcessorCtor, ErrorResult& aRv) {
+  TRACE_COMMENT("AudioWorkletGlobalScope::RegisterProcessor", "%s",
+                NS_ConvertUTF16toUTF8(aName).get());
+
+  JS::Rooted<JSObject*> processorConstructor(aCx,
+                                             aProcessorCtor.CallableOrNull());
+
+  /**
+   * 1. If the name is the empty string, throw a NotSupportedError
+   *    exception and abort these steps because the empty string is not
+   *    a valid key.
+   */
+  if (aName.IsEmpty()) {
+    aRv.ThrowNotSupportedError("Argument 1 should not be an empty string.");
+    return;
+  }
+
+  /**
+   * 2. If the name exists as a key in the node name to processor
+   *    definition map, throw a NotSupportedError exception and abort
+   *    these steps because registering a definition with a duplicated
+   *    key is not allowed.
+   */
+  if (mNameToProcessorMap.GetWeak(aName)) {
+    // Duplicate names are not allowed
+    aRv.ThrowNotSupportedError(
+        "Argument 1 is invalid: a class with the same name is already "
+        "registered.");
+    return;
+  }
+
+  // We know processorConstructor is callable, so not a WindowProxy or Location.
+  JS::Rooted<JSObject*> constructorUnwrapped(
+      aCx, js::CheckedUnwrapStatic(processorConstructor));
+  if (!constructorUnwrapped) {
+    // If the caller's compartment does not have permission to access the
+    // unwrapped constructor then throw.
+    aRv.ThrowSecurityError("Constructor cannot be called");
+    return;
+  }
+
+  /**
+   * 3. If the result of IsConstructor(argument=processorCtor) is false,
+   *    throw a TypeError and abort these steps.
+   */
+  if (!JS::IsConstructor(constructorUnwrapped)) {
+    aRv.ThrowTypeError<MSG_NOT_CONSTRUCTOR>("Argument 2");
+    return;
+  }
+
+  /**
+   * 4. Let prototype be the result of Get(O=processorCtor, P="prototype").
+   */
+  // The .prototype on the constructor passed could be an "expando" of a
+  // wrapper. So we should get it from wrapper instead of the underlying
+  // object.
+  JS::Rooted<JS::Value> prototype(aCx);
+  if (!JS_GetProperty(aCx, processorConstructor, "prototype", &prototype)) {
+    aRv.NoteJSContextException(aCx);
+    return;
+  }
+
+  /**
+   * 5. If the result of Type(argument=prototype) is not Object, throw a
+   *    TypeError and abort all these steps.
+   */
+  if (!prototype.isObject()) {
+    aRv.ThrowTypeError<MSG_NOT_OBJECT>("processorCtor.prototype");
+    return;
+  }
+  /**
+   * 6. Let parameterDescriptorsValue be the result of Get(O=processorCtor,
+   *    P="parameterDescriptors").
+   */
+  JS::Rooted<JS::Value> descriptors(aCx);
+  if (!JS_GetProperty(aCx, processorConstructor, "parameterDescriptors",
+                      &descriptors)) {
+    aRv.NoteJSContextException(aCx);
+    return;
+  }
+
+  AudioParamDescriptorMap map;
+  /*
+   * 7. If parameterDescriptorsValue is not undefined
+   */
+  if (!descriptors.isUndefined()) {
+    /*
+     * 7.1. Let parameterDescriptorSequence be the result of the conversion
+     *    from parameterDescriptorsValue to an IDL value of type
+     *    sequence<AudioParamDescriptor>.
+     */
+    JS::Rooted<JS::Value> objectValue(aCx, descriptors);
+    JS::ForOfIterator iter(aCx);
+    if (!iter.init(objectValue, JS::ForOfIterator::AllowNonIterable)) {
+      aRv.NoteJSContextException(aCx);
+      return;
+    }
+    if (!iter.valueIsIterable()) {
+      aRv.ThrowTypeError<MSG_CONVERSION_ERROR>(
+          "AudioWorkletProcessor.parameterDescriptors", "sequence");
+      return;
+    }
+    /*
+     * 7.2 and 7.3 (and substeps)
+     */
+    map = DescriptorsFromJS(aCx, &iter, aRv);
+    if (aRv.Failed()) {
+      return;
+    }
+  }
+
+  /**
+   * 8. Append the key-value pair name → processorCtor to node name to processor
+   * constructor map of the associated AudioWorkletGlobalScope.
+   */
+  if (!mNameToProcessorMap.InsertOrUpdate(aName, RefPtr{&aProcessorCtor},
+                                          fallible)) {
+    aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+    return;
+  }
+
+  /**
+   * 9. Queue a task to the control thread to add the key-value pair
+   *     (name - descriptors) to the node name to parameter descriptor
+   *     map of the associated BaseAudioContext.
+   */
+  NS_DispatchToMainThread(NS_NewRunnableFunction(
+      "AudioWorkletGlobalScope: parameter descriptors",
+      [impl = RefPtr{Impl()}, name = nsString(aName),
+       map = std::move(map)]() mutable {
+        AudioNode* destinationNode =
+            impl->DestinationTrack()->Engine()->NodeMainThread();
+        if (!destinationNode) {
+          return;
+        }
+        destinationNode->Context()->SetParamMapForWorkletName(name, &map);
+      }));
+}
+
+uint64_t AudioWorkletGlobalScope::CurrentFrame() const {
+  AudioNodeTrack* destinationTrack = Impl()->DestinationTrack();
+  GraphTime processedTime = destinationTrack->Graph()->ProcessedTime();
+  return destinationTrack->GraphTimeToTrackTime(processedTime);
+}
+
+double AudioWorkletGlobalScope::CurrentTime() const {
+  return static_cast<double>(CurrentFrame()) / SampleRate();
+}
+
+float AudioWorkletGlobalScope::SampleRate() const {
+  return static_cast<float>(Impl()->DestinationTrack()->mSampleRate);
+}
+
+AudioParamDescriptorMap AudioWorkletGlobalScope::DescriptorsFromJS(
+    JSContext* aCx, JS::ForOfIterator* aIter, ErrorResult& aRv) {
+  AudioParamDescriptorMap res;
+  // To check for duplicates
+  nsTHashSet<nsString> namesSet;
+
+  JS::Rooted<JS::Value> nextValue(aCx);
+  bool done = false;
+  size_t i = 0;
+  while (true) {
+    if (!aIter->next(&nextValue, &done)) {
+      aRv.NoteJSContextException(aCx);
+      return AudioParamDescriptorMap();
+    }
+    if (done) {
+      break;
+    }
+
+    BindingCallContext callCx(aCx, "AudioWorkletGlobalScope.registerProcessor");
+    nsPrintfCString sourceDescription("Element %zu in parameterDescriptors", i);
+    i++;
+    AudioParamDescriptor* descriptor = res.AppendElement(fallible);
+    if (!descriptor) {
+      aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+      return AudioParamDescriptorMap();
+    }
+    if (!descriptor->Init(callCx, nextValue, sourceDescription.get())) {
+      aRv.NoteJSContextException(aCx);
+      return AudioParamDescriptorMap();
+    }
+  }
+
+  for (const auto& descriptor : res) {
+    if (namesSet.Contains(descriptor.mName)) {
+      aRv.ThrowNotSupportedError("Duplicated name \""_ns +
+                                 NS_ConvertUTF16toUTF8(descriptor.mName) +
+                                 "\" in parameterDescriptors."_ns);
+      return AudioParamDescriptorMap();
+    }
+
+    if (!namesSet.Insert(descriptor.mName, fallible)) {
+      aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+      return AudioParamDescriptorMap();
+    }
+
+    if (descriptor.mMinValue > descriptor.mMaxValue) {
+      aRv.ThrowInvalidStateError(
+          "In parameterDescriptors, "_ns +
+          NS_ConvertUTF16toUTF8(descriptor.mName) +
+          " minValue should be smaller than maxValue."_ns);
+      return AudioParamDescriptorMap();
+    }
+
+    if (descriptor.mDefaultValue < descriptor.mMinValue ||
+        descriptor.mDefaultValue > descriptor.mMaxValue) {
+      aRv.ThrowInvalidStateError(
+          "In parameterDescriptors, "_ns +
+          NS_ConvertUTF16toUTF8(descriptor.mName) +
+          nsLiteralCString(" defaultValue is out of the range defined by "
+                           "minValue and maxValue."));
+      return AudioParamDescriptorMap();
+    }
+  }
+
+  return res;
+}
+
+bool AudioWorkletGlobalScope::ConstructProcessor(
+    JSContext* aCx, const nsAString& aName,
+    NotNull<StructuredCloneHolder*> aSerializedOptions,
+    UniqueMessagePortId& aPortIdentifier,
+    JS::MutableHandle<JSObject*> aRetProcessor) {
+  TRACE_COMMENT("AudioWorkletProcessor::ConstructProcessor", "%s",
+                NS_ConvertUTF16toUTF8(aName).get());
+  /**
+   * See
+   * https://webaudio.github.io/web-audio-api/#AudioWorkletProcessor-instantiation
+   */
+  ErrorResult rv;
+  /**
+   * 4. Let deserializedPort be the result of
+   *    StructuredDeserialize(serializedPort, the current Realm).
+   */
+  RefPtr<MessagePort> deserializedPort =
+      MessagePort::Create(this, aPortIdentifier, rv);
+  if (NS_WARN_IF(rv.MaybeSetPendingException(aCx))) {
+    return false;
+  }
+  /**
+   * 5. Let deserializedOptions be the result of
+   *    StructuredDeserialize(serializedOptions, the current Realm).
+   */
+  JS::CloneDataPolicy cloneDataPolicy;
+  cloneDataPolicy.allowIntraClusterClonableSharedObjects();
+  cloneDataPolicy.allowSharedMemoryObjects();
+
+  JS::Rooted<JS::Value> deserializedOptions(aCx);
+  aSerializedOptions->Read(this, aCx, &deserializedOptions, cloneDataPolicy,
+                           rv);
+  if (rv.MaybeSetPendingException(aCx)) {
+    return false;
+  }
+  /**
+   * 6. Let processorCtor be the result of looking up processorName on the
+   *    AudioWorkletGlobalScope's node name to processor definition map.
+   */
+  RefPtr<AudioWorkletProcessorConstructor> processorCtor =
+      mNameToProcessorMap.Get(aName);
+  // AudioWorkletNode has already checked the definition exists.
+  // See also https://github.com/WebAudio/web-audio-api/issues/1854
+  MOZ_ASSERT(processorCtor);
+  /**
+   * 7. Store nodeReference and deserializedPort to node reference and
+   *    transferred port of this AudioWorkletGlobalScope's pending processor
+   *    construction data respectively.
+   */
+  // |nodeReference| is not required here because the "processorerror" event
+  // is thrown by WorkletNodeEngine::ConstructProcessor().
+  mPortForProcessor = std::move(deserializedPort);
+  /**
+   * 8. Construct a callback function from processorCtor with the argument
+   *    of deserializedOptions.
+   */
+  // The options were an object before serialization and so will be an object
+  // if deserialization succeeded above.  toObject() asserts.
+  JS::Rooted<JSObject*> options(aCx, &deserializedOptions.toObject());
+  RefPtr<AudioWorkletProcessor> processor = processorCtor->Construct(
+      options, rv, "AudioWorkletProcessor construction",
+      CallbackFunction::eRethrowExceptions);
+  // https://github.com/WebAudio/web-audio-api/issues/2096
+  mPortForProcessor = nullptr;
+  if (rv.MaybeSetPendingException(aCx)) {
+    return false;
+  }
+  JS::Rooted<JS::Value> processorVal(aCx);
+  if (NS_WARN_IF(!ToJSValue(aCx, processor, &processorVal))) {
+    return false;
+  }
+  MOZ_ASSERT(processorVal.isObject());
+  aRetProcessor.set(&processorVal.toObject());
+  return true;
+}
+
+RefPtr<MessagePort> AudioWorkletGlobalScope::TakePortForProcessorCtor() {
+  return std::move(mPortForProcessor);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioWorkletGlobalScope.h b/dom/media/webaudio/AudioWorkletGlobalScope.h
new file mode 100644
index 0000000000..faaa731fa6
--- /dev/null
+++ b/dom/media/webaudio/AudioWorkletGlobalScope.h
@@ -0,0 +1,84 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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/. */
+
+#ifndef mozilla_dom_AudioWorkletGlobalScope_h
+#define mozilla_dom_AudioWorkletGlobalScope_h
+
+#include "mozilla/dom/AudioParamDescriptorMap.h"
+#include "mozilla/dom/FunctionBinding.h"
+#include "mozilla/dom/WorkletGlobalScope.h"
+#include "js/ForOfIterator.h"
+#include "nsRefPtrHashtable.h"
+
+namespace mozilla {
+
+class AudioWorkletImpl;
+
+namespace dom {
+
+class AudioWorkletProcessorConstructor;
+class MessagePort;
+class StructuredCloneHolder;
+class UniqueMessagePortId;
+
+class AudioWorkletGlobalScope final : public WorkletGlobalScope {
+ public:
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioWorkletGlobalScope,
+                                           WorkletGlobalScope);
+
+  explicit AudioWorkletGlobalScope(AudioWorkletImpl* aImpl);
+
+  bool WrapGlobalObject(JSContext* aCx,
+                        JS::MutableHandle<JSObject*> aReflector) override;
+
+  void RegisterProcessor(JSContext* aCx, const nsAString& aName,
+                         AudioWorkletProcessorConstructor& aProcessorCtor,
+                         ErrorResult& aRv);
+
+  AudioWorkletImpl* Impl() const;
+
+  uint64_t CurrentFrame() const;
+
+  double CurrentTime() const;
+
+  float SampleRate() const;
+
+  // If successful, returns true and sets aRetProcessor, which will be in the
+  // compartment for the realm of this global.  Returns false on failure.
+  MOZ_CAN_RUN_SCRIPT
+  bool ConstructProcessor(JSContext* aCx, const nsAString& aName,
+                          NotNull<StructuredCloneHolder*> aSerializedOptions,
+                          UniqueMessagePortId& aPortIdentifier,
+                          JS::MutableHandle<JSObject*> aRetProcessor);
+
+  // Returns null if not called during ConstructProcessor() or if the port has
+  // already been taken.
+  RefPtr<MessagePort> TakePortForProcessorCtor();
+
+ private:
+  ~AudioWorkletGlobalScope() = default;
+
+  // Returns an AudioParamDescriptorMap filled with AudioParamDescriptor
+  // objects, extracted from JS. Returns an empty map in case of error and set
+  // aRv accordingly.
+  AudioParamDescriptorMap DescriptorsFromJS(JSContext* aCx,
+                                            JS::ForOfIterator* aIter,
+                                            ErrorResult& aRv);
+
+  typedef nsRefPtrHashtable<nsStringHashKey, AudioWorkletProcessorConstructor>
+      NodeNameToProcessorDefinitionMap;
+  NodeNameToProcessorDefinitionMap mNameToProcessorMap;
+  // https://webaudio.github.io/web-audio-api/#pending-processor-construction-data-transferred-port
+  // This does not need to be traversed during cycle-collection because it is
+  // only set while this AudioWorkletGlobalScope is on the stack.
+  RefPtr<MessagePort> mPortForProcessor;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+
+#endif  // mozilla_dom_AudioWorkletGlobalScope_h
diff --git a/dom/media/webaudio/AudioWorkletImpl.cpp b/dom/media/webaudio/AudioWorkletImpl.cpp
new file mode 100644
index 0000000000..5eaa128c8a
--- /dev/null
+++ b/dom/media/webaudio/AudioWorkletImpl.cpp
@@ -0,0 +1,88 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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 https://mozilla.org/MPL/2.0/. */
+
+#include "AudioWorkletImpl.h"
+
+#include "AudioContext.h"
+#include "AudioNodeTrack.h"
+#include "GeckoProfiler.h"
+#include "mozilla/dom/AudioWorkletBinding.h"
+#include "mozilla/dom/AudioWorkletGlobalScope.h"
+#include "mozilla/dom/Worklet.h"
+#include "mozilla/dom/WorkletThread.h"
+#include "nsGlobalWindowInner.h"
+
+namespace mozilla {
+
+/* static */ already_AddRefed<dom::Worklet> AudioWorkletImpl::CreateWorklet(
+    dom::AudioContext* aContext, ErrorResult& aRv) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  nsCOMPtr<nsPIDOMWindowInner> window = aContext->GetOwner();
+  if (NS_WARN_IF(!window)) {
+    aRv.Throw(NS_ERROR_FAILURE);
+    return nullptr;
+  }
+  nsCOMPtr<nsIPrincipal> principal =
+      nsGlobalWindowInner::Cast(window)->GetPrincipal();
+  if (NS_WARN_IF(!principal)) {
+    aRv.Throw(NS_ERROR_FAILURE);
+    return nullptr;
+  }
+
+  RefPtr<AudioWorkletImpl> impl =
+      new AudioWorkletImpl(window, principal, aContext->DestinationTrack());
+
+  // The Worklet owns a reference to the AudioContext so as to keep the graph
+  // thread running as long as the Worklet is alive by keeping the
+  // AudioDestinationNode alive.
+  return MakeAndAddRef<dom::Worklet>(window, std::move(impl),
+                                     ToSupports(aContext));
+}
+
+AudioWorkletImpl::AudioWorkletImpl(nsPIDOMWindowInner* aWindow,
+                                   nsIPrincipal* aPrincipal,
+                                   AudioNodeTrack* aDestinationTrack)
+    : WorkletImpl(aWindow, aPrincipal), mDestinationTrack(aDestinationTrack) {}
+
+AudioWorkletImpl::~AudioWorkletImpl() = default;
+
+JSObject* AudioWorkletImpl::WrapWorklet(JSContext* aCx, dom::Worklet* aWorklet,
+                                        JS::Handle<JSObject*> aGivenProto) {
+  MOZ_ASSERT(NS_IsMainThread());
+  return dom::AudioWorklet_Binding::Wrap(aCx, aWorklet, aGivenProto);
+}
+
+nsresult AudioWorkletImpl::SendControlMessage(
+    already_AddRefed<nsIRunnable> aRunnable) {
+  mDestinationTrack->SendRunnable(std::move(aRunnable));
+  return NS_OK;
+}
+
+void AudioWorkletImpl::OnAddModuleStarted() const {
+#ifdef MOZ_GECKO_PROFILER
+  profiler_add_marker(ProfilerStringView("AudioWorklet.addModule"),
+                      geckoprofiler::category::MEDIA_RT,
+                      {MarkerTiming::IntervalStart()});
+#endif
+}
+
+void AudioWorkletImpl::OnAddModulePromiseSettled() const {
+#ifdef MOZ_GECKO_PROFILER
+  profiler_add_marker(ProfilerStringView("AudioWorklet.addModule"),
+                      geckoprofiler::category::MEDIA_RT,
+                      {MarkerTiming::IntervalEnd()});
+#endif
+}
+
+already_AddRefed<dom::WorkletGlobalScope>
+AudioWorkletImpl::ConstructGlobalScope() {
+  dom::WorkletThread::AssertIsOnWorkletThread();
+
+  return MakeAndAddRef<dom::AudioWorkletGlobalScope>(this);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webaudio/AudioWorkletImpl.h b/dom/media/webaudio/AudioWorkletImpl.h
new file mode 100644
index 0000000000..39f347fdb2
--- /dev/null
+++ b/dom/media/webaudio/AudioWorkletImpl.h
@@ -0,0 +1,63 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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 https://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioWorkletImpl_h
+#define AudioWorkletImpl_h
+
+#include "mozilla/dom/WorkletImpl.h"
+#include "mozilla/dom/AudioWorkletGlobalScope.h"
+
+namespace mozilla {
+
+class AudioNodeTrack;
+
+namespace dom {
+class AudioContext;
+}
+
+class AudioWorkletImpl final : public WorkletImpl {
+ public:
+  // Methods for parent thread only:
+
+  static already_AddRefed<dom::Worklet> CreateWorklet(
+      dom::AudioContext* aContext, ErrorResult& aRv);
+
+  JSObject* WrapWorklet(JSContext* aCx, dom::Worklet* aWorklet,
+                        JS::Handle<JSObject*> aGivenProto) override;
+
+  nsresult SendControlMessage(already_AddRefed<nsIRunnable> aRunnable) override;
+
+  nsContentPolicyType ContentPolicyType() const override {
+    return nsIContentPolicy::TYPE_INTERNAL_AUDIOWORKLET;
+  }
+
+  // Execution thread only.
+  dom::AudioWorkletGlobalScope* GetGlobalScope() {
+    return static_cast<dom::AudioWorkletGlobalScope*>(
+        WorkletImpl::GetGlobalScope());
+  }
+
+  // Any thread:
+  AudioNodeTrack* DestinationTrack() const { return mDestinationTrack; }
+
+  void OnAddModuleStarted() const override;
+  void OnAddModulePromiseSettled() const override;
+
+ protected:
+  // Execution thread only.
+  already_AddRefed<dom::WorkletGlobalScope> ConstructGlobalScope() override;
+
+ private:
+  AudioWorkletImpl(nsPIDOMWindowInner* aWindow, nsIPrincipal* aPrincipal,
+                   AudioNodeTrack* aDestinationTrack);
+  ~AudioWorkletImpl();
+
+  const RefPtr<AudioNodeTrack> mDestinationTrack;
+};
+
+}  // namespace mozilla
+
+#endif  // AudioWorkletImpl_h
diff --git a/dom/media/webaudio/AudioWorkletNode.cpp b/dom/media/webaudio/AudioWorkletNode.cpp
new file mode 100644
index 0000000000..76a3185f2c
--- /dev/null
+++ b/dom/media/webaudio/AudioWorkletNode.cpp
@@ -0,0 +1,899 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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 https://mozilla.org/MPL/2.0/. */
+
+#include "AudioWorkletNode.h"
+
+#include "AudioNodeEngine.h"
+#include "AudioParamMap.h"
+#include "AudioWorkletImpl.h"
+#include "js/Array.h"  // JS::{Get,Set}ArrayLength, JS::NewArrayLength
+#include "js/CallAndConstruct.h"  // JS::Call, JS::IsCallable
+#include "js/Exception.h"
+#include "js/experimental/TypedData.h"  // JS_NewFloat32Array, JS_GetFloat32ArrayData, JS_GetTypedArrayLength, JS_GetArrayBufferViewBuffer
+#include "js/PropertyAndElement.h"  // JS_DefineElement, JS_DefineUCProperty, JS_GetProperty
+#include "mozilla/dom/AudioWorkletNodeBinding.h"
+#include "mozilla/dom/AudioParamMapBinding.h"
+#include "mozilla/dom/AutoEntryScript.h"
+#include "mozilla/dom/RootedDictionary.h"
+#include "mozilla/dom/ErrorEvent.h"
+#include "mozilla/dom/Worklet.h"
+#include "nsIScriptGlobalObject.h"
+#include "AudioParam.h"
+#include "AudioDestinationNode.h"
+#include "mozilla/dom/MessageChannel.h"
+#include "mozilla/dom/MessagePort.h"
+#include "mozilla/ScopeExit.h"
+#include "nsReadableUtils.h"
+#include "mozilla/Span.h"
+#include "PlayingRefChangeHandler.h"
+#include "nsPrintfCString.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_ISUPPORTS_CYCLE_COLLECTION_INHERITED_0(AudioWorkletNode, AudioNode)
+NS_IMPL_CYCLE_COLLECTION_INHERITED(AudioWorkletNode, AudioNode, mPort,
+                                   mParameters)
+
+struct NamedAudioParamTimeline {
+  explicit NamedAudioParamTimeline(const AudioParamDescriptor& aParamDescriptor)
+      : mName(aParamDescriptor.mName),
+        mTimeline(aParamDescriptor.mDefaultValue) {}
+  const nsString mName;
+  AudioParamTimeline mTimeline;
+};
+
+struct ProcessorErrorDetails {
+  ProcessorErrorDetails() : mLineno(0), mColno(0) {}
+  // Line number (1-origin).
+  unsigned mLineno;
+  // Column number in UTF-16 code units (1-origin).
+  unsigned mColno;
+  nsString mFilename;
+  nsString mMessage;
+};
+
+class WorkletNodeEngine final : public AudioNodeEngine {
+ public:
+  WorkletNodeEngine(AudioWorkletNode* aNode,
+                    AudioDestinationNode* aDestinationNode,
+                    nsTArray<NamedAudioParamTimeline>&& aParamTimelines,
+                    const Optional<Sequence<uint32_t>>& aOutputChannelCount)
+      : AudioNodeEngine(aNode),
+        mDestination(aDestinationNode->Track()),
+        mParamTimelines(std::move(aParamTimelines)) {
+    if (aOutputChannelCount.WasPassed()) {
+      mOutputChannelCount = aOutputChannelCount.Value();
+    }
+  }
+
+  MOZ_CAN_RUN_SCRIPT
+  void ConstructProcessor(AudioWorkletImpl* aWorkletImpl,
+                          const nsAString& aName,
+                          NotNull<StructuredCloneHolder*> aSerializedOptions,
+                          UniqueMessagePortId& aPortIdentifier,
+                          AudioNodeTrack* aTrack);
+
+  void RecvTimelineEvent(uint32_t aIndex, AudioParamEvent& aEvent) override {
+    MOZ_ASSERT(mDestination);
+    aEvent.ConvertToTicks(mDestination);
+
+    if (aIndex < mParamTimelines.Length()) {
+      mParamTimelines[aIndex].mTimeline.InsertEvent<int64_t>(aEvent);
+    } else {
+      NS_ERROR("Bad WorkletNodeEngine timeline event index");
+    }
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    MOZ_ASSERT(InputCount() <= 1);
+    MOZ_ASSERT(OutputCount() <= 1);
+    TRACE("WorkletNodeEngine::ProcessBlock");
+    ProcessBlocksOnPorts(aTrack, aFrom, Span(&aInput, InputCount()),
+                         Span(aOutput, OutputCount()), aFinished);
+  }
+
+  void ProcessBlocksOnPorts(AudioNodeTrack* aTrack, GraphTime aFrom,
+                            Span<const AudioBlock> aInput,
+                            Span<AudioBlock> aOutput, bool* aFinished) override;
+
+  void OnGraphThreadDone() override { ReleaseJSResources(); }
+
+  bool IsActive() const override { return mKeepEngineActive; }
+
+  // Vector<T> supports non-memmovable types such as PersistentRooted
+  // (without any need to jump through hoops like
+  // MOZ_DECLARE_RELOCATE_USING_MOVE_CONSTRUCTOR_FOR_TEMPLATE for nsTArray).
+  // PersistentRooted is used because these AudioWorkletGlobalScope scope
+  // objects may be kept alive as long as the AudioWorkletNode in the
+  // main-thread global.
+  struct Channels {
+    Vector<JS::PersistentRooted<JSObject*>, GUESS_AUDIO_CHANNELS>
+        mFloat32Arrays;
+    JS::PersistentRooted<JSObject*> mJSArray;
+    // For SetArrayElements():
+    operator JS::Handle<JSObject*>() const { return mJSArray; }
+  };
+  struct Ports {
+    Vector<Channels, 1> mPorts;
+    JS::PersistentRooted<JSObject*> mJSArray;
+  };
+  struct ParameterValues {
+    Vector<JS::PersistentRooted<JSObject*>> mFloat32Arrays;
+    JS::PersistentRooted<JSObject*> mJSObject;
+  };
+
+ private:
+  size_t ParameterCount() { return mParamTimelines.Length(); }
+  void SendProcessorError(AudioNodeTrack* aTrack, JSContext* aCx);
+  bool CallProcess(AudioNodeTrack* aTrack, JSContext* aCx,
+                   JS::Handle<JS::Value> aCallable);
+  void ProduceSilence(AudioNodeTrack* aTrack, Span<AudioBlock> aOutput);
+  void SendErrorToMainThread(AudioNodeTrack* aTrack,
+                             const ProcessorErrorDetails& aDetails);
+
+  void ReleaseJSResources() {
+    mInputs.mPorts.clearAndFree();
+    mOutputs.mPorts.clearAndFree();
+    mParameters.mFloat32Arrays.clearAndFree();
+    mInputs.mJSArray.reset();
+    mOutputs.mJSArray.reset();
+    mParameters.mJSObject.reset();
+    mGlobal = nullptr;
+    // This is equivalent to setting [[callable process]] to false.
+    mProcessor.reset();
+  }
+
+  nsCString mProcessorName;
+  RefPtr<AudioNodeTrack> mDestination;
+  nsTArray<uint32_t> mOutputChannelCount;
+  nsTArray<NamedAudioParamTimeline> mParamTimelines;
+  // The AudioWorkletGlobalScope-associated objects referenced from
+  // WorkletNodeEngine are typically kept alive as long as the
+  // AudioWorkletNode in the main-thread global.  The objects must be released
+  // on the rendering thread, which usually happens simply because
+  // AudioWorkletNode is such that the last AudioNodeTrack reference is
+  // released by the MTG.  That occurs on the rendering thread except during
+  // process shutdown, in which case NotifyForcedShutdown() is called on the
+  // rendering thread.
+  //
+  // mInputs, mOutputs and mParameters keep references to all objects passed to
+  // process(), for reuse of the same objects.  The JS objects are all in the
+  // compartment of the realm of mGlobal.  Properties on the objects may be
+  // replaced by script, so don't assume that getting indexed properties on the
+  // JS arrays will return the same objects.  Only objects and buffers created
+  // by the implementation are modified or read by the implementation.
+  Ports mInputs;
+  Ports mOutputs;
+  ParameterValues mParameters;
+
+  RefPtr<AudioWorkletGlobalScope> mGlobal;
+  JS::PersistentRooted<JSObject*> mProcessor;
+
+  // mProcessorIsActive is named [[active source]] in the spec.
+  // It is initially true and so at least the first process()
+  // call will not be skipped when there are no active inputs.
+  bool mProcessorIsActive = true;
+  // mKeepEngineActive ensures another call to ProcessBlocksOnPorts(), even if
+  // there are no active inputs.  Its transitions to false lag those of
+  // mProcessorIsActive by one call to ProcessBlocksOnPorts() so that
+  // downstream engines can addref their nodes before this engine's node is
+  // released.
+  bool mKeepEngineActive = true;
+};
+
+void WorkletNodeEngine::SendErrorToMainThread(
+    AudioNodeTrack* aTrack, const ProcessorErrorDetails& aDetails) {
+  RefPtr<AudioNodeTrack> track = aTrack;
+  NS_DispatchToMainThread(NS_NewRunnableFunction(
+      "WorkletNodeEngine::SendProcessorError",
+      [track = std::move(track), aDetails]() mutable {
+        AudioWorkletNode* node =
+            static_cast<AudioWorkletNode*>(track->Engine()->NodeMainThread());
+        if (!node) {
+          return;
+        }
+        node->DispatchProcessorErrorEvent(aDetails);
+      }));
+}
+
+void WorkletNodeEngine::SendProcessorError(AudioNodeTrack* aTrack,
+                                           JSContext* aCx) {
+  // Note that once an exception is thrown, the processor will output silence
+  // throughout its lifetime.
+  ReleaseJSResources();
+  // The processor errored out while getting a context, try to tell the node
+  // anyways.
+  if (!aCx || !JS_IsExceptionPending(aCx)) {
+    ProcessorErrorDetails details;
+    details.mMessage.Assign(u"Unknown processor error");
+    SendErrorToMainThread(aTrack, details);
+    return;
+  }
+
+  JS::ExceptionStack exnStack(aCx);
+  if (JS::StealPendingExceptionStack(aCx, &exnStack)) {
+    JS::ErrorReportBuilder jsReport(aCx);
+    if (!jsReport.init(aCx, exnStack,
+                       JS::ErrorReportBuilder::WithSideEffects)) {
+      ProcessorErrorDetails details;
+      details.mMessage.Assign(u"Unknown processor error");
+      SendErrorToMainThread(aTrack, details);
+      // Set the exception and stack back to have it in the console with a stack
+      // trace.
+      JS::SetPendingExceptionStack(aCx, exnStack);
+      return;
+    }
+
+    ProcessorErrorDetails details;
+
+    CopyUTF8toUTF16(
+        mozilla::MakeStringSpan(jsReport.report()->filename.c_str()),
+        details.mFilename);
+
+    xpc::ErrorReport::ErrorReportToMessageString(jsReport.report(),
+                                                 details.mMessage);
+    details.mLineno = jsReport.report()->lineno;
+    details.mColno = jsReport.report()->column.oneOriginValue();
+    MOZ_ASSERT(!jsReport.report()->isMuted);
+
+    SendErrorToMainThread(aTrack, details);
+
+    // Set the exception and stack back to have it in the console with a stack
+    // trace.
+    JS::SetPendingExceptionStack(aCx, exnStack);
+  } else {
+    NS_WARNING("No exception, but processor errored out?");
+  }
+}
+
+void WorkletNodeEngine::ConstructProcessor(
+    AudioWorkletImpl* aWorkletImpl, const nsAString& aName,
+    NotNull<StructuredCloneHolder*> aSerializedOptions,
+    UniqueMessagePortId& aPortIdentifier, AudioNodeTrack* aTrack) {
+  MOZ_ASSERT(mInputs.mPorts.empty() && mOutputs.mPorts.empty());
+  RefPtr<AudioWorkletGlobalScope> global = aWorkletImpl->GetGlobalScope();
+  if (!global) {
+    // A global was previously used to register this kind of processor.  If it
+    // no longer exists now, that is because the document is going away and so
+    // there is no need to send an error.
+    return;
+  }
+  AutoJSAPI api;
+  if (NS_WARN_IF(!api.Init(global))) {
+    SendProcessorError(aTrack, nullptr);
+    return;
+  }
+  mProcessorName = NS_ConvertUTF16toUTF8(aName);
+  JSContext* cx = api.cx();
+  mProcessor.init(cx);
+  if (!global->ConstructProcessor(cx, aName, aSerializedOptions,
+                                  aPortIdentifier, &mProcessor) ||
+      // mInputs and mOutputs outer arrays are fixed length and so much of the
+      // initialization need only be performed once (i.e. here).
+      NS_WARN_IF(!mInputs.mPorts.growBy(InputCount())) ||
+      NS_WARN_IF(!mOutputs.mPorts.growBy(OutputCount()))) {
+    SendProcessorError(aTrack, cx);
+    return;
+  }
+  mGlobal = std::move(global);
+  mInputs.mJSArray.init(cx);
+  mOutputs.mJSArray.init(cx);
+  for (auto& port : mInputs.mPorts) {
+    port.mJSArray.init(cx);
+  }
+  for (auto& port : mOutputs.mPorts) {
+    port.mJSArray.init(cx);
+  }
+  JSObject* object = JS_NewPlainObject(cx);
+  if (NS_WARN_IF(!object)) {
+    SendProcessorError(aTrack, cx);
+    return;
+  }
+
+  mParameters.mJSObject.init(cx, object);
+  if (NS_WARN_IF(!mParameters.mFloat32Arrays.growBy(ParameterCount()))) {
+    SendProcessorError(aTrack, cx);
+    return;
+  }
+  for (size_t i = 0; i < mParamTimelines.Length(); i++) {
+    auto& float32ArraysRef = mParameters.mFloat32Arrays;
+    float32ArraysRef[i].init(cx);
+    JSObject* array = JS_NewFloat32Array(cx, WEBAUDIO_BLOCK_SIZE);
+    if (NS_WARN_IF(!array)) {
+      SendProcessorError(aTrack, cx);
+      return;
+    }
+
+    float32ArraysRef[i] = array;
+    if (NS_WARN_IF(!JS_DefineUCProperty(
+            cx, mParameters.mJSObject, mParamTimelines[i].mName.get(),
+            mParamTimelines[i].mName.Length(), float32ArraysRef[i],
+            JSPROP_ENUMERATE))) {
+      SendProcessorError(aTrack, cx);
+      return;
+    }
+  }
+  if (NS_WARN_IF(!JS_FreezeObject(cx, mParameters.mJSObject))) {
+    SendProcessorError(aTrack, cx);
+    return;
+  }
+}
+
+// Type T should support the length() and operator[]() methods and the return
+// type of |operator[]() const| should support conversion to Handle<JSObject*>.
+template <typename T>
+static bool SetArrayElements(JSContext* aCx, const T& aElements,
+                             JS::Handle<JSObject*> aArray) {
+  for (size_t i = 0; i < aElements.length(); ++i) {
+    if (!JS_DefineElement(aCx, aArray, i, aElements[i], JSPROP_ENUMERATE)) {
+      return false;
+    }
+  }
+
+  return true;
+}
+
+template <typename T>
+static bool PrepareArray(JSContext* aCx, const T& aElements,
+                         JS::MutableHandle<JSObject*> aArray) {
+  size_t length = aElements.length();
+  if (aArray) {
+    // Attempt to reuse.
+    uint32_t oldLength;
+    if (JS::GetArrayLength(aCx, aArray, &oldLength) &&
+        (oldLength == length || JS::SetArrayLength(aCx, aArray, length)) &&
+        SetArrayElements(aCx, aElements, aArray)) {
+      return true;
+    }
+    // Script may have frozen the array.  Try again with a new Array.
+    JS_ClearPendingException(aCx);
+  }
+  JSObject* array = JS::NewArrayObject(aCx, length);
+  if (NS_WARN_IF(!array)) {
+    return false;
+  }
+  aArray.set(array);
+  return SetArrayElements(aCx, aElements, aArray);
+}
+
+enum class ArrayElementInit { None, Zero };
+
+// Exactly when to create new Float32Array and Array objects is not specified.
+// This approach aims to minimize garbage creation, while continuing to
+// function after objects are modified by content.
+// See https://github.com/WebAudio/web-audio-api/issues/1934 and
+// https://github.com/WebAudio/web-audio-api/issues/1933
+static bool PrepareBufferArrays(JSContext* aCx, Span<const AudioBlock> aBlocks,
+                                WorkletNodeEngine::Ports* aPorts,
+                                ArrayElementInit aInit) {
+  MOZ_ASSERT(aBlocks.Length() == aPorts->mPorts.length());
+  for (size_t i = 0; i < aBlocks.Length(); ++i) {
+    size_t channelCount = aBlocks[i].ChannelCount();
+    WorkletNodeEngine::Channels& portRef = aPorts->mPorts[i];
+
+    auto& float32ArraysRef = portRef.mFloat32Arrays;
+    for (auto& channelRef : float32ArraysRef) {
+      size_t length = JS_GetTypedArrayLength(channelRef);
+      if (length != WEBAUDIO_BLOCK_SIZE) {
+        // Script has detached array buffers.  Create new objects.
+        JSObject* array = JS_NewFloat32Array(aCx, WEBAUDIO_BLOCK_SIZE);
+        if (NS_WARN_IF(!array)) {
+          return false;
+        }
+        channelRef = array;
+      } else if (aInit == ArrayElementInit::Zero) {
+        // Need only zero existing arrays as new arrays are already zeroed.
+        JS::AutoCheckCannotGC nogc;
+        bool isShared;
+        float* elementData =
+            JS_GetFloat32ArrayData(channelRef, &isShared, nogc);
+        MOZ_ASSERT(!isShared);  // Was created as unshared
+        std::fill_n(elementData, WEBAUDIO_BLOCK_SIZE, 0.0f);
+      }
+    }
+    // Enlarge if necessary...
+    if (NS_WARN_IF(!float32ArraysRef.reserve(channelCount))) {
+      return false;
+    }
+    while (float32ArraysRef.length() < channelCount) {
+      JSObject* array = JS_NewFloat32Array(aCx, WEBAUDIO_BLOCK_SIZE);
+      if (NS_WARN_IF(!array)) {
+        return false;
+      }
+      float32ArraysRef.infallibleEmplaceBack(aCx, array);
+    }
+    // ... or shrink if necessary.
+    float32ArraysRef.shrinkTo(channelCount);
+
+    if (NS_WARN_IF(!PrepareArray(aCx, float32ArraysRef, &portRef.mJSArray))) {
+      return false;
+    }
+  }
+
+  return !(NS_WARN_IF(!PrepareArray(aCx, aPorts->mPorts, &aPorts->mJSArray)));
+}
+
+// This runs JS script.  MediaTrackGraph control messages, which would
+// potentially destroy the WorkletNodeEngine and its AudioNodeTrack, cannot
+// be triggered by script.  They are not run from an nsIThread event loop and
+// do not run until after ProcessBlocksOnPorts() has returned.
+bool WorkletNodeEngine::CallProcess(AudioNodeTrack* aTrack, JSContext* aCx,
+                                    JS::Handle<JS::Value> aCallable) {
+  TRACE_COMMENT("AudioWorkletNodeEngine::CallProcess", mProcessorName.get());
+
+  JS::RootedVector<JS::Value> argv(aCx);
+  if (NS_WARN_IF(!argv.resize(3))) {
+    return false;
+  }
+  argv[0].setObject(*mInputs.mJSArray);
+  argv[1].setObject(*mOutputs.mJSArray);
+  argv[2].setObject(*mParameters.mJSObject);
+  JS::Rooted<JS::Value> rval(aCx);
+  if (!JS::Call(aCx, mProcessor, aCallable, argv, &rval)) {
+    return false;
+  }
+
+  mProcessorIsActive = JS::ToBoolean(rval);
+  // Transitions of mProcessorIsActive to false do not trigger
+  // PlayingRefChangeHandler::RELEASE until silence is produced in the next
+  // block.  This allows downstream engines receiving this non-silence block
+  // to take a reference to their nodes before this engine's node releases its
+  // down node references.
+  if (mProcessorIsActive && !mKeepEngineActive) {
+    mKeepEngineActive = true;
+    RefPtr<PlayingRefChangeHandler> refchanged =
+        new PlayingRefChangeHandler(aTrack, PlayingRefChangeHandler::ADDREF);
+    aTrack->Graph()->DispatchToMainThreadStableState(refchanged.forget());
+  }
+  return true;
+}
+
+void WorkletNodeEngine::ProduceSilence(AudioNodeTrack* aTrack,
+                                       Span<AudioBlock> aOutput) {
+  if (mKeepEngineActive) {
+    mKeepEngineActive = false;
+    aTrack->ScheduleCheckForInactive();
+    RefPtr<PlayingRefChangeHandler> refchanged =
+        new PlayingRefChangeHandler(aTrack, PlayingRefChangeHandler::RELEASE);
+    aTrack->Graph()->DispatchToMainThreadStableState(refchanged.forget());
+  }
+  for (AudioBlock& output : aOutput) {
+    output.SetNull(WEBAUDIO_BLOCK_SIZE);
+  }
+}
+
+void WorkletNodeEngine::ProcessBlocksOnPorts(AudioNodeTrack* aTrack,
+                                             GraphTime aFrom,
+                                             Span<const AudioBlock> aInput,
+                                             Span<AudioBlock> aOutput,
+                                             bool* aFinished) {
+  MOZ_ASSERT(aInput.Length() == InputCount());
+  MOZ_ASSERT(aOutput.Length() == OutputCount());
+  TRACE("WorkletNodeEngine::ProcessBlocksOnPorts");
+
+  bool isSilent = true;
+  if (mProcessor) {
+    if (mProcessorIsActive) {
+      isSilent = false;  // call process()
+    } else {             // [[active source]] is false.
+      // Call process() only if an input is actively processing.
+      for (const AudioBlock& input : aInput) {
+        if (!input.IsNull()) {
+          isSilent = false;
+          break;
+        }
+      }
+    }
+  }
+  if (isSilent) {
+    ProduceSilence(aTrack, aOutput);
+    return;
+  }
+
+  if (!mOutputChannelCount.IsEmpty()) {
+    MOZ_ASSERT(mOutputChannelCount.Length() == aOutput.Length());
+    for (size_t o = 0; o < aOutput.Length(); ++o) {
+      aOutput[o].AllocateChannels(mOutputChannelCount[o]);
+    }
+  } else if (aInput.Length() == 1 && aOutput.Length() == 1) {
+    uint32_t channelCount = std::max(aInput[0].ChannelCount(), 1U);
+    aOutput[0].AllocateChannels(channelCount);
+  } else {
+    for (AudioBlock& output : aOutput) {
+      output.AllocateChannels(1);
+    }
+  }
+
+  AutoEntryScript aes(mGlobal, "Worklet Process");
+  JSContext* cx = aes.cx();
+  auto produceSilenceWithError = MakeScopeExit([this, aTrack, cx, &aOutput] {
+    SendProcessorError(aTrack, cx);
+    ProduceSilence(aTrack, aOutput);
+  });
+
+  JS::Rooted<JS::Value> process(cx);
+  if (!JS_GetProperty(cx, mProcessor, "process", &process) ||
+      !process.isObject() || !JS::IsCallable(&process.toObject()) ||
+      !PrepareBufferArrays(cx, aInput, &mInputs, ArrayElementInit::None) ||
+      !PrepareBufferArrays(cx, aOutput, &mOutputs, ArrayElementInit::Zero)) {
+    // process() not callable or OOM.
+    return;
+  }
+
+  // Copy input values to JS objects.
+  for (size_t i = 0; i < aInput.Length(); ++i) {
+    const AudioBlock& input = aInput[i];
+    size_t channelCount = input.ChannelCount();
+    if (channelCount == 0) {
+      // Null blocks have AUDIO_FORMAT_SILENCE.
+      // Don't call ChannelData<float>().
+      continue;
+    }
+    float volume = input.mVolume;
+    const auto& channelData = input.ChannelData<float>();
+    const auto& float32Arrays = mInputs.mPorts[i].mFloat32Arrays;
+    JS::AutoCheckCannotGC nogc;
+    for (size_t c = 0; c < channelCount; ++c) {
+      bool isShared;
+      float* dest = JS_GetFloat32ArrayData(float32Arrays[c], &isShared, nogc);
+      MOZ_ASSERT(!isShared);  // Was created as unshared
+      AudioBlockCopyChannelWithScale(channelData[c], volume, dest);
+    }
+  }
+
+  TrackTime tick = mDestination->GraphTimeToTrackTime(aFrom);
+  // Compute and copy parameter values to JS objects.
+  for (size_t i = 0; i < mParamTimelines.Length(); ++i) {
+    const auto& float32Arrays = mParameters.mFloat32Arrays[i];
+    size_t length = JS_GetTypedArrayLength(float32Arrays);
+
+    // If the Float32Array that is supposed to hold the values for a particular
+    // AudioParam has been detached, error out. This is being worked on in
+    // https://github.com/WebAudio/web-audio-api/issues/1933 and
+    // https://bugzilla.mozilla.org/show_bug.cgi?id=1619486
+    if (length != WEBAUDIO_BLOCK_SIZE) {
+      return;
+    }
+    JS::AutoCheckCannotGC nogc;
+    bool isShared;
+    float* dest = JS_GetFloat32ArrayData(float32Arrays, &isShared, nogc);
+    MOZ_ASSERT(!isShared);  // Was created as unshared
+
+    size_t frames =
+        mParamTimelines[i].mTimeline.HasSimpleValue() ? 1 : WEBAUDIO_BLOCK_SIZE;
+    mParamTimelines[i].mTimeline.GetValuesAtTime(tick, dest, frames);
+    // https://bugzilla.mozilla.org/show_bug.cgi?id=1616599
+    if (frames == 1) {
+      std::fill_n(dest + 1, WEBAUDIO_BLOCK_SIZE - 1, dest[0]);
+    }
+  }
+
+  if (!CallProcess(aTrack, cx, process)) {
+    // An exception occurred.
+    /**
+     * https://webaudio.github.io/web-audio-api/#dom-audioworkletnode-onprocessorerror
+     * Note that once an exception is thrown, the processor will output silence
+     * throughout its lifetime.
+     */
+    return;
+  }
+
+  // Copy output values from JS objects.
+  for (size_t o = 0; o < aOutput.Length(); ++o) {
+    AudioBlock* output = &aOutput[o];
+    size_t channelCount = output->ChannelCount();
+    const auto& float32Arrays = mOutputs.mPorts[o].mFloat32Arrays;
+    for (size_t c = 0; c < channelCount; ++c) {
+      size_t length = JS_GetTypedArrayLength(float32Arrays[c]);
+      if (length != WEBAUDIO_BLOCK_SIZE) {
+        // ArrayBuffer has been detached.  Behavior is unspecified.
+        // https://github.com/WebAudio/web-audio-api/issues/1933 and
+        // https://bugzilla.mozilla.org/show_bug.cgi?id=1619486
+        return;
+      }
+      JS::AutoCheckCannotGC nogc;
+      bool isShared;
+      const float* src =
+          JS_GetFloat32ArrayData(float32Arrays[c], &isShared, nogc);
+      MOZ_ASSERT(!isShared);  // Was created as unshared
+      PodCopy(output->ChannelFloatsForWrite(c), src, WEBAUDIO_BLOCK_SIZE);
+    }
+  }
+
+  produceSilenceWithError.release();  // have output and no error
+}
+
+AudioWorkletNode::AudioWorkletNode(AudioContext* aAudioContext,
+                                   const nsAString& aName,
+                                   const AudioWorkletNodeOptions& aOptions)
+    : AudioNode(aAudioContext, 2, ChannelCountMode::Max,
+                ChannelInterpretation::Speakers),
+      mNodeName(aName),
+      mInputCount(aOptions.mNumberOfInputs),
+      mOutputCount(aOptions.mNumberOfOutputs) {}
+
+void AudioWorkletNode::InitializeParameters(
+    nsTArray<NamedAudioParamTimeline>* aParamTimelines, ErrorResult& aRv) {
+  MOZ_ASSERT(!mParameters, "Only initialize the `parameters` attribute once.");
+  MOZ_ASSERT(aParamTimelines);
+
+  AudioContext* context = Context();
+  const AudioParamDescriptorMap* parameterDescriptors =
+      context->GetParamMapForWorkletName(mNodeName);
+  MOZ_ASSERT(parameterDescriptors);
+
+  size_t audioParamIndex = 0;
+  aParamTimelines->SetCapacity(parameterDescriptors->Length());
+
+  for (size_t i = 0; i < parameterDescriptors->Length(); i++) {
+    auto& paramEntry = (*parameterDescriptors)[i];
+    CreateAudioParam(audioParamIndex++, paramEntry.mName,
+                     paramEntry.mDefaultValue, paramEntry.mMinValue,
+                     paramEntry.mMaxValue);
+    aParamTimelines->AppendElement(paramEntry);
+  }
+}
+
+void AudioWorkletNode::SendParameterData(
+    const Optional<Record<nsString, double>>& aParameterData) {
+  MOZ_ASSERT(mTrack, "This method only works if the track has been created.");
+  nsAutoString name;
+  if (aParameterData.WasPassed()) {
+    const auto& paramData = aParameterData.Value();
+    for (const auto& paramDataEntry : paramData.Entries()) {
+      for (auto& audioParam : mParams) {
+        audioParam->GetName(name);
+        if (paramDataEntry.mKey.Equals(name)) {
+          audioParam->SetInitialValue(paramDataEntry.mValue);
+        }
+      }
+    }
+  }
+}
+
+/* static */
+already_AddRefed<AudioWorkletNode> AudioWorkletNode::Constructor(
+    const GlobalObject& aGlobal, AudioContext& aAudioContext,
+    const nsAString& aName, const AudioWorkletNodeOptions& aOptions,
+    ErrorResult& aRv) {
+  TRACE_COMMENT("AudioWorkletNode::Constructor", "%s",
+                NS_ConvertUTF16toUTF8(aName).get());
+  /**
+   * 1. If nodeName does not exist as a key in the BaseAudioContext’s node
+   *    name to parameter descriptor map, throw a InvalidStateError exception
+   *    and abort these steps.
+   */
+  const AudioParamDescriptorMap* parameterDescriptors =
+      aAudioContext.GetParamMapForWorkletName(aName);
+  if (!parameterDescriptors) {
+    // Not using nsPrintfCString in case aName has embedded nulls.
+    aRv.ThrowInvalidStateError("Unknown AudioWorklet name '"_ns +
+                               NS_ConvertUTF16toUTF8(aName) + "'"_ns);
+    return nullptr;
+  }
+
+  // See https://github.com/WebAudio/web-audio-api/issues/2074 for ordering.
+  RefPtr<AudioWorkletNode> audioWorkletNode =
+      new AudioWorkletNode(&aAudioContext, aName, aOptions);
+  audioWorkletNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  /**
+   * 3. Configure input, output and output channels of node with options.
+   */
+  if (aOptions.mNumberOfInputs == 0 && aOptions.mNumberOfOutputs == 0) {
+    aRv.ThrowNotSupportedError(
+        "Must have nonzero numbers of inputs or outputs");
+    return nullptr;
+  }
+
+  if (aOptions.mOutputChannelCount.WasPassed()) {
+    /**
+     * 1. If any value in outputChannelCount is zero or greater than the
+     *    implementation’s maximum number of channels, throw a
+     *    NotSupportedError and abort the remaining steps.
+     */
+    for (uint32_t channelCount : aOptions.mOutputChannelCount.Value()) {
+      if (channelCount == 0 || channelCount > WebAudioUtils::MaxChannelCount) {
+        aRv.ThrowNotSupportedError(
+            nsPrintfCString("Channel count (%u) must be in the range [1, max "
+                            "supported channel count]",
+                            channelCount));
+        return nullptr;
+      }
+    }
+    /**
+     * 2. If the length of outputChannelCount does not equal numberOfOutputs,
+     *    throw an IndexSizeError and abort the remaining steps.
+     */
+    if (aOptions.mOutputChannelCount.Value().Length() !=
+        aOptions.mNumberOfOutputs) {
+      aRv.ThrowIndexSizeError(
+          nsPrintfCString("Length of outputChannelCount (%zu) does not match "
+                          "numberOfOutputs (%u)",
+                          aOptions.mOutputChannelCount.Value().Length(),
+                          aOptions.mNumberOfOutputs));
+      return nullptr;
+    }
+  }
+  // MTG does not support more than UINT16_MAX inputs or outputs.
+  if (aOptions.mNumberOfInputs > UINT16_MAX) {
+    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>("numberOfInputs");
+    return nullptr;
+  }
+  if (aOptions.mNumberOfOutputs > UINT16_MAX) {
+    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>("numberOfOutputs");
+    return nullptr;
+  }
+
+  /**
+   * 4. Let messageChannel be a new MessageChannel.
+   */
+  RefPtr<MessageChannel> messageChannel =
+      MessageChannel::Constructor(aGlobal, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+  /* 5. Let nodePort be the value of messageChannel’s port1 attribute.
+   * 6. Let processorPortOnThisSide be the value of messageChannel’s port2
+   *    attribute.
+   * 7. Let serializedProcessorPort be the result of
+   *    StructuredSerializeWithTransfer(processorPortOnThisSide,
+   *                                    « processorPortOnThisSide »).
+   */
+  UniqueMessagePortId processorPortId;
+  messageChannel->Port2()->CloneAndDisentangle(processorPortId);
+  /**
+   * 8. Convert options dictionary to optionsObject.
+   */
+  JSContext* cx = aGlobal.Context();
+  JS::Rooted<JS::Value> optionsVal(cx);
+  if (NS_WARN_IF(!ToJSValue(cx, aOptions, &optionsVal))) {
+    aRv.NoteJSContextException(cx);
+    return nullptr;
+  }
+
+  /**
+   * 9. Let serializedOptions be the result of
+   *    StructuredSerialize(optionsObject).
+   */
+
+  // This context and the worklet are part of the same agent cluster and they
+  // can share memory.
+  JS::CloneDataPolicy cloneDataPolicy;
+  cloneDataPolicy.allowIntraClusterClonableSharedObjects();
+  cloneDataPolicy.allowSharedMemoryObjects();
+
+  // StructuredCloneHolder does not have a move constructor.  Instead allocate
+  // memory so that the pointer can be passed to the rendering thread.
+  UniquePtr<StructuredCloneHolder> serializedOptions =
+      MakeUnique<StructuredCloneHolder>(
+          StructuredCloneHolder::CloningSupported,
+          StructuredCloneHolder::TransferringNotSupported,
+          JS::StructuredCloneScope::SameProcess);
+  serializedOptions->Write(cx, optionsVal, JS::UndefinedHandleValue,
+                           cloneDataPolicy, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+  /**
+   * 10. Set node’s port to nodePort.
+   */
+  audioWorkletNode->mPort = messageChannel->Port1();
+
+  /**
+   * 11. Let parameterDescriptors be the result of retrieval of nodeName from
+   * node name to parameter descriptor map.
+   */
+  nsTArray<NamedAudioParamTimeline> paramTimelines;
+  audioWorkletNode->InitializeParameters(&paramTimelines, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  auto engine = new WorkletNodeEngine(
+      audioWorkletNode, aAudioContext.Destination(), std::move(paramTimelines),
+      aOptions.mOutputChannelCount);
+  audioWorkletNode->mTrack = AudioNodeTrack::Create(
+      &aAudioContext, engine, AudioNodeTrack::NO_TRACK_FLAGS,
+      aAudioContext.Graph());
+
+  audioWorkletNode->SendParameterData(aOptions.mParameterData);
+
+  /**
+   * 12. Queue a control message to invoke the constructor of the
+   *     corresponding AudioWorkletProcessor with the processor construction
+   *     data that consists of: nodeName, node, serializedOptions, and
+   *     serializedProcessorPort.
+   */
+  Worklet* worklet = aAudioContext.GetAudioWorklet(aRv);
+  MOZ_ASSERT(worklet, "Worklet already existed and so getter shouldn't fail.");
+  auto workletImpl = static_cast<AudioWorkletImpl*>(worklet->Impl());
+  audioWorkletNode->mTrack->SendRunnable(NS_NewRunnableFunction(
+      "WorkletNodeEngine::ConstructProcessor",
+  // MOZ_CAN_RUN_SCRIPT_BOUNDARY until Runnable::Run is MOZ_CAN_RUN_SCRIPT.
+  // See bug 1535398.
+  //
+  // Note that clang and gcc have mutually incompatible rules about whether
+  // attributes should come before or after the `mutable` keyword here, so
+  // use a compatibility hack until we can switch to the standardized
+  // [[attr]] syntax (bug 1627007).
+#ifdef __clang__
+#  define AND_MUTABLE(macro) macro mutable
+#else
+#  define AND_MUTABLE(macro) mutable macro
+#endif
+      [track = audioWorkletNode->mTrack,
+       workletImpl = RefPtr<AudioWorkletImpl>(workletImpl),
+       name = nsString(aName), options = std::move(serializedOptions),
+       portId = std::move(processorPortId)]()
+          AND_MUTABLE(MOZ_CAN_RUN_SCRIPT_BOUNDARY) {
+            auto engine = static_cast<WorkletNodeEngine*>(track->Engine());
+            engine->ConstructProcessor(
+                workletImpl, name, WrapNotNull(options.get()), portId, track);
+          }));
+#undef AND_MUTABLE
+
+  // [[active source]] is initially true and so at least the first process()
+  // call will not be skipped when there are no active inputs.
+  audioWorkletNode->MarkActive();
+
+  return audioWorkletNode.forget();
+}
+
+AudioParamMap* AudioWorkletNode::GetParameters(ErrorResult& aRv) {
+  if (!mParameters) {
+    RefPtr<AudioParamMap> parameters = new AudioParamMap(this);
+    nsAutoString name;
+    for (const auto& audioParam : mParams) {
+      audioParam->GetName(name);
+      AudioParamMap_Binding::MaplikeHelpers::Set(parameters, name, *audioParam,
+                                                 aRv);
+      if (NS_WARN_IF(aRv.Failed())) {
+        return nullptr;
+      }
+    }
+    mParameters = std::move(parameters);
+  }
+  return mParameters.get();
+}
+
+void AudioWorkletNode::DispatchProcessorErrorEvent(
+    const ProcessorErrorDetails& aDetails) {
+  TRACE("AudioWorkletNode::DispatchProcessorErrorEvent");
+  if (HasListenersFor(nsGkAtoms::onprocessorerror)) {
+    RootedDictionary<ErrorEventInit> init(RootingCx());
+    init.mMessage = aDetails.mMessage;
+    init.mFilename = aDetails.mFilename;
+    init.mLineno = aDetails.mLineno;
+    init.mColno = aDetails.mColno;
+    RefPtr<ErrorEvent> errorEvent =
+        ErrorEvent::Constructor(this, u"processorerror"_ns, init);
+    MOZ_ASSERT(errorEvent);
+    DispatchTrustedEvent(errorEvent);
+  }
+}
+
+JSObject* AudioWorkletNode::WrapObject(JSContext* aCx,
+                                       JS::Handle<JSObject*> aGivenProto) {
+  return AudioWorkletNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+size_t AudioWorkletNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t AudioWorkletNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioWorkletNode.h b/dom/media/webaudio/AudioWorkletNode.h
new file mode 100644
index 0000000000..22e089f8cd
--- /dev/null
+++ b/dom/media/webaudio/AudioWorkletNode.h
@@ -0,0 +1,68 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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 https://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioWorkletNode_h_
+#define AudioWorkletNode_h_
+
+#include "AudioNode.h"
+
+namespace mozilla::dom {
+
+class AudioParamMap;
+struct AudioWorkletNodeOptions;
+class MessagePort;
+struct NamedAudioParamTimeline;
+struct ProcessorErrorDetails;
+template <typename KeyType, typename ValueType>
+class Record;
+
+class AudioWorkletNode : public AudioNode {
+ public:
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioWorkletNode, AudioNode)
+
+  IMPL_EVENT_HANDLER(processorerror)
+
+  static already_AddRefed<AudioWorkletNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const nsAString& aName, const AudioWorkletNodeOptions& aOptions,
+      ErrorResult& aRv);
+
+  AudioParamMap* GetParameters(ErrorResult& aRv);
+
+  MessagePort* Port() const { return mPort; };
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  // AudioNode methods
+  uint16_t NumberOfInputs() const override { return mInputCount; }
+  uint16_t NumberOfOutputs() const override { return mOutputCount; }
+  const char* NodeType() const override { return "AudioWorkletNode"; }
+  void DispatchProcessorErrorEvent(const ProcessorErrorDetails& aDetails);
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ private:
+  AudioWorkletNode(AudioContext* aAudioContext, const nsAString& aName,
+                   const AudioWorkletNodeOptions& aOptions);
+  ~AudioWorkletNode() = default;
+  void InitializeParameters(nsTArray<NamedAudioParamTimeline>* aParamTimelines,
+                            ErrorResult& aRv);
+  void SendParameterData(
+      const Optional<Record<nsString, double>>& aParameterData);
+
+  nsString mNodeName;
+  RefPtr<MessagePort> mPort;
+  RefPtr<AudioParamMap> mParameters;
+  uint16_t mInputCount;
+  uint16_t mOutputCount;
+};
+
+}  // namespace mozilla::dom
+
+#endif  // AudioWorkletNode_h_
diff --git a/dom/media/webaudio/AudioWorkletProcessor.cpp b/dom/media/webaudio/AudioWorkletProcessor.cpp
new file mode 100644
index 0000000000..194cc5bf97
--- /dev/null
+++ b/dom/media/webaudio/AudioWorkletProcessor.cpp
@@ -0,0 +1,49 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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 https://mozilla.org/MPL/2.0/. */
+
+#include "AudioWorkletProcessor.h"
+
+#include "mozilla/dom/AudioWorkletNodeBinding.h"
+#include "mozilla/dom/AudioWorkletProcessorBinding.h"
+#include "mozilla/dom/AudioWorkletGlobalScope.h"
+#include "mozilla/dom/MessagePort.h"
+#include "mozilla/dom/WorkletGlobalScope.h"
+#include "nsIGlobalObject.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE(AudioWorkletProcessor, mParent, mPort)
+
+AudioWorkletProcessor::AudioWorkletProcessor(nsIGlobalObject* aParent,
+                                             MessagePort* aPort)
+    : mParent(aParent), mPort(aPort) {}
+
+AudioWorkletProcessor::~AudioWorkletProcessor() = default;
+
+/* static */
+already_AddRefed<AudioWorkletProcessor> AudioWorkletProcessor::Constructor(
+    const GlobalObject& aGlobal, ErrorResult& aRv) {
+  nsCOMPtr<WorkletGlobalScope> global =
+      do_QueryInterface(aGlobal.GetAsSupports());
+  MOZ_ASSERT(global);
+  RefPtr<MessagePort> port = static_cast<AudioWorkletGlobalScope*>(global.get())
+                                 ->TakePortForProcessorCtor();
+  if (!port) {
+    aRv.ThrowTypeError<MSG_ILLEGAL_CONSTRUCTOR>();
+    return nullptr;
+  }
+  RefPtr<AudioWorkletProcessor> audioWorkletProcessor =
+      new AudioWorkletProcessor(global, port);
+
+  return audioWorkletProcessor.forget();
+}
+
+JSObject* AudioWorkletProcessor::WrapObject(JSContext* aCx,
+                                            JS::Handle<JSObject*> aGivenProto) {
+  return AudioWorkletProcessor_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/AudioWorkletProcessor.h b/dom/media/webaudio/AudioWorkletProcessor.h
new file mode 100644
index 0000000000..3d37d61498
--- /dev/null
+++ b/dom/media/webaudio/AudioWorkletProcessor.h
@@ -0,0 +1,50 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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 https://mozilla.org/MPL/2.0/. */
+
+#ifndef AudioWorkletProcessor_h_
+#define AudioWorkletProcessor_h_
+
+#include "nsCOMPtr.h"
+#include "nsWrapperCache.h"
+
+class nsIGlobalObject;
+
+namespace mozilla {
+
+class ErrorResult;
+
+namespace dom {
+
+struct AudioWorkletNodeOptions;
+class GlobalObject;
+class MessagePort;
+
+class AudioWorkletProcessor final : public nsWrapperCache {
+ public:
+  NS_INLINE_DECL_CYCLE_COLLECTING_NATIVE_REFCOUNTING(AudioWorkletProcessor)
+  NS_DECL_CYCLE_COLLECTION_NATIVE_WRAPPERCACHE_CLASS(AudioWorkletProcessor)
+
+  static already_AddRefed<AudioWorkletProcessor> Constructor(
+      const GlobalObject& aGlobal, ErrorResult& aRv);
+
+  nsIGlobalObject* GetParentObject() const { return mParent; }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  MessagePort* Port() const { return mPort; };
+
+ private:
+  explicit AudioWorkletProcessor(nsIGlobalObject* aParent, MessagePort* aPort);
+  ~AudioWorkletProcessor();
+  nsCOMPtr<nsIGlobalObject> mParent;
+  RefPtr<MessagePort> mPort;
+};
+
+}  // namespace dom
+}  // namespace mozilla
+
+#endif  // AudioWorkletProcessor_h_
diff --git a/dom/media/webaudio/BiquadFilterNode.cpp b/dom/media/webaudio/BiquadFilterNode.cpp
new file mode 100644
index 0000000000..602dbf9cc1
--- /dev/null
+++ b/dom/media/webaudio/BiquadFilterNode.cpp
@@ -0,0 +1,358 @@
+/* -*- 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 "BiquadFilterNode.h"
+#include <algorithm>
+#include "AlignmentUtils.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "AudioDestinationNode.h"
+#include "PlayingRefChangeHandler.h"
+#include "WebAudioUtils.h"
+#include "blink/Biquad.h"
+#include "mozilla/UniquePtr.h"
+#include "mozilla/ErrorResult.h"
+#include "AudioParamTimeline.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(BiquadFilterNode, AudioNode, mFrequency,
+                                   mDetune, mQ, mGain)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(BiquadFilterNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(BiquadFilterNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(BiquadFilterNode, AudioNode)
+
+static void SetParamsOnBiquad(WebCore::Biquad& aBiquad, float aSampleRate,
+                              BiquadFilterType aType, double aFrequency,
+                              double aQ, double aGain, double aDetune) {
+  const double nyquist = aSampleRate * 0.5;
+  double normalizedFrequency = aFrequency / nyquist;
+
+  if (aDetune) {
+    normalizedFrequency *= fdlibm_exp2(aDetune / 1200);
+  }
+
+  switch (aType) {
+    case BiquadFilterType::Lowpass:
+      aBiquad.setLowpassParams(normalizedFrequency, aQ);
+      break;
+    case BiquadFilterType::Highpass:
+      aBiquad.setHighpassParams(normalizedFrequency, aQ);
+      break;
+    case BiquadFilterType::Bandpass:
+      aBiquad.setBandpassParams(normalizedFrequency, aQ);
+      break;
+    case BiquadFilterType::Lowshelf:
+      aBiquad.setLowShelfParams(normalizedFrequency, aGain);
+      break;
+    case BiquadFilterType::Highshelf:
+      aBiquad.setHighShelfParams(normalizedFrequency, aGain);
+      break;
+    case BiquadFilterType::Peaking:
+      aBiquad.setPeakingParams(normalizedFrequency, aQ, aGain);
+      break;
+    case BiquadFilterType::Notch:
+      aBiquad.setNotchParams(normalizedFrequency, aQ);
+      break;
+    case BiquadFilterType::Allpass:
+      aBiquad.setAllpassParams(normalizedFrequency, aQ);
+      break;
+    default:
+      MOZ_ASSERT_UNREACHABLE("We should never see the alternate names here");
+      break;
+  }
+}
+
+class BiquadFilterNodeEngine final : public AudioNodeEngine {
+ public:
+  BiquadFilterNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination,
+                         uint64_t aWindowID)
+      : AudioNodeEngine(aNode),
+        mDestination(aDestination->Track())
+        // Keep the default values in sync with the default values in
+        // BiquadFilterNode::BiquadFilterNode
+        ,
+        mType(BiquadFilterType::Lowpass),
+        mFrequency(350.f),
+        mDetune(0.f),
+        mQ(1.f),
+        mGain(0.f),
+        mWindowID(aWindowID) {}
+
+  enum Parameters { TYPE, FREQUENCY, DETUNE, Q, GAIN };
+  void SetInt32Parameter(uint32_t aIndex, int32_t aValue) override {
+    switch (aIndex) {
+      case TYPE:
+        mType = static_cast<BiquadFilterType>(aValue);
+        break;
+      default:
+        NS_ERROR("Bad BiquadFilterNode Int32Parameter");
+    }
+  }
+  void RecvTimelineEvent(uint32_t aIndex, AudioParamEvent& aEvent) override {
+    MOZ_ASSERT(mDestination);
+
+    aEvent.ConvertToTicks(mDestination);
+
+    switch (aIndex) {
+      case FREQUENCY:
+        mFrequency.InsertEvent<int64_t>(aEvent);
+        break;
+      case DETUNE:
+        mDetune.InsertEvent<int64_t>(aEvent);
+        break;
+      case Q:
+        mQ.InsertEvent<int64_t>(aEvent);
+        break;
+      case GAIN:
+        mGain.InsertEvent<int64_t>(aEvent);
+        break;
+      default:
+        NS_ERROR("Bad BiquadFilterNodeEngine TimelineParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    TRACE("BiquadFilterNode::ProcessBlock");
+    float inputBuffer[WEBAUDIO_BLOCK_SIZE + 4];
+    float* alignedInputBuffer = ALIGNED16(inputBuffer);
+    ASSERT_ALIGNED16(alignedInputBuffer);
+
+    if (aInput.IsNull()) {
+      bool hasTail = false;
+      for (uint32_t i = 0; i < mBiquads.Length(); ++i) {
+        if (mBiquads[i].hasTail()) {
+          hasTail = true;
+          break;
+        }
+      }
+      if (!hasTail) {
+        if (!mBiquads.IsEmpty()) {
+          mBiquads.Clear();
+          aTrack->ScheduleCheckForInactive();
+
+          RefPtr<PlayingRefChangeHandler> refchanged =
+              new PlayingRefChangeHandler(aTrack,
+                                          PlayingRefChangeHandler::RELEASE);
+          aTrack->Graph()->DispatchToMainThreadStableState(refchanged.forget());
+        }
+
+        aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+        return;
+      }
+
+      PodArrayZero(inputBuffer);
+
+    } else if (mBiquads.Length() != aInput.ChannelCount()) {
+      if (mBiquads.IsEmpty()) {
+        RefPtr<PlayingRefChangeHandler> refchanged =
+            new PlayingRefChangeHandler(aTrack,
+                                        PlayingRefChangeHandler::ADDREF);
+        aTrack->Graph()->DispatchToMainThreadStableState(refchanged.forget());
+      } else {  // Help people diagnose bug 924718
+        WebAudioUtils::LogToDeveloperConsole(
+            mWindowID, "BiquadFilterChannelCountChangeWarning");
+      }
+
+      // Adjust the number of biquads based on the number of channels
+      mBiquads.SetLength(aInput.ChannelCount());
+    }
+
+    uint32_t numberOfChannels = mBiquads.Length();
+    aOutput->AllocateChannels(numberOfChannels);
+
+    TrackTime pos = mDestination->GraphTimeToTrackTime(aFrom);
+
+    double freq = mFrequency.GetValueAtTime(pos);
+    double q = mQ.GetValueAtTime(pos);
+    double gain = mGain.GetValueAtTime(pos);
+    double detune = mDetune.GetValueAtTime(pos);
+
+    for (uint32_t i = 0; i < numberOfChannels; ++i) {
+      const float* input;
+      if (aInput.IsNull()) {
+        input = alignedInputBuffer;
+      } else {
+        input = static_cast<const float*>(aInput.mChannelData[i]);
+        if (aInput.mVolume != 1.0) {
+          AudioBlockCopyChannelWithScale(input, aInput.mVolume,
+                                         alignedInputBuffer);
+          input = alignedInputBuffer;
+        }
+      }
+      SetParamsOnBiquad(mBiquads[i], aTrack->mSampleRate, mType, freq, q, gain,
+                        detune);
+
+      mBiquads[i].process(input, aOutput->ChannelFloatsForWrite(i),
+                          aInput.GetDuration());
+    }
+  }
+
+  bool IsActive() const override { return !mBiquads.IsEmpty(); }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    // Not owned:
+    // - mDestination - probably not owned
+    // - AudioParamTimelines - counted in the AudioNode
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    amount += mBiquads.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  RefPtr<AudioNodeTrack> mDestination;
+  BiquadFilterType mType;
+  AudioParamTimeline mFrequency;
+  AudioParamTimeline mDetune;
+  AudioParamTimeline mQ;
+  AudioParamTimeline mGain;
+  nsTArray<WebCore::Biquad> mBiquads;
+  uint64_t mWindowID;
+};
+
+BiquadFilterNode::BiquadFilterNode(AudioContext* aContext)
+    : AudioNode(aContext, 2, ChannelCountMode::Max,
+                ChannelInterpretation::Speakers),
+      mType(BiquadFilterType::Lowpass) {
+  mFrequency = CreateAudioParam(
+      BiquadFilterNodeEngine::FREQUENCY, u"frequency"_ns, 350.f,
+      -(aContext->SampleRate() / 2), aContext->SampleRate() / 2);
+  mDetune = CreateAudioParam(BiquadFilterNodeEngine::DETUNE, u"detune"_ns, 0.f);
+  mQ = CreateAudioParam(BiquadFilterNodeEngine::Q, u"Q"_ns, 1.f);
+  mGain = CreateAudioParam(BiquadFilterNodeEngine::GAIN, u"gain"_ns, 0.f);
+
+  uint64_t windowID = 0;
+  if (aContext->GetParentObject()) {
+    windowID = aContext->GetParentObject()->WindowID();
+  }
+  BiquadFilterNodeEngine* engine =
+      new BiquadFilterNodeEngine(this, aContext->Destination(), windowID);
+  mTrack = AudioNodeTrack::Create(
+      aContext, engine, AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+}
+
+/* static */
+already_AddRefed<BiquadFilterNode> BiquadFilterNode::Create(
+    AudioContext& aAudioContext, const BiquadFilterOptions& aOptions,
+    ErrorResult& aRv) {
+  RefPtr<BiquadFilterNode> audioNode = new BiquadFilterNode(&aAudioContext);
+
+  audioNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  audioNode->SetType(aOptions.mType);
+  audioNode->Q()->SetInitialValue(aOptions.mQ);
+  audioNode->Detune()->SetInitialValue(aOptions.mDetune);
+  audioNode->Frequency()->SetInitialValue(aOptions.mFrequency);
+  audioNode->Gain()->SetInitialValue(aOptions.mGain);
+
+  return audioNode.forget();
+}
+
+size_t BiquadFilterNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+
+  if (mFrequency) {
+    amount += mFrequency->SizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  if (mDetune) {
+    amount += mDetune->SizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  if (mQ) {
+    amount += mQ->SizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  if (mGain) {
+    amount += mGain->SizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  return amount;
+}
+
+size_t BiquadFilterNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject* BiquadFilterNode::WrapObject(JSContext* aCx,
+                                       JS::Handle<JSObject*> aGivenProto) {
+  return BiquadFilterNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void BiquadFilterNode::SetType(BiquadFilterType aType) {
+  mType = aType;
+  SendInt32ParameterToTrack(BiquadFilterNodeEngine::TYPE,
+                            static_cast<int32_t>(aType));
+}
+
+void BiquadFilterNode::GetFrequencyResponse(const Float32Array& aFrequencyHz,
+                                            const Float32Array& aMagResponse,
+                                            const Float32Array& aPhaseResponse,
+                                            ErrorResult& aRv) {
+  UniquePtr<float[]> frequencies;
+  size_t length;
+  const double currentTime = Context()->CurrentTime();
+  aFrequencyHz.ProcessData([&](const Span<float>& aFrequencyData,
+                               JS::AutoCheckCannotGC&&) {
+    aMagResponse.ProcessData([&](const Span<float>& aMagData,
+                                 JS::AutoCheckCannotGC&&) {
+      aPhaseResponse.ProcessData([&](const Span<float>& aPhaseData,
+                                     JS::AutoCheckCannotGC&&) {
+        length = aFrequencyData.Length();
+        if (length != aMagData.Length() || length != aPhaseData.Length()) {
+          aRv.ThrowInvalidAccessError("Parameter lengths must match");
+          return;
+        }
+
+        if (length == 0) {
+          return;
+        }
+
+        frequencies = MakeUniqueForOverwriteFallible<float[]>(length);
+        if (!frequencies) {
+          aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+          return;
+        }
+
+        const double nyquist = Context()->SampleRate() * 0.5;
+        std::transform(aFrequencyData.begin(), aFrequencyData.end(),
+                       frequencies.get(), [&](float aFrequency) {
+                         if (aFrequency >= 0 && aFrequency <= nyquist) {
+                           return static_cast<float>(aFrequency / nyquist);
+                         }
+
+                         return std::numeric_limits<float>::quiet_NaN();
+                       });
+
+        double freq = mFrequency->GetValueAtTime(currentTime);
+        double q = mQ->GetValueAtTime(currentTime);
+        double gain = mGain->GetValueAtTime(currentTime);
+        double detune = mDetune->GetValueAtTime(currentTime);
+
+        WebCore::Biquad biquad;
+        SetParamsOnBiquad(biquad, Context()->SampleRate(), mType, freq, q, gain,
+                          detune);
+        biquad.getFrequencyResponse(int(length), frequencies.get(),
+                                    aMagData.Elements(), aPhaseData.Elements());
+      });
+    });
+  });
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/BiquadFilterNode.h b/dom/media/webaudio/BiquadFilterNode.h
new file mode 100644
index 0000000000..31b08c00c8
--- /dev/null
+++ b/dom/media/webaudio/BiquadFilterNode.h
@@ -0,0 +1,71 @@
+/* -*- 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/. */
+
+#ifndef BiquadFilterNode_h_
+#define BiquadFilterNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+#include "mozilla/dom/BiquadFilterNodeBinding.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct BiquadFilterOptions;
+
+class BiquadFilterNode final : public AudioNode {
+ public:
+  static already_AddRefed<BiquadFilterNode> Create(
+      AudioContext& aAudioContext, const BiquadFilterOptions& aOptions,
+      ErrorResult& aRv);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(BiquadFilterNode, AudioNode)
+
+  static already_AddRefed<BiquadFilterNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const BiquadFilterOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  BiquadFilterType Type() const { return mType; }
+  void SetType(BiquadFilterType aType);
+
+  AudioParam* Frequency() const { return mFrequency; }
+
+  AudioParam* Detune() const { return mDetune; }
+
+  AudioParam* Q() const { return mQ; }
+
+  AudioParam* Gain() const { return mGain; }
+
+  void GetFrequencyResponse(const Float32Array& aFrequencyHz,
+                            const Float32Array& aMagResponse,
+                            const Float32Array& aPhaseResponse,
+                            ErrorResult& aRv);
+
+  const char* NodeType() const override { return "BiquadFilterNode"; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ private:
+  explicit BiquadFilterNode(AudioContext* aContext);
+  ~BiquadFilterNode() = default;
+
+  BiquadFilterType mType;
+  RefPtr<AudioParam> mFrequency;
+  RefPtr<AudioParam> mDetune;
+  RefPtr<AudioParam> mQ;
+  RefPtr<AudioParam> mGain;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/ChannelMergerNode.cpp b/dom/media/webaudio/ChannelMergerNode.cpp
new file mode 100644
index 0000000000..4ffae02591
--- /dev/null
+++ b/dom/media/webaudio/ChannelMergerNode.cpp
@@ -0,0 +1,100 @@
+/* -*- 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 "mozilla/dom/ChannelMergerNode.h"
+#include "mozilla/dom/ChannelMergerNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "nsPrintfCString.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+class ChannelMergerNodeEngine final : public AudioNodeEngine {
+ public:
+  explicit ChannelMergerNodeEngine(ChannelMergerNode* aNode)
+      : AudioNodeEngine(aNode) {
+    MOZ_ASSERT(NS_IsMainThread());
+  }
+
+  void ProcessBlocksOnPorts(AudioNodeTrack* aTrack, GraphTime aFrom,
+                            Span<const AudioBlock> aInput,
+                            Span<AudioBlock> aOutput,
+                            bool* aFinished) override {
+    MOZ_ASSERT(aInput.Length() == InputCount());
+    MOZ_ASSERT(aOutput.Length() == 1, "Should have only one output port");
+    TRACE("ChannelMergerNodeEngine::ProcessBlocksOnPorts");
+
+    // Get the number of output channels, and allocate it
+    size_t channelCount = InputCount();
+    bool allNull = true;
+    for (size_t i = 0; i < channelCount; ++i) {
+      allNull &= aInput[i].IsNull();
+    }
+    if (allNull) {
+      aOutput[0].SetNull(WEBAUDIO_BLOCK_SIZE);
+      return;
+    }
+
+    aOutput[0].AllocateChannels(channelCount);
+
+    for (size_t i = 0; i < channelCount; ++i) {
+      float* output = aOutput[0].ChannelFloatsForWrite(i);
+      if (aInput[i].IsNull()) {
+        PodZero(output, WEBAUDIO_BLOCK_SIZE);
+      } else {
+        AudioBlockCopyChannelWithScale(
+            static_cast<const float*>(aInput[i].mChannelData[0]),
+            aInput[i].mVolume, output);
+      }
+    }
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+};
+
+ChannelMergerNode::ChannelMergerNode(AudioContext* aContext,
+                                     uint16_t aInputCount)
+    : AudioNode(aContext, 1, ChannelCountMode::Explicit,
+                ChannelInterpretation::Speakers),
+      mInputCount(aInputCount) {
+  mTrack =
+      AudioNodeTrack::Create(aContext, new ChannelMergerNodeEngine(this),
+                             AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+}
+
+/* static */
+already_AddRefed<ChannelMergerNode> ChannelMergerNode::Create(
+    AudioContext& aAudioContext, const ChannelMergerOptions& aOptions,
+    ErrorResult& aRv) {
+  if (aOptions.mNumberOfInputs == 0 ||
+      aOptions.mNumberOfInputs > WebAudioUtils::MaxChannelCount) {
+    aRv.ThrowIndexSizeError(
+        nsPrintfCString("Number of inputs (%u) must be in the range [1, number "
+                        "of supported channels]",
+                        aOptions.mNumberOfInputs));
+    return nullptr;
+  }
+
+  RefPtr<ChannelMergerNode> audioNode =
+      new ChannelMergerNode(&aAudioContext, aOptions.mNumberOfInputs);
+
+  audioNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  return audioNode.forget();
+}
+
+JSObject* ChannelMergerNode::WrapObject(JSContext* aCx,
+                                        JS::Handle<JSObject*> aGivenProto) {
+  return ChannelMergerNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/ChannelMergerNode.h b/dom/media/webaudio/ChannelMergerNode.h
new file mode 100644
index 0000000000..92327c8b1e
--- /dev/null
+++ b/dom/media/webaudio/ChannelMergerNode.h
@@ -0,0 +1,67 @@
+/* -*- 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/. */
+
+#ifndef ChannelMergerNode_h_
+#define ChannelMergerNode_h_
+
+#include "AudioNode.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct ChannelMergerOptions;
+
+class ChannelMergerNode final : public AudioNode {
+ public:
+  static already_AddRefed<ChannelMergerNode> Create(
+      AudioContext& aAudioContext, const ChannelMergerOptions& aOptions,
+      ErrorResult& aRv);
+
+  NS_INLINE_DECL_REFCOUNTING_INHERITED(ChannelMergerNode, AudioNode)
+
+  static already_AddRefed<ChannelMergerNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const ChannelMergerOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  uint16_t NumberOfInputs() const override { return mInputCount; }
+
+  const char* NodeType() const override { return "ChannelMergerNode"; }
+
+  virtual void SetChannelCount(uint32_t aChannelCount,
+                               ErrorResult& aRv) override {
+    if (aChannelCount != 1) {
+      aRv.ThrowInvalidStateError(
+          "Cannot change channel count of ChannelMergerNode");
+    }
+  }
+
+  virtual void SetChannelCountModeValue(ChannelCountMode aMode,
+                                        ErrorResult& aRv) override {
+    if (aMode != ChannelCountMode::Explicit) {
+      aRv.ThrowInvalidStateError(
+          "Cannot change channel count mode of ChannelMergerNode");
+    }
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  ChannelMergerNode(AudioContext* aContext, uint16_t aInputCount);
+  ~ChannelMergerNode() = default;
+
+  const uint16_t mInputCount;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/ChannelSplitterNode.cpp b/dom/media/webaudio/ChannelSplitterNode.cpp
new file mode 100644
index 0000000000..52d3ae948f
--- /dev/null
+++ b/dom/media/webaudio/ChannelSplitterNode.cpp
@@ -0,0 +1,105 @@
+/* -*- 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 "mozilla/dom/ChannelSplitterNode.h"
+#include "mozilla/dom/ChannelSplitterNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+class ChannelSplitterNodeEngine final : public AudioNodeEngine {
+ public:
+  explicit ChannelSplitterNodeEngine(ChannelSplitterNode* aNode)
+      : AudioNodeEngine(aNode) {
+    MOZ_ASSERT(NS_IsMainThread());
+  }
+
+  void ProcessBlocksOnPorts(AudioNodeTrack* aTrack, GraphTime aFrom,
+                            Span<const AudioBlock> aInput,
+                            Span<AudioBlock> aOutput,
+                            bool* aFinished) override {
+    MOZ_ASSERT(aInput.Length() == 1, "Should only have one input port");
+    MOZ_ASSERT(aOutput.Length() == OutputCount());
+    TRACE("ChannelSplitterNodeEngine::ProcessBlocksOnPorts");
+
+    for (uint16_t i = 0; i < OutputCount(); ++i) {
+      if (i < aInput[0].ChannelCount()) {
+        // Split out existing channels
+        aOutput[i].AllocateChannels(1);
+        AudioBlockCopyChannelWithScale(
+            static_cast<const float*>(aInput[0].mChannelData[i]),
+            aInput[0].mVolume, aOutput[i].ChannelFloatsForWrite(0));
+      } else {
+        // Pad with silent channels if needed
+        aOutput[i].SetNull(WEBAUDIO_BLOCK_SIZE);
+      }
+    }
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+};
+
+ChannelSplitterNode::ChannelSplitterNode(AudioContext* aContext,
+                                         uint16_t aOutputCount)
+    : AudioNode(aContext, aOutputCount, ChannelCountMode::Explicit,
+                ChannelInterpretation::Discrete),
+      mOutputCount(aOutputCount) {
+  mTrack =
+      AudioNodeTrack::Create(aContext, new ChannelSplitterNodeEngine(this),
+                             AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+}
+
+/* static */
+already_AddRefed<ChannelSplitterNode> ChannelSplitterNode::Create(
+    AudioContext& aAudioContext, const ChannelSplitterOptions& aOptions,
+    ErrorResult& aRv) {
+  if (aOptions.mNumberOfOutputs == 0 ||
+      aOptions.mNumberOfOutputs > WebAudioUtils::MaxChannelCount) {
+    aRv.ThrowIndexSizeError(nsPrintfCString(
+        "%u is not a valid number of outputs", aOptions.mNumberOfOutputs));
+    return nullptr;
+  }
+
+  RefPtr<ChannelSplitterNode> audioNode =
+      new ChannelSplitterNode(&aAudioContext, aOptions.mNumberOfOutputs);
+
+  // Manually check that the other options are valid, this node has
+  // channelCount, channelCountMode and channelInterpretation constraints: they
+  // cannot be changed from the default.
+  if (aOptions.mChannelCount.WasPassed()) {
+    audioNode->SetChannelCount(aOptions.mChannelCount.Value(), aRv);
+    if (aRv.Failed()) {
+      return nullptr;
+    }
+  }
+  if (aOptions.mChannelInterpretation.WasPassed()) {
+    audioNode->SetChannelInterpretationValue(
+        aOptions.mChannelInterpretation.Value(), aRv);
+    if (aRv.Failed()) {
+      return nullptr;
+    }
+  }
+  if (aOptions.mChannelCountMode.WasPassed()) {
+    audioNode->SetChannelCountModeValue(aOptions.mChannelCountMode.Value(),
+                                        aRv);
+    if (aRv.Failed()) {
+      return nullptr;
+    }
+  }
+
+  return audioNode.forget();
+}
+
+JSObject* ChannelSplitterNode::WrapObject(JSContext* aCx,
+                                          JS::Handle<JSObject*> aGivenProto) {
+  return ChannelSplitterNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/ChannelSplitterNode.h b/dom/media/webaudio/ChannelSplitterNode.h
new file mode 100644
index 0000000000..c13eaa8f22
--- /dev/null
+++ b/dom/media/webaudio/ChannelSplitterNode.h
@@ -0,0 +1,72 @@
+/* -*- 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/. */
+
+#ifndef ChannelSplitterNode_h_
+#define ChannelSplitterNode_h_
+
+#include "AudioNode.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct ChannelSplitterOptions;
+
+class ChannelSplitterNode final : public AudioNode {
+ public:
+  static already_AddRefed<ChannelSplitterNode> Create(
+      AudioContext& aAudioContext, const ChannelSplitterOptions& aOptions,
+      ErrorResult& aRv);
+
+  NS_INLINE_DECL_REFCOUNTING_INHERITED(ChannelSplitterNode, AudioNode)
+
+  static already_AddRefed<ChannelSplitterNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const ChannelSplitterOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  void SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv) override {
+    if (aChannelCount != ChannelCount()) {
+      aRv.ThrowInvalidStateError(
+          "Cannot change channel count of ChannelSplitterNode");
+    }
+  }
+  void SetChannelCountModeValue(ChannelCountMode aMode,
+                                ErrorResult& aRv) override {
+    if (aMode != ChannelCountModeValue()) {
+      aRv.ThrowInvalidStateError(
+          "Cannot change channel count mode of ChannelSplitterNode");
+    }
+  }
+  void SetChannelInterpretationValue(ChannelInterpretation aMode,
+                                     ErrorResult& aRv) override {
+    if (aMode != ChannelInterpretationValue()) {
+      aRv.ThrowInvalidStateError(
+          "Cannot change channel interpretation of ChannelSplitterNode");
+    }
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  uint16_t NumberOfOutputs() const override { return mOutputCount; }
+
+  const char* NodeType() const override { return "ChannelSplitterNode"; }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  ChannelSplitterNode(AudioContext* aContext, uint16_t aOutputCount);
+  ~ChannelSplitterNode() = default;
+
+  const uint16_t mOutputCount;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/ConstantSourceNode.cpp b/dom/media/webaudio/ConstantSourceNode.cpp
new file mode 100644
index 0000000000..6e74c3e219
--- /dev/null
+++ b/dom/media/webaudio/ConstantSourceNode.cpp
@@ -0,0 +1,279 @@
+/* -*- 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 "ConstantSourceNode.h"
+
+#include "AudioDestinationNode.h"
+#include "nsContentUtils.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(ConstantSourceNode, AudioScheduledSourceNode,
+                                   mOffset)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(ConstantSourceNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioScheduledSourceNode)
+
+NS_IMPL_ADDREF_INHERITED(ConstantSourceNode, AudioScheduledSourceNode)
+NS_IMPL_RELEASE_INHERITED(ConstantSourceNode, AudioScheduledSourceNode)
+
+class ConstantSourceNodeEngine final : public AudioNodeEngine {
+ public:
+  ConstantSourceNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination)
+      : AudioNodeEngine(aNode),
+        mSource(nullptr),
+        mDestination(aDestination->Track()),
+        mStart(-1),
+        mStop(TRACK_TIME_MAX)
+        // Keep the default values in sync with
+        // ConstantSourceNode::ConstantSourceNode.
+        ,
+        mOffset(1.0f) {
+    MOZ_ASSERT(NS_IsMainThread());
+  }
+
+  void SetSourceTrack(AudioNodeTrack* aSource) { mSource = aSource; }
+
+  enum Parameters {
+    OFFSET,
+    START,
+    STOP,
+  };
+  void RecvTimelineEvent(uint32_t aIndex, AudioParamEvent& aEvent) override {
+    MOZ_ASSERT(mDestination);
+
+    aEvent.ConvertToTicks(mDestination);
+
+    switch (aIndex) {
+      case OFFSET:
+        mOffset.InsertEvent<int64_t>(aEvent);
+        break;
+      default:
+        NS_ERROR("Bad ConstantSourceNodeEngine TimelineParameter");
+    }
+  }
+
+  void SetTrackTimeParameter(uint32_t aIndex, TrackTime aParam) override {
+    switch (aIndex) {
+      case START:
+        mStart = aParam;
+        mSource->SetActive();
+        break;
+      case STOP:
+        mStop = aParam;
+        break;
+      default:
+        NS_ERROR("Bad ConstantSourceNodeEngine TrackTimeParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    MOZ_ASSERT(mSource == aTrack, "Invalid source track");
+    TRACE("ConstantSourceNodeEngine::ProcessBlock");
+
+    TrackTime ticks = mDestination->GraphTimeToTrackTime(aFrom);
+    if (mStart == -1) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      return;
+    }
+
+    if (ticks + WEBAUDIO_BLOCK_SIZE <= mStart || ticks >= mStop ||
+        mStop <= mStart) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+    } else {
+      aOutput->AllocateChannels(1);
+      float* output = aOutput->ChannelFloatsForWrite(0);
+      uint32_t writeOffset = 0;
+
+      if (ticks < mStart) {
+        MOZ_ASSERT(mStart - ticks <= WEBAUDIO_BLOCK_SIZE);
+        uint32_t count = mStart - ticks;
+        std::fill_n(output, count, 0.0f);
+        writeOffset += count;
+      }
+
+      MOZ_ASSERT(ticks + writeOffset >= mStart);
+      MOZ_ASSERT(mStop - ticks >= writeOffset);
+      uint32_t count =
+          std::min<TrackTime>(WEBAUDIO_BLOCK_SIZE, mStop - ticks) - writeOffset;
+
+      if (mOffset.HasSimpleValue()) {
+        float value = mOffset.GetValue();
+        std::fill_n(output + writeOffset, count, value);
+      } else {
+        mOffset.GetValuesAtTime(ticks + writeOffset, output + writeOffset,
+                                count);
+      }
+
+      writeOffset += count;
+
+      std::fill_n(output + writeOffset, WEBAUDIO_BLOCK_SIZE - writeOffset,
+                  0.0f);
+    }
+
+    if (ticks + WEBAUDIO_BLOCK_SIZE >= mStop) {
+      // We've finished playing.
+      *aFinished = true;
+    }
+  }
+
+  bool IsActive() const override {
+    // start() has been called.
+    return mStart != -1;
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+
+    // Not owned:
+    // - mSource
+    // - mDestination
+    // - mOffset (internal ref owned by node)
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  // mSource deletes the engine in its destructor.
+  AudioNodeTrack* MOZ_NON_OWNING_REF mSource;
+  RefPtr<AudioNodeTrack> mDestination;
+  TrackTime mStart;
+  TrackTime mStop;
+  AudioParamTimeline mOffset;
+};
+
+ConstantSourceNode::ConstantSourceNode(AudioContext* aContext)
+    : AudioScheduledSourceNode(aContext, 2, ChannelCountMode::Max,
+                               ChannelInterpretation::Speakers),
+      mStartCalled(false) {
+  mOffset =
+      CreateAudioParam(ConstantSourceNodeEngine::OFFSET, u"offset"_ns, 1.0f);
+  ConstantSourceNodeEngine* engine =
+      new ConstantSourceNodeEngine(this, aContext->Destination());
+  mTrack = AudioNodeTrack::Create(aContext, engine,
+                                  AudioNodeTrack::NEED_MAIN_THREAD_ENDED,
+                                  aContext->Graph());
+  engine->SetSourceTrack(mTrack);
+  mTrack->AddMainThreadListener(this);
+}
+
+ConstantSourceNode::~ConstantSourceNode() = default;
+
+size_t ConstantSourceNode::SizeOfExcludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+
+  amount += mOffset->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t ConstantSourceNode::SizeOfIncludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject* ConstantSourceNode::WrapObject(JSContext* aCx,
+                                         JS::Handle<JSObject*> aGivenProto) {
+  return ConstantSourceNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+already_AddRefed<ConstantSourceNode> ConstantSourceNode::Constructor(
+    const GlobalObject& aGlobal, AudioContext& aContext,
+    const ConstantSourceOptions& aOptions) {
+  RefPtr<ConstantSourceNode> object = new ConstantSourceNode(&aContext);
+  object->mOffset->SetInitialValue(aOptions.mOffset);
+  return object.forget();
+}
+
+void ConstantSourceNode::DestroyMediaTrack() {
+  if (mTrack) {
+    mTrack->RemoveMainThreadListener(this);
+  }
+  AudioNode::DestroyMediaTrack();
+}
+
+void ConstantSourceNode::Start(double aWhen, ErrorResult& aRv) {
+  if (!WebAudioUtils::IsTimeValid(aWhen)) {
+    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>("start time");
+    return;
+  }
+
+  if (mStartCalled) {
+    aRv.ThrowInvalidStateError("Can't call start() more than once");
+    return;
+  }
+  mStartCalled = true;
+
+  if (!mTrack) {
+    return;
+  }
+
+  mTrack->SetTrackTimeParameter(ConstantSourceNodeEngine::START, Context(),
+                                aWhen);
+
+  MarkActive();
+  Context()->StartBlockedAudioContextIfAllowed();
+}
+
+void ConstantSourceNode::Stop(double aWhen, ErrorResult& aRv) {
+  if (!WebAudioUtils::IsTimeValid(aWhen)) {
+    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>("stop time");
+    return;
+  }
+
+  if (!mStartCalled) {
+    aRv.ThrowInvalidStateError("Can't call stop() without calling start()");
+    return;
+  }
+
+  if (!mTrack || !Context()) {
+    return;
+  }
+
+  mTrack->SetTrackTimeParameter(ConstantSourceNodeEngine::STOP, Context(),
+                                std::max(0.0, aWhen));
+}
+
+void ConstantSourceNode::NotifyMainThreadTrackEnded() {
+  MOZ_ASSERT(mTrack->IsEnded());
+
+  class EndedEventDispatcher final : public Runnable {
+   public:
+    explicit EndedEventDispatcher(ConstantSourceNode* 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(u"ended"_ns);
+      // Release track resources.
+      mNode->DestroyMediaTrack();
+      return NS_OK;
+    }
+
+   private:
+    RefPtr<ConstantSourceNode> mNode;
+  };
+
+  Context()->Dispatch(do_AddRef(new EndedEventDispatcher(this)));
+
+  // Drop the playing reference
+  // Warning: The below line might delete this.
+  MarkInactive();
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/ConstantSourceNode.h b/dom/media/webaudio/ConstantSourceNode.h
new file mode 100644
index 0000000000..61f939a7e6
--- /dev/null
+++ b/dom/media/webaudio/ConstantSourceNode.h
@@ -0,0 +1,60 @@
+/* -*- 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/. */
+
+#ifndef ConstantSourceNode_h_
+#define ConstantSourceNode_h_
+
+#include "AudioScheduledSourceNode.h"
+#include "AudioParam.h"
+#include "mozilla/dom/ConstantSourceNodeBinding.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+
+class ConstantSourceNode final : public AudioScheduledSourceNode,
+                                 public MainThreadMediaTrackListener {
+ public:
+  explicit ConstantSourceNode(AudioContext* aContext);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(ConstantSourceNode,
+                                           AudioScheduledSourceNode)
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  static already_AddRefed<ConstantSourceNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aContext,
+      const ConstantSourceOptions& aOptions);
+
+  void DestroyMediaTrack() override;
+
+  uint16_t NumberOfInputs() const final { return 0; }
+
+  AudioParam* Offset() const { return mOffset; }
+
+  void Start(double aWhen, ErrorResult& rv) override;
+  void Stop(double aWhen, ErrorResult& rv) override;
+
+  void NotifyMainThreadTrackEnded() override;
+
+  const char* NodeType() const override { return "ConstantSourceNode"; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ protected:
+  virtual ~ConstantSourceNode();
+
+ private:
+  RefPtr<AudioParam> mOffset;
+  bool mStartCalled;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/ConvolverNode.cpp b/dom/media/webaudio/ConvolverNode.cpp
new file mode 100644
index 0000000000..65562ae6d0
--- /dev/null
+++ b/dom/media/webaudio/ConvolverNode.cpp
@@ -0,0 +1,479 @@
+/* -*- 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 "ConvolverNode.h"
+#include "mozilla/dom/ConvolverNodeBinding.h"
+#include "AlignmentUtils.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "blink/Reverb.h"
+#include "PlayingRefChangeHandler.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(ConvolverNode, AudioNode, mBuffer)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(ConvolverNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(ConvolverNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(ConvolverNode, AudioNode)
+
+class ConvolverNodeEngine final : public AudioNodeEngine {
+  typedef PlayingRefChangeHandler PlayingRefChanged;
+
+ public:
+  ConvolverNodeEngine(AudioNode* aNode, bool aNormalize)
+      : AudioNodeEngine(aNode) {}
+
+  // Indicates how the right output channel is generated.
+  enum class RightConvolverMode {
+    // A right convolver is always used when there is more than one impulse
+    // response channel.
+    Always,
+    // With a single response channel, the mode may be either Direct or
+    // Difference.  The decision on which to use is made when stereo input is
+    // received.  Once the right convolver is in use, convolver state is
+    // suitable only for the selected mode, and so the mode cannot change
+    // until the right convolver contains only silent history.
+    //
+    // With Direct mode, each convolver processes a corresponding channel.
+    // This mode is selected when input is initially stereo or
+    // channelInterpretation is "discrete" at the time or starting the right
+    // convolver when input changes from non-silent mono to stereo.
+    Direct,
+    // Difference mode is selected if channelInterpretation is "speakers" at
+    // the time starting the right convolver when the input changes from mono
+    // to stereo.
+    //
+    // When non-silent input is initially mono, with a single response
+    // channel, the right output channel is not produced until input becomes
+    // stereo.  Only a single convolver is used for mono processing.  When
+    // stereo input arrives after mono input, output must be as if the mono
+    // signal remaining in the left convolver is up-mixed, but the right
+    // convolver has not been initialized with the history of the mono input.
+    // Copying the state of the left convolver into the right convolver is not
+    // desirable, because there is considerable state to copy, and the
+    // different convolvers are intended to process out of phase, which means
+    // that state from one convolver would not directly map to state in
+    // another convolver.
+    //
+    // Instead the distributive property of convolution is used to generate
+    // the right output channel using information in the left output channel.
+    // Using l and r to denote the left and right channel input signals, g the
+    // impulse response, and * convolution, the convolution of the right
+    // channel can be given by
+    //
+    //   r * g = (l + (r - l)) * g
+    //         = l * g + (r - l) * g
+    //
+    // The left convolver continues to process the left channel l to produce
+    // l * g.  The right convolver processes the difference of input channel
+    // signals r - l to produce (r - l) * g.  The outputs of the two
+    // convolvers are added to generate the right channel output r * g.
+    //
+    // The benefit of doing this is that the history of the r - l input for a
+    // "speakers" up-mixed mono signal is zero, and so an empty convolver
+    // already has exactly the right history for mixing the previous mono
+    // signal with the new stereo signal.
+    Difference
+  };
+
+  void SetReverb(WebCore::Reverb* aReverb,
+                 uint32_t aImpulseChannelCount) override {
+    mRemainingLeftOutput = INT32_MIN;
+    mRemainingRightOutput = 0;
+    mRemainingRightHistory = 0;
+
+    // Assume for now that convolution of channel difference is not required.
+    // Direct may change to Difference during processing.
+    if (aReverb) {
+      mRightConvolverMode = aImpulseChannelCount == 1
+                                ? RightConvolverMode::Direct
+                                : RightConvolverMode::Always;
+    } else {
+      mRightConvolverMode = RightConvolverMode::Always;
+    }
+
+    mReverb.reset(aReverb);
+  }
+
+  void AllocateReverbInput(const AudioBlock& aInput,
+                           uint32_t aTotalChannelCount) {
+    uint32_t inputChannelCount = aInput.ChannelCount();
+    MOZ_ASSERT(inputChannelCount <= aTotalChannelCount);
+    mReverbInput.AllocateChannels(aTotalChannelCount);
+    // Pre-multiply the input's volume
+    for (uint32_t i = 0; i < inputChannelCount; ++i) {
+      const float* src = static_cast<const float*>(aInput.mChannelData[i]);
+      float* dest = mReverbInput.ChannelFloatsForWrite(i);
+      AudioBlockCopyChannelWithScale(src, aInput.mVolume, dest);
+    }
+    // Fill remaining channels with silence
+    for (uint32_t i = inputChannelCount; i < aTotalChannelCount; ++i) {
+      float* dest = mReverbInput.ChannelFloatsForWrite(i);
+      std::fill_n(dest, WEBAUDIO_BLOCK_SIZE, 0.0f);
+    }
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override;
+
+  bool IsActive() const override { return mRemainingLeftOutput != INT32_MIN; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+
+    amount += mReverbInput.SizeOfExcludingThis(aMallocSizeOf, false);
+
+    if (mReverb) {
+      amount += mReverb->sizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  // Keeping mReverbInput across process calls avoids unnecessary reallocation.
+  AudioBlock mReverbInput;
+  UniquePtr<WebCore::Reverb> mReverb;
+  // Tracks samples of the tail remaining to be output.  INT32_MIN is a
+  // special value to indicate that the end of any previous tail has been
+  // handled.
+  int32_t mRemainingLeftOutput = INT32_MIN;
+  // mRemainingRightOutput and mRemainingRightHistory are only used when
+  // mRightOutputMode != Always.  There is no special handling required at the
+  // end of tail times and so INT32_MIN is not used.
+  // mRemainingRightOutput tracks how much longer this node needs to continue
+  // to produce a right output channel.
+  int32_t mRemainingRightOutput = 0;
+  // mRemainingRightHistory tracks how much silent input would be required to
+  // drain the right convolver, which may sometimes be longer than the period
+  // a right output channel is required.
+  int32_t mRemainingRightHistory = 0;
+  RightConvolverMode mRightConvolverMode = RightConvolverMode::Always;
+};
+
+static void AddScaledLeftToRight(AudioBlock* aBlock, float aScale) {
+  const float* left = static_cast<const float*>(aBlock->mChannelData[0]);
+  float* right = aBlock->ChannelFloatsForWrite(1);
+  AudioBlockAddChannelWithScale(left, aScale, right);
+}
+
+void ConvolverNodeEngine::ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                                       const AudioBlock& aInput,
+                                       AudioBlock* aOutput, bool* aFinished) {
+  TRACE("ConvolverNodeEngine::ProcessBlock");
+  if (!mReverb) {
+    aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  uint32_t inputChannelCount = aInput.ChannelCount();
+  if (aInput.IsNull()) {
+    if (mRemainingLeftOutput > 0) {
+      mRemainingLeftOutput -= WEBAUDIO_BLOCK_SIZE;
+      AllocateReverbInput(aInput, 1);  // floats for silence
+    } else {
+      if (mRemainingLeftOutput != INT32_MIN) {
+        mRemainingLeftOutput = INT32_MIN;
+        MOZ_ASSERT(mRemainingRightOutput <= 0);
+        MOZ_ASSERT(mRemainingRightHistory <= 0);
+        aTrack->ScheduleCheckForInactive();
+        RefPtr<PlayingRefChanged> refchanged =
+            new PlayingRefChanged(aTrack, PlayingRefChanged::RELEASE);
+        aTrack->Graph()->DispatchToMainThreadStableState(refchanged.forget());
+      }
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      return;
+    }
+  } else {
+    if (mRemainingLeftOutput <= 0) {
+      RefPtr<PlayingRefChanged> refchanged =
+          new PlayingRefChanged(aTrack, PlayingRefChanged::ADDREF);
+      aTrack->Graph()->DispatchToMainThreadStableState(refchanged.forget());
+    }
+
+    // Use mVolume as a flag to detect whether AllocateReverbInput() gets
+    // called.
+    mReverbInput.mVolume = 0.0f;
+
+    // Special handling of input channel count changes is used when there is
+    // only a single impulse response channel.  See RightConvolverMode.
+    if (mRightConvolverMode != RightConvolverMode::Always) {
+      ChannelInterpretation channelInterpretation =
+          aTrack->GetChannelInterpretation();
+      if (inputChannelCount == 2) {
+        if (mRemainingRightHistory <= 0) {
+          // Will start the second convolver.  Choose to convolve the right
+          // channel directly if there is no left tail to up-mix or up-mixing
+          // is "discrete".
+          mRightConvolverMode =
+              (mRemainingLeftOutput <= 0 ||
+               channelInterpretation == ChannelInterpretation::Discrete)
+                  ? RightConvolverMode::Direct
+                  : RightConvolverMode::Difference;
+        }
+        // The extra WEBAUDIO_BLOCK_SIZE is subtracted below.
+        mRemainingRightOutput =
+            mReverb->impulseResponseLength() + WEBAUDIO_BLOCK_SIZE;
+        mRemainingRightHistory = mRemainingRightOutput;
+        if (mRightConvolverMode == RightConvolverMode::Difference) {
+          AllocateReverbInput(aInput, 2);
+          // Subtract left from right.
+          AddScaledLeftToRight(&mReverbInput, -1.0f);
+        }
+      } else if (mRemainingRightHistory > 0) {
+        // There is one channel of input, but a second convolver also
+        // requires input.  Up-mix appropriately for the second convolver.
+        if ((mRightConvolverMode == RightConvolverMode::Difference) ^
+            (channelInterpretation == ChannelInterpretation::Discrete)) {
+          MOZ_ASSERT(
+              (mRightConvolverMode == RightConvolverMode::Difference &&
+               channelInterpretation == ChannelInterpretation::Speakers) ||
+              (mRightConvolverMode == RightConvolverMode::Direct &&
+               channelInterpretation == ChannelInterpretation::Discrete));
+          // The state is one of the following combinations:
+          // 1) Difference and speakers.
+          //    Up-mixing gives r = l.
+          //    The input to the second convolver is r - l.
+          // 2) Direct and discrete.
+          //    Up-mixing gives r = 0.
+          //    The input to the second convolver is r.
+          //
+          // In each case the input for the second convolver is silence, which
+          // will drain the convolver.
+          AllocateReverbInput(aInput, 2);
+        } else {
+          if (channelInterpretation == ChannelInterpretation::Discrete) {
+            MOZ_ASSERT(mRightConvolverMode == RightConvolverMode::Difference);
+            // channelInterpretation has changed since the second convolver
+            // was added.  "discrete" up-mixing of input would produce a
+            // silent right channel r = 0, but the second convolver needs
+            // r - l for RightConvolverMode::Difference.
+            AllocateReverbInput(aInput, 2);
+            AddScaledLeftToRight(&mReverbInput, -1.0f);
+          } else {
+            MOZ_ASSERT(channelInterpretation ==
+                       ChannelInterpretation::Speakers);
+            MOZ_ASSERT(mRightConvolverMode == RightConvolverMode::Direct);
+            // The Reverb will essentially up-mix the single input channel by
+            // feeding it into both convolvers.
+          }
+          // The second convolver does not have silent input, and so it will
+          // not drain.  It will need to continue processing up-mixed input
+          // because the next input block may be stereo, which would be mixed
+          // with the signal remaining in the convolvers.
+          // The extra WEBAUDIO_BLOCK_SIZE is subtracted below.
+          mRemainingRightHistory =
+              mReverb->impulseResponseLength() + WEBAUDIO_BLOCK_SIZE;
+        }
+      }
+    }
+
+    if (mReverbInput.mVolume == 0.0f) {  // not yet set
+      if (aInput.mVolume != 1.0f) {
+        AllocateReverbInput(aInput, inputChannelCount);  // pre-multiply
+      } else {
+        mReverbInput = aInput;
+      }
+    }
+
+    mRemainingLeftOutput = mReverb->impulseResponseLength();
+    MOZ_ASSERT(mRemainingLeftOutput > 0);
+  }
+
+  // "The ConvolverNode produces a mono output only in the single case where
+  // there is a single input channel and a single-channel buffer."
+  uint32_t outputChannelCount = 2;
+  uint32_t reverbOutputChannelCount = 2;
+  if (mRightConvolverMode != RightConvolverMode::Always) {
+    // When the input changes from stereo to mono, the output continues to be
+    // stereo for the length of the tail time, during which the two channels
+    // may differ.
+    if (mRemainingRightOutput > 0) {
+      MOZ_ASSERT(mRemainingRightHistory > 0);
+      mRemainingRightOutput -= WEBAUDIO_BLOCK_SIZE;
+    } else {
+      outputChannelCount = 1;
+    }
+    // The second convolver keeps processing until it drains.
+    if (mRemainingRightHistory > 0) {
+      mRemainingRightHistory -= WEBAUDIO_BLOCK_SIZE;
+    } else {
+      reverbOutputChannelCount = 1;
+    }
+  }
+
+  // If there are two convolvers, then they each need an output buffer, even
+  // if the second convolver is only processing to keep history of up-mixed
+  // input.
+  aOutput->AllocateChannels(reverbOutputChannelCount);
+
+  mReverb->process(&mReverbInput, aOutput);
+
+  if (mRightConvolverMode == RightConvolverMode::Difference &&
+      outputChannelCount == 2) {
+    // Add left to right.
+    AddScaledLeftToRight(aOutput, 1.0f);
+  } else {
+    // Trim if outputChannelCount < reverbOutputChannelCount
+    aOutput->mChannelData.TruncateLength(outputChannelCount);
+  }
+}
+
+ConvolverNode::ConvolverNode(AudioContext* aContext)
+    : AudioNode(aContext, 2, ChannelCountMode::Clamped_max,
+                ChannelInterpretation::Speakers),
+      mNormalize(true) {
+  ConvolverNodeEngine* engine = new ConvolverNodeEngine(this, mNormalize);
+  mTrack = AudioNodeTrack::Create(
+      aContext, engine, AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+}
+
+/* static */
+already_AddRefed<ConvolverNode> ConvolverNode::Create(
+    JSContext* aCx, AudioContext& aAudioContext,
+    const ConvolverOptions& aOptions, ErrorResult& aRv) {
+  RefPtr<ConvolverNode> audioNode = new ConvolverNode(&aAudioContext);
+
+  audioNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  // This must be done before setting the buffer.
+  audioNode->SetNormalize(!aOptions.mDisableNormalization);
+
+  if (aOptions.mBuffer.WasPassed()) {
+    MOZ_ASSERT(aCx);
+    audioNode->SetBuffer(aCx, aOptions.mBuffer.Value(), aRv);
+    if (NS_WARN_IF(aRv.Failed())) {
+      return nullptr;
+    }
+  }
+
+  return audioNode.forget();
+}
+
+size_t ConvolverNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  if (mBuffer) {
+    // NB: mBuffer might be shared with the associated engine, by convention
+    //     the AudioNode will report.
+    amount += mBuffer->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  return amount;
+}
+
+size_t ConvolverNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject* ConvolverNode::WrapObject(JSContext* aCx,
+                                    JS::Handle<JSObject*> aGivenProto) {
+  return ConvolverNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void ConvolverNode::SetBuffer(JSContext* aCx, AudioBuffer* aBuffer,
+                              ErrorResult& aRv) {
+  if (aBuffer) {
+    switch (aBuffer->NumberOfChannels()) {
+      case 1:
+      case 2:
+      case 4:
+        // Supported number of channels
+        break;
+      default:
+        aRv.ThrowNotSupportedError(
+            nsPrintfCString("%u is not a supported number of channels",
+                            aBuffer->NumberOfChannels()));
+        return;
+    }
+  }
+
+  if (aBuffer && (aBuffer->SampleRate() != Context()->SampleRate())) {
+    aRv.ThrowNotSupportedError(nsPrintfCString(
+        "Buffer sample rate (%g) does not match AudioContext sample rate (%g)",
+        aBuffer->SampleRate(), Context()->SampleRate()));
+    return;
+  }
+
+  // Send the buffer to the track
+  AudioNodeTrack* ns = mTrack;
+  MOZ_ASSERT(ns, "Why don't we have a track here?");
+  if (aBuffer) {
+    AudioChunk data = aBuffer->GetThreadSharedChannelsForRate(aCx);
+    if (data.mBufferFormat == AUDIO_FORMAT_S16) {
+      // Reverb expects data in float format.
+      // Convert on the main thread so as to minimize allocations on the audio
+      // thread.
+      // Reverb will dispose of the buffer once initialized, so convert here
+      // and leave the smaller arrays in the AudioBuffer.
+      // There is currently no value in providing 16/32-byte aligned data
+      // because PadAndMakeScaledDFT() will copy the data (without SIMD
+      // instructions) to aligned arrays for the FFT.
+      CheckedInt<size_t> bufferSize(sizeof(float));
+      bufferSize *= data.mDuration;
+      bufferSize *= data.ChannelCount();
+      RefPtr<SharedBuffer> floatBuffer =
+          SharedBuffer::Create(bufferSize, fallible);
+      if (!floatBuffer) {
+        aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+        return;
+      }
+      auto floatData = static_cast<float*>(floatBuffer->Data());
+      for (size_t i = 0; i < data.ChannelCount(); ++i) {
+        ConvertAudioSamples(data.ChannelData<int16_t>()[i], floatData,
+                            data.mDuration);
+        data.mChannelData[i] = floatData;
+        floatData += data.mDuration;
+      }
+      data.mBuffer = std::move(floatBuffer);
+      data.mBufferFormat = AUDIO_FORMAT_FLOAT32;
+    } else if (data.mBufferFormat == AUDIO_FORMAT_SILENCE) {
+      // This is valid, but a signal convolved by a silent signal is silent, set
+      // the reverb to nullptr and return.
+      ns->SetReverb(nullptr, 0);
+      mBuffer = aBuffer;
+      return;
+    }
+
+    // Note about empirical tuning (this is copied from Blink)
+    // The maximum FFT size affects reverb performance and accuracy.
+    // If the reverb is single-threaded and processes entirely in the real-time
+    // audio thread, it's important not to make this too high.  In this case
+    // 8192 is a good value. But, the Reverb object is multi-threaded, so we
+    // want this as high as possible without losing too much accuracy. Very
+    // large FFTs will have worse phase errors. Given these constraints 32768 is
+    // a good compromise.
+    const size_t MaxFFTSize = 32768;
+
+    bool allocationFailure = false;
+    UniquePtr<WebCore::Reverb> reverb(new WebCore::Reverb(
+        data, MaxFFTSize, !Context()->IsOffline(), mNormalize,
+        aBuffer->SampleRate(), &allocationFailure));
+    if (!allocationFailure) {
+      ns->SetReverb(reverb.release(), data.ChannelCount());
+    } else {
+      aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+      return;
+    }
+  } else {
+    ns->SetReverb(nullptr, 0);
+  }
+  mBuffer = aBuffer;
+}
+
+void ConvolverNode::SetNormalize(bool aNormalize) { mNormalize = aNormalize; }
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/ConvolverNode.h b/dom/media/webaudio/ConvolverNode.h
new file mode 100644
index 0000000000..897f8d6547
--- /dev/null
+++ b/dom/media/webaudio/ConvolverNode.h
@@ -0,0 +1,77 @@
+/* -*- 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/. */
+
+#ifndef ConvolverNode_h_
+#define ConvolverNode_h_
+
+#include "AudioNode.h"
+#include "AudioBuffer.h"
+#include "nsPrintfCString.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct ConvolverOptions;
+
+class ConvolverNode final : public AudioNode {
+ public:
+  static already_AddRefed<ConvolverNode> Create(
+      JSContext* aCx, AudioContext& aAudioContext,
+      const ConvolverOptions& aOptions, ErrorResult& aRv);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(ConvolverNode, AudioNode);
+
+  static already_AddRefed<ConvolverNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const ConvolverOptions& aOptions, ErrorResult& aRv) {
+    return Create(aGlobal.Context(), aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  AudioBuffer* GetBuffer(JSContext* aCx) const { return mBuffer; }
+
+  void SetBuffer(JSContext* aCx, AudioBuffer* aBuffer, ErrorResult& aRv);
+
+  bool Normalize() const { return mNormalize; }
+
+  void SetNormalize(bool aNormal);
+
+  void SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv) override {
+    if (aChannelCount > 2) {
+      aRv.ThrowNotSupportedError(
+          nsPrintfCString("%u is greater than 2", aChannelCount));
+      return;
+    }
+    AudioNode::SetChannelCount(aChannelCount, aRv);
+  }
+  void SetChannelCountModeValue(ChannelCountMode aMode,
+                                ErrorResult& aRv) override {
+    if (aMode == ChannelCountMode::Max) {
+      aRv.ThrowNotSupportedError("Cannot set channel count mode to \"max\"");
+      return;
+    }
+    AudioNode::SetChannelCountModeValue(aMode, aRv);
+  }
+
+  const char* NodeType() const override { return "ConvolverNode"; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ private:
+  explicit ConvolverNode(AudioContext* aContext);
+  ~ConvolverNode() = default;
+
+  RefPtr<AudioBuffer> mBuffer;
+  bool mNormalize;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/DelayBuffer.cpp b/dom/media/webaudio/DelayBuffer.cpp
new file mode 100644
index 0000000000..52c8d4184b
--- /dev/null
+++ b/dom/media/webaudio/DelayBuffer.cpp
@@ -0,0 +1,236 @@
+/* -*- 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 "DelayBuffer.h"
+
+#include "mozilla/PodOperations.h"
+#include "AudioChannelFormat.h"
+#include "AudioNodeEngine.h"
+
+namespace mozilla {
+
+size_t DelayBuffer::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = 0;
+  amount += mChunks.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < mChunks.Length(); i++) {
+    amount += mChunks[i].SizeOfExcludingThis(aMallocSizeOf, false);
+  }
+
+  amount += mUpmixChannels.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+void DelayBuffer::Write(const AudioBlock& aInputChunk) {
+  // We must have a reference to the buffer if there are channels
+  MOZ_ASSERT(aInputChunk.IsNull() == !aInputChunk.ChannelCount());
+#ifdef DEBUG
+  MOZ_ASSERT(!mHaveWrittenBlock);
+  mHaveWrittenBlock = true;
+#endif
+
+  if (!EnsureBuffer()) {
+    return;
+  }
+
+  if (mCurrentChunk == mLastReadChunk) {
+    mLastReadChunk = -1;  // invalidate cache
+  }
+  mChunks[mCurrentChunk] = aInputChunk.AsAudioChunk();
+}
+
+void DelayBuffer::Read(const float aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
+                       AudioBlock* aOutputChunk,
+                       ChannelInterpretation aChannelInterpretation) {
+  int chunkCount = mChunks.Length();
+  if (!chunkCount) {
+    aOutputChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  // Find the maximum number of contributing channels to determine the output
+  // channel count that retains all signal information.  Buffered blocks will
+  // be upmixed if necessary.
+  //
+  // First find the range of "delay" offsets backwards from the current
+  // position.  Note that these may be negative for frames that are after the
+  // current position (including i).
+  float minDelay = aPerFrameDelays[0];
+  float maxDelay = minDelay;
+  for (unsigned i = 1; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+    minDelay = std::min(minDelay, aPerFrameDelays[i] - i);
+    maxDelay = std::max(maxDelay, aPerFrameDelays[i] - i);
+  }
+
+  // Now find the chunks touched by this range and check their channel counts.
+  int oldestChunk = ChunkForDelay(std::ceil(maxDelay));
+  int youngestChunk = ChunkForDelay(std::floor(minDelay));
+
+  uint32_t channelCount = 0;
+  for (int i = oldestChunk; true; i = (i + 1) % chunkCount) {
+    channelCount =
+        GetAudioChannelsSuperset(channelCount, mChunks[i].ChannelCount());
+    if (i == youngestChunk) {
+      break;
+    }
+  }
+
+  if (channelCount) {
+    aOutputChunk->AllocateChannels(channelCount);
+    ReadChannels(aPerFrameDelays, aOutputChunk, 0, channelCount,
+                 aChannelInterpretation);
+  } else {
+    aOutputChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
+  }
+}
+
+void DelayBuffer::ReadChannel(const float aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
+                              AudioBlock* aOutputChunk, uint32_t aChannel,
+                              ChannelInterpretation aChannelInterpretation) {
+  if (!mChunks.Length()) {
+    float* outputChannel = aOutputChunk->ChannelFloatsForWrite(aChannel);
+    PodZero(outputChannel, WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  ReadChannels(aPerFrameDelays, aOutputChunk, aChannel, 1,
+               aChannelInterpretation);
+}
+
+void DelayBuffer::ReadChannels(const float aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
+                               AudioBlock* aOutputChunk, uint32_t aFirstChannel,
+                               uint32_t aNumChannelsToRead,
+                               ChannelInterpretation aChannelInterpretation) {
+  uint32_t totalChannelCount = aOutputChunk->ChannelCount();
+  uint32_t readChannelsEnd = aFirstChannel + aNumChannelsToRead;
+  MOZ_ASSERT(readChannelsEnd <= totalChannelCount);
+
+  if (mUpmixChannels.Length() != totalChannelCount) {
+    mLastReadChunk = -1;  // invalidate cache
+  }
+
+  for (uint32_t channel = aFirstChannel; channel < readChannelsEnd; ++channel) {
+    PodZero(aOutputChunk->ChannelFloatsForWrite(channel), WEBAUDIO_BLOCK_SIZE);
+  }
+
+  for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+    float currentDelay = aPerFrameDelays[i];
+    MOZ_ASSERT(currentDelay >= 0.0f);
+    MOZ_ASSERT(currentDelay <= (mChunks.Length() - 1) * WEBAUDIO_BLOCK_SIZE);
+
+    // Interpolate two input frames in case the read position does not match
+    // an integer index.
+    // Use the larger delay, for the older frame, first, as this is more
+    // likely to use the cached upmixed channel arrays.
+    int floorDelay = int(currentDelay);
+    float interpolationFactor = currentDelay - floorDelay;
+    int positions[2];
+    positions[1] = PositionForDelay(floorDelay) + i;
+    positions[0] = positions[1] - 1;
+
+    for (unsigned tick = 0; tick < ArrayLength(positions); ++tick) {
+      int readChunk = ChunkForPosition(positions[tick]);
+      // The zero check on interpolationFactor is important because, when
+      // currentDelay is integer, positions[0] may be outside the range
+      // considered for determining totalChannelCount.
+      // mVolume is not set on default initialized chunks so also handle null
+      // chunks specially.
+      if (interpolationFactor != 0.0f && !mChunks[readChunk].IsNull()) {
+        int readOffset = OffsetForPosition(positions[tick]);
+        UpdateUpmixChannels(readChunk, totalChannelCount,
+                            aChannelInterpretation);
+        float multiplier = interpolationFactor * mChunks[readChunk].mVolume;
+        for (uint32_t channel = aFirstChannel; channel < readChannelsEnd;
+             ++channel) {
+          aOutputChunk->ChannelFloatsForWrite(channel)[i] +=
+              multiplier * mUpmixChannels[channel][readOffset];
+        }
+      }
+
+      interpolationFactor = 1.0f - interpolationFactor;
+    }
+  }
+}
+
+void DelayBuffer::Read(float aDelayTicks, AudioBlock* aOutputChunk,
+                       ChannelInterpretation aChannelInterpretation) {
+  float computedDelay[WEBAUDIO_BLOCK_SIZE];
+
+  for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+    computedDelay[i] = aDelayTicks;
+  }
+
+  Read(computedDelay, aOutputChunk, aChannelInterpretation);
+}
+
+bool DelayBuffer::EnsureBuffer() {
+  if (mChunks.Length() == 0) {
+    // The length of the buffer is at least one block greater than the maximum
+    // delay so that writing an input block does not overwrite the block that
+    // would subsequently be read at maximum delay.  Also round up to the next
+    // block size, so that no block of writes will need to wrap.
+    const int chunkCount = (mMaxDelayTicks + 2 * WEBAUDIO_BLOCK_SIZE - 1) >>
+                           WEBAUDIO_BLOCK_SIZE_BITS;
+    if (!mChunks.SetLength(chunkCount, fallible)) {
+      return false;
+    }
+
+    mLastReadChunk = -1;
+  }
+  return true;
+}
+
+int DelayBuffer::PositionForDelay(int aDelay) {
+  // Adding mChunks.Length() keeps integers positive for defined and
+  // appropriate bitshift, remainder, and bitwise operations.
+  return ((mCurrentChunk + mChunks.Length()) * WEBAUDIO_BLOCK_SIZE) - aDelay;
+}
+
+int DelayBuffer::ChunkForPosition(int aPosition) {
+  MOZ_ASSERT(aPosition >= 0);
+  return (aPosition >> WEBAUDIO_BLOCK_SIZE_BITS) % mChunks.Length();
+}
+
+int DelayBuffer::OffsetForPosition(int aPosition) {
+  MOZ_ASSERT(aPosition >= 0);
+  return aPosition & (WEBAUDIO_BLOCK_SIZE - 1);
+}
+
+int DelayBuffer::ChunkForDelay(int aDelay) {
+  return ChunkForPosition(PositionForDelay(aDelay));
+}
+
+void DelayBuffer::UpdateUpmixChannels(
+    int aNewReadChunk, uint32_t aChannelCount,
+    ChannelInterpretation aChannelInterpretation) {
+  if (aNewReadChunk == mLastReadChunk) {
+    MOZ_ASSERT(mUpmixChannels.Length() == aChannelCount);
+    return;
+  }
+
+  NS_WARNING_ASSERTION(mHaveWrittenBlock || aNewReadChunk != mCurrentChunk,
+                       "Smoothing is making feedback delay too small.");
+
+  mLastReadChunk = aNewReadChunk;
+  mUpmixChannels.ClearAndRetainStorage();
+  mUpmixChannels.AppendElements(mChunks[aNewReadChunk].ChannelData<float>());
+  MOZ_ASSERT(mUpmixChannels.Length() <= aChannelCount);
+  if (mUpmixChannels.Length() < aChannelCount) {
+    if (aChannelInterpretation == ChannelInterpretation::Speakers) {
+      AudioChannelsUpMix(&mUpmixChannels, aChannelCount,
+                         SilentChannel::ZeroChannel<float>());
+      MOZ_ASSERT(mUpmixChannels.Length() == aChannelCount,
+                 "We called GetAudioChannelsSuperset to avoid this");
+    } else {
+      // Fill up the remaining channels with zeros
+      for (uint32_t channel = mUpmixChannels.Length(); channel < aChannelCount;
+           ++channel) {
+        mUpmixChannels.AppendElement(SilentChannel::ZeroChannel<float>());
+      }
+    }
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webaudio/DelayBuffer.h b/dom/media/webaudio/DelayBuffer.h
new file mode 100644
index 0000000000..4b1135fa56
--- /dev/null
+++ b/dom/media/webaudio/DelayBuffer.h
@@ -0,0 +1,105 @@
+/* -*- 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/. */
+
+#ifndef DelayBuffer_h_
+#define DelayBuffer_h_
+
+#include "nsTArray.h"
+#include "AudioBlock.h"
+#include "AudioSegment.h"
+#include "mozilla/dom/AudioNodeBinding.h"  // for ChannelInterpretation
+
+namespace mozilla {
+
+class DelayBuffer final {
+  typedef dom::ChannelInterpretation ChannelInterpretation;
+
+ public:
+  explicit DelayBuffer(float aMaxDelayTicks)
+      // Round the maximum delay up to the next tick.
+      : mMaxDelayTicks(std::ceil(aMaxDelayTicks)),
+        mCurrentChunk(0)
+  // mLastReadChunk is initialized in EnsureBuffer
+#ifdef DEBUG
+        ,
+        mHaveWrittenBlock(false)
+#endif
+  {
+    // The 180 second limit in AudioContext::CreateDelay() and the
+    // 1 << MEDIA_TIME_FRAC_BITS limit on sample rate provide a limit on the
+    // maximum delay.
+    MOZ_ASSERT(aMaxDelayTicks <=
+               float(std::numeric_limits<decltype(mMaxDelayTicks)>::max()));
+  }
+
+  // Write a WEBAUDIO_BLOCK_SIZE block for aChannelCount channels.
+  void Write(const AudioBlock& aInputChunk);
+
+  // Read a block with an array of delays, in ticks, for each sample frame.
+  // Each delay should be >= 0 and <= MaxDelayTicks().
+  void Read(const float aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
+            AudioBlock* aOutputChunk,
+            ChannelInterpretation aChannelInterpretation);
+  // Read a block with a constant delay. The delay should be >= 0 and
+  // <= MaxDelayTicks().
+  void Read(float aDelayTicks, AudioBlock* aOutputChunk,
+            ChannelInterpretation aChannelInterpretation);
+
+  // Read into one of the channels of aOutputChunk, given an array of
+  // delays in ticks.  This is useful when delays are different on different
+  // channels.  aOutputChunk must have already been allocated with at least as
+  // many channels as were in any of the blocks passed to Write().
+  void ReadChannel(const float aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
+                   AudioBlock* aOutputChunk, uint32_t aChannel,
+                   ChannelInterpretation aChannelInterpretation);
+
+  // Advance the buffer pointer
+  void NextBlock() {
+    mCurrentChunk = (mCurrentChunk + 1) % mChunks.Length();
+#ifdef DEBUG
+    MOZ_ASSERT(mHaveWrittenBlock);
+    mHaveWrittenBlock = false;
+#endif
+  }
+
+  void Reset() { mChunks.Clear(); };
+
+  int MaxDelayTicks() const { return mMaxDelayTicks; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const;
+
+ private:
+  void ReadChannels(const float aPerFrameDelays[WEBAUDIO_BLOCK_SIZE],
+                    AudioBlock* aOutputChunk, uint32_t aFirstChannel,
+                    uint32_t aNumChannelsToRead,
+                    ChannelInterpretation aChannelInterpretation);
+  bool EnsureBuffer();
+  int PositionForDelay(int aDelay);
+  int ChunkForPosition(int aPosition);
+  int OffsetForPosition(int aPosition);
+  int ChunkForDelay(int aDelay);
+  void UpdateUpmixChannels(int aNewReadChunk, uint32_t channelCount,
+                           ChannelInterpretation aChannelInterpretation);
+
+  // Circular buffer for capturing delayed samples.
+  FallibleTArray<AudioChunk> mChunks;
+  // Cached upmixed channel arrays, to avoid repeated allocations.
+  CopyableAutoTArray<const float*, GUESS_AUDIO_CHANNELS> mUpmixChannels;
+  // Maximum delay, in ticks
+  int mMaxDelayTicks;
+  // The current position in the circular buffer.  The next write will be to
+  // this chunk, and the next read may begin before this chunk.
+  int mCurrentChunk;
+  // The chunk owning the pointers in mUpmixChannels
+  int mLastReadChunk;
+#ifdef DEBUG
+  bool mHaveWrittenBlock;
+#endif
+};
+
+}  // namespace mozilla
+
+#endif  // DelayBuffer_h_
diff --git a/dom/media/webaudio/DelayNode.cpp b/dom/media/webaudio/DelayNode.cpp
new file mode 100644
index 0000000000..c531756794
--- /dev/null
+++ b/dom/media/webaudio/DelayNode.cpp
@@ -0,0 +1,223 @@
+/* -*- 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 "DelayNode.h"
+#include "mozilla/dom/DelayNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "AudioDestinationNode.h"
+#include "WebAudioUtils.h"
+#include "DelayBuffer.h"
+#include "PlayingRefChangeHandler.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(DelayNode, AudioNode, mDelay)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(DelayNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(DelayNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(DelayNode, AudioNode)
+
+class DelayNodeEngine final : public AudioNodeEngine {
+  typedef PlayingRefChangeHandler PlayingRefChanged;
+
+ public:
+  DelayNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination,
+                  float aMaxDelayTicks)
+      : AudioNodeEngine(aNode),
+        mDestination(aDestination->Track())
+        // Keep the default value in sync with the default value in
+        // DelayNode::DelayNode.
+        ,
+        mDelay(0.f)
+        // Use a smoothing range of 20ms
+        ,
+        mBuffer(
+            std::max(aMaxDelayTicks, static_cast<float>(WEBAUDIO_BLOCK_SIZE))),
+        mMaxDelay(aMaxDelayTicks),
+        mHaveProducedBeforeInput(false),
+        mLeftOverData(INT32_MIN) {}
+
+  DelayNodeEngine* AsDelayNodeEngine() override { return this; }
+
+  enum Parameters {
+    DELAY,
+  };
+  void RecvTimelineEvent(uint32_t aIndex, AudioParamEvent& aEvent) override {
+    MOZ_ASSERT(mDestination);
+    aEvent.ConvertToTicks(mDestination);
+
+    switch (aIndex) {
+      case DELAY:
+        mDelay.InsertEvent<int64_t>(aEvent);
+        break;
+      default:
+        NS_ERROR("Bad DelayNodeEngine TimelineParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    MOZ_ASSERT(aTrack->mSampleRate == mDestination->mSampleRate);
+    TRACE("DelayNodeEngine::ProcessBlock");
+
+    if (!aInput.IsSilentOrSubnormal()) {
+      if (mLeftOverData <= 0) {
+        RefPtr<PlayingRefChanged> refchanged =
+            new PlayingRefChanged(aTrack, PlayingRefChanged::ADDREF);
+        aTrack->Graph()->DispatchToMainThreadStableState(refchanged.forget());
+      }
+      mLeftOverData = mBuffer.MaxDelayTicks();
+    } else if (mLeftOverData > 0) {
+      mLeftOverData -= WEBAUDIO_BLOCK_SIZE;
+    } else {
+      if (mLeftOverData != INT32_MIN) {
+        mLeftOverData = INT32_MIN;
+        aTrack->ScheduleCheckForInactive();
+
+        // Delete our buffered data now we no longer need it
+        mBuffer.Reset();
+
+        RefPtr<PlayingRefChanged> refchanged =
+            new PlayingRefChanged(aTrack, PlayingRefChanged::RELEASE);
+        aTrack->Graph()->DispatchToMainThreadStableState(refchanged.forget());
+      }
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      return;
+    }
+
+    mBuffer.Write(aInput);
+
+    // Skip output update if mLastChunks has already been set by
+    // ProduceBlockBeforeInput() when in a cycle.
+    if (!mHaveProducedBeforeInput) {
+      UpdateOutputBlock(aTrack, aFrom, aOutput, 0.0);
+    }
+    mHaveProducedBeforeInput = false;
+    mBuffer.NextBlock();
+  }
+
+  void UpdateOutputBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                         AudioBlock* aOutput, float minDelay) {
+    float maxDelay = mMaxDelay;
+    float sampleRate = aTrack->mSampleRate;
+    ChannelInterpretation channelInterpretation =
+        aTrack->GetChannelInterpretation();
+    if (mDelay.HasSimpleValue()) {
+      // If this DelayNode is in a cycle, make sure the delay value is at least
+      // one block, even if that is greater than maxDelay.
+      float delayFrames = mDelay.GetValue() * sampleRate;
+      float delayFramesClamped =
+          std::max(minDelay, std::min(delayFrames, maxDelay));
+      mBuffer.Read(delayFramesClamped, aOutput, channelInterpretation);
+    } else {
+      // Compute the delay values for the duration of the input AudioChunk
+      // If this DelayNode is in a cycle, make sure the delay value is at least
+      // one block.
+      TrackTime tick = mDestination->GraphTimeToTrackTime(aFrom);
+      float values[WEBAUDIO_BLOCK_SIZE];
+      mDelay.GetValuesAtTime(tick, values, WEBAUDIO_BLOCK_SIZE);
+
+      float computedDelay[WEBAUDIO_BLOCK_SIZE];
+      for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
+        float delayAtTick = values[counter] * sampleRate;
+        float delayAtTickClamped =
+            std::max(minDelay, std::min(delayAtTick, maxDelay));
+        computedDelay[counter] = delayAtTickClamped;
+      }
+      mBuffer.Read(computedDelay, aOutput, channelInterpretation);
+    }
+  }
+
+  void ProduceBlockBeforeInput(AudioNodeTrack* aTrack, GraphTime aFrom,
+                               AudioBlock* aOutput) override {
+    if (mLeftOverData <= 0) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+    } else {
+      UpdateOutputBlock(aTrack, aFrom, aOutput, WEBAUDIO_BLOCK_SIZE);
+    }
+    mHaveProducedBeforeInput = true;
+  }
+
+  bool IsActive() const override { return mLeftOverData != INT32_MIN; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    // Not owned:
+    // - mDestination - probably not owned
+    // - mDelay - shares ref with AudioNode, don't count
+    amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  RefPtr<AudioNodeTrack> mDestination;
+  AudioParamTimeline mDelay;
+  DelayBuffer mBuffer;
+  float mMaxDelay;
+  bool mHaveProducedBeforeInput;
+  // How much data we have in our buffer which needs to be flushed out when our
+  // inputs finish.
+  int32_t mLeftOverData;
+};
+
+DelayNode::DelayNode(AudioContext* aContext, double aMaxDelay)
+    : AudioNode(aContext, 2, ChannelCountMode::Max,
+                ChannelInterpretation::Speakers) {
+  mDelay = CreateAudioParam(DelayNodeEngine::DELAY, u"delayTime"_ns, 0.0f, 0.f,
+                            aMaxDelay);
+  DelayNodeEngine* engine = new DelayNodeEngine(
+      this, aContext->Destination(), aContext->SampleRate() * aMaxDelay);
+  mTrack = AudioNodeTrack::Create(
+      aContext, engine, AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+}
+
+/* static */
+already_AddRefed<DelayNode> DelayNode::Create(AudioContext& aAudioContext,
+                                              const DelayOptions& aOptions,
+                                              ErrorResult& aRv) {
+  if (aOptions.mMaxDelayTime <= 0. || aOptions.mMaxDelayTime >= 180.) {
+    aRv.ThrowNotSupportedError(
+        nsPrintfCString("\"maxDelayTime\" (%g) is not in the range (0,180)",
+                        aOptions.mMaxDelayTime));
+    return nullptr;
+  }
+
+  RefPtr<DelayNode> audioNode =
+      new DelayNode(&aAudioContext, aOptions.mMaxDelayTime);
+
+  audioNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  audioNode->DelayTime()->SetInitialValue(aOptions.mDelayTime);
+  return audioNode.forget();
+}
+
+size_t DelayNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mDelay->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t DelayNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject* DelayNode::WrapObject(JSContext* aCx,
+                                JS::Handle<JSObject*> aGivenProto) {
+  return DelayNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/DelayNode.h b/dom/media/webaudio/DelayNode.h
new file mode 100644
index 0000000000..bdfe9eba68
--- /dev/null
+++ b/dom/media/webaudio/DelayNode.h
@@ -0,0 +1,55 @@
+/* -*- 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/. */
+
+#ifndef DelayNode_h_
+#define DelayNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct DelayOptions;
+
+class DelayNode final : public AudioNode {
+ public:
+  static already_AddRefed<DelayNode> Create(AudioContext& aAudioContext,
+                                            const DelayOptions& aOptions,
+                                            ErrorResult& aRv);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(DelayNode, AudioNode)
+
+  static already_AddRefed<DelayNode> Constructor(const GlobalObject& aGlobal,
+                                                 AudioContext& aAudioContext,
+                                                 const DelayOptions& aOptions,
+                                                 ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  AudioParam* DelayTime() const { return mDelay; }
+
+  const char* NodeType() const override { return "DelayNode"; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ private:
+  DelayNode(AudioContext* aContext, double aMaxDelay);
+  ~DelayNode() = default;
+
+  friend class DelayNodeEngine;
+
+  RefPtr<AudioParam> mDelay;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/DynamicsCompressorNode.cpp b/dom/media/webaudio/DynamicsCompressorNode.cpp
new file mode 100644
index 0000000000..7ec27d9495
--- /dev/null
+++ b/dom/media/webaudio/DynamicsCompressorNode.cpp
@@ -0,0 +1,226 @@
+/* -*- 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 "DynamicsCompressorNode.h"
+#include "mozilla/dom/DynamicsCompressorNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "AudioDestinationNode.h"
+#include "WebAudioUtils.h"
+#include "blink/DynamicsCompressor.h"
+#include "Tracing.h"
+
+using WebCore::DynamicsCompressor;
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(DynamicsCompressorNode, AudioNode,
+                                   mThreshold, mKnee, mRatio, mAttack, mRelease)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(DynamicsCompressorNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(DynamicsCompressorNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(DynamicsCompressorNode, AudioNode)
+
+class DynamicsCompressorNodeEngine final : public AudioNodeEngine {
+ public:
+  explicit DynamicsCompressorNodeEngine(AudioNode* aNode,
+                                        AudioDestinationNode* aDestination)
+      : AudioNodeEngine(aNode),
+        mDestination(aDestination->Track())
+        // Keep the default value in sync with the default value in
+        // DynamicsCompressorNode::DynamicsCompressorNode.
+        ,
+        mThreshold(-24.f),
+        mKnee(30.f),
+        mRatio(12.f),
+        mAttack(0.003f),
+        mRelease(0.25f),
+        mCompressor(new DynamicsCompressor(mDestination->mSampleRate, 2)) {}
+
+  enum Parameters { THRESHOLD, KNEE, RATIO, ATTACK, RELEASE };
+  void RecvTimelineEvent(uint32_t aIndex, AudioParamEvent& aEvent) override {
+    MOZ_ASSERT(mDestination);
+
+    aEvent.ConvertToTicks(mDestination);
+
+    switch (aIndex) {
+      case THRESHOLD:
+        mThreshold.InsertEvent<int64_t>(aEvent);
+        break;
+      case KNEE:
+        mKnee.InsertEvent<int64_t>(aEvent);
+        break;
+      case RATIO:
+        mRatio.InsertEvent<int64_t>(aEvent);
+        break;
+      case ATTACK:
+        mAttack.InsertEvent<int64_t>(aEvent);
+        break;
+      case RELEASE:
+        mRelease.InsertEvent<int64_t>(aEvent);
+        break;
+      default:
+        NS_ERROR("Bad DynamicsCompresssorNodeEngine TimelineParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    TRACE("DynamicsCompressorNodeEngine::ProcessBlock");
+    if (aInput.IsNull()) {
+      // Just output silence
+      *aOutput = aInput;
+      return;
+    }
+
+    const uint32_t channelCount = aInput.ChannelCount();
+    if (mCompressor->numberOfChannels() != channelCount) {
+      // Create a new compressor object with a new channel count
+      mCompressor = MakeUnique<WebCore::DynamicsCompressor>(
+          aTrack->mSampleRate, aInput.ChannelCount());
+    }
+
+    TrackTime pos = mDestination->GraphTimeToTrackTime(aFrom);
+    mCompressor->setParameterValue(DynamicsCompressor::ParamThreshold,
+                                   mThreshold.GetValueAtTime(pos));
+    mCompressor->setParameterValue(DynamicsCompressor::ParamKnee,
+                                   mKnee.GetValueAtTime(pos));
+    mCompressor->setParameterValue(DynamicsCompressor::ParamRatio,
+                                   mRatio.GetValueAtTime(pos));
+    mCompressor->setParameterValue(DynamicsCompressor::ParamAttack,
+                                   mAttack.GetValueAtTime(pos));
+    mCompressor->setParameterValue(DynamicsCompressor::ParamRelease,
+                                   mRelease.GetValueAtTime(pos));
+
+    aOutput->AllocateChannels(channelCount);
+    mCompressor->process(&aInput, aOutput, aInput.GetDuration());
+
+    SendReductionParamToMainThread(
+        aTrack,
+        mCompressor->parameterValue(DynamicsCompressor::ParamReduction));
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    // Not owned:
+    // - mDestination (probably)
+    // - Don't count the AudioParamTimelines, their inner refs are owned by the
+    // AudioNode.
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    amount += mCompressor->sizeOfIncludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  void SendReductionParamToMainThread(AudioNodeTrack* aTrack,
+                                      float aReduction) {
+    MOZ_ASSERT(!NS_IsMainThread());
+
+    class Command final : public Runnable {
+     public:
+      Command(AudioNodeTrack* aTrack, float aReduction)
+          : mozilla::Runnable("Command"),
+            mTrack(aTrack),
+            mReduction(aReduction) {}
+
+      NS_IMETHOD Run() override {
+        RefPtr<DynamicsCompressorNode> node =
+            static_cast<DynamicsCompressorNode*>(
+                mTrack->Engine()->NodeMainThread());
+        if (node) {
+          node->SetReduction(mReduction);
+        }
+        return NS_OK;
+      }
+
+     private:
+      RefPtr<AudioNodeTrack> mTrack;
+      float mReduction;
+    };
+
+    AbstractThread::MainThread()->Dispatch(
+        do_AddRef(new Command(aTrack, aReduction)));
+  }
+
+ private:
+  RefPtr<AudioNodeTrack> mDestination;
+  AudioParamTimeline mThreshold;
+  AudioParamTimeline mKnee;
+  AudioParamTimeline mRatio;
+  AudioParamTimeline mAttack;
+  AudioParamTimeline mRelease;
+  UniquePtr<DynamicsCompressor> mCompressor;
+};
+
+DynamicsCompressorNode::DynamicsCompressorNode(AudioContext* aContext)
+    : AudioNode(aContext, 2, ChannelCountMode::Clamped_max,
+                ChannelInterpretation::Speakers),
+      mReduction(0) {
+  mThreshold = CreateAudioParam(DynamicsCompressorNodeEngine::THRESHOLD,
+                                u"threshold"_ns, -24.f, -100.f, 0.f);
+  mKnee = CreateAudioParam(DynamicsCompressorNodeEngine::KNEE, u"knee"_ns, 30.f,
+                           0.f, 40.f);
+  mRatio = CreateAudioParam(DynamicsCompressorNodeEngine::RATIO, u"ratio"_ns,
+                            12.f, 1.f, 20.f);
+  mAttack = CreateAudioParam(DynamicsCompressorNodeEngine::ATTACK, u"attack"_ns,
+                             0.003f, 0.f, 1.f);
+  mRelease = CreateAudioParam(DynamicsCompressorNodeEngine::RELEASE,
+                              u"release"_ns, 0.25f, 0.f, 1.f);
+  DynamicsCompressorNodeEngine* engine =
+      new DynamicsCompressorNodeEngine(this, aContext->Destination());
+  mTrack = AudioNodeTrack::Create(
+      aContext, engine, AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+}
+
+/* static */
+already_AddRefed<DynamicsCompressorNode> DynamicsCompressorNode::Create(
+    AudioContext& aAudioContext, const DynamicsCompressorOptions& aOptions,
+    ErrorResult& aRv) {
+  RefPtr<DynamicsCompressorNode> audioNode =
+      new DynamicsCompressorNode(&aAudioContext);
+
+  audioNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  audioNode->Attack()->SetInitialValue(aOptions.mAttack);
+  audioNode->Knee()->SetInitialValue(aOptions.mKnee);
+  audioNode->Ratio()->SetInitialValue(aOptions.mRatio);
+  audioNode->GetRelease()->SetInitialValue(aOptions.mRelease);
+  audioNode->Threshold()->SetInitialValue(aOptions.mThreshold);
+
+  return audioNode.forget();
+}
+
+size_t DynamicsCompressorNode::SizeOfExcludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mThreshold->SizeOfIncludingThis(aMallocSizeOf);
+  amount += mKnee->SizeOfIncludingThis(aMallocSizeOf);
+  amount += mRatio->SizeOfIncludingThis(aMallocSizeOf);
+  amount += mAttack->SizeOfIncludingThis(aMallocSizeOf);
+  amount += mRelease->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t DynamicsCompressorNode::SizeOfIncludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject* DynamicsCompressorNode::WrapObject(
+    JSContext* aCx, JS::Handle<JSObject*> aGivenProto) {
+  return DynamicsCompressorNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/DynamicsCompressorNode.h b/dom/media/webaudio/DynamicsCompressorNode.h
new file mode 100644
index 0000000000..1e0f9cda62
--- /dev/null
+++ b/dom/media/webaudio/DynamicsCompressorNode.h
@@ -0,0 +1,91 @@
+/* -*- 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/. */
+
+#ifndef DynamicsCompressorNode_h_
+#define DynamicsCompressorNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct DynamicsCompressorOptions;
+
+class DynamicsCompressorNode final : public AudioNode {
+ public:
+  static already_AddRefed<DynamicsCompressorNode> Create(
+      AudioContext& aAudioContext, const DynamicsCompressorOptions& aOptions,
+      ErrorResult& aRv);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(DynamicsCompressorNode, AudioNode)
+
+  static already_AddRefed<DynamicsCompressorNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const DynamicsCompressorOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  AudioParam* Threshold() const { return mThreshold; }
+
+  AudioParam* Knee() const { return mKnee; }
+
+  AudioParam* Ratio() const { return mRatio; }
+
+  AudioParam* Attack() const { return mAttack; }
+
+  // Called GetRelease to prevent clashing with the nsISupports::Release name
+  AudioParam* GetRelease() const { return mRelease; }
+
+  float Reduction() const { return mReduction; }
+
+  const char* NodeType() const override { return "DynamicsCompressorNode"; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+  void SetReduction(float aReduction) {
+    MOZ_ASSERT(NS_IsMainThread());
+    mReduction = aReduction;
+  }
+
+  void SetChannelCountModeValue(ChannelCountMode aMode,
+                                ErrorResult& aRv) override {
+    if (aMode == ChannelCountMode::Max) {
+      aRv.ThrowNotSupportedError("Cannot set channel count mode to \"max\"");
+      return;
+    }
+    AudioNode::SetChannelCountModeValue(aMode, aRv);
+  }
+
+  void SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv) override {
+    if (aChannelCount > 2) {
+      aRv.ThrowNotSupportedError(
+          nsPrintfCString("%u is greater than 2", aChannelCount));
+      return;
+    }
+    AudioNode::SetChannelCount(aChannelCount, aRv);
+  }
+
+ private:
+  explicit DynamicsCompressorNode(AudioContext* aContext);
+  ~DynamicsCompressorNode() = default;
+
+  RefPtr<AudioParam> mThreshold;
+  RefPtr<AudioParam> mKnee;
+  RefPtr<AudioParam> mRatio;
+  float mReduction;
+  RefPtr<AudioParam> mAttack;
+  RefPtr<AudioParam> mRelease;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/FFTBlock.cpp b/dom/media/webaudio/FFTBlock.cpp
new file mode 100644
index 0000000000..6789ca0264
--- /dev/null
+++ b/dom/media/webaudio/FFTBlock.cpp
@@ -0,0 +1,231 @@
+/* -*- Mode: C++; tab-width: 4; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=4 sw=2 sts=2 et cindent: */
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "FFTBlock.h"
+
+#include <complex>
+
+namespace mozilla {
+
+typedef std::complex<double> Complex;
+
+#ifdef MOZ_LIBAV_FFT
+FFmpegRDFTFuncs FFTBlock::sRDFTFuncs;
+#endif
+
+static double fdlibm_cabs(const Complex& z) {
+  return fdlibm_hypot(real(z), imag(z));
+}
+
+static double fdlibm_carg(const Complex& z) {
+  return fdlibm_atan2(imag(z), real(z));
+}
+
+FFTBlock* FFTBlock::CreateInterpolatedBlock(const FFTBlock& block0,
+                                            const FFTBlock& block1,
+                                            double interp) {
+  FFTBlock* newBlock = new FFTBlock(block0.FFTSize());
+
+  newBlock->InterpolateFrequencyComponents(block0, block1, interp);
+
+  // In the time-domain, the 2nd half of the response must be zero, to avoid
+  // circular convolution aliasing...
+  int fftSize = newBlock->FFTSize();
+  AlignedTArray<float> buffer(fftSize);
+  newBlock->GetInverseWithoutScaling(buffer.Elements());
+  AudioBufferInPlaceScale(buffer.Elements(), 1.0f / fftSize, fftSize / 2);
+  PodZero(buffer.Elements() + fftSize / 2, fftSize / 2);
+
+  // Put back into frequency domain.
+  newBlock->PerformFFT(buffer.Elements());
+
+  return newBlock;
+}
+
+void FFTBlock::InterpolateFrequencyComponents(const FFTBlock& block0,
+                                              const FFTBlock& block1,
+                                              double interp) {
+  // FIXME : with some work, this method could be optimized
+
+  ComplexU* dft = mOutputBuffer.Elements();
+
+  const ComplexU* dft1 = block0.mOutputBuffer.Elements();
+  const ComplexU* dft2 = block1.mOutputBuffer.Elements();
+
+  MOZ_ASSERT(mFFTSize == block0.FFTSize());
+  MOZ_ASSERT(mFFTSize == block1.FFTSize());
+  double s1base = (1.0 - interp);
+  double s2base = interp;
+
+  double phaseAccum = 0.0;
+  double lastPhase1 = 0.0;
+  double lastPhase2 = 0.0;
+
+  int n = mFFTSize / 2;
+
+  dft[0].r = static_cast<float>(s1base * dft1[0].r + s2base * dft2[0].r);
+  dft[n].r = static_cast<float>(s1base * dft1[n].r + s2base * dft2[n].r);
+
+  for (int i = 1; i < n; ++i) {
+    Complex c1(dft1[i].r, dft1[i].i);
+    Complex c2(dft2[i].r, dft2[i].i);
+
+    double mag1 = fdlibm_cabs(c1);
+    double mag2 = fdlibm_cabs(c2);
+
+    // Interpolate magnitudes in decibels
+    double mag1db = 20.0 * fdlibm_log10(mag1);
+    double mag2db = 20.0 * fdlibm_log10(mag2);
+
+    double s1 = s1base;
+    double s2 = s2base;
+
+    double magdbdiff = mag1db - mag2db;
+
+    // Empirical tweak to retain higher-frequency zeroes
+    double threshold = (i > 16) ? 5.0 : 2.0;
+
+    if (magdbdiff < -threshold && mag1db < 0.0) {
+      s1 = fdlibm_pow(s1, 0.75);
+      s2 = 1.0 - s1;
+    } else if (magdbdiff > threshold && mag2db < 0.0) {
+      s2 = fdlibm_pow(s2, 0.75);
+      s1 = 1.0 - s2;
+    }
+
+    // Average magnitude by decibels instead of linearly
+    double magdb = s1 * mag1db + s2 * mag2db;
+    double mag = fdlibm_pow(10.0, 0.05 * magdb);
+
+    // Now, deal with phase
+    double phase1 = fdlibm_carg(c1);
+    double phase2 = fdlibm_carg(c2);
+
+    double deltaPhase1 = phase1 - lastPhase1;
+    double deltaPhase2 = phase2 - lastPhase2;
+    lastPhase1 = phase1;
+    lastPhase2 = phase2;
+
+    // Unwrap phase deltas
+    if (deltaPhase1 > M_PI) deltaPhase1 -= 2.0 * M_PI;
+    if (deltaPhase1 < -M_PI) deltaPhase1 += 2.0 * M_PI;
+    if (deltaPhase2 > M_PI) deltaPhase2 -= 2.0 * M_PI;
+    if (deltaPhase2 < -M_PI) deltaPhase2 += 2.0 * M_PI;
+
+    // Blend group-delays
+    double deltaPhaseBlend;
+
+    if (deltaPhase1 - deltaPhase2 > M_PI)
+      deltaPhaseBlend = s1 * deltaPhase1 + s2 * (2.0 * M_PI + deltaPhase2);
+    else if (deltaPhase2 - deltaPhase1 > M_PI)
+      deltaPhaseBlend = s1 * (2.0 * M_PI + deltaPhase1) + s2 * deltaPhase2;
+    else
+      deltaPhaseBlend = s1 * deltaPhase1 + s2 * deltaPhase2;
+
+    phaseAccum += deltaPhaseBlend;
+
+    // Unwrap
+    if (phaseAccum > M_PI) phaseAccum -= 2.0 * M_PI;
+    if (phaseAccum < -M_PI) phaseAccum += 2.0 * M_PI;
+
+    dft[i].r = static_cast<float>(mag * fdlibm_cos(phaseAccum));
+    dft[i].i = static_cast<float>(mag * fdlibm_sin(phaseAccum));
+  }
+}
+
+double FFTBlock::ExtractAverageGroupDelay() {
+  ComplexU* dft = mOutputBuffer.Elements();
+
+  double aveSum = 0.0;
+  double weightSum = 0.0;
+  double lastPhase = 0.0;
+
+  int halfSize = FFTSize() / 2;
+
+  const double kSamplePhaseDelay = (2.0 * M_PI) / double(FFTSize());
+
+  // Remove DC offset
+  dft[0].r = 0.0f;
+
+  // Calculate weighted average group delay
+  for (int i = 1; i < halfSize; i++) {
+    Complex c(dft[i].r, dft[i].i);
+    double mag = fdlibm_cabs(c);
+    double phase = fdlibm_carg(c);
+
+    double deltaPhase = phase - lastPhase;
+    lastPhase = phase;
+
+    // Unwrap
+    if (deltaPhase < -M_PI) deltaPhase += 2.0 * M_PI;
+    if (deltaPhase > M_PI) deltaPhase -= 2.0 * M_PI;
+
+    aveSum += mag * deltaPhase;
+    weightSum += mag;
+  }
+
+  // Note how we invert the phase delta wrt frequency since this is how group
+  // delay is defined
+  double ave = aveSum / weightSum;
+  double aveSampleDelay = -ave / kSamplePhaseDelay;
+
+  // Leave 20 sample headroom (for leading edge of impulse)
+  aveSampleDelay -= 20.0;
+  if (aveSampleDelay <= 0.0) return 0.0;
+
+  // Remove average group delay (minus 20 samples for headroom)
+  AddConstantGroupDelay(-aveSampleDelay);
+
+  return aveSampleDelay;
+}
+
+void FFTBlock::AddConstantGroupDelay(double sampleFrameDelay) {
+  int halfSize = FFTSize() / 2;
+
+  ComplexU* dft = mOutputBuffer.Elements();
+
+  const double kSamplePhaseDelay = (2.0 * M_PI) / double(FFTSize());
+
+  double phaseAdj = -sampleFrameDelay * kSamplePhaseDelay;
+
+  // Add constant group delay
+  for (int i = 1; i < halfSize; i++) {
+    Complex c(dft[i].r, dft[i].i);
+    double mag = fdlibm_cabs(c);
+    double phase = fdlibm_carg(c);
+
+    phase += i * phaseAdj;
+
+    dft[i].r = static_cast<float>(mag * fdlibm_cos(phase));
+    dft[i].i = static_cast<float>(mag * fdlibm_sin(phase));
+  }
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webaudio/FFTBlock.h b/dom/media/webaudio/FFTBlock.h
new file mode 100644
index 0000000000..6af882f9af
--- /dev/null
+++ b/dom/media/webaudio/FFTBlock.h
@@ -0,0 +1,348 @@
+/* -*- 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/. */
+
+#ifndef FFTBlock_h_
+#define FFTBlock_h_
+
+#ifdef BUILD_ARM_NEON
+#  include <cmath>
+#  include "mozilla/arm.h"
+#  include "dl/sp/api/omxSP.h"
+#endif
+
+#include "AlignedTArray.h"
+#include "AudioNodeEngine.h"
+#if defined(MOZ_LIBAV_FFT)
+#  include "FFmpegRDFTTypes.h"
+#  include "FFVPXRuntimeLinker.h"
+#else
+#  include "kiss_fft/kiss_fftr.h"
+#endif
+
+namespace mozilla {
+
+// This class defines an FFT block, loosely modeled after Blink's FFTFrame
+// class to make sharing code with Blink easy.
+// Currently it's implemented on top of KissFFT on all platforms.
+class FFTBlock final {
+  union ComplexU {
+#if !defined(MOZ_LIBAV_FFT)
+    kiss_fft_cpx c;
+#endif
+    float f[2];
+    struct {
+      float r;
+      float i;
+    };
+  };
+
+ public:
+  static void MainThreadInit() {
+#ifdef MOZ_LIBAV_FFT
+    FFVPXRuntimeLinker::Init();
+    if (!sRDFTFuncs.init) {
+      FFVPXRuntimeLinker::GetRDFTFuncs(&sRDFTFuncs);
+    }
+#endif
+  }
+
+  explicit FFTBlock(uint32_t aFFTSize)
+#if defined(MOZ_LIBAV_FFT)
+      : mAvRDFT(nullptr),
+        mAvIRDFT(nullptr)
+#else
+      : mKissFFT(nullptr),
+        mKissIFFT(nullptr)
+#  ifdef BUILD_ARM_NEON
+        ,
+        mOmxFFT(nullptr),
+        mOmxIFFT(nullptr)
+#  endif
+#endif
+  {
+    MOZ_COUNT_CTOR(FFTBlock);
+    SetFFTSize(aFFTSize);
+  }
+  ~FFTBlock() {
+    MOZ_COUNT_DTOR(FFTBlock);
+    Clear();
+  }
+
+  // Return a new FFTBlock with frequency components interpolated between
+  // |block0| and |block1| with |interp| between 0.0 and 1.0.
+  static FFTBlock* CreateInterpolatedBlock(const FFTBlock& block0,
+                                           const FFTBlock& block1,
+                                           double interp);
+
+  // Transform FFTSize() points of aData and store the result internally.
+  void PerformFFT(const float* aData) {
+    if (!EnsureFFT()) {
+      return;
+    }
+
+#if defined(MOZ_LIBAV_FFT)
+    PodCopy(mOutputBuffer.Elements()->f, aData, mFFTSize);
+    sRDFTFuncs.calc(mAvRDFT, mOutputBuffer.Elements()->f);
+    // Recover packed Nyquist.
+    mOutputBuffer[mFFTSize / 2].r = mOutputBuffer[0].i;
+    mOutputBuffer[0].i = 0.0f;
+#else
+#  ifdef BUILD_ARM_NEON
+    if (mozilla::supports_neon()) {
+      omxSP_FFTFwd_RToCCS_F32_Sfs(aData, mOutputBuffer.Elements()->f, mOmxFFT);
+    } else
+#  endif
+    {
+      kiss_fftr(mKissFFT, aData, &(mOutputBuffer.Elements()->c));
+    }
+#endif
+  }
+  // Inverse-transform internal data and store the resulting FFTSize()
+  // points in aDataOut.
+  void GetInverse(float* aDataOut) {
+    GetInverseWithoutScaling(aDataOut);
+    AudioBufferInPlaceScale(aDataOut, 1.0f / mFFTSize, mFFTSize);
+  }
+
+  // Inverse-transform internal frequency data and store the resulting
+  // FFTSize() points in |aDataOut|.  If frequency data has not already been
+  // scaled, then the output will need scaling by 1/FFTSize().
+  void GetInverseWithoutScaling(float* aDataOut) {
+    if (!EnsureIFFT()) {
+      std::fill_n(aDataOut, mFFTSize, 0.0f);
+      return;
+    };
+
+#if defined(MOZ_LIBAV_FFT)
+    {
+      // Even though this function doesn't scale, the libav forward transform
+      // gives a value that needs scaling by 2 in order for things to turn out
+      // similar to how we expect from kissfft/openmax.
+      AudioBufferCopyWithScale(mOutputBuffer.Elements()->f, 2.0f, aDataOut,
+                               mFFTSize);
+      aDataOut[1] = 2.0f * mOutputBuffer[mFFTSize / 2].r;  // Packed Nyquist
+      sRDFTFuncs.calc(mAvIRDFT, aDataOut);
+    }
+#else
+#  ifdef BUILD_ARM_NEON
+    if (mozilla::supports_neon()) {
+      omxSP_FFTInv_CCSToR_F32_Sfs_unscaled(mOutputBuffer.Elements()->f,
+                                           aDataOut, mOmxIFFT);
+    } else
+#  endif
+    {
+      kiss_fftri(mKissIFFT, &(mOutputBuffer.Elements()->c), aDataOut);
+    }
+#endif
+  }
+
+  void Multiply(const FFTBlock& aFrame) {
+    uint32_t halfSize = mFFTSize / 2;
+    // DFTs are not packed.
+    MOZ_ASSERT(mOutputBuffer[0].i == 0);
+    MOZ_ASSERT(aFrame.mOutputBuffer[0].i == 0);
+
+    BufferComplexMultiply(mOutputBuffer.Elements()->f,
+                          aFrame.mOutputBuffer.Elements()->f,
+                          mOutputBuffer.Elements()->f, halfSize);
+    mOutputBuffer[halfSize].r *= aFrame.mOutputBuffer[halfSize].r;
+    // This would have been set to NaN if either real component was NaN.
+    mOutputBuffer[0].i = 0.0f;
+  }
+
+  // Perform a forward FFT on |aData|, assuming zeros after dataSize samples,
+  // and pre-scale the generated internal frequency domain coefficients so
+  // that GetInverseWithoutScaling() can be used to transform to the time
+  // domain.  This is useful for convolution kernels.
+  void PadAndMakeScaledDFT(const float* aData, size_t dataSize) {
+    MOZ_ASSERT(dataSize <= FFTSize());
+    AlignedTArray<float> paddedData;
+    paddedData.SetLength(FFTSize());
+    AudioBufferCopyWithScale(aData, 1.0f / FFTSize(), paddedData.Elements(),
+                             dataSize);
+    PodZero(paddedData.Elements() + dataSize, mFFTSize - dataSize);
+    PerformFFT(paddedData.Elements());
+  }
+
+  // aSize must be a power of 2
+  void SetFFTSize(uint32_t aSize) {
+    MOZ_ASSERT(CountPopulation32(aSize) == 1);
+    mFFTSize = aSize;
+    mOutputBuffer.SetLength(aSize / 2 + 1);
+    PodZero(mOutputBuffer.Elements(), aSize / 2 + 1);
+    Clear();
+  }
+
+  // Return the average group delay and removes this from the frequency data.
+  double ExtractAverageGroupDelay();
+
+  uint32_t FFTSize() const { return mFFTSize; }
+  float RealData(uint32_t aIndex) const { return mOutputBuffer[aIndex].r; }
+  float& RealData(uint32_t aIndex) { return mOutputBuffer[aIndex].r; }
+  float ImagData(uint32_t aIndex) const { return mOutputBuffer[aIndex].i; }
+  float& ImagData(uint32_t aIndex) { return mOutputBuffer[aIndex].i; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+    size_t amount = 0;
+
+#if defined(MOZ_LIBAV_FFT)
+    auto ComputedSizeOfContextIfSet = [this](void* aContext) -> size_t {
+      if (!aContext) {
+        return 0;
+      }
+      // RDFTContext is only forward declared in public headers, but this is
+      // an estimate based on a value of 231 seen requested from
+      // _aligned_alloc on Win64.  Don't use malloc_usable_size() because the
+      // context pointer is not necessarily from malloc.
+      size_t amount = 232;
+      // Add size of allocations performed in ff_fft_init().
+      // The maximum FFT size used is 32768 = 2^15 and so revtab32 is not
+      // allocated.
+      MOZ_ASSERT(mFFTSize <= 32768);
+      amount += mFFTSize * (sizeof(uint16_t) + 2 * sizeof(float));
+
+      return amount;
+    };
+
+    amount += ComputedSizeOfContextIfSet(mAvRDFT);
+    amount += ComputedSizeOfContextIfSet(mAvIRDFT);
+#else
+#  ifdef BUILD_ARM_NEON
+    amount += aMallocSizeOf(mOmxFFT);
+    amount += aMallocSizeOf(mOmxIFFT);
+#  endif
+#  ifdef USE_SIMD
+#    error kiss fft uses malloc only when USE_SIMD is not defined
+#  endif
+    amount += aMallocSizeOf(mKissFFT);
+    amount += aMallocSizeOf(mKissIFFT);
+#endif
+    amount += mOutputBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  FFTBlock(const FFTBlock& other) = delete;
+  void operator=(const FFTBlock& other) = delete;
+
+  bool EnsureFFT() {
+#if defined(MOZ_LIBAV_FFT)
+    if (!mAvRDFT) {
+      if (!sRDFTFuncs.init) {
+        return false;
+      }
+
+      mAvRDFT = sRDFTFuncs.init(FloorLog2(mFFTSize), DFT_R2C);
+    }
+#else
+#  ifdef BUILD_ARM_NEON
+    if (mozilla::supports_neon()) {
+      if (!mOmxFFT) {
+        mOmxFFT = createOmxFFT(mFFTSize);
+      }
+    } else
+#  endif
+    {
+      if (!mKissFFT) {
+        mKissFFT = kiss_fftr_alloc(mFFTSize, 0, nullptr, nullptr);
+      }
+    }
+#endif
+    return true;
+  }
+
+  bool EnsureIFFT() {
+#if defined(MOZ_LIBAV_FFT)
+    if (!mAvIRDFT) {
+      if (!sRDFTFuncs.init) {
+        return false;
+      }
+
+      mAvIRDFT = sRDFTFuncs.init(FloorLog2(mFFTSize), IDFT_C2R);
+    }
+#else
+#  ifdef BUILD_ARM_NEON
+    if (mozilla::supports_neon()) {
+      if (!mOmxIFFT) {
+        mOmxIFFT = createOmxFFT(mFFTSize);
+      }
+    } else
+#  endif
+    {
+      if (!mKissIFFT) {
+        mKissIFFT = kiss_fftr_alloc(mFFTSize, 1, nullptr, nullptr);
+      }
+    }
+#endif
+    return true;
+  }
+
+#ifdef BUILD_ARM_NEON
+  static OMXFFTSpec_R_F32* createOmxFFT(uint32_t aFFTSize) {
+    MOZ_ASSERT((aFFTSize & (aFFTSize - 1)) == 0);
+    OMX_INT bufSize;
+    OMX_INT order = FloorLog2(aFFTSize);
+    MOZ_ASSERT(aFFTSize >> order == 1);
+    OMXResult status = omxSP_FFTGetBufSize_R_F32(order, &bufSize);
+    if (status == OMX_Sts_NoErr) {
+      OMXFFTSpec_R_F32* context =
+          static_cast<OMXFFTSpec_R_F32*>(malloc(bufSize));
+      if (omxSP_FFTInit_R_F32(context, order) != OMX_Sts_NoErr) {
+        return nullptr;
+      }
+      return context;
+    }
+    return nullptr;
+  }
+#endif
+
+  void Clear() {
+#if defined(MOZ_LIBAV_FFT)
+    if (mAvRDFT) {
+      sRDFTFuncs.end(mAvRDFT);
+      mAvRDFT = nullptr;
+    }
+    if (mAvIRDFT) {
+      sRDFTFuncs.end(mAvIRDFT);
+      mAvIRDFT = nullptr;
+    }
+#else
+#  ifdef BUILD_ARM_NEON
+    free(mOmxFFT);
+    free(mOmxIFFT);
+    mOmxFFT = mOmxIFFT = nullptr;
+#  endif
+    free(mKissFFT);
+    free(mKissIFFT);
+    mKissFFT = mKissIFFT = nullptr;
+#endif
+  }
+  void AddConstantGroupDelay(double sampleFrameDelay);
+  void InterpolateFrequencyComponents(const FFTBlock& block0,
+                                      const FFTBlock& block1, double interp);
+#if defined(MOZ_LIBAV_FFT)
+  static FFmpegRDFTFuncs sRDFTFuncs;
+  RDFTContext* mAvRDFT;
+  RDFTContext* mAvIRDFT;
+#else
+  kiss_fftr_cfg mKissFFT;
+  kiss_fftr_cfg mKissIFFT;
+#  ifdef BUILD_ARM_NEON
+  OMXFFTSpec_R_F32* mOmxFFT;
+  OMXFFTSpec_R_F32* mOmxIFFT;
+#  endif
+#endif
+  AlignedTArray<ComplexU> mOutputBuffer;
+  uint32_t mFFTSize;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/GainNode.cpp b/dom/media/webaudio/GainNode.cpp
new file mode 100644
index 0000000000..ed998ee2a9
--- /dev/null
+++ b/dom/media/webaudio/GainNode.cpp
@@ -0,0 +1,149 @@
+/* -*- 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 "GainNode.h"
+#include "mozilla/dom/GainNodeBinding.h"
+#include "AlignmentUtils.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "AudioDestinationNode.h"
+#include "WebAudioUtils.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(GainNode, AudioNode, mGain)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(GainNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(GainNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(GainNode, AudioNode)
+
+class GainNodeEngine final : public AudioNodeEngine {
+ public:
+  GainNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination)
+      : AudioNodeEngine(aNode),
+        mDestination(aDestination->Track())
+        // Keep the default value in sync with the default value in
+        // GainNode::GainNode.
+        ,
+        mGain(1.f) {}
+
+  enum Parameters { GAIN };
+  void RecvTimelineEvent(uint32_t aIndex, AudioParamEvent& aEvent) override {
+    MOZ_ASSERT(mDestination);
+    aEvent.ConvertToTicks(mDestination);
+
+    switch (aIndex) {
+      case GAIN:
+        mGain.InsertEvent<int64_t>(aEvent);
+        break;
+      default:
+        NS_ERROR("Bad GainNodeEngine TimelineParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    TRACE("GainNodeEngine::ProcessBlock");
+    if (aInput.IsNull()) {
+      // If input is silent, so is the output
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+    } else if (mGain.HasSimpleValue()) {
+      // Optimize the case where we only have a single value set as the volume
+      float gain = mGain.GetValue();
+      if (gain == 0.0f) {
+        aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      } else {
+        *aOutput = aInput;
+        aOutput->mVolume *= gain;
+      }
+    } else {
+      // First, compute a vector of gains for each track tick based on the
+      // timeline at hand, and then for each channel, multiply the values
+      // in the buffer with the gain vector.
+      aOutput->AllocateChannels(aInput.ChannelCount());
+
+      // Compute the gain values for the duration of the input AudioChunk
+      TrackTime tick = mDestination->GraphTimeToTrackTime(aFrom);
+      float computedGain[WEBAUDIO_BLOCK_SIZE + 4];
+      float* alignedComputedGain = ALIGNED16(computedGain);
+      ASSERT_ALIGNED16(alignedComputedGain);
+      mGain.GetValuesAtTime(tick, alignedComputedGain, WEBAUDIO_BLOCK_SIZE);
+
+      for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
+        alignedComputedGain[counter] *= aInput.mVolume;
+      }
+
+      // Apply the gain to the output buffer
+      for (size_t channel = 0; channel < aOutput->ChannelCount(); ++channel) {
+        const float* inputBuffer =
+            static_cast<const float*>(aInput.mChannelData[channel]);
+        float* buffer = aOutput->ChannelFloatsForWrite(channel);
+        AudioBlockCopyChannelWithScale(inputBuffer, alignedComputedGain,
+                                       buffer);
+      }
+    }
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    // Not owned:
+    // - mDestination - MediaTrackGraphImpl::CollectSizesForMemoryReport()
+    // accounts for mDestination.
+    // - mGain - Internal ref owned by AudioNode
+    return AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  RefPtr<AudioNodeTrack> mDestination;
+  AudioParamTimeline mGain;
+};
+
+GainNode::GainNode(AudioContext* aContext)
+    : AudioNode(aContext, 2, ChannelCountMode::Max,
+                ChannelInterpretation::Speakers) {
+  mGain = CreateAudioParam(GainNodeEngine::GAIN, u"gain"_ns, 1.0f);
+  GainNodeEngine* engine = new GainNodeEngine(this, aContext->Destination());
+  mTrack = AudioNodeTrack::Create(
+      aContext, engine, AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+}
+
+/* static */
+already_AddRefed<GainNode> GainNode::Create(AudioContext& aAudioContext,
+                                            const GainOptions& aOptions,
+                                            ErrorResult& aRv) {
+  RefPtr<GainNode> audioNode = new GainNode(&aAudioContext);
+
+  audioNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  audioNode->Gain()->SetInitialValue(aOptions.mGain);
+  return audioNode.forget();
+}
+
+size_t GainNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mGain->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t GainNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject* GainNode::WrapObject(JSContext* aCx,
+                               JS::Handle<JSObject*> aGivenProto) {
+  return GainNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/GainNode.h b/dom/media/webaudio/GainNode.h
new file mode 100644
index 0000000000..d2790af746
--- /dev/null
+++ b/dom/media/webaudio/GainNode.h
@@ -0,0 +1,53 @@
+/* -*- 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/. */
+
+#ifndef GainNode_h_
+#define GainNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct GainOptions;
+
+class GainNode final : public AudioNode {
+ public:
+  static already_AddRefed<GainNode> Create(AudioContext& aAudioContext,
+                                           const GainOptions& aOptions,
+                                           ErrorResult& aRv);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(GainNode, AudioNode)
+
+  static already_AddRefed<GainNode> Constructor(const GlobalObject& aGlobal,
+                                                AudioContext& aAudioContext,
+                                                const GainOptions& aOptions,
+                                                ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  AudioParam* Gain() const { return mGain; }
+
+  const char* NodeType() const override { return "GainNode"; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ private:
+  explicit GainNode(AudioContext* aContext);
+  ~GainNode() = default;
+
+  RefPtr<AudioParam> mGain;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/IIRFilterNode.cpp b/dom/media/webaudio/IIRFilterNode.cpp
new file mode 100644
index 0000000000..fe87744d4d
--- /dev/null
+++ b/dom/media/webaudio/IIRFilterNode.cpp
@@ -0,0 +1,266 @@
+/* -*- 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 "IIRFilterNode.h"
+#include "AudioNodeEngine.h"
+#include "AudioDestinationNode.h"
+#include "blink/IIRFilter.h"
+#include "PlayingRefChangeHandler.h"
+#include "AlignmentUtils.h"
+#include "nsPrintfCString.h"
+
+#include "nsGkAtoms.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+class IIRFilterNodeEngine final : public AudioNodeEngine {
+ public:
+  IIRFilterNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination,
+                      const AudioDoubleArray& aFeedforward,
+                      const AudioDoubleArray& aFeedback, uint64_t aWindowID)
+      : AudioNodeEngine(aNode),
+        mDestination(aDestination->Track()),
+        mFeedforward(aFeedforward.Clone()),
+        mFeedback(aFeedback.Clone()),
+        mWindowID(aWindowID) {}
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    TRACE("IIRFilterNodeEngine::ProcessBlock");
+    float inputBuffer[WEBAUDIO_BLOCK_SIZE + 4];
+    float* alignedInputBuffer = ALIGNED16(inputBuffer);
+    ASSERT_ALIGNED16(alignedInputBuffer);
+
+    if (aInput.IsNull()) {
+      if (!mIIRFilters.IsEmpty()) {
+        bool allZero = true;
+        for (uint32_t i = 0; i < mIIRFilters.Length(); ++i) {
+          allZero &= mIIRFilters[i]->buffersAreZero();
+        }
+
+        // all filter buffer values are zero, so the output will be zero
+        // as well.
+        if (allZero) {
+          mIIRFilters.Clear();
+          aTrack->ScheduleCheckForInactive();
+
+          RefPtr<PlayingRefChangeHandler> refchanged =
+              new PlayingRefChangeHandler(aTrack,
+                                          PlayingRefChangeHandler::RELEASE);
+          aTrack->Graph()->DispatchToMainThreadStableState(refchanged.forget());
+
+          aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+          return;
+        }
+
+        PodZero(alignedInputBuffer, WEBAUDIO_BLOCK_SIZE);
+      }
+    } else if (mIIRFilters.Length() != aInput.ChannelCount()) {
+      if (mIIRFilters.IsEmpty()) {
+        RefPtr<PlayingRefChangeHandler> refchanged =
+            new PlayingRefChangeHandler(aTrack,
+                                        PlayingRefChangeHandler::ADDREF);
+        aTrack->Graph()->DispatchToMainThreadStableState(refchanged.forget());
+      } else {
+        WebAudioUtils::LogToDeveloperConsole(
+            mWindowID, "IIRFilterChannelCountChangeWarning");
+      }
+
+      // Adjust the number of filters based on the number of channels
+      mIIRFilters.SetLength(aInput.ChannelCount());
+      for (size_t i = 0; i < aInput.ChannelCount(); ++i) {
+        mIIRFilters[i] =
+            MakeUnique<blink::IIRFilter>(&mFeedforward, &mFeedback);
+      }
+    }
+
+    uint32_t numberOfChannels = mIIRFilters.Length();
+    aOutput->AllocateChannels(numberOfChannels);
+
+    for (uint32_t i = 0; i < numberOfChannels; ++i) {
+      const float* input;
+      if (aInput.IsNull()) {
+        input = alignedInputBuffer;
+      } else {
+        input = static_cast<const float*>(aInput.mChannelData[i]);
+        if (aInput.mVolume != 1.0) {
+          AudioBlockCopyChannelWithScale(input, aInput.mVolume,
+                                         alignedInputBuffer);
+          input = alignedInputBuffer;
+        }
+      }
+
+      mIIRFilters[i]->process(input, aOutput->ChannelFloatsForWrite(i),
+                              aInput.GetDuration());
+    }
+  }
+
+  bool IsActive() const override { return !mIIRFilters.IsEmpty(); }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    // Not owned:
+    // - mDestination - probably not owned
+    // - AudioParamTimelines - counted in the AudioNode
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    amount += mIIRFilters.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  RefPtr<AudioNodeTrack> mDestination;
+  nsTArray<UniquePtr<blink::IIRFilter>> mIIRFilters;
+  AudioDoubleArray mFeedforward;
+  AudioDoubleArray mFeedback;
+  uint64_t mWindowID;
+};
+
+IIRFilterNode::IIRFilterNode(AudioContext* aContext,
+                             const Sequence<double>& aFeedforward,
+                             const Sequence<double>& aFeedback)
+    : AudioNode(aContext, 2, ChannelCountMode::Max,
+                ChannelInterpretation::Speakers) {
+  mFeedforward.SetLength(aFeedforward.Length());
+  PodCopy(mFeedforward.Elements(), aFeedforward.Elements(),
+          aFeedforward.Length());
+  mFeedback.SetLength(aFeedback.Length());
+  PodCopy(mFeedback.Elements(), aFeedback.Elements(), aFeedback.Length());
+
+  // Scale coefficients -- we guarantee that mFeedback != 0 when creating
+  // the IIRFilterNode.
+  double scale = mFeedback[0];
+  double* elements = mFeedforward.Elements();
+  for (size_t i = 0; i < mFeedforward.Length(); ++i) {
+    elements[i] /= scale;
+  }
+
+  elements = mFeedback.Elements();
+  for (size_t i = 0; i < mFeedback.Length(); ++i) {
+    elements[i] /= scale;
+  }
+
+  // We check that this is exactly equal to one later in blink/IIRFilter.cpp
+  elements[0] = 1.0;
+
+  uint64_t windowID = 0;
+  if (aContext->GetParentObject()) {
+    windowID = aContext->GetParentObject()->WindowID();
+  }
+  IIRFilterNodeEngine* engine = new IIRFilterNodeEngine(
+      this, aContext->Destination(), mFeedforward, mFeedback, windowID);
+  mTrack = AudioNodeTrack::Create(
+      aContext, engine, AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+}
+
+/* static */
+already_AddRefed<IIRFilterNode> IIRFilterNode::Create(
+    AudioContext& aAudioContext, const IIRFilterOptions& aOptions,
+    ErrorResult& aRv) {
+  if (aOptions.mFeedforward.Length() == 0 ||
+      aOptions.mFeedforward.Length() > 20) {
+    aRv.ThrowNotSupportedError(
+        nsPrintfCString("\"feedforward\" length %zu is not in the range [1,20]",
+                        aOptions.mFeedforward.Length()));
+    return nullptr;
+  }
+
+  if (aOptions.mFeedback.Length() == 0 || aOptions.mFeedback.Length() > 20) {
+    aRv.ThrowNotSupportedError(
+        nsPrintfCString("\"feedback\" length %zu is not in the range [1,20]",
+                        aOptions.mFeedback.Length()));
+    return nullptr;
+  }
+
+  bool feedforwardAllZeros = true;
+  for (size_t i = 0; i < aOptions.mFeedforward.Length(); ++i) {
+    if (aOptions.mFeedforward.Elements()[i] != 0.0) {
+      feedforwardAllZeros = false;
+      break;
+    }
+  }
+
+  if (feedforwardAllZeros) {
+    aRv.ThrowInvalidStateError(
+        "\"feedforward\" must contain some nonzero values");
+    return nullptr;
+  }
+
+  if (aOptions.mFeedback[0] == 0.0) {
+    aRv.ThrowInvalidStateError("First value in \"feedback\" must be nonzero");
+    return nullptr;
+  }
+
+  RefPtr<IIRFilterNode> audioNode = new IIRFilterNode(
+      &aAudioContext, aOptions.mFeedforward, aOptions.mFeedback);
+
+  audioNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  return audioNode.forget();
+}
+
+size_t IIRFilterNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t IIRFilterNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject* IIRFilterNode::WrapObject(JSContext* aCx,
+                                    JS::Handle<JSObject*> aGivenProto) {
+  return IIRFilterNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void IIRFilterNode::GetFrequencyResponse(const Float32Array& aFrequencyHz,
+                                         const Float32Array& aMagResponse,
+                                         const Float32Array& aPhaseResponse) {
+  aFrequencyHz.ProcessData([&](const Span<float>& aFrequencyData,
+                               JS::AutoCheckCannotGC&&) {
+    aMagResponse.ProcessData([&](const Span<float>& aMagData,
+                                 JS::AutoCheckCannotGC&&) {
+      aPhaseResponse.ProcessData([&](const Span<float>& aPhaseData,
+                                     JS::AutoCheckCannotGC&&) {
+        uint32_t length = std::min(
+            {aFrequencyData.Length(), aMagData.Length(), aPhaseData.Length()});
+        if (!length) {
+          return;
+        }
+
+        auto frequencies = MakeUniqueForOverwriteFallible<float[]>(length);
+        if (!frequencies) {
+          return;
+        }
+
+        const double nyquist = Context()->SampleRate() * 0.5;
+
+        // Normalize the frequencies
+        std::transform(aFrequencyData.begin(), aFrequencyData.begin() + length,
+                       frequencies.get(), [&](float aFrequency) {
+                         if (aFrequency >= 0 && aFrequency <= nyquist) {
+                           return static_cast<float>(aFrequency / nyquist);
+                         }
+
+                         return std::numeric_limits<float>::quiet_NaN();
+                       });
+
+        blink::IIRFilter filter(&mFeedforward, &mFeedback);
+        filter.getFrequencyResponse(int(length), frequencies.get(),
+                                    aMagData.Elements(), aPhaseData.Elements());
+      });
+    });
+  });
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/IIRFilterNode.h b/dom/media/webaudio/IIRFilterNode.h
new file mode 100644
index 0000000000..d79f87f632
--- /dev/null
+++ b/dom/media/webaudio/IIRFilterNode.h
@@ -0,0 +1,56 @@
+/* -*- 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/. */
+
+#ifndef IIRFilterNode_h_
+#define IIRFilterNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+#include "mozilla/dom/IIRFilterNodeBinding.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct IIRFilterOptions;
+
+class IIRFilterNode final : public AudioNode {
+ public:
+  static already_AddRefed<IIRFilterNode> Create(
+      AudioContext& aAudioContext, const IIRFilterOptions& aOptions,
+      ErrorResult& aRv);
+
+  NS_INLINE_DECL_REFCOUNTING_INHERITED(IIRFilterNode, AudioNode)
+
+  static already_AddRefed<IIRFilterNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const IIRFilterOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  void GetFrequencyResponse(const Float32Array& aFrequencyHz,
+                            const Float32Array& aMagResponse,
+                            const Float32Array& aPhaseResponse);
+
+  const char* NodeType() const override { return "IIRFilterNode"; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ private:
+  IIRFilterNode(AudioContext* aContext, const Sequence<double>& aFeedforward,
+                const Sequence<double>& aFeedback);
+  ~IIRFilterNode() = default;
+
+  nsTArray<double> mFeedback;
+  nsTArray<double> mFeedforward;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/MediaBufferDecoder.cpp b/dom/media/webaudio/MediaBufferDecoder.cpp
new file mode 100644
index 0000000000..425c0639b5
--- /dev/null
+++ b/dom/media/webaudio/MediaBufferDecoder.cpp
@@ -0,0 +1,764 @@
+/* -*- 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 "MediaBufferDecoder.h"
+
+#include <speex/speex_resampler.h>
+
+#include "AudioBuffer.h"
+#include "AudioContext.h"
+#include "AudioNodeEngine.h"
+#include "BufferMediaResource.h"
+#include "DecoderTraits.h"
+#include "MediaContainerType.h"
+#include "MediaDataDecoderProxy.h"
+#include "MediaDataDemuxer.h"
+#include "MediaQueue.h"
+#include "PDMFactory.h"
+#include "VideoUtils.h"
+#include "WebAudioUtils.h"
+#include "js/MemoryFunctions.h"
+#include "mozilla/AbstractThread.h"
+#include "mozilla/Logging.h"
+#include "mozilla/StaticPrefs_media.h"
+#include "mozilla/Telemetry.h"
+#include "mozilla/dom/AudioContextBinding.h"
+#include "mozilla/dom/BaseAudioContextBinding.h"
+#include "mozilla/dom/DOMException.h"
+#include "mozilla/dom/Promise.h"
+#include "mozilla/dom/ScriptSettings.h"
+#include "nsComponentManagerUtils.h"
+#include "nsContentUtils.h"
+#include "nsIScriptError.h"
+#include "nsIScriptObjectPrincipal.h"
+#include "nsMimeTypes.h"
+#include "nsPrintfCString.h"
+#include "nsXPCOMCIDInternal.h"
+
+namespace mozilla {
+
+extern LazyLogModule gMediaDecoderLog;
+
+#define LOG(x, ...) \
+  MOZ_LOG(gMediaDecoderLog, LogLevel::Debug, (x, ##__VA_ARGS__))
+
+using namespace dom;
+
+class ReportResultTask final : public Runnable {
+ public:
+  ReportResultTask(WebAudioDecodeJob& aDecodeJob,
+                   WebAudioDecodeJob::ResultFn aFunction,
+                   WebAudioDecodeJob::ErrorCode aErrorCode)
+      : Runnable("ReportResultTask"),
+        mDecodeJob(aDecodeJob),
+        mFunction(aFunction),
+        mErrorCode(aErrorCode) {
+    MOZ_ASSERT(aFunction);
+  }
+
+  NS_IMETHOD Run() override {
+    MOZ_ASSERT(NS_IsMainThread());
+
+    (mDecodeJob.*mFunction)(mErrorCode);
+
+    return NS_OK;
+  }
+
+ private:
+  // Note that the mDecodeJob member will probably die when mFunction is run.
+  // Therefore, it is not safe to do anything fancy with it in this class.
+  // Really, this class is only used because nsRunnableMethod doesn't support
+  // methods accepting arguments.
+  WebAudioDecodeJob& mDecodeJob;
+  WebAudioDecodeJob::ResultFn mFunction;
+  WebAudioDecodeJob::ErrorCode mErrorCode;
+};
+
+enum class PhaseEnum : int { Decode, AllocateBuffer, Done };
+
+class MediaDecodeTask final : public Runnable {
+ public:
+  MediaDecodeTask(const MediaContainerType& aContainerType, uint8_t* aBuffer,
+                  uint32_t aLength, WebAudioDecodeJob& aDecodeJob)
+      : Runnable("MediaDecodeTask"),
+        mContainerType(aContainerType),
+        mBuffer(aBuffer),
+        mLength(aLength),
+        mBatchSize(StaticPrefs::media_rdd_webaudio_batch_size()),
+        mDecodeJob(aDecodeJob),
+        mPhase(PhaseEnum::Decode) {
+    MOZ_ASSERT(aBuffer);
+    MOZ_ASSERT(NS_IsMainThread());
+  }
+
+  // MOZ_CAN_RUN_SCRIPT_BOUNDARY until Runnable::Run is MOZ_CAN_RUN_SCRIPT.  See
+  // bug 1535398.
+  MOZ_CAN_RUN_SCRIPT_BOUNDARY
+  NS_IMETHOD Run() override;
+  bool Init();
+  TaskQueue* PSupervisorTaskQueue() { return mPSupervisorTaskQueue; }
+  bool OnPSupervisorTaskQueue() const {
+    return mPSupervisorTaskQueue->IsCurrentThreadIn();
+  }
+
+ private:
+  MOZ_CAN_RUN_SCRIPT
+  void ReportFailureOnMainThread(WebAudioDecodeJob::ErrorCode aErrorCode) {
+    if (NS_IsMainThread()) {
+      Cleanup();
+      mDecodeJob.OnFailure(aErrorCode);
+    } else {
+      // Take extra care to cleanup on the main thread
+      mMainThread->Dispatch(NewRunnableMethod("MediaDecodeTask::Cleanup", this,
+                                              &MediaDecodeTask::Cleanup));
+
+      nsCOMPtr<nsIRunnable> event = new ReportResultTask(
+          mDecodeJob, &WebAudioDecodeJob::OnFailure, aErrorCode);
+      mMainThread->Dispatch(event.forget());
+    }
+  }
+
+  void Decode();
+
+  void OnCreateDecoderCompleted(RefPtr<MediaDataDecoder> aDecoder);
+  MOZ_CAN_RUN_SCRIPT void OnCreateDecoderFailed(const MediaResult& aError);
+
+  MOZ_CAN_RUN_SCRIPT void OnInitDemuxerCompleted();
+  MOZ_CAN_RUN_SCRIPT void OnInitDemuxerFailed(const MediaResult& aError);
+
+  void InitDecoder();
+  void OnInitDecoderCompleted();
+  MOZ_CAN_RUN_SCRIPT void OnInitDecoderFailed();
+
+  void DoDemux();
+  void OnAudioDemuxCompleted(RefPtr<MediaTrackDemuxer::SamplesHolder> aSamples);
+  MOZ_CAN_RUN_SCRIPT void OnAudioDemuxFailed(const MediaResult& aError);
+
+  void DoDecode();
+  void OnAudioDecodeCompleted(MediaDataDecoder::DecodedData&& aResults);
+  MOZ_CAN_RUN_SCRIPT void OnAudioDecodeFailed(const MediaResult& aError);
+
+  void DoDrain();
+  MOZ_CAN_RUN_SCRIPT void OnAudioDrainCompleted(
+      MediaDataDecoder::DecodedData&& aResults);
+  MOZ_CAN_RUN_SCRIPT void OnAudioDrainFailed(const MediaResult& aError);
+
+  void ShutdownDecoder();
+
+  MOZ_CAN_RUN_SCRIPT void FinishDecode();
+  MOZ_CAN_RUN_SCRIPT void AllocateBuffer();
+  MOZ_CAN_RUN_SCRIPT void CallbackTheResult();
+
+  void Cleanup() {
+    MOZ_ASSERT(NS_IsMainThread());
+    JS_free(nullptr, mBuffer);
+    if (mTrackDemuxer) {
+      mTrackDemuxer->BreakCycles();
+    }
+    mTrackDemuxer = nullptr;
+    mDemuxer = nullptr;
+    mPSupervisorTaskQueue = nullptr;
+    mPDecoderTaskQueue = nullptr;
+  }
+
+ private:
+  MediaContainerType mContainerType;
+  uint8_t* mBuffer;
+  const uint32_t mLength;
+  const uint32_t mBatchSize;
+  WebAudioDecodeJob& mDecodeJob;
+  PhaseEnum mPhase;
+  RefPtr<TaskQueue> mPSupervisorTaskQueue;
+  RefPtr<TaskQueue> mPDecoderTaskQueue;
+  RefPtr<MediaDataDemuxer> mDemuxer;
+  RefPtr<MediaTrackDemuxer> mTrackDemuxer;
+  RefPtr<MediaDataDecoder> mDecoder;
+  nsTArray<RefPtr<MediaRawData>> mRawSamples;
+  MediaInfo mMediaInfo;
+  MediaQueue<AudioData> mAudioQueue;
+  RefPtr<AbstractThread> mMainThread;
+};
+
+NS_IMETHODIMP
+MediaDecodeTask::Run() {
+  switch (mPhase) {
+    case PhaseEnum::Decode:
+      Decode();
+      break;
+    case PhaseEnum::AllocateBuffer:
+      AllocateBuffer();
+      break;
+    case PhaseEnum::Done:
+      break;
+  }
+
+  return NS_OK;
+}
+
+bool MediaDecodeTask::Init() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  RefPtr<BufferMediaResource> resource =
+      new BufferMediaResource(static_cast<uint8_t*>(mBuffer), mLength);
+
+  mMainThread = AbstractThread::MainThread();
+
+  mPSupervisorTaskQueue =
+      TaskQueue::Create(GetMediaThreadPool(MediaThreadType::SUPERVISOR),
+                        "MediaBufferDecoder::mPSupervisorTaskQueue");
+  mPDecoderTaskQueue =
+      TaskQueue::Create(GetMediaThreadPool(MediaThreadType::PLATFORM_DECODER),
+                        "MediaBufferDecoder::mPDecoderTaskQueue");
+
+  // If you change this list to add support for new decoders, please consider
+  // updating HTMLMediaElement::CreateDecoder as well.
+  mDemuxer = DecoderTraits::CreateDemuxer(mContainerType, resource);
+  if (!mDemuxer) {
+    return false;
+  }
+
+  return true;
+}
+
+class AutoResampler final {
+ public:
+  AutoResampler() : mResampler(nullptr) {}
+  ~AutoResampler() {
+    if (mResampler) {
+      speex_resampler_destroy(mResampler);
+    }
+  }
+  operator SpeexResamplerState*() const {
+    MOZ_ASSERT(mResampler);
+    return mResampler;
+  }
+  void operator=(SpeexResamplerState* aResampler) { mResampler = aResampler; }
+
+ private:
+  SpeexResamplerState* mResampler;
+};
+
+void MediaDecodeTask::Decode() {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  mDemuxer->Init()->Then(PSupervisorTaskQueue(), __func__, this,
+                         &MediaDecodeTask::OnInitDemuxerCompleted,
+                         &MediaDecodeTask::OnInitDemuxerFailed);
+}
+
+void MediaDecodeTask::OnInitDemuxerCompleted() {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  if (!!mDemuxer->GetNumberTracks(TrackInfo::kAudioTrack)) {
+    mTrackDemuxer = mDemuxer->GetTrackDemuxer(TrackInfo::kAudioTrack, 0);
+    if (!mTrackDemuxer) {
+      LOG("MediaDecodeTask: Could not get a track demuxer.");
+      ReportFailureOnMainThread(WebAudioDecodeJob::UnknownContent);
+      return;
+    }
+
+    RefPtr<PDMFactory> platform = new PDMFactory();
+    UniquePtr<TrackInfo> audioInfo = mTrackDemuxer->GetInfo();
+    // We actively ignore audio tracks that we know we can't play.
+    if (audioInfo && audioInfo->IsValid() &&
+        !platform->SupportsMimeType(audioInfo->mMimeType).isEmpty()) {
+      mMediaInfo.mAudio = *audioInfo->GetAsAudioInfo();
+    }
+  }
+
+  RefPtr<PDMFactory> pdm = new PDMFactory();
+  pdm->CreateDecoder(
+         {*mMediaInfo.mAudio.GetAsAudioInfo(), TrackInfo::kAudioTrack})
+      ->Then(PSupervisorTaskQueue(), __func__, this,
+             &MediaDecodeTask::OnCreateDecoderCompleted,
+             &MediaDecodeTask::OnCreateDecoderFailed);
+}
+
+void MediaDecodeTask::OnCreateDecoderCompleted(
+    RefPtr<MediaDataDecoder> aDecoder) {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  mDecoder = new MediaDataDecoderProxy(aDecoder.forget(),
+                                       do_AddRef(mPDecoderTaskQueue.get()));
+  InitDecoder();
+}
+
+void MediaDecodeTask::OnCreateDecoderFailed(const MediaResult& aError) {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  LOG("MediaDecodeTask: Could not create a decoder.");
+  ReportFailureOnMainThread(WebAudioDecodeJob::UnknownContent);
+}
+
+void MediaDecodeTask::OnInitDemuxerFailed(const MediaResult& aError) {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  LOG("MediaDecodeTask: Could not initialize the demuxer.");
+  ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
+}
+
+void MediaDecodeTask::InitDecoder() {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  mDecoder->Init()->Then(PSupervisorTaskQueue(), __func__, this,
+                         &MediaDecodeTask::OnInitDecoderCompleted,
+                         &MediaDecodeTask::OnInitDecoderFailed);
+}
+
+void MediaDecodeTask::OnInitDecoderCompleted() {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  DoDemux();
+}
+
+void MediaDecodeTask::OnInitDecoderFailed() {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  ShutdownDecoder();
+  LOG("MediaDecodeTask: Could not initialize the decoder");
+  ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
+}
+
+void MediaDecodeTask::DoDemux() {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  mTrackDemuxer->GetSamples(mBatchSize)
+      ->Then(PSupervisorTaskQueue(), __func__, this,
+             &MediaDecodeTask::OnAudioDemuxCompleted,
+             &MediaDecodeTask::OnAudioDemuxFailed);
+}
+
+void MediaDecodeTask::OnAudioDemuxCompleted(
+    RefPtr<MediaTrackDemuxer::SamplesHolder> aSamples) {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  mRawSamples.AppendElements(aSamples->GetSamples());
+
+  DoDemux();
+}
+
+void MediaDecodeTask::OnAudioDemuxFailed(const MediaResult& aError) {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  if (aError.Code() == NS_ERROR_DOM_MEDIA_END_OF_STREAM) {
+    DoDecode();
+  } else {
+    ShutdownDecoder();
+    LOG("MediaDecodeTask: Audio demux failed");
+    ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
+  }
+}
+
+void MediaDecodeTask::DoDecode() {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  if (mRawSamples.IsEmpty()) {
+    DoDrain();
+    return;
+  }
+
+  if (mBatchSize > 1 && mDecoder->CanDecodeBatch()) {
+    nsTArray<RefPtr<MediaRawData>> rawSampleBatch;
+    const int batchSize = std::min((unsigned long)mBatchSize,
+                                   (unsigned long)mRawSamples.Length());
+    for (int i = 0; i < batchSize; ++i) {
+      rawSampleBatch.AppendElement(std::move(mRawSamples[i]));
+    }
+
+    mDecoder->DecodeBatch(std::move(rawSampleBatch))
+        ->Then(PSupervisorTaskQueue(), __func__, this,
+               &MediaDecodeTask::OnAudioDecodeCompleted,
+               &MediaDecodeTask::OnAudioDecodeFailed);
+
+    mRawSamples.RemoveElementsAt(0, batchSize);
+  } else {
+    RefPtr<MediaRawData> sample = std::move(mRawSamples[0]);
+
+    mDecoder->Decode(sample)->Then(PSupervisorTaskQueue(), __func__, this,
+                                   &MediaDecodeTask::OnAudioDecodeCompleted,
+                                   &MediaDecodeTask::OnAudioDecodeFailed);
+
+    mRawSamples.RemoveElementAt(0);
+  }
+}
+
+void MediaDecodeTask::OnAudioDecodeCompleted(
+    MediaDataDecoder::DecodedData&& aResults) {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  for (auto&& sample : aResults) {
+    MOZ_ASSERT(sample->mType == MediaData::Type::AUDIO_DATA);
+    RefPtr<AudioData> audioData = sample->As<AudioData>();
+
+    mMediaInfo.mAudio.mRate = audioData->mRate;
+    mMediaInfo.mAudio.mChannels = audioData->mChannels;
+
+    mAudioQueue.Push(audioData.forget());
+  }
+
+  DoDecode();
+}
+
+void MediaDecodeTask::OnAudioDecodeFailed(const MediaResult& aError) {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  ShutdownDecoder();
+  LOG("MediaDecodeTask: decode audio failed.");
+  ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
+}
+
+void MediaDecodeTask::DoDrain() {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  mDecoder->Drain()->Then(PSupervisorTaskQueue(), __func__, this,
+                          &MediaDecodeTask::OnAudioDrainCompleted,
+                          &MediaDecodeTask::OnAudioDrainFailed);
+}
+
+void MediaDecodeTask::OnAudioDrainCompleted(
+    MediaDataDecoder::DecodedData&& aResults) {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  if (aResults.IsEmpty()) {
+    FinishDecode();
+    return;
+  }
+
+  for (auto&& sample : aResults) {
+    MOZ_ASSERT(sample->mType == MediaData::Type::AUDIO_DATA);
+    RefPtr<AudioData> audioData = sample->As<AudioData>();
+
+    mAudioQueue.Push(audioData.forget());
+  }
+  DoDrain();
+}
+
+void MediaDecodeTask::OnAudioDrainFailed(const MediaResult& aError) {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  ShutdownDecoder();
+  LOG("MediaDecodeTask: Drain audio failed");
+  ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
+}
+
+void MediaDecodeTask::ShutdownDecoder() {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  if (!mDecoder) {
+    return;
+  }
+
+  RefPtr<MediaDecodeTask> self = this;
+  mDecoder->Shutdown();
+  mDecoder = nullptr;
+}
+
+void MediaDecodeTask::FinishDecode() {
+  MOZ_ASSERT(OnPSupervisorTaskQueue());
+
+  ShutdownDecoder();
+
+  uint32_t frameCount = mAudioQueue.AudioFramesCount();
+  uint32_t channelCount = mMediaInfo.mAudio.mChannels;
+  uint32_t sampleRate = mMediaInfo.mAudio.mRate;
+
+  if (!frameCount || !channelCount || !sampleRate) {
+    LOG("MediaDecodeTask: invalid content frame count, channel count or "
+        "sample-rate");
+    ReportFailureOnMainThread(WebAudioDecodeJob::InvalidContent);
+    return;
+  }
+
+  const uint32_t destSampleRate = mDecodeJob.mContext->SampleRate();
+  AutoResampler resampler;
+
+  uint32_t resampledFrames = frameCount;
+  if (sampleRate != destSampleRate) {
+    resampledFrames = static_cast<uint32_t>(
+        static_cast<uint64_t>(destSampleRate) *
+        static_cast<uint64_t>(frameCount) / static_cast<uint64_t>(sampleRate));
+
+    resampler = speex_resampler_init(channelCount, sampleRate, destSampleRate,
+                                     SPEEX_RESAMPLER_QUALITY_DEFAULT, nullptr);
+    speex_resampler_skip_zeros(resampler);
+    resampledFrames += speex_resampler_get_output_latency(resampler);
+  }
+
+  // Allocate contiguous channel buffers.  Note that if we end up resampling,
+  // we may write fewer bytes than mResampledFrames to the output buffer, in
+  // which case writeIndex will tell us how many valid samples we have.
+  mDecodeJob.mBuffer.mChannelData.SetLength(channelCount);
+#if AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_FLOAT32
+  // This buffer has separate channel arrays that could be transferred to
+  // JS::NewArrayBufferWithContents(), but AudioBuffer::RestoreJSChannelData()
+  // does not yet take advantage of this.
+  RefPtr<ThreadSharedFloatArrayBufferList> buffer =
+      ThreadSharedFloatArrayBufferList::Create(channelCount, resampledFrames,
+                                               fallible);
+  if (!buffer) {
+    LOG("MediaDecodeTask: Could not create final buffer (f32)");
+    ReportFailureOnMainThread(WebAudioDecodeJob::UnknownError);
+    return;
+  }
+  for (uint32_t i = 0; i < channelCount; ++i) {
+    mDecodeJob.mBuffer.mChannelData[i] = buffer->GetData(i);
+  }
+#else
+  CheckedInt<size_t> bufferSize(sizeof(AudioDataValue));
+  bufferSize *= resampledFrames;
+  bufferSize *= channelCount;
+  RefPtr<SharedBuffer> buffer = SharedBuffer::Create(bufferSize);
+  if (!buffer) {
+    LOG("MediaDecodeTask: Could not create final buffer (i16)");
+    ReportFailureOnMainThread(WebAudioDecodeJob::UnknownError);
+    return;
+  }
+  auto data = static_cast<AudioDataValue*>(floatBuffer->Data());
+  for (uint32_t i = 0; i < channelCount; ++i) {
+    mDecodeJob.mBuffer.mChannelData[i] = data;
+    data += resampledFrames;
+  }
+#endif
+  mDecodeJob.mBuffer.mBuffer = std::move(buffer);
+  mDecodeJob.mBuffer.mVolume = 1.0f;
+  mDecodeJob.mBuffer.mBufferFormat = AUDIO_OUTPUT_FORMAT;
+
+  uint32_t writeIndex = 0;
+  RefPtr<AudioData> audioData;
+  while ((audioData = mAudioQueue.PopFront())) {
+    if (!audioData->Frames()) {
+      // The packet contains no audio frames, skip it.
+      continue;
+    }
+    audioData->EnsureAudioBuffer();  // could lead to a copy :(
+    const AudioDataValue* bufferData =
+        static_cast<AudioDataValue*>(audioData->mAudioBuffer->Data());
+
+    if (sampleRate != destSampleRate) {
+      const uint32_t maxOutSamples = resampledFrames - writeIndex;
+
+      for (uint32_t i = 0; i < audioData->mChannels; ++i) {
+        uint32_t inSamples = audioData->Frames();
+        uint32_t outSamples = maxOutSamples;
+        AudioDataValue* outData =
+            mDecodeJob.mBuffer.ChannelDataForWrite<AudioDataValue>(i) +
+            writeIndex;
+
+        WebAudioUtils::SpeexResamplerProcess(
+            resampler, i, &bufferData[i * audioData->Frames()], &inSamples,
+            outData, &outSamples);
+
+        if (i == audioData->mChannels - 1) {
+          writeIndex += outSamples;
+          MOZ_ASSERT(writeIndex <= resampledFrames);
+          MOZ_ASSERT(inSamples == audioData->Frames());
+        }
+      }
+    } else {
+      for (uint32_t i = 0; i < audioData->mChannels; ++i) {
+        AudioDataValue* outData =
+            mDecodeJob.mBuffer.ChannelDataForWrite<AudioDataValue>(i) +
+            writeIndex;
+        PodCopy(outData, &bufferData[i * audioData->Frames()],
+                audioData->Frames());
+
+        if (i == audioData->mChannels - 1) {
+          writeIndex += audioData->Frames();
+        }
+      }
+    }
+  }
+
+  if (sampleRate != destSampleRate) {
+    uint32_t inputLatency = speex_resampler_get_input_latency(resampler);
+    const uint32_t maxOutSamples = resampledFrames - writeIndex;
+    for (uint32_t i = 0; i < channelCount; ++i) {
+      uint32_t inSamples = inputLatency;
+      uint32_t outSamples = maxOutSamples;
+      AudioDataValue* outData =
+          mDecodeJob.mBuffer.ChannelDataForWrite<AudioDataValue>(i) +
+          writeIndex;
+
+      WebAudioUtils::SpeexResamplerProcess(resampler, i,
+                                           (AudioDataValue*)nullptr, &inSamples,
+                                           outData, &outSamples);
+
+      if (i == channelCount - 1) {
+        writeIndex += outSamples;
+        MOZ_ASSERT(writeIndex <= resampledFrames);
+        MOZ_ASSERT(inSamples == inputLatency);
+      }
+    }
+  }
+
+  mDecodeJob.mBuffer.mDuration = writeIndex;
+  mPhase = PhaseEnum::AllocateBuffer;
+  mMainThread->Dispatch(do_AddRef(this));
+}
+
+void MediaDecodeTask::AllocateBuffer() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  if (!mDecodeJob.AllocateBuffer()) {
+    LOG("MediaDecodeTask: Could not allocate final buffer");
+    ReportFailureOnMainThread(WebAudioDecodeJob::UnknownError);
+    return;
+  }
+
+  mPhase = PhaseEnum::Done;
+  CallbackTheResult();
+}
+
+void MediaDecodeTask::CallbackTheResult() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  Cleanup();
+
+  // Now, we're ready to call the script back with the resulting buffer
+  mDecodeJob.OnSuccess(WebAudioDecodeJob::NoError);
+}
+
+bool WebAudioDecodeJob::AllocateBuffer() {
+  MOZ_ASSERT(!mOutput);
+  MOZ_ASSERT(NS_IsMainThread());
+
+  // Now create the AudioBuffer
+  mOutput = AudioBuffer::Create(mContext->GetOwner(), mContext->SampleRate(),
+                                std::move(mBuffer));
+  return mOutput != nullptr;
+}
+
+void AsyncDecodeWebAudio(const char* aContentType, uint8_t* aBuffer,
+                         uint32_t aLength, WebAudioDecodeJob& aDecodeJob) {
+  Maybe<MediaContainerType> containerType =
+      MakeMediaContainerType(aContentType);
+  // Do not attempt to decode the media if we were not successful at sniffing
+  // the container type.
+  if (!*aContentType || strcmp(aContentType, APPLICATION_OCTET_STREAM) == 0 ||
+      !containerType) {
+    nsCOMPtr<nsIRunnable> event =
+        new ReportResultTask(aDecodeJob, &WebAudioDecodeJob::OnFailure,
+                             WebAudioDecodeJob::UnknownContent);
+    JS_free(nullptr, aBuffer);
+    aDecodeJob.mContext->Dispatch(event.forget());
+    return;
+  }
+
+  RefPtr<MediaDecodeTask> task =
+      new MediaDecodeTask(*containerType, aBuffer, aLength, aDecodeJob);
+  if (!task->Init()) {
+    nsCOMPtr<nsIRunnable> event =
+        new ReportResultTask(aDecodeJob, &WebAudioDecodeJob::OnFailure,
+                             WebAudioDecodeJob::UnknownError);
+    aDecodeJob.mContext->Dispatch(event.forget());
+  } else {
+    nsresult rv = task->PSupervisorTaskQueue()->Dispatch(task.forget());
+    MOZ_DIAGNOSTIC_ASSERT(NS_SUCCEEDED(rv));
+    Unused << rv;
+  }
+}
+
+WebAudioDecodeJob::WebAudioDecodeJob(AudioContext* aContext, Promise* aPromise,
+                                     DecodeSuccessCallback* aSuccessCallback,
+                                     DecodeErrorCallback* aFailureCallback)
+    : mContext(aContext),
+      mPromise(aPromise),
+      mSuccessCallback(aSuccessCallback),
+      mFailureCallback(aFailureCallback) {
+  MOZ_ASSERT(aContext);
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_COUNT_CTOR(WebAudioDecodeJob);
+}
+
+WebAudioDecodeJob::~WebAudioDecodeJob() {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_COUNT_DTOR(WebAudioDecodeJob);
+}
+
+void WebAudioDecodeJob::OnSuccess(ErrorCode aErrorCode) {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(aErrorCode == NoError);
+
+  RefPtr<AudioBuffer> output(mOutput);
+  if (mSuccessCallback) {
+    RefPtr<DecodeSuccessCallback> callback(mSuccessCallback);
+    // Ignore errors in calling the callback, since there is not much that we
+    // can do about it here.
+    callback->Call(*output);
+  }
+  mPromise->MaybeResolve(output);
+
+  mContext->RemoveFromDecodeQueue(this);
+}
+
+void WebAudioDecodeJob::OnFailure(ErrorCode aErrorCode) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  const char* errorMessage;
+  switch (aErrorCode) {
+    case UnknownContent:
+      errorMessage =
+          "The buffer passed to decodeAudioData contains an unknown content "
+          "type.";
+      break;
+    case InvalidContent:
+      errorMessage =
+          "The buffer passed to decodeAudioData contains invalid content which "
+          "cannot be decoded successfully.";
+      break;
+    case NoAudio:
+      errorMessage =
+          "The buffer passed to decodeAudioData does not contain any audio.";
+      break;
+    case NoError:
+      MOZ_FALLTHROUGH_ASSERT("Who passed NoError to OnFailure?");
+      // Fall through to get some sort of a sane error message if this actually
+      // happens at runtime.
+    case UnknownError:
+      [[fallthrough]];
+    default:
+      errorMessage =
+          "An unknown error occurred while processing decodeAudioData.";
+      break;
+  }
+
+  // Ignore errors in calling the callback, since there is not much that we can
+  // do about it here.
+  nsAutoCString errorString(errorMessage);
+  if (mFailureCallback) {
+    RefPtr<DOMException> exception = DOMException::Create(
+        NS_ERROR_DOM_ENCODING_NOT_SUPPORTED_ERR, errorString);
+    RefPtr<DecodeErrorCallback> callback(mFailureCallback);
+    callback->Call(*exception);
+  }
+
+  mPromise->MaybeRejectWithEncodingError(errorString);
+
+  mContext->RemoveFromDecodeQueue(this);
+}
+
+size_t WebAudioDecodeJob::SizeOfExcludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  size_t amount = 0;
+  if (mSuccessCallback) {
+    amount += mSuccessCallback->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  if (mFailureCallback) {
+    amount += mFailureCallback->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  if (mOutput) {
+    amount += mOutput->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf, false);
+  return amount;
+}
+
+size_t WebAudioDecodeJob::SizeOfIncludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+}  // namespace mozilla
diff --git a/dom/media/webaudio/MediaBufferDecoder.h b/dom/media/webaudio/MediaBufferDecoder.h
new file mode 100644
index 0000000000..7af1acfd06
--- /dev/null
+++ b/dom/media/webaudio/MediaBufferDecoder.h
@@ -0,0 +1,70 @@
+/* -*- 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/. */
+
+#ifndef MediaBufferDecoder_h_
+#define MediaBufferDecoder_h_
+
+#include "AudioSegment.h"
+#include "nsWrapperCache.h"
+#include "nsCOMPtr.h"
+#include "nsString.h"
+#include "nsTArray.h"
+#include "mozilla/Attributes.h"
+#include "mozilla/dom/TypedArray.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace mozilla {
+
+class ThreadSharedFloatArrayBufferList;
+
+namespace dom {
+class AudioBuffer;
+class AudioContext;
+class DecodeErrorCallback;
+class DecodeSuccessCallback;
+class Promise;
+}  // namespace dom
+
+struct WebAudioDecodeJob final {
+  // You may omit both the success and failure callback, or you must pass both.
+  // The callbacks are only necessary for asynchronous operation.
+  WebAudioDecodeJob(dom::AudioContext* aContext, dom::Promise* aPromise,
+                    dom::DecodeSuccessCallback* aSuccessCallback = nullptr,
+                    dom::DecodeErrorCallback* aFailureCallback = nullptr);
+  ~WebAudioDecodeJob();
+
+  enum ErrorCode {
+    NoError,
+    UnknownContent,
+    UnknownError,
+    InvalidContent,
+    NoAudio
+  };
+
+  typedef void (WebAudioDecodeJob::*ResultFn)(ErrorCode);
+
+  MOZ_CAN_RUN_SCRIPT void OnSuccess(ErrorCode /* ignored */);
+  MOZ_CAN_RUN_SCRIPT void OnFailure(ErrorCode aErrorCode);
+
+  bool AllocateBuffer();
+
+  size_t SizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+  size_t SizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+  AudioChunk mBuffer;
+  RefPtr<dom::AudioContext> mContext;
+  RefPtr<dom::Promise> mPromise;
+  RefPtr<dom::DecodeSuccessCallback> mSuccessCallback;
+  RefPtr<dom::DecodeErrorCallback> mFailureCallback;  // can be null
+  RefPtr<dom::AudioBuffer> mOutput;
+};
+
+void AsyncDecodeWebAudio(const char* aContentType, uint8_t* aBuffer,
+                         uint32_t aLength, WebAudioDecodeJob& aDecodeJob);
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/MediaElementAudioSourceNode.cpp b/dom/media/webaudio/MediaElementAudioSourceNode.cpp
new file mode 100644
index 0000000000..953fee66d9
--- /dev/null
+++ b/dom/media/webaudio/MediaElementAudioSourceNode.cpp
@@ -0,0 +1,100 @@
+/* -*- 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 "MediaElementAudioSourceNode.h"
+#include "mozilla/dom/MediaElementAudioSourceNodeBinding.h"
+#include "AudioDestinationNode.h"
+#include "AudioNodeTrack.h"
+#include "MediaStreamTrack.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(MediaElementAudioSourceNode)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(MediaElementAudioSourceNode)
+  tmp->Destroy();
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mElement)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END_INHERITED(AudioNode)
+
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(MediaElementAudioSourceNode,
+                                                  MediaStreamAudioSourceNode)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mElement)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(MediaElementAudioSourceNode)
+NS_INTERFACE_MAP_END_INHERITING(MediaStreamAudioSourceNode)
+
+NS_IMPL_ADDREF_INHERITED(MediaElementAudioSourceNode,
+                         MediaStreamAudioSourceNode)
+NS_IMPL_RELEASE_INHERITED(MediaElementAudioSourceNode,
+                          MediaStreamAudioSourceNode)
+
+MediaElementAudioSourceNode::MediaElementAudioSourceNode(
+    AudioContext* aContext, HTMLMediaElement* aElement)
+    : MediaStreamAudioSourceNode(aContext, TrackChangeBehavior::FollowChanges),
+      mElement(aElement) {
+  MOZ_ASSERT(aElement);
+}
+
+/* static */
+already_AddRefed<MediaElementAudioSourceNode>
+MediaElementAudioSourceNode::Create(
+    AudioContext& aAudioContext, const MediaElementAudioSourceOptions& aOptions,
+    ErrorResult& aRv) {
+  // The spec has a pointless check here.  See
+  // https://github.com/WebAudio/web-audio-api/issues/2149
+  MOZ_RELEASE_ASSERT(!aAudioContext.IsOffline(), "Bindings messed up?");
+
+  RefPtr<MediaElementAudioSourceNode> node =
+      new MediaElementAudioSourceNode(&aAudioContext, aOptions.mMediaElement);
+
+  RefPtr<DOMMediaStream> stream = aOptions.mMediaElement->CaptureAudio(
+      aRv, aAudioContext.Destination()->Track()->Graph());
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+  MOZ_ASSERT(stream, "CaptureAudio should report failure via aRv!");
+
+  node->Init(*stream, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+
+  node->ListenForAllowedToPlay(aOptions);
+  return node.forget();
+}
+
+JSObject* MediaElementAudioSourceNode::WrapObject(
+    JSContext* aCx, JS::Handle<JSObject*> aGivenProto) {
+  return MediaElementAudioSourceNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void MediaElementAudioSourceNode::ListenForAllowedToPlay(
+    const MediaElementAudioSourceOptions& aOptions) {
+  aOptions.mMediaElement->GetAllowedToPlayPromise()
+      ->Then(
+          AbstractThread::MainThread(), __func__,
+          // Capture by reference to bypass the mozilla-refcounted-inside-lambda
+          // static analysis. We capture a non-owning reference so as to allow
+          // cycle collection of the node. The reference is cleared via
+          // DisconnectIfExists() from Destroy() when the node is collected.
+          [&self = *this]() {
+            self.Context()->StartBlockedAudioContextIfAllowed();
+            self.mAllowedToPlayRequest.Complete();
+          })
+      ->Track(mAllowedToPlayRequest);
+}
+
+void MediaElementAudioSourceNode::Destroy() {
+  mAllowedToPlayRequest.DisconnectIfExists();
+  MediaStreamAudioSourceNode::Destroy();
+}
+
+HTMLMediaElement* MediaElementAudioSourceNode::MediaElement() {
+  return mElement;
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/MediaElementAudioSourceNode.h b/dom/media/webaudio/MediaElementAudioSourceNode.h
new file mode 100644
index 0000000000..6ba21279fe
--- /dev/null
+++ b/dom/media/webaudio/MediaElementAudioSourceNode.h
@@ -0,0 +1,67 @@
+/* -*- 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/. */
+
+#ifndef MediaElementAudioSourceNode_h_
+#define MediaElementAudioSourceNode_h_
+
+#include "MediaStreamAudioSourceNode.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct MediaElementAudioSourceOptions;
+
+class MediaElementAudioSourceNode final : public MediaStreamAudioSourceNode {
+ public:
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(MediaElementAudioSourceNode,
+                                           MediaStreamAudioSourceNode)
+  static already_AddRefed<MediaElementAudioSourceNode> Create(
+      AudioContext& aAudioContext,
+      const MediaElementAudioSourceOptions& aOptions, ErrorResult& aRv);
+
+  static already_AddRefed<MediaElementAudioSourceNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const MediaElementAudioSourceOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  const char* NodeType() const override {
+    return "MediaElementAudioSourceNode";
+  }
+
+  const char* CrossOriginErrorString() const override {
+    return "MediaElementAudioSourceNodeCrossOrigin";
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  HTMLMediaElement* MediaElement();
+
+ private:
+  explicit MediaElementAudioSourceNode(AudioContext* aContext,
+                                       HTMLMediaElement* aElement);
+  ~MediaElementAudioSourceNode() = default;
+
+  void Destroy() override;
+
+  // If AudioContext was not allowed to start, we would try to start it when
+  // source starts.
+  void ListenForAllowedToPlay(const MediaElementAudioSourceOptions& aOptions);
+
+  MozPromiseRequestHolder<GenericNonExclusivePromise> mAllowedToPlayRequest;
+
+  RefPtr<HTMLMediaElement> mElement;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/MediaStreamAudioDestinationNode.cpp b/dom/media/webaudio/MediaStreamAudioDestinationNode.cpp
new file mode 100644
index 0000000000..704adaa304
--- /dev/null
+++ b/dom/media/webaudio/MediaStreamAudioDestinationNode.cpp
@@ -0,0 +1,150 @@
+/* -*- 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 "MediaStreamAudioDestinationNode.h"
+#include "mozilla/dom/Document.h"
+#include "mozilla/dom/MediaStreamAudioDestinationNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "AudioStreamTrack.h"
+#include "DOMMediaStream.h"
+#include "ForwardedInputTrack.h"
+
+namespace mozilla::dom {
+
+class AudioDestinationTrackSource final : public MediaStreamTrackSource {
+ public:
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(AudioDestinationTrackSource,
+                                           MediaStreamTrackSource)
+
+  AudioDestinationTrackSource(MediaStreamAudioDestinationNode* aNode,
+                              mozilla::MediaTrack* aInputTrack,
+                              ProcessedMediaTrack* aTrack,
+                              nsIPrincipal* aPrincipal)
+      : MediaStreamTrackSource(
+            aPrincipal, nsString(),
+            // We pass 0 here because tracking ids are video only for now.
+            TrackingId(TrackingId::Source::AudioDestinationNode, 0)),
+        mTrack(aTrack),
+        mPort(mTrack->AllocateInputPort(aInputTrack)),
+        mNode(aNode) {}
+
+  void Destroy() override {
+    if (!mTrack->IsDestroyed()) {
+      mTrack->Destroy();
+      mPort->Destroy();
+    }
+    if (mNode) {
+      mNode->DestroyMediaTrack();
+      mNode = nullptr;
+    }
+  }
+
+  MediaSourceEnum GetMediaSource() const override {
+    return MediaSourceEnum::AudioCapture;
+  }
+
+  void Stop() override { Destroy(); }
+
+  void Disable() override {}
+
+  void Enable() override {}
+
+  const RefPtr<ProcessedMediaTrack> mTrack;
+  const RefPtr<MediaInputPort> mPort;
+
+ private:
+  ~AudioDestinationTrackSource() = default;
+
+  RefPtr<MediaStreamAudioDestinationNode> mNode;
+};
+
+NS_IMPL_ADDREF_INHERITED(AudioDestinationTrackSource, MediaStreamTrackSource)
+NS_IMPL_RELEASE_INHERITED(AudioDestinationTrackSource, MediaStreamTrackSource)
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(AudioDestinationTrackSource)
+NS_INTERFACE_MAP_END_INHERITING(MediaStreamTrackSource)
+NS_IMPL_CYCLE_COLLECTION_INHERITED(AudioDestinationTrackSource,
+                                   MediaStreamTrackSource, mNode)
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(MediaStreamAudioDestinationNode, AudioNode,
+                                   mDOMStream)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(MediaStreamAudioDestinationNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(MediaStreamAudioDestinationNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(MediaStreamAudioDestinationNode, AudioNode)
+
+MediaStreamAudioDestinationNode::MediaStreamAudioDestinationNode(
+    AudioContext* aContext)
+    : AudioNode(aContext, 2, ChannelCountMode::Explicit,
+                ChannelInterpretation::Speakers),
+      mDOMStream(MakeAndAddRef<DOMMediaStream>(GetOwner())) {
+  // Ensure an audio track with the correct ID is exposed to JS. If we can't get
+  // a principal here because the document is not available, pass in a null
+  // principal. This happens in edge cases when the document is being unloaded
+  // and it does not matter too much to have something working as long as it's
+  // not dangerous.
+  nsCOMPtr<nsIPrincipal> principal = nullptr;
+  if (aContext->GetParentObject()) {
+    Document* doc = aContext->GetParentObject()->GetExtantDoc();
+    principal = doc->NodePrincipal();
+  }
+  mTrack = AudioNodeTrack::Create(aContext, new AudioNodeEngine(this),
+                                  AudioNodeTrack::EXTERNAL_OUTPUT,
+                                  aContext->Graph());
+  auto source = MakeRefPtr<AudioDestinationTrackSource>(
+      this, mTrack,
+      aContext->Graph()->CreateForwardedInputTrack(MediaSegment::AUDIO),
+      principal);
+  auto track = MakeRefPtr<AudioStreamTrack>(GetOwner(), source->mTrack, source);
+  mDOMStream->AddTrackInternal(track);
+}
+
+/* static */
+already_AddRefed<MediaStreamAudioDestinationNode>
+MediaStreamAudioDestinationNode::Create(AudioContext& aAudioContext,
+                                        const AudioNodeOptions& aOptions,
+                                        ErrorResult& aRv) {
+  // The spec has a pointless check here.  See
+  // https://github.com/WebAudio/web-audio-api/issues/2149
+  MOZ_RELEASE_ASSERT(!aAudioContext.IsOffline(), "Bindings messed up?");
+
+  RefPtr<MediaStreamAudioDestinationNode> audioNode =
+      new MediaStreamAudioDestinationNode(&aAudioContext);
+
+  audioNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  return audioNode.forget();
+}
+
+size_t MediaStreamAudioDestinationNode::SizeOfExcludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  // Future:
+  // - mDOMStream
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t MediaStreamAudioDestinationNode::SizeOfIncludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+void MediaStreamAudioDestinationNode::DestroyMediaTrack() {
+  AudioNode::DestroyMediaTrack();
+}
+
+JSObject* MediaStreamAudioDestinationNode::WrapObject(
+    JSContext* aCx, JS::Handle<JSObject*> aGivenProto) {
+  return MediaStreamAudioDestinationNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/MediaStreamAudioDestinationNode.h b/dom/media/webaudio/MediaStreamAudioDestinationNode.h
new file mode 100644
index 0000000000..24923955df
--- /dev/null
+++ b/dom/media/webaudio/MediaStreamAudioDestinationNode.h
@@ -0,0 +1,58 @@
+/* -*- 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/. */
+
+#ifndef MediaStreamAudioDestinationNode_h_
+#define MediaStreamAudioDestinationNode_h_
+
+#include "AudioNode.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct AudioNodeOptions;
+
+class MediaStreamAudioDestinationNode final : public AudioNode {
+ public:
+  static already_AddRefed<MediaStreamAudioDestinationNode> Create(
+      AudioContext& aAudioContext, const AudioNodeOptions& aOptions,
+      ErrorResult& aRv);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(MediaStreamAudioDestinationNode,
+                                           AudioNode)
+
+  static already_AddRefed<MediaStreamAudioDestinationNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const AudioNodeOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  uint16_t NumberOfOutputs() const final { return 0; }
+
+  void DestroyMediaTrack() override;
+
+  DOMMediaStream* DOMStream() const { return mDOMStream; }
+
+  const char* NodeType() const override {
+    return "MediaStreamAudioDestinationNode";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ private:
+  explicit MediaStreamAudioDestinationNode(AudioContext* aContext);
+  ~MediaStreamAudioDestinationNode() = default;
+
+  RefPtr<DOMMediaStream> mDOMStream;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/MediaStreamAudioSourceNode.cpp b/dom/media/webaudio/MediaStreamAudioSourceNode.cpp
new file mode 100644
index 0000000000..a915e78859
--- /dev/null
+++ b/dom/media/webaudio/MediaStreamAudioSourceNode.cpp
@@ -0,0 +1,278 @@
+/* -*- 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 "MediaStreamAudioSourceNode.h"
+#include "mozilla/dom/MediaStreamAudioSourceNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeExternalInputTrack.h"
+#include "AudioStreamTrack.h"
+#include "mozilla/dom/Document.h"
+#include "nsContentUtils.h"
+#include "nsIScriptError.h"
+#include "nsID.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(MediaStreamAudioSourceNode)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(MediaStreamAudioSourceNode)
+  tmp->Destroy();
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mInputStream)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mInputTrack)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END_INHERITED(AudioNode)
+
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(MediaStreamAudioSourceNode,
+                                                  AudioNode)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mInputStream)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mInputTrack)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(MediaStreamAudioSourceNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(MediaStreamAudioSourceNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(MediaStreamAudioSourceNode, AudioNode)
+
+MediaStreamAudioSourceNode::MediaStreamAudioSourceNode(
+    AudioContext* aContext, TrackChangeBehavior aBehavior)
+    : AudioNode(aContext, 2, ChannelCountMode::Max,
+                ChannelInterpretation::Speakers),
+      mBehavior(aBehavior) {}
+
+/* static */
+already_AddRefed<MediaStreamAudioSourceNode> MediaStreamAudioSourceNode::Create(
+    AudioContext& aAudioContext, const MediaStreamAudioSourceOptions& aOptions,
+    ErrorResult& aRv) {
+  // The spec has a pointless check here.  See
+  // https://github.com/WebAudio/web-audio-api/issues/2149
+  MOZ_RELEASE_ASSERT(!aAudioContext.IsOffline(), "Bindings messed up?");
+
+  RefPtr<MediaStreamAudioSourceNode> node =
+      new MediaStreamAudioSourceNode(&aAudioContext, LockOnTrackPicked);
+
+  // aOptions.mMediaStream is not nullable.
+  node->Init(*aOptions.mMediaStream, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+
+  return node.forget();
+}
+
+void MediaStreamAudioSourceNode::Init(DOMMediaStream& aMediaStream,
+                                      ErrorResult& aRv) {
+  mInputStream = &aMediaStream;
+  AudioNodeEngine* engine = new MediaStreamAudioSourceNodeEngine(this);
+  mTrack = AudioNodeExternalInputTrack::Create(Context()->Graph(), engine);
+  mInputStream->AddConsumerToKeepAlive(ToSupports(this));
+
+  mInputStream->RegisterTrackListener(this);
+  if (mInputStream->Audible()) {
+    NotifyAudible();
+  }
+  AttachToRightTrack(mInputStream, aRv);
+}
+
+void MediaStreamAudioSourceNode::Destroy() {
+  if (mInputStream) {
+    mInputStream->UnregisterTrackListener(this);
+    mInputStream = nullptr;
+  }
+  DetachFromTrack();
+}
+
+MediaStreamAudioSourceNode::~MediaStreamAudioSourceNode() { Destroy(); }
+
+void MediaStreamAudioSourceNode::AttachToTrack(
+    const RefPtr<MediaStreamTrack>& aTrack, ErrorResult& aRv) {
+  MOZ_ASSERT(!mInputTrack);
+  MOZ_ASSERT(aTrack->AsAudioStreamTrack());
+  MOZ_DIAGNOSTIC_ASSERT(!aTrack->Ended());
+
+  if (!mTrack) {
+    return;
+  }
+
+  if (NS_WARN_IF(Context()->Graph() != aTrack->Graph())) {
+    nsCOMPtr<nsPIDOMWindowInner> pWindow = Context()->GetParentObject();
+    Document* document = pWindow ? pWindow->GetExtantDoc() : nullptr;
+    nsContentUtils::ReportToConsole(nsIScriptError::warningFlag, "Web Audio"_ns,
+                                    document, nsContentUtils::eDOM_PROPERTIES,
+                                    "MediaStreamAudioSourceNodeDifferentRate");
+    // This is not a spec-required exception, just a limitation of our
+    // implementation.
+    aRv.ThrowNotSupportedError(
+        "Connecting AudioNodes from AudioContexts with different sample-rate "
+        "is currently not supported.");
+    return;
+  }
+
+  mInputTrack = aTrack;
+  ProcessedMediaTrack* outputTrack =
+      static_cast<ProcessedMediaTrack*>(mTrack.get());
+  mInputPort = mInputTrack->ForwardTrackContentsTo(outputTrack);
+  PrincipalChanged(mInputTrack);  // trigger enabling/disabling of the connector
+  mInputTrack->AddPrincipalChangeObserver(this);
+  MarkActive();
+}
+
+void MediaStreamAudioSourceNode::DetachFromTrack() {
+  if (mInputTrack) {
+    mInputTrack->RemovePrincipalChangeObserver(this);
+    mInputTrack = nullptr;
+  }
+  if (mInputPort) {
+    mInputPort->Destroy();
+    mInputPort = nullptr;
+  }
+}
+
+static int AudioTrackCompare(const RefPtr<AudioStreamTrack>& aLhs,
+                             const RefPtr<AudioStreamTrack>& aRhs) {
+  nsAutoStringN<NSID_LENGTH> IDLhs;
+  nsAutoStringN<NSID_LENGTH> IDRhs;
+  aLhs->GetId(IDLhs);
+  aRhs->GetId(IDRhs);
+  return Compare(NS_ConvertUTF16toUTF8(IDLhs), NS_ConvertUTF16toUTF8(IDRhs));
+}
+
+void MediaStreamAudioSourceNode::AttachToRightTrack(
+    const RefPtr<DOMMediaStream>& aMediaStream, ErrorResult& aRv) {
+  nsTArray<RefPtr<AudioStreamTrack>> tracks;
+  aMediaStream->GetAudioTracks(tracks);
+
+  if (tracks.IsEmpty() && mBehavior == LockOnTrackPicked) {
+    aRv.ThrowInvalidStateError("No audio tracks in MediaStream");
+    return;
+  }
+
+  // Sort the track to have a stable order, on their ID by lexicographic
+  // ordering on sequences of code unit values.
+  tracks.Sort(AudioTrackCompare);
+
+  for (const RefPtr<AudioStreamTrack>& track : tracks) {
+    if (mBehavior == FollowChanges) {
+      if (track->Ended()) {
+        continue;
+      }
+    }
+
+    if (!track->Ended()) {
+      AttachToTrack(track, aRv);
+    }
+    return;
+  }
+
+  // There was no track available. We'll allow the node to be garbage collected.
+  MarkInactive();
+}
+
+void MediaStreamAudioSourceNode::NotifyTrackAdded(
+    const RefPtr<MediaStreamTrack>& aTrack) {
+  if (mBehavior != FollowChanges) {
+    return;
+  }
+  if (mInputTrack) {
+    return;
+  }
+
+  if (!aTrack->AsAudioStreamTrack()) {
+    return;
+  }
+
+  AttachToTrack(aTrack, IgnoreErrors());
+}
+
+void MediaStreamAudioSourceNode::NotifyTrackRemoved(
+    const RefPtr<MediaStreamTrack>& aTrack) {
+  if (mBehavior == FollowChanges) {
+    if (aTrack != mInputTrack) {
+      return;
+    }
+
+    DetachFromTrack();
+    AttachToRightTrack(mInputStream, IgnoreErrors());
+  }
+}
+
+void MediaStreamAudioSourceNode::NotifyAudible() {
+  MOZ_ASSERT(mInputStream);
+  Context()->StartBlockedAudioContextIfAllowed();
+}
+
+/**
+ * Changes the principal. Note that this will be called on the main thread, but
+ * changes will be enacted on the MediaTrackGraph thread. If the principal
+ * change results in the document principal losing access to the stream, then
+ * there needs to be other measures in place to ensure that any media that is
+ * governed by the new stream principal is not available to the MediaTrackGraph
+ * before this change completes. Otherwise, a site could get access to
+ * media that they are not authorized to receive.
+ *
+ * One solution is to block the altered content, call this method, then dispatch
+ * another change request to the MediaTrackGraph thread that allows the content
+ * under the new principal to flow. This might be unnecessary if the principal
+ * change is changing to be the document principal.
+ */
+void MediaStreamAudioSourceNode::PrincipalChanged(
+    MediaStreamTrack* aMediaStreamTrack) {
+  MOZ_ASSERT(aMediaStreamTrack == mInputTrack);
+
+  bool subsumes = false;
+  Document* doc = nullptr;
+  if (nsPIDOMWindowInner* parent = Context()->GetParentObject()) {
+    doc = parent->GetExtantDoc();
+    if (doc) {
+      nsIPrincipal* docPrincipal = doc->NodePrincipal();
+      nsIPrincipal* trackPrincipal = aMediaStreamTrack->GetPrincipal();
+      if (!trackPrincipal ||
+          NS_FAILED(docPrincipal->Subsumes(trackPrincipal, &subsumes))) {
+        subsumes = false;
+      }
+    }
+  }
+  auto track = static_cast<AudioNodeExternalInputTrack*>(mTrack.get());
+  bool enabled = subsumes;
+  track->SetInt32Parameter(MediaStreamAudioSourceNodeEngine::ENABLE, enabled);
+
+  if (!enabled && doc) {
+    nsContentUtils::ReportToConsole(nsIScriptError::warningFlag, "Web Audio"_ns,
+                                    doc, nsContentUtils::eDOM_PROPERTIES,
+                                    CrossOriginErrorString());
+  }
+}
+
+size_t MediaStreamAudioSourceNode::SizeOfExcludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  // Future:
+  // - mInputStream
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  if (mInputPort) {
+    amount += mInputPort->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  return amount;
+}
+
+size_t MediaStreamAudioSourceNode::SizeOfIncludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+void MediaStreamAudioSourceNode::DestroyMediaTrack() {
+  if (mInputPort) {
+    mInputPort->Destroy();
+    mInputPort = nullptr;
+  }
+  AudioNode::DestroyMediaTrack();
+}
+
+JSObject* MediaStreamAudioSourceNode::WrapObject(
+    JSContext* aCx, JS::Handle<JSObject*> aGivenProto) {
+  return MediaStreamAudioSourceNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/MediaStreamAudioSourceNode.h b/dom/media/webaudio/MediaStreamAudioSourceNode.h
new file mode 100644
index 0000000000..1875fc2e83
--- /dev/null
+++ b/dom/media/webaudio/MediaStreamAudioSourceNode.h
@@ -0,0 +1,127 @@
+/* -*- 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/. */
+
+#ifndef MediaStreamAudioSourceNode_h_
+#define MediaStreamAudioSourceNode_h_
+
+#include "AudioNode.h"
+#include "AudioNodeEngine.h"
+#include "DOMMediaStream.h"
+#include "PrincipalChangeObserver.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct MediaStreamAudioSourceOptions;
+
+class MediaStreamAudioSourceNodeEngine final : public AudioNodeEngine {
+ public:
+  explicit MediaStreamAudioSourceNodeEngine(AudioNode* aNode)
+      : AudioNodeEngine(aNode), mEnabled(false) {}
+
+  bool IsEnabled() const { return mEnabled; }
+  enum Parameters { ENABLE };
+  void SetInt32Parameter(uint32_t aIndex, int32_t aValue) override {
+    switch (aIndex) {
+      case ENABLE:
+        mEnabled = !!aValue;
+        break;
+      default:
+        NS_ERROR("MediaStreamAudioSourceNodeEngine bad parameter index");
+    }
+  }
+
+ private:
+  bool mEnabled;
+};
+
+class MediaStreamAudioSourceNode
+    : public AudioNode,
+      public DOMMediaStream::TrackListener,
+      public PrincipalChangeObserver<MediaStreamTrack> {
+ public:
+  static already_AddRefed<MediaStreamAudioSourceNode> Create(
+      AudioContext& aContext, const MediaStreamAudioSourceOptions& aOptions,
+      ErrorResult& aRv);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(MediaStreamAudioSourceNode,
+                                           AudioNode)
+
+  static already_AddRefed<MediaStreamAudioSourceNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const MediaStreamAudioSourceOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  void DestroyMediaTrack() override;
+
+  uint16_t NumberOfInputs() const override { return 0; }
+
+  DOMMediaStream* GetMediaStream() { return mInputStream; }
+
+  const char* NodeType() const override { return "MediaStreamAudioSourceNode"; }
+
+  virtual const char* CrossOriginErrorString() const {
+    return "MediaStreamAudioSourceNodeCrossOrigin";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+  // Attaches to aTrack so that its audio content will be used as input.
+  void AttachToTrack(const RefPtr<MediaStreamTrack>& aTrack, ErrorResult& aRv);
+
+  // Detaches from the currently attached track if there is one.
+  void DetachFromTrack();
+
+  // Attaches to the first audio track in the MediaStream, when the tracks are
+  // ordered by id.
+  void AttachToRightTrack(const RefPtr<DOMMediaStream>& aMediaStream,
+                          ErrorResult& aRv);
+
+  // From DOMMediaStream::TrackListener.
+  void NotifyTrackAdded(const RefPtr<MediaStreamTrack>& aTrack) override;
+  void NotifyTrackRemoved(const RefPtr<MediaStreamTrack>& aTrack) override;
+  void NotifyAudible() override;
+
+  // From PrincipalChangeObserver<MediaStreamTrack>.
+  void PrincipalChanged(MediaStreamTrack* aMediaStreamTrack) override;
+
+  // This allows implementing the correct behaviour for both
+  // MediaElementAudioSourceNode and MediaStreamAudioSourceNode, that have most
+  // of their behaviour shared.
+  enum TrackChangeBehavior {
+    // MediaStreamAudioSourceNode locks on the track it picked, and never
+    // changes.
+    LockOnTrackPicked,
+    // MediaElementAudioSourceNode can change track, depending on what the
+    // HTMLMediaElement does.
+    FollowChanges
+  };
+
+ protected:
+  MediaStreamAudioSourceNode(AudioContext* aContext,
+                             TrackChangeBehavior aBehavior);
+  void Init(DOMMediaStream& aMediaStream, ErrorResult& aRv);
+  virtual void Destroy();
+  virtual ~MediaStreamAudioSourceNode();
+
+ private:
+  const TrackChangeBehavior mBehavior;
+  RefPtr<MediaInputPort> mInputPort;
+  RefPtr<DOMMediaStream> mInputStream;
+
+  // On construction we set this to the first audio track of mInputStream.
+  RefPtr<MediaStreamTrack> mInputTrack;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/MediaStreamTrackAudioSourceNode.cpp b/dom/media/webaudio/MediaStreamTrackAudioSourceNode.cpp
new file mode 100644
index 0000000000..f3948a33ad
--- /dev/null
+++ b/dom/media/webaudio/MediaStreamTrackAudioSourceNode.cpp
@@ -0,0 +1,200 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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 "MediaStreamTrackAudioSourceNode.h"
+#include "mozilla/dom/MediaStreamTrackAudioSourceNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeExternalInputTrack.h"
+#include "AudioStreamTrack.h"
+#include "mozilla/dom/Document.h"
+#include "nsContentUtils.h"
+#include "nsIScriptError.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(MediaStreamTrackAudioSourceNode)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(MediaStreamTrackAudioSourceNode)
+  tmp->Destroy();
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mInputTrack)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_WEAK_PTR
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END_INHERITED(AudioNode)
+
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(
+    MediaStreamTrackAudioSourceNode, AudioNode)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mInputTrack)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(MediaStreamTrackAudioSourceNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(MediaStreamTrackAudioSourceNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(MediaStreamTrackAudioSourceNode, AudioNode)
+
+MediaStreamTrackAudioSourceNode::MediaStreamTrackAudioSourceNode(
+    AudioContext* aContext)
+    : AudioNode(aContext, 2, ChannelCountMode::Max,
+                ChannelInterpretation::Speakers),
+      mTrackListener(this) {}
+
+/* static */ already_AddRefed<MediaStreamTrackAudioSourceNode>
+MediaStreamTrackAudioSourceNode::Create(
+    AudioContext& aAudioContext,
+    const MediaStreamTrackAudioSourceOptions& aOptions, ErrorResult& aRv) {
+  // The spec has a pointless check here.  See
+  // https://github.com/WebAudio/web-audio-api/issues/2149
+  MOZ_RELEASE_ASSERT(!aAudioContext.IsOffline(), "Bindings messed up?");
+
+  if (!aOptions.mMediaStreamTrack->Ended() &&
+      aAudioContext.Graph() != aOptions.mMediaStreamTrack->Graph()) {
+    nsCOMPtr<nsPIDOMWindowInner> pWindow = aAudioContext.GetParentObject();
+    Document* document = pWindow ? pWindow->GetExtantDoc() : nullptr;
+    nsContentUtils::ReportToConsole(nsIScriptError::warningFlag, "Web Audio"_ns,
+                                    document, nsContentUtils::eDOM_PROPERTIES,
+                                    "MediaStreamAudioSourceNodeDifferentRate");
+    // This is not a spec-required exception, just a limitation of our
+    // implementation.
+    aRv.ThrowNotSupportedError(
+        "Connecting AudioNodes from AudioContexts with different sample-rate "
+        "is currently not supported.");
+    return nullptr;
+  }
+
+  RefPtr<MediaStreamTrackAudioSourceNode> node =
+      new MediaStreamTrackAudioSourceNode(&aAudioContext);
+
+  node->Init(aOptions.mMediaStreamTrack, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+
+  return node.forget();
+}
+
+void MediaStreamTrackAudioSourceNode::Init(MediaStreamTrack* aMediaStreamTrack,
+                                           ErrorResult& aRv) {
+  MOZ_ASSERT(aMediaStreamTrack);
+
+  if (!aMediaStreamTrack->AsAudioStreamTrack()) {
+    aRv.ThrowInvalidStateError("\"mediaStreamTrack\" must be an audio track");
+    return;
+  }
+
+  if (aMediaStreamTrack->Ended()) {
+    // The track is ended and will never produce any data. Pretend like this is
+    // fine.
+    return;
+  }
+
+  MarkActive();
+
+  MediaTrackGraph* graph = Context()->Graph();
+
+  AudioNodeEngine* engine = new MediaStreamTrackAudioSourceNodeEngine(this);
+  mTrack = AudioNodeExternalInputTrack::Create(graph, engine);
+
+  MOZ_ASSERT(mTrack);
+
+  mInputTrack = aMediaStreamTrack;
+  ProcessedMediaTrack* outputTrack =
+      static_cast<ProcessedMediaTrack*>(mTrack.get());
+  mInputPort = mInputTrack->ForwardTrackContentsTo(outputTrack);
+  PrincipalChanged(mInputTrack);  // trigger enabling/disabling of the connector
+  mInputTrack->AddPrincipalChangeObserver(this);
+
+  mInputTrack->AddConsumer(&mTrackListener);
+}
+
+void MediaStreamTrackAudioSourceNode::Destroy() {
+  if (mInputTrack) {
+    mTrackListener.NotifyEnded(mInputTrack);
+    mInputTrack->RemovePrincipalChangeObserver(this);
+    mInputTrack->RemoveConsumer(&mTrackListener);
+    mInputTrack = nullptr;
+  }
+
+  if (mInputPort) {
+    mInputPort->Destroy();
+    mInputPort = nullptr;
+  }
+}
+
+MediaStreamTrackAudioSourceNode::~MediaStreamTrackAudioSourceNode() {
+  Destroy();
+}
+
+/**
+ * Changes the principal. Note that this will be called on the main thread, but
+ * changes will be enacted on the MediaTrackGraph thread. If the principal
+ * change results in the document principal losing access to the track, then
+ * there needs to be other measures in place to ensure that any media that is
+ * governed by the new track principal is not available to the MediaTrackGraph
+ * before this change completes. Otherwise, a site could get access to
+ * media that they are not authorized to receive.
+ *
+ * One solution is to block the altered content, call this method, then dispatch
+ * another change request to the MediaTrackGraph thread that allows the content
+ * under the new principal to flow. This might be unnecessary if the principal
+ * change is changing to be the document principal.
+ */
+void MediaStreamTrackAudioSourceNode::PrincipalChanged(
+    MediaStreamTrack* aMediaStreamTrack) {
+  MOZ_ASSERT(aMediaStreamTrack == mInputTrack);
+
+  bool subsumes = false;
+  Document* doc = nullptr;
+  if (nsPIDOMWindowInner* parent = Context()->GetParentObject()) {
+    doc = parent->GetExtantDoc();
+    if (doc) {
+      nsIPrincipal* docPrincipal = doc->NodePrincipal();
+      nsIPrincipal* trackPrincipal = aMediaStreamTrack->GetPrincipal();
+      if (!trackPrincipal ||
+          NS_FAILED(docPrincipal->Subsumes(trackPrincipal, &subsumes))) {
+        subsumes = false;
+      }
+    }
+  }
+  auto track = static_cast<AudioNodeExternalInputTrack*>(mTrack.get());
+  bool enabled = subsumes;
+  track->SetInt32Parameter(MediaStreamTrackAudioSourceNodeEngine::ENABLE,
+                           enabled);
+  fprintf(stderr, "NOW: %s", enabled ? "enabled" : "disabled");
+
+  if (!enabled && doc) {
+    nsContentUtils::ReportToConsole(nsIScriptError::warningFlag, "Web Audio"_ns,
+                                    doc, nsContentUtils::eDOM_PROPERTIES,
+                                    CrossOriginErrorString());
+  }
+}
+
+size_t MediaStreamTrackAudioSourceNode::SizeOfExcludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  if (mInputPort) {
+    amount += mInputPort->SizeOfIncludingThis(aMallocSizeOf);
+  }
+  return amount;
+}
+
+size_t MediaStreamTrackAudioSourceNode::SizeOfIncludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+void MediaStreamTrackAudioSourceNode::DestroyMediaTrack() {
+  if (mInputPort) {
+    mInputPort->Destroy();
+    mInputPort = nullptr;
+  }
+  AudioNode::DestroyMediaTrack();
+}
+
+JSObject* MediaStreamTrackAudioSourceNode::WrapObject(
+    JSContext* aCx, JS::Handle<JSObject*> aGivenProto) {
+  return MediaStreamTrackAudioSourceNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/MediaStreamTrackAudioSourceNode.h b/dom/media/webaudio/MediaStreamTrackAudioSourceNode.h
new file mode 100644
index 0000000000..bb3d5d6c94
--- /dev/null
+++ b/dom/media/webaudio/MediaStreamTrackAudioSourceNode.h
@@ -0,0 +1,113 @@
+/* -*- Mode: C++; tab-width: 8; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim: set ts=8 sts=2 et sw=2 tw=80: */
+/* 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/. */
+
+#ifndef MediaStreamTrackAudioSourceNode_h_
+#define MediaStreamTrackAudioSourceNode_h_
+
+#include "AudioNode.h"
+#include "AudioNodeEngine.h"
+#include "mozilla/dom/MediaStreamTrack.h"
+#include "mozilla/WeakPtr.h"
+#include "PrincipalChangeObserver.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct MediaStreamTrackAudioSourceOptions;
+
+class MediaStreamTrackAudioSourceNodeEngine final : public AudioNodeEngine {
+ public:
+  explicit MediaStreamTrackAudioSourceNodeEngine(AudioNode* aNode)
+      : AudioNodeEngine(aNode), mEnabled(false) {}
+
+  bool IsEnabled() const { return mEnabled; }
+  enum Parameters { ENABLE };
+  void SetInt32Parameter(uint32_t aIndex, int32_t aValue) override {
+    switch (aIndex) {
+      case ENABLE:
+        mEnabled = !!aValue;
+        break;
+      default:
+        NS_ERROR("MediaStreamTrackAudioSourceNodeEngine bad parameter index");
+    }
+  }
+
+ private:
+  bool mEnabled;
+};
+
+class MediaStreamTrackAudioSourceNode
+    : public AudioNode,
+      public PrincipalChangeObserver<MediaStreamTrack>,
+      public SupportsWeakPtr {
+ public:
+  static already_AddRefed<MediaStreamTrackAudioSourceNode> Create(
+      AudioContext& aContext,
+      const MediaStreamTrackAudioSourceOptions& aOptions, ErrorResult& aRv);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(MediaStreamTrackAudioSourceNode,
+                                           AudioNode)
+
+  static already_AddRefed<MediaStreamTrackAudioSourceNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const MediaStreamTrackAudioSourceOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  void DestroyMediaTrack() override;
+
+  uint16_t NumberOfInputs() const override { return 0; }
+
+  const char* NodeType() const override {
+    return "MediaStreamTrackAudioSourceNode";
+  }
+
+  virtual const char* CrossOriginErrorString() const {
+    return "MediaStreamTrackAudioSourceNodeCrossOrigin";
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+  // From PrincipalChangeObserver<MediaStreamTrack>.
+  void PrincipalChanged(MediaStreamTrack* aMediaStreamTrack) override;
+
+ protected:
+  explicit MediaStreamTrackAudioSourceNode(AudioContext* aContext);
+  void Init(MediaStreamTrack* aMediaStreamTrack, ErrorResult& aRv);
+  void Destroy();
+  virtual ~MediaStreamTrackAudioSourceNode();
+
+  class TrackListener : public MediaStreamTrackConsumer {
+   public:
+    explicit TrackListener(MediaStreamTrackAudioSourceNode* aNode)
+        : mNode(aNode) {}
+
+    void NotifyEnded(MediaStreamTrack* aTrack) override {
+      if (mNode) {
+        mNode->MarkInactive();
+        mNode->DestroyMediaTrack();
+        mNode = nullptr;
+      }
+    }
+
+   private:
+    WeakPtr<MediaStreamTrackAudioSourceNode> mNode;
+  };
+
+ private:
+  RefPtr<MediaInputPort> mInputPort;
+  RefPtr<MediaStreamTrack> mInputTrack;
+  TrackListener mTrackListener;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/OscillatorNode.cpp b/dom/media/webaudio/OscillatorNode.cpp
new file mode 100644
index 0000000000..70592f6e96
--- /dev/null
+++ b/dom/media/webaudio/OscillatorNode.cpp
@@ -0,0 +1,560 @@
+/* -*- 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 "OscillatorNode.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "AudioDestinationNode.h"
+#include "nsContentUtils.h"
+#include "WebAudioUtils.h"
+#include "blink/PeriodicWave.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(OscillatorNode, AudioScheduledSourceNode,
+                                   mPeriodicWave, mFrequency, mDetune)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(OscillatorNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioScheduledSourceNode)
+
+NS_IMPL_ADDREF_INHERITED(OscillatorNode, AudioScheduledSourceNode)
+NS_IMPL_RELEASE_INHERITED(OscillatorNode, AudioScheduledSourceNode)
+
+class OscillatorNodeEngine final : public AudioNodeEngine {
+ public:
+  OscillatorNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination)
+      : AudioNodeEngine(aNode),
+        mSource(nullptr),
+        mDestination(aDestination->Track()),
+        mStart(-1),
+        mStop(TRACK_TIME_MAX)
+        // Keep the default values in sync with OscillatorNode::OscillatorNode.
+        ,
+        mFrequency(440.f),
+        mDetune(0.f),
+        mType(OscillatorType::Sine),
+        mPhase(0.),
+        mFinalFrequency(0.),
+        mPhaseIncrement(0.),
+        mRecomputeParameters(true),
+        mCustomDisableNormalization(false) {
+    MOZ_ASSERT(NS_IsMainThread());
+    mBasicWaveFormCache = aDestination->Context()->GetBasicWaveFormCache();
+  }
+
+  void SetSourceTrack(AudioNodeTrack* aSource) { mSource = aSource; }
+
+  enum Parameters {
+    FREQUENCY,
+    DETUNE,
+    TYPE,
+    DISABLE_NORMALIZATION,
+    START,
+    STOP,
+  };
+  void RecvTimelineEvent(uint32_t aIndex, AudioParamEvent& aEvent) override {
+    mRecomputeParameters = true;
+
+    MOZ_ASSERT(mDestination);
+
+    aEvent.ConvertToTicks(mDestination);
+
+    switch (aIndex) {
+      case FREQUENCY:
+        mFrequency.InsertEvent<int64_t>(aEvent);
+        break;
+      case DETUNE:
+        mDetune.InsertEvent<int64_t>(aEvent);
+        break;
+      default:
+        NS_ERROR("Bad OscillatorNodeEngine TimelineParameter");
+    }
+  }
+
+  void SetTrackTimeParameter(uint32_t aIndex, TrackTime aParam) override {
+    switch (aIndex) {
+      case START:
+        mStart = aParam;
+        mSource->SetActive();
+        break;
+      case STOP:
+        mStop = aParam;
+        break;
+      default:
+        NS_ERROR("Bad OscillatorNodeEngine TrackTimeParameter");
+    }
+  }
+
+  void SetInt32Parameter(uint32_t aIndex, int32_t aParam) override {
+    switch (aIndex) {
+      case TYPE:
+        // Set the new type.
+        mType = static_cast<OscillatorType>(aParam);
+        if (mType == OscillatorType::Sine) {
+          // Forget any previous custom data.
+          mCustomDisableNormalization = false;
+          mPeriodicWave = nullptr;
+          mRecomputeParameters = true;
+        }
+        switch (mType) {
+          case OscillatorType::Sine:
+            mPhase = 0.0;
+            break;
+          case OscillatorType::Square:
+          case OscillatorType::Triangle:
+          case OscillatorType::Sawtooth:
+            mPeriodicWave = mBasicWaveFormCache->GetBasicWaveForm(mType);
+            break;
+          case OscillatorType::Custom:
+            break;
+          default:
+            NS_ERROR("Bad OscillatorNodeEngine type parameter.");
+        }
+        // End type switch.
+        break;
+      case DISABLE_NORMALIZATION:
+        MOZ_ASSERT(aParam >= 0, "negative custom array length");
+        mCustomDisableNormalization = static_cast<uint32_t>(aParam);
+        break;
+      default:
+        NS_ERROR("Bad OscillatorNodeEngine Int32Parameter.");
+    }
+    // End index switch.
+  }
+
+  void SetBuffer(AudioChunk&& aBuffer) override {
+    MOZ_ASSERT(aBuffer.ChannelCount() == 2,
+               "PeriodicWave should have sent two channels");
+    MOZ_ASSERT(aBuffer.mVolume == 1.0f);
+    mPeriodicWave = WebCore::PeriodicWave::create(
+        mSource->mSampleRate, aBuffer.ChannelData<float>()[0],
+        aBuffer.ChannelData<float>()[1], aBuffer.mDuration,
+        mCustomDisableNormalization);
+  }
+
+  void IncrementPhase() {
+    const float twoPiFloat = float(2 * M_PI);
+    mPhase += mPhaseIncrement;
+    if (mPhase > twoPiFloat) {
+      mPhase -= twoPiFloat;
+    } else if (mPhase < -twoPiFloat) {
+      mPhase += twoPiFloat;
+    }
+  }
+
+  // Returns true if the final frequency (and thus the phase increment) changed,
+  // false otherwise. This allow some optimizations at callsite.
+  bool UpdateParametersIfNeeded(size_t aIndexInBlock,
+                                const float aFrequency[WEBAUDIO_BLOCK_SIZE],
+                                const float aDetune[WEBAUDIO_BLOCK_SIZE]) {
+    // Shortcut if frequency-related AudioParam are not automated, and we
+    // already have computed the frequency information and related parameters.
+    if (!ParametersMayNeedUpdate()) {
+      return false;
+    }
+
+    float detune = aDetune[aIndexInBlock];
+    if (detune != mLastDetune) {
+      mLastDetune = detune;
+      // Single-precision fdlibm_exp2f() would sometimes amplify rounding
+      // error in the division for large detune.
+      // https://bugzilla.mozilla.org/show_bug.cgi?id=1849806#c4
+      mDetuneRatio = fdlibm_exp2(detune / 1200.);
+    }
+    float finalFrequency = aFrequency[aIndexInBlock] * mDetuneRatio;
+    mRecomputeParameters = false;
+
+    if (finalFrequency == mFinalFrequency) {
+      return false;
+    }
+
+    mFinalFrequency = finalFrequency;
+    mPhaseIncrement = 2 * M_PI * finalFrequency / mSource->mSampleRate;
+    return true;
+  }
+
+  void FillBounds(float* output, TrackTime ticks, uint32_t& start,
+                  uint32_t& end) {
+    MOZ_ASSERT(output);
+    static_assert(TrackTime(WEBAUDIO_BLOCK_SIZE) < UINT_MAX,
+                  "WEBAUDIO_BLOCK_SIZE overflows interator bounds.");
+    start = 0;
+    if (ticks < mStart) {
+      start = mStart - ticks;
+      for (uint32_t i = 0; i < start; ++i) {
+        output[i] = 0.0;
+      }
+    }
+    end = WEBAUDIO_BLOCK_SIZE;
+    if (ticks + end > mStop) {
+      end = mStop - ticks;
+      for (uint32_t i = end; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+        output[i] = 0.0;
+      }
+    }
+  }
+
+  void ComputeSine(float* aOutput, uint32_t aStart, uint32_t aEnd,
+                   const float aFrequency[WEBAUDIO_BLOCK_SIZE],
+                   const float aDetune[WEBAUDIO_BLOCK_SIZE]) {
+    for (uint32_t i = aStart; i < aEnd; ++i) {
+      // We ignore the return value, changing the frequency has no impact on
+      // performances here.
+      UpdateParametersIfNeeded(i, aFrequency, aDetune);
+
+      aOutput[i] = fdlibm_sinf(mPhase);
+
+      IncrementPhase();
+    }
+  }
+
+  bool ParametersMayNeedUpdate() {
+    return !mDetune.HasSimpleValue() || !mFrequency.HasSimpleValue() ||
+           mRecomputeParameters;
+  }
+
+  void ComputeCustom(float* aOutput, uint32_t aStart, uint32_t aEnd,
+                     const float aFrequency[WEBAUDIO_BLOCK_SIZE],
+                     const float aDetune[WEBAUDIO_BLOCK_SIZE]) {
+    MOZ_ASSERT(mPeriodicWave, "No custom waveform data");
+
+    uint32_t periodicWaveSize = mPeriodicWave->periodicWaveSize();
+    // Mask to wrap wave data indices into the range [0,periodicWaveSize).
+    uint32_t indexMask = periodicWaveSize - 1;
+    MOZ_ASSERT(periodicWaveSize && (periodicWaveSize & indexMask) == 0,
+               "periodicWaveSize must be power of 2");
+    float* higherWaveData = nullptr;
+    float* lowerWaveData = nullptr;
+    float tableInterpolationFactor;
+    // Phase increment at frequency of 1 Hz.
+    // mPhase runs [0,periodicWaveSize) here instead of [0,2*M_PI).
+    float basePhaseIncrement = mPeriodicWave->rateScale();
+
+    bool needToFetchWaveData =
+        UpdateParametersIfNeeded(aStart, aFrequency, aDetune);
+
+    bool parametersMayNeedUpdate = ParametersMayNeedUpdate();
+    mPeriodicWave->waveDataForFundamentalFrequency(
+        mFinalFrequency, lowerWaveData, higherWaveData,
+        tableInterpolationFactor);
+
+    for (uint32_t i = aStart; i < aEnd; ++i) {
+      if (parametersMayNeedUpdate) {
+        if (needToFetchWaveData) {
+          mPeriodicWave->waveDataForFundamentalFrequency(
+              mFinalFrequency, lowerWaveData, higherWaveData,
+              tableInterpolationFactor);
+        }
+        needToFetchWaveData = UpdateParametersIfNeeded(i, aFrequency, aDetune);
+      }
+      // Bilinear interpolation between adjacent samples in each table.
+      float floorPhase = floorf(mPhase);
+      int j1Signed = static_cast<int>(floorPhase);
+      uint32_t j1 = j1Signed & indexMask;
+      uint32_t j2 = j1 + 1;
+      j2 &= indexMask;
+
+      float sampleInterpolationFactor = mPhase - floorPhase;
+
+      float lower = (1.0f - sampleInterpolationFactor) * lowerWaveData[j1] +
+                    sampleInterpolationFactor * lowerWaveData[j2];
+      float higher = (1.0f - sampleInterpolationFactor) * higherWaveData[j1] +
+                     sampleInterpolationFactor * higherWaveData[j2];
+      aOutput[i] = (1.0f - tableInterpolationFactor) * lower +
+                   tableInterpolationFactor * higher;
+
+      // Calculate next phase position from wrapped value j1 to avoid loss of
+      // precision at large values.
+      mPhase =
+          j1 + sampleInterpolationFactor + basePhaseIncrement * mFinalFrequency;
+    }
+  }
+
+  void ComputeSilence(AudioBlock* aOutput) {
+    aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    MOZ_ASSERT(mSource == aTrack, "Invalid source track");
+    TRACE("OscillatorNodeEngine::ProcessBlock");
+
+    TrackTime ticks = mDestination->GraphTimeToTrackTime(aFrom);
+    if (mStart == -1) {
+      ComputeSilence(aOutput);
+      return;
+    }
+
+    if (ticks + WEBAUDIO_BLOCK_SIZE <= mStart || ticks >= mStop ||
+        mStop <= mStart) {
+      ComputeSilence(aOutput);
+
+    } else {
+      aOutput->AllocateChannels(1);
+      float* output = aOutput->ChannelFloatsForWrite(0);
+
+      uint32_t start, end;
+      FillBounds(output, ticks, start, end);
+      MOZ_ASSERT(start < end);
+
+      float frequency[WEBAUDIO_BLOCK_SIZE];
+      float detune[WEBAUDIO_BLOCK_SIZE];
+      if (ParametersMayNeedUpdate()) {
+        if (mFrequency.HasSimpleValue()) {
+          std::fill_n(frequency, WEBAUDIO_BLOCK_SIZE, mFrequency.GetValue());
+        } else {
+          mFrequency.GetValuesAtTime(ticks + start, frequency + start,
+                                     end - start);
+        }
+        if (mDetune.HasSimpleValue()) {
+          std::fill_n(detune, WEBAUDIO_BLOCK_SIZE, mDetune.GetValue());
+        } else {
+          mDetune.GetValuesAtTime(ticks + start, detune + start, end - start);
+        }
+      }
+
+      // Synthesize the correct waveform.
+      switch (mType) {
+        case OscillatorType::Sine:
+          ComputeSine(output, start, end, frequency, detune);
+          break;
+        case OscillatorType::Square:
+        case OscillatorType::Triangle:
+        case OscillatorType::Sawtooth:
+        case OscillatorType::Custom:
+          ComputeCustom(output, start, end, frequency, detune);
+          break;
+        case OscillatorType::EndGuard_:
+          MOZ_ASSERT_UNREACHABLE("end guard");
+          // Avoid `default:` so that `-Wswitch` catches missing enumerators
+          // at compile time.
+      };
+    }
+
+    if (ticks + WEBAUDIO_BLOCK_SIZE >= mStop) {
+      // We've finished playing.
+      *aFinished = true;
+    }
+  }
+
+  bool IsActive() const override {
+    // start() has been called.
+    return mStart != -1;
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+
+    // Not owned:
+    // - mSource
+    // - mDestination
+    // - mFrequency (internal ref owned by node)
+    // - mDetune (internal ref owned by node)
+
+    if (mPeriodicWave) {
+      amount += mPeriodicWave->sizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  // mSource deletes this engine in its destructor
+  AudioNodeTrack* MOZ_NON_OWNING_REF mSource;
+  RefPtr<AudioNodeTrack> mDestination;
+  TrackTime mStart;
+  TrackTime mStop;
+  AudioParamTimeline mFrequency;
+  AudioParamTimeline mDetune;
+  OscillatorType mType;
+  float mPhase;
+  float mFinalFrequency;
+  float mPhaseIncrement;
+  float mLastDetune = 0.f;
+  float mDetuneRatio = 1.f;  // 2^(mLastDetune/1200)
+  bool mRecomputeParameters;
+  RefPtr<BasicWaveFormCache> mBasicWaveFormCache;
+  bool mCustomDisableNormalization;
+  RefPtr<WebCore::PeriodicWave> mPeriodicWave;
+};
+
+OscillatorNode::OscillatorNode(AudioContext* aContext)
+    : AudioScheduledSourceNode(aContext, 2, ChannelCountMode::Max,
+                               ChannelInterpretation::Speakers),
+      mType(OscillatorType::Sine),
+      mStartCalled(false) {
+  mFrequency = CreateAudioParam(
+      OscillatorNodeEngine::FREQUENCY, u"frequency"_ns, 440.0f,
+      -(aContext->SampleRate() / 2), aContext->SampleRate() / 2);
+  mDetune = CreateAudioParam(OscillatorNodeEngine::DETUNE, u"detune"_ns, 0.0f);
+  OscillatorNodeEngine* engine =
+      new OscillatorNodeEngine(this, aContext->Destination());
+  mTrack = AudioNodeTrack::Create(aContext, engine,
+                                  AudioNodeTrack::NEED_MAIN_THREAD_ENDED,
+                                  aContext->Graph());
+  engine->SetSourceTrack(mTrack);
+  mTrack->AddMainThreadListener(this);
+}
+
+/* static */
+already_AddRefed<OscillatorNode> OscillatorNode::Create(
+    AudioContext& aAudioContext, const OscillatorOptions& aOptions,
+    ErrorResult& aRv) {
+  RefPtr<OscillatorNode> audioNode = new OscillatorNode(&aAudioContext);
+
+  audioNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  audioNode->Frequency()->SetInitialValue(aOptions.mFrequency);
+  audioNode->Detune()->SetInitialValue(aOptions.mDetune);
+
+  if (aOptions.mPeriodicWave.WasPassed()) {
+    audioNode->SetPeriodicWave(aOptions.mPeriodicWave.Value());
+  } else {
+    audioNode->SetType(aOptions.mType, aRv);
+    if (NS_WARN_IF(aRv.Failed())) {
+      return nullptr;
+    }
+  }
+
+  return audioNode.forget();
+}
+
+size_t OscillatorNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+
+  // For now only report if we know for sure that it's not shared.
+  if (mPeriodicWave) {
+    amount += mPeriodicWave->SizeOfIncludingThisIfNotShared(aMallocSizeOf);
+  }
+  amount += mFrequency->SizeOfIncludingThis(aMallocSizeOf);
+  amount += mDetune->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t OscillatorNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject* OscillatorNode::WrapObject(JSContext* aCx,
+                                     JS::Handle<JSObject*> aGivenProto) {
+  return OscillatorNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void OscillatorNode::DestroyMediaTrack() {
+  if (mTrack) {
+    mTrack->RemoveMainThreadListener(this);
+  }
+  AudioNode::DestroyMediaTrack();
+}
+
+void OscillatorNode::SendTypeToTrack() {
+  if (!mTrack) {
+    return;
+  }
+  if (mType == OscillatorType::Custom) {
+    // The engine assumes we'll send the custom data before updating the type.
+    SendPeriodicWaveToTrack();
+  }
+  SendInt32ParameterToTrack(OscillatorNodeEngine::TYPE,
+                            static_cast<int32_t>(mType));
+}
+
+void OscillatorNode::SendPeriodicWaveToTrack() {
+  NS_ASSERTION(mType == OscillatorType::Custom,
+               "Sending custom waveform to engine thread with non-custom type");
+  MOZ_ASSERT(mTrack, "Missing node track.");
+  MOZ_ASSERT(mPeriodicWave, "Send called without PeriodicWave object.");
+  SendInt32ParameterToTrack(OscillatorNodeEngine::DISABLE_NORMALIZATION,
+                            mPeriodicWave->DisableNormalization());
+  AudioChunk data = mPeriodicWave->GetThreadSharedBuffer();
+  mTrack->SetBuffer(std::move(data));
+}
+
+void OscillatorNode::Start(double aWhen, ErrorResult& aRv) {
+  if (!WebAudioUtils::IsTimeValid(aWhen)) {
+    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>("start time");
+    return;
+  }
+
+  if (mStartCalled) {
+    aRv.ThrowInvalidStateError("Can't call start() more than once");
+    return;
+  }
+  mStartCalled = true;
+
+  if (!mTrack) {
+    // Nothing to play, or we're already dead for some reason
+    return;
+  }
+
+  // TODO: Perhaps we need to do more here.
+  mTrack->SetTrackTimeParameter(OscillatorNodeEngine::START, Context(), aWhen);
+
+  MarkActive();
+  Context()->StartBlockedAudioContextIfAllowed();
+}
+
+void OscillatorNode::Stop(double aWhen, ErrorResult& aRv) {
+  if (!WebAudioUtils::IsTimeValid(aWhen)) {
+    aRv.ThrowRangeError<MSG_VALUE_OUT_OF_RANGE>("stop time");
+    return;
+  }
+
+  if (!mStartCalled) {
+    aRv.ThrowInvalidStateError("Can't call stop() without calling start()");
+    return;
+  }
+
+  if (!mTrack || !Context()) {
+    // We've already stopped and had our track shut down
+    return;
+  }
+
+  // TODO: Perhaps we need to do more here.
+  mTrack->SetTrackTimeParameter(OscillatorNodeEngine::STOP, Context(),
+                                std::max(0.0, aWhen));
+}
+
+void OscillatorNode::NotifyMainThreadTrackEnded() {
+  MOZ_ASSERT(mTrack->IsEnded());
+
+  class EndedEventDispatcher final : public Runnable {
+   public:
+    explicit EndedEventDispatcher(OscillatorNode* 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(u"ended"_ns);
+      // Release track resources.
+      mNode->DestroyMediaTrack();
+      return NS_OK;
+    }
+
+   private:
+    RefPtr<OscillatorNode> mNode;
+  };
+
+  Context()->Dispatch(do_AddRef(new EndedEventDispatcher(this)));
+
+  // Drop the playing reference
+  // Warning: The below line might delete this.
+  MarkInactive();
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/OscillatorNode.h b/dom/media/webaudio/OscillatorNode.h
new file mode 100644
index 0000000000..1e207b9d67
--- /dev/null
+++ b/dom/media/webaudio/OscillatorNode.h
@@ -0,0 +1,92 @@
+/* -*- 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/. */
+
+#ifndef OscillatorNode_h_
+#define OscillatorNode_h_
+
+#include "AudioScheduledSourceNode.h"
+#include "AudioParam.h"
+#include "PeriodicWave.h"
+#include "mozilla/dom/OscillatorNodeBinding.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct OscillatorOptions;
+
+class OscillatorNode final : public AudioScheduledSourceNode,
+                             public MainThreadMediaTrackListener {
+ public:
+  static already_AddRefed<OscillatorNode> Create(
+      AudioContext& aAudioContext, const OscillatorOptions& aOptions,
+      ErrorResult& aRv);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(OscillatorNode,
+                                           AudioScheduledSourceNode)
+
+  static already_AddRefed<OscillatorNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const OscillatorOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  void DestroyMediaTrack() override;
+
+  uint16_t NumberOfInputs() const final { return 0; }
+
+  OscillatorType Type() const { return mType; }
+  void SetType(OscillatorType aType, ErrorResult& aRv) {
+    if (aType == OscillatorType::Custom) {
+      // ::Custom can only be set by setPeriodicWave().
+      // https://github.com/WebAudio/web-audio-api/issues/105 for exception.
+      aRv.ThrowInvalidStateError("Can't set type to \"custom\"");
+      return;
+    }
+    mType = aType;
+    SendTypeToTrack();
+  }
+
+  AudioParam* Frequency() const { return mFrequency; }
+  AudioParam* Detune() const { return mDetune; }
+
+  void Start(double aWhen, ErrorResult& aRv) override;
+  void Stop(double aWhen, ErrorResult& aRv) override;
+
+  void SetPeriodicWave(PeriodicWave& aPeriodicWave) {
+    mPeriodicWave = &aPeriodicWave;
+    // SendTypeToTrack will call SendPeriodicWaveToTrack for us.
+    mType = OscillatorType::Custom;
+    SendTypeToTrack();
+  }
+
+  void NotifyMainThreadTrackEnded() override;
+
+  const char* NodeType() const override { return "OscillatorNode"; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ private:
+  explicit OscillatorNode(AudioContext* aContext);
+  ~OscillatorNode() = default;
+
+  void SendTypeToTrack();
+  void SendPeriodicWaveToTrack();
+
+  OscillatorType mType;
+  RefPtr<PeriodicWave> mPeriodicWave;
+  RefPtr<AudioParam> mFrequency;
+  RefPtr<AudioParam> mDetune;
+  bool mStartCalled;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/PannerNode.cpp b/dom/media/webaudio/PannerNode.cpp
new file mode 100644
index 0000000000..c6a17e5943
--- /dev/null
+++ b/dom/media/webaudio/PannerNode.cpp
@@ -0,0 +1,730 @@
+/* -*- 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 "PannerNode.h"
+#include "AlignmentUtils.h"
+#include "AudioDestinationNode.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "AudioListener.h"
+#include "PanningUtils.h"
+#include "AudioBufferSourceNode.h"
+#include "PlayingRefChangeHandler.h"
+#include "blink/HRTFPanner.h"
+#include "blink/HRTFDatabaseLoader.h"
+#include "Tracing.h"
+
+using WebCore::HRTFDatabaseLoader;
+using WebCore::HRTFPanner;
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(PannerNode)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(PannerNode, AudioNode)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK(mPositionX, mPositionY, mPositionZ,
+                                  mOrientationX, mOrientationY, mOrientationZ)
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(PannerNode, AudioNode)
+  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mPositionX, mPositionY, mPositionZ,
+                                    mOrientationX, mOrientationY, mOrientationZ)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(PannerNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(PannerNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(PannerNode, AudioNode)
+
+class PannerNodeEngine final : public AudioNodeEngine {
+ public:
+  explicit PannerNodeEngine(AudioNode* aNode,
+                            AudioDestinationNode* aDestination,
+                            AudioListenerEngine* aListenerEngine)
+      : AudioNodeEngine(aNode),
+        mDestination(aDestination->Track()),
+        mListenerEngine(aListenerEngine)
+        // Please keep these default values consistent with
+        // PannerNode::PannerNode below.
+        ,
+        mPanningModelFunction(&PannerNodeEngine::EqualPowerPanningFunction),
+        mDistanceModelFunction(&PannerNodeEngine::InverseGainFunction),
+        mPositionX(0.),
+        mPositionY(0.),
+        mPositionZ(0.),
+        mOrientationX(1.),
+        mOrientationY(0.),
+        mOrientationZ(0.),
+        mRefDistance(1.),
+        mMaxDistance(10000.),
+        mRolloffFactor(1.),
+        mConeInnerAngle(360.),
+        mConeOuterAngle(360.),
+        mConeOuterGain(0.),
+        mLeftOverData(INT_MIN) {}
+
+  void RecvTimelineEvent(uint32_t aIndex, AudioParamEvent& aEvent) override {
+    MOZ_ASSERT(mDestination);
+    aEvent.ConvertToTicks(mDestination);
+    switch (aIndex) {
+      case PannerNode::POSITIONX:
+        mPositionX.InsertEvent<int64_t>(aEvent);
+        break;
+      case PannerNode::POSITIONY:
+        mPositionY.InsertEvent<int64_t>(aEvent);
+        break;
+      case PannerNode::POSITIONZ:
+        mPositionZ.InsertEvent<int64_t>(aEvent);
+        break;
+      case PannerNode::ORIENTATIONX:
+        mOrientationX.InsertEvent<int64_t>(aEvent);
+        break;
+      case PannerNode::ORIENTATIONY:
+        mOrientationY.InsertEvent<int64_t>(aEvent);
+        break;
+      case PannerNode::ORIENTATIONZ:
+        mOrientationZ.InsertEvent<int64_t>(aEvent);
+        break;
+      default:
+        NS_ERROR("Bad PannerNode TimelineParameter");
+    }
+  }
+
+  void CreateHRTFPanner() {
+    MOZ_ASSERT(NS_IsMainThread());
+    if (mHRTFPanner) {
+      return;
+    }
+    // HRTFDatabaseLoader needs to be fetched on the main thread.
+    RefPtr<HRTFDatabaseLoader> loader =
+        HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(
+            NodeMainThread()->Context()->SampleRate());
+    mHRTFPanner = MakeUnique<HRTFPanner>(
+        NodeMainThread()->Context()->SampleRate(), loader.forget());
+  }
+
+  void SetInt32Parameter(uint32_t aIndex, int32_t aParam) override {
+    switch (aIndex) {
+      case PannerNode::PANNING_MODEL:
+        switch (PanningModelType(aParam)) {
+          case PanningModelType::Equalpower:
+            mPanningModelFunction =
+                &PannerNodeEngine::EqualPowerPanningFunction;
+            break;
+          case PanningModelType::HRTF:
+            mPanningModelFunction = &PannerNodeEngine::HRTFPanningFunction;
+            break;
+          default:
+            MOZ_ASSERT_UNREACHABLE("We should never see alternate names here");
+            break;
+        }
+        break;
+      case PannerNode::DISTANCE_MODEL:
+        switch (DistanceModelType(aParam)) {
+          case DistanceModelType::Inverse:
+            mDistanceModelFunction = &PannerNodeEngine::InverseGainFunction;
+            break;
+          case DistanceModelType::Linear:
+            mDistanceModelFunction = &PannerNodeEngine::LinearGainFunction;
+            break;
+          case DistanceModelType::Exponential:
+            mDistanceModelFunction = &PannerNodeEngine::ExponentialGainFunction;
+            break;
+          default:
+            MOZ_ASSERT_UNREACHABLE("We should never see alternate names here");
+            break;
+        }
+        break;
+      default:
+        NS_ERROR("Bad PannerNodeEngine Int32Parameter");
+    }
+  }
+  void SetDoubleParameter(uint32_t aIndex, double aParam) override {
+    switch (aIndex) {
+      case PannerNode::REF_DISTANCE:
+        mRefDistance = aParam;
+        break;
+      case PannerNode::MAX_DISTANCE:
+        mMaxDistance = aParam;
+        break;
+      case PannerNode::ROLLOFF_FACTOR:
+        mRolloffFactor = aParam;
+        break;
+      case PannerNode::CONE_INNER_ANGLE:
+        mConeInnerAngle = aParam;
+        break;
+      case PannerNode::CONE_OUTER_ANGLE:
+        mConeOuterAngle = aParam;
+        break;
+      case PannerNode::CONE_OUTER_GAIN:
+        mConeOuterGain = aParam;
+        break;
+      default:
+        NS_ERROR("Bad PannerNodeEngine DoubleParameter");
+    }
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    TRACE("PannerNodeEngine::ProcessBlock");
+
+    if (aInput.IsNull()) {
+      // mLeftOverData != INT_MIN means that the panning model was HRTF and a
+      // tail-time reference was added.  Even if the model is now equalpower,
+      // the reference will need to be removed.
+      if (mLeftOverData > 0 &&
+          mPanningModelFunction == &PannerNodeEngine::HRTFPanningFunction) {
+        mLeftOverData -= WEBAUDIO_BLOCK_SIZE;
+      } else {
+        if (mLeftOverData != INT_MIN) {
+          mLeftOverData = INT_MIN;
+          aTrack->ScheduleCheckForInactive();
+          mHRTFPanner->reset();
+
+          RefPtr<PlayingRefChangeHandler> refchanged =
+              new PlayingRefChangeHandler(aTrack,
+                                          PlayingRefChangeHandler::RELEASE);
+          aTrack->Graph()->DispatchToMainThreadStableState(refchanged.forget());
+        }
+        aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+        return;
+      }
+    } else if (mPanningModelFunction ==
+               &PannerNodeEngine::HRTFPanningFunction) {
+      if (mLeftOverData == INT_MIN) {
+        RefPtr<PlayingRefChangeHandler> refchanged =
+            new PlayingRefChangeHandler(aTrack,
+                                        PlayingRefChangeHandler::ADDREF);
+        aTrack->Graph()->DispatchToMainThreadStableState(refchanged.forget());
+      }
+      mLeftOverData = mHRTFPanner->maxTailFrames();
+    }
+
+    TrackTime tick = mDestination->GraphTimeToTrackTime(aFrom);
+    (this->*mPanningModelFunction)(aInput, aOutput, tick);
+  }
+
+  bool IsActive() const override { return mLeftOverData != INT_MIN; }
+
+  void ComputeAzimuthAndElevation(const ThreeDPoint& position, float& aAzimuth,
+                                  float& aElevation);
+  float ComputeConeGain(const ThreeDPoint& position,
+                        const ThreeDPoint& orientation);
+  // Compute how much the distance contributes to the gain reduction.
+  double ComputeDistanceGain(const ThreeDPoint& position);
+
+  void EqualPowerPanningFunction(const AudioBlock& aInput, AudioBlock* aOutput,
+                                 TrackTime tick);
+  void HRTFPanningFunction(const AudioBlock& aInput, AudioBlock* aOutput,
+                           TrackTime tick);
+
+  float LinearGainFunction(double aDistance);
+  float InverseGainFunction(double aDistance);
+  float ExponentialGainFunction(double aDistance);
+
+  ThreeDPoint ConvertAudioParamTimelineTo3DP(AudioParamTimeline& aX,
+                                             AudioParamTimeline& aY,
+                                             AudioParamTimeline& aZ,
+                                             TrackTime& tick);
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    if (mHRTFPanner) {
+      amount += mHRTFPanner->sizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  RefPtr<AudioNodeTrack> mDestination;
+  // This member is set on the main thread, but is not accessed on the rendering
+  // thread untile mPanningModelFunction has changed, and this happens strictly
+  // later, via a MediaTrackGraph ControlMessage.
+  UniquePtr<HRTFPanner> mHRTFPanner;
+  RefPtr<AudioListenerEngine> mListenerEngine;
+  using PanningModelFunction = void (PannerNodeEngine::*)(const AudioBlock&,
+                                                          AudioBlock*,
+                                                          TrackTime);
+  PanningModelFunction mPanningModelFunction;
+  using DistanceModelFunction = float (PannerNodeEngine::*)(double);
+  DistanceModelFunction mDistanceModelFunction;
+  AudioParamTimeline mPositionX;
+  AudioParamTimeline mPositionY;
+  AudioParamTimeline mPositionZ;
+  AudioParamTimeline mOrientationX;
+  AudioParamTimeline mOrientationY;
+  AudioParamTimeline mOrientationZ;
+  double mRefDistance;
+  double mMaxDistance;
+  double mRolloffFactor;
+  double mConeInnerAngle;
+  double mConeOuterAngle;
+  double mConeOuterGain;
+  int mLeftOverData;
+};
+
+PannerNode::PannerNode(AudioContext* aContext)
+    : AudioNode(aContext, 2, ChannelCountMode::Clamped_max,
+                ChannelInterpretation::Speakers)
+      // Please keep these default values consistent with
+      // PannerNodeEngine::PannerNodeEngine above.
+      ,
+      mPanningModel(PanningModelType::Equalpower),
+      mDistanceModel(DistanceModelType::Inverse),
+      mRefDistance(1.),
+      mMaxDistance(10000.),
+      mRolloffFactor(1.),
+      mConeInnerAngle(360.),
+      mConeOuterAngle(360.),
+      mConeOuterGain(0.) {
+  mPositionX = CreateAudioParam(PannerNode::POSITIONX, u"PositionX"_ns, 0.f);
+  mPositionY = CreateAudioParam(PannerNode::POSITIONY, u"PositionY"_ns, 0.f);
+  mPositionZ = CreateAudioParam(PannerNode::POSITIONZ, u"PositionZ"_ns, 0.f);
+  mOrientationX =
+      CreateAudioParam(PannerNode::ORIENTATIONX, u"OrientationX"_ns, 1.0f);
+  mOrientationY =
+      CreateAudioParam(PannerNode::ORIENTATIONY, u"OrientationY"_ns, 0.f);
+  mOrientationZ =
+      CreateAudioParam(PannerNode::ORIENTATIONZ, u"OrientationZ"_ns, 0.f);
+  mTrack = AudioNodeTrack::Create(
+      aContext,
+      new PannerNodeEngine(this, aContext->Destination(),
+                           aContext->Listener()->Engine()),
+      AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+}
+
+/* static */
+already_AddRefed<PannerNode> PannerNode::Create(AudioContext& aAudioContext,
+                                                const PannerOptions& aOptions,
+                                                ErrorResult& aRv) {
+  RefPtr<PannerNode> audioNode = new PannerNode(&aAudioContext);
+
+  audioNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  audioNode->SetPanningModel(aOptions.mPanningModel);
+  audioNode->SetDistanceModel(aOptions.mDistanceModel);
+  audioNode->mPositionX->SetInitialValue(aOptions.mPositionX);
+  audioNode->mPositionY->SetInitialValue(aOptions.mPositionY);
+  audioNode->mPositionZ->SetInitialValue(aOptions.mPositionZ);
+  audioNode->mOrientationX->SetInitialValue(aOptions.mOrientationX);
+  audioNode->mOrientationY->SetInitialValue(aOptions.mOrientationY);
+  audioNode->mOrientationZ->SetInitialValue(aOptions.mOrientationZ);
+  audioNode->SetRefDistance(aOptions.mRefDistance, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+  audioNode->SetMaxDistance(aOptions.mMaxDistance, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+  audioNode->SetRolloffFactor(aOptions.mRolloffFactor, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+  audioNode->SetConeInnerAngle(aOptions.mConeInnerAngle);
+  audioNode->SetConeOuterAngle(aOptions.mConeOuterAngle);
+  audioNode->SetConeOuterGain(aOptions.mConeOuterGain, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  return audioNode.forget();
+}
+
+void PannerNode::SetPanningModel(PanningModelType aPanningModel) {
+  mPanningModel = aPanningModel;
+  if (mPanningModel == PanningModelType::HRTF) {
+    // We can set the engine's `mHRTFPanner` member here from the main thread,
+    // because the engine will not touch it from the MediaTrackGraph
+    // thread until the PANNING_MODEL message sent below is received.
+    static_cast<PannerNodeEngine*>(mTrack->Engine())->CreateHRTFPanner();
+  }
+  SendInt32ParameterToTrack(PANNING_MODEL, int32_t(mPanningModel));
+}
+
+static bool SetParamFromDouble(AudioParam* aParam, double aValue,
+                               const char (&aParamName)[2], ErrorResult& aRv) {
+  float value = static_cast<float>(aValue);
+  if (!std::isfinite(value)) {
+    aRv.ThrowTypeError<MSG_NOT_FINITE>(aParamName);
+    return false;
+  }
+  aParam->SetValue(value, aRv);
+  return !aRv.Failed();
+}
+
+void PannerNode::SetPosition(double aX, double aY, double aZ,
+                             ErrorResult& aRv) {
+  if (!SetParamFromDouble(mPositionX, aX, "x", aRv)) {
+    return;
+  }
+  if (!SetParamFromDouble(mPositionY, aY, "y", aRv)) {
+    return;
+  }
+  SetParamFromDouble(mPositionZ, aZ, "z", aRv);
+}
+
+void PannerNode::SetOrientation(double aX, double aY, double aZ,
+                                ErrorResult& aRv) {
+  if (!SetParamFromDouble(mOrientationX, aX, "x", aRv)) {
+    return;
+  }
+  if (!SetParamFromDouble(mOrientationY, aY, "y", aRv)) {
+    return;
+  }
+  SetParamFromDouble(mOrientationZ, aZ, "z", aRv);
+}
+
+size_t PannerNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  return AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+}
+
+size_t PannerNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject* PannerNode::WrapObject(JSContext* aCx,
+                                 JS::Handle<JSObject*> aGivenProto) {
+  return PannerNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+// Those three functions are described in the spec.
+float PannerNodeEngine::LinearGainFunction(double aDistance) {
+  double clampedRollof = std::clamp(mRolloffFactor, 0.0, 1.0);
+  return AssertedCast<float>(
+      1.0 - clampedRollof *
+                (std::max(std::min(aDistance, mMaxDistance), mRefDistance) -
+                 mRefDistance) /
+                (mMaxDistance - mRefDistance));
+}
+
+float PannerNodeEngine::InverseGainFunction(double aDistance) {
+  return mRefDistance /
+         (mRefDistance +
+          mRolloffFactor * (std::max(aDistance, mRefDistance) - mRefDistance));
+}
+
+float PannerNodeEngine::ExponentialGainFunction(double aDistance) {
+  return fdlibm_pow(std::max(aDistance, mRefDistance) / mRefDistance,
+                    -mRolloffFactor);
+}
+
+void PannerNodeEngine::HRTFPanningFunction(const AudioBlock& aInput,
+                                           AudioBlock* aOutput,
+                                           TrackTime tick) {
+  // The output of this node is always stereo, no matter what the inputs are.
+  aOutput->AllocateChannels(2);
+
+  float azimuth, elevation;
+
+  ThreeDPoint position =
+      ConvertAudioParamTimelineTo3DP(mPositionX, mPositionY, mPositionZ, tick);
+  ThreeDPoint orientation = ConvertAudioParamTimelineTo3DP(
+      mOrientationX, mOrientationY, mOrientationZ, tick);
+  if (!orientation.IsZero()) {
+    orientation.Normalize();
+  }
+  ComputeAzimuthAndElevation(position, azimuth, elevation);
+
+  AudioBlock input = aInput;
+  // Gain is applied before the delay and convolution of the HRTF.
+  input.mVolume *=
+      ComputeConeGain(position, orientation) * ComputeDistanceGain(position);
+
+  mHRTFPanner->pan(azimuth, elevation, &input, aOutput);
+}
+
+ThreeDPoint PannerNodeEngine::ConvertAudioParamTimelineTo3DP(
+    AudioParamTimeline& aX, AudioParamTimeline& aY, AudioParamTimeline& aZ,
+    TrackTime& tick) {
+  return ThreeDPoint(aX.GetValueAtTime(tick), aY.GetValueAtTime(tick),
+                     aZ.GetValueAtTime(tick));
+}
+
+void PannerNodeEngine::EqualPowerPanningFunction(const AudioBlock& aInput,
+                                                 AudioBlock* aOutput,
+                                                 TrackTime tick) {
+  float azimuth, elevation, gainL, gainR, normalizedAzimuth, distanceGain,
+      coneGain;
+  int inputChannels = aInput.ChannelCount();
+
+  // Optimize the case where the position and orientation is constant for this
+  // processing block: we can just apply a constant gain on the left and right
+  // channel
+  if (mPositionX.HasSimpleValue() && mPositionY.HasSimpleValue() &&
+      mPositionZ.HasSimpleValue() && mOrientationX.HasSimpleValue() &&
+      mOrientationY.HasSimpleValue() && mOrientationZ.HasSimpleValue()) {
+    ThreeDPoint position(mPositionX.GetValue(), mPositionY.GetValue(),
+                         mPositionZ.GetValue());
+    ThreeDPoint orientation(mOrientationX.GetValue(), mOrientationY.GetValue(),
+                            mOrientationZ.GetValue());
+    if (!orientation.IsZero()) {
+      orientation.Normalize();
+    }
+
+    // For a stereo source, when both the listener and the panner are in
+    // the same spot, and no cone gain is specified, this node is noop.
+    if (inputChannels == 2 && mListenerEngine->Position() == position &&
+        mConeInnerAngle == 360 && mConeOuterAngle == 360) {
+      *aOutput = aInput;
+      return;
+    }
+
+    ComputeAzimuthAndElevation(position, azimuth, elevation);
+    coneGain = ComputeConeGain(position, orientation);
+
+    // The following algorithm is described in the spec.
+    // Clamp azimuth in the [-90, 90] range.
+    azimuth = std::min(180.f, std::max(-180.f, azimuth));
+
+    // Wrap around
+    if (azimuth < -90.f) {
+      azimuth = -180.f - azimuth;
+    } else if (azimuth > 90) {
+      azimuth = 180.f - azimuth;
+    }
+
+    // Normalize the value in the [0, 1] range.
+    if (inputChannels == 1) {
+      normalizedAzimuth = (azimuth + 90.f) / 180.f;
+    } else {
+      if (azimuth <= 0) {
+        normalizedAzimuth = (azimuth + 90.f) / 90.f;
+      } else {
+        normalizedAzimuth = azimuth / 90.f;
+      }
+    }
+
+    distanceGain = ComputeDistanceGain(position);
+
+    // Actually compute the left and right gain.
+    gainL = fdlibm_cos(0.5 * M_PI * normalizedAzimuth);
+    gainR = fdlibm_sin(0.5 * M_PI * normalizedAzimuth);
+
+    // Compute the output.
+    ApplyStereoPanning(aInput, aOutput, gainL, gainR, azimuth <= 0);
+
+    aOutput->mVolume *= distanceGain * coneGain;
+  } else {
+    float positionX[WEBAUDIO_BLOCK_SIZE];
+    float positionY[WEBAUDIO_BLOCK_SIZE];
+    float positionZ[WEBAUDIO_BLOCK_SIZE];
+    float orientationX[WEBAUDIO_BLOCK_SIZE];
+    float orientationY[WEBAUDIO_BLOCK_SIZE];
+    float orientationZ[WEBAUDIO_BLOCK_SIZE];
+
+    if (!mPositionX.HasSimpleValue()) {
+      mPositionX.GetValuesAtTime(tick, positionX, WEBAUDIO_BLOCK_SIZE);
+    } else {
+      positionX[0] = mPositionX.GetValue();
+    }
+    if (!mPositionY.HasSimpleValue()) {
+      mPositionY.GetValuesAtTime(tick, positionY, WEBAUDIO_BLOCK_SIZE);
+    } else {
+      positionY[0] = mPositionY.GetValue();
+    }
+    if (!mPositionZ.HasSimpleValue()) {
+      mPositionZ.GetValuesAtTime(tick, positionZ, WEBAUDIO_BLOCK_SIZE);
+    } else {
+      positionZ[0] = mPositionZ.GetValue();
+    }
+    if (!mOrientationX.HasSimpleValue()) {
+      mOrientationX.GetValuesAtTime(tick, orientationX, WEBAUDIO_BLOCK_SIZE);
+    } else {
+      orientationX[0] = mOrientationX.GetValue();
+    }
+    if (!mOrientationY.HasSimpleValue()) {
+      mOrientationY.GetValuesAtTime(tick, orientationY, WEBAUDIO_BLOCK_SIZE);
+    } else {
+      orientationY[0] = mOrientationY.GetValue();
+    }
+    if (!mOrientationZ.HasSimpleValue()) {
+      mOrientationZ.GetValuesAtTime(tick, orientationZ, WEBAUDIO_BLOCK_SIZE);
+    } else {
+      orientationZ[0] = mOrientationZ.GetValue();
+    }
+
+    float buffer[3 * WEBAUDIO_BLOCK_SIZE + 4];
+    bool onLeft[WEBAUDIO_BLOCK_SIZE];
+
+    float* alignedPanningL = ALIGNED16(buffer);
+    float* alignedPanningR = alignedPanningL + WEBAUDIO_BLOCK_SIZE;
+    float* alignedGain = alignedPanningR + WEBAUDIO_BLOCK_SIZE;
+    ASSERT_ALIGNED16(alignedPanningL);
+    ASSERT_ALIGNED16(alignedPanningR);
+    ASSERT_ALIGNED16(alignedGain);
+
+    for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
+      ThreeDPoint position(
+          mPositionX.HasSimpleValue() ? positionX[0] : positionX[counter],
+          mPositionY.HasSimpleValue() ? positionY[0] : positionY[counter],
+          mPositionZ.HasSimpleValue() ? positionZ[0] : positionZ[counter]);
+      ThreeDPoint orientation(
+          mOrientationX.HasSimpleValue() ? orientationX[0]
+                                         : orientationX[counter],
+          mOrientationY.HasSimpleValue() ? orientationY[0]
+                                         : orientationY[counter],
+          mOrientationZ.HasSimpleValue() ? orientationZ[0]
+                                         : orientationZ[counter]);
+      if (!orientation.IsZero()) {
+        orientation.Normalize();
+      }
+
+      ComputeAzimuthAndElevation(position, azimuth, elevation);
+      coneGain = ComputeConeGain(position, orientation);
+
+      // The following algorithm is described in the spec.
+      // Clamp azimuth in the [-90, 90] range.
+      azimuth = std::min(180.f, std::max(-180.f, azimuth));
+
+      // Wrap around
+      if (azimuth < -90.f) {
+        azimuth = -180.f - azimuth;
+      } else if (azimuth > 90) {
+        azimuth = 180.f - azimuth;
+      }
+
+      // Normalize the value in the [0, 1] range.
+      if (inputChannels == 1) {
+        normalizedAzimuth = (azimuth + 90.f) / 180.f;
+      } else {
+        if (azimuth <= 0) {
+          normalizedAzimuth = (azimuth + 90.f) / 90.f;
+        } else {
+          normalizedAzimuth = azimuth / 90.f;
+        }
+      }
+
+      distanceGain = ComputeDistanceGain(position);
+
+      // Actually compute the left and right gain.
+      float gainL = fdlibm_cos(0.5 * M_PI * normalizedAzimuth);
+      float gainR = fdlibm_sin(0.5 * M_PI * normalizedAzimuth);
+
+      alignedPanningL[counter] = gainL;
+      alignedPanningR[counter] = gainR;
+      alignedGain[counter] = distanceGain * coneGain;
+      onLeft[counter] = azimuth <= 0;
+    }
+
+    // Apply the panning to the output buffer
+    ApplyStereoPanning(aInput, aOutput, alignedPanningL, alignedPanningR,
+                       onLeft);
+
+    // Apply the input volume, cone and distance gain to the output buffer.
+    float* outputL = aOutput->ChannelFloatsForWrite(0);
+    float* outputR = aOutput->ChannelFloatsForWrite(1);
+    AudioBlockInPlaceScale(outputL, alignedGain);
+    AudioBlockInPlaceScale(outputR, alignedGain);
+  }
+}
+
+// This algorithm is specified in the webaudio spec.
+void PannerNodeEngine::ComputeAzimuthAndElevation(const ThreeDPoint& position,
+                                                  float& aAzimuth,
+                                                  float& aElevation) {
+  ThreeDPoint sourceListener = position - mListenerEngine->Position();
+  if (sourceListener.IsZero()) {
+    aAzimuth = 0.0;
+    aElevation = 0.0;
+    return;
+  }
+
+  sourceListener.Normalize();
+
+  // Project the source-listener vector on the x-z plane.
+  const ThreeDPoint& listenerFront = mListenerEngine->FrontVector();
+  const ThreeDPoint& listenerRight = mListenerEngine->RightVector();
+  ThreeDPoint up = listenerRight.CrossProduct(listenerFront);
+
+  double upProjection = sourceListener.DotProduct(up);
+  aElevation = 90 - 180 * fdlibm_acos(upProjection) / M_PI;
+
+  if (aElevation > 90) {
+    aElevation = 180 - aElevation;
+  } else if (aElevation < -90) {
+    aElevation = -180 - aElevation;
+  }
+
+  ThreeDPoint projectedSource = sourceListener - up * upProjection;
+  if (projectedSource.IsZero()) {
+    // source - listener direction is up or down.
+    aAzimuth = 0.0;
+    return;
+  }
+  projectedSource.Normalize();
+
+  // Actually compute the angle, and convert to degrees
+  double projection = projectedSource.DotProduct(listenerRight);
+  aAzimuth = 180 * fdlibm_acos(projection) / M_PI;
+
+  // Compute whether the source is in front or behind the listener.
+  double frontBack = projectedSource.DotProduct(listenerFront);
+  if (frontBack < 0) {
+    aAzimuth = 360 - aAzimuth;
+  }
+  // Rotate the azimuth so it is relative to the listener front vector instead
+  // of the right vector.
+  if ((aAzimuth >= 0) && (aAzimuth <= 270)) {
+    aAzimuth = 90 - aAzimuth;
+  } else {
+    aAzimuth = 450 - aAzimuth;
+  }
+}
+
+// This algorithm is described in the WebAudio spec.
+float PannerNodeEngine::ComputeConeGain(const ThreeDPoint& position,
+                                        const ThreeDPoint& orientation) {
+  // Omnidirectional source
+  if (orientation.IsZero() ||
+      ((mConeInnerAngle == 360) && (mConeOuterAngle == 360))) {
+    return 1;
+  }
+
+  // Normalized source-listener vector
+  ThreeDPoint sourceToListener = mListenerEngine->Position() - position;
+  sourceToListener.Normalize();
+
+  // Angle between the source orientation vector and the source-listener vector
+  double dotProduct = sourceToListener.DotProduct(orientation);
+  double angle = 180 * fdlibm_acos(dotProduct) / M_PI;
+  double absAngle = fabs(angle);
+
+  // Divide by 2 here since API is entire angle (not half-angle)
+  double absInnerAngle = fabs(mConeInnerAngle) / 2;
+  double absOuterAngle = fabs(mConeOuterAngle) / 2;
+  double gain = 1;
+
+  if (absAngle <= absInnerAngle) {
+    // No attenuation
+    gain = 1;
+  } else if (absAngle >= absOuterAngle) {
+    // Max attenuation
+    gain = mConeOuterGain;
+  } else {
+    // Between inner and outer cones
+    // inner -> outer, x goes from 0 -> 1
+    double x = (absAngle - absInnerAngle) / (absOuterAngle - absInnerAngle);
+    gain = (1 - x) + mConeOuterGain * x;
+  }
+
+  return gain;
+}
+
+double PannerNodeEngine::ComputeDistanceGain(const ThreeDPoint& position) {
+  ThreeDPoint distanceVec = position - mListenerEngine->Position();
+  float distance = sqrt(distanceVec.DotProduct(distanceVec));
+  return std::max(0.0f, (this->*mDistanceModelFunction)(distance));
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/PannerNode.h b/dom/media/webaudio/PannerNode.h
new file mode 100644
index 0000000000..94876c371c
--- /dev/null
+++ b/dom/media/webaudio/PannerNode.h
@@ -0,0 +1,221 @@
+/* -*- 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/. */
+
+#ifndef PannerNode_h_
+#define PannerNode_h_
+
+#include "AudioNode.h"
+#include "AudioParam.h"
+#include "nsPrintfCString.h"
+#include "mozilla/dom/PannerNodeBinding.h"
+#include "ThreeDPoint.h"
+#include <limits>
+#include <set>
+
+namespace mozilla::dom {
+
+class AudioContext;
+class AudioBufferSourceNode;
+struct PannerOptions;
+
+class PannerNode final : public AudioNode {
+ public:
+  static already_AddRefed<PannerNode> Create(AudioContext& aAudioContext,
+                                             const PannerOptions& aOptions,
+                                             ErrorResult& aRv);
+
+  static already_AddRefed<PannerNode> Constructor(const GlobalObject& aGlobal,
+                                                  AudioContext& aAudioContext,
+                                                  const PannerOptions& aOptions,
+                                                  ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  void SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv) override {
+    if (aChannelCount > 2) {
+      aRv.ThrowNotSupportedError(
+          nsPrintfCString("%u is greater than 2", aChannelCount));
+      return;
+    }
+    AudioNode::SetChannelCount(aChannelCount, aRv);
+  }
+  void SetChannelCountModeValue(ChannelCountMode aMode,
+                                ErrorResult& aRv) override {
+    if (aMode == ChannelCountMode::Max) {
+      aRv.ThrowNotSupportedError("Cannot set channel count mode to \"max\"");
+      return;
+    }
+    AudioNode::SetChannelCountModeValue(aMode, aRv);
+  }
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(PannerNode, AudioNode)
+
+  PanningModelType PanningModel() const { return mPanningModel; }
+  void SetPanningModel(PanningModelType aPanningModel);
+
+  DistanceModelType DistanceModel() const { return mDistanceModel; }
+  void SetDistanceModel(DistanceModelType aDistanceModel) {
+    mDistanceModel = aDistanceModel;
+    SendInt32ParameterToTrack(DISTANCE_MODEL, int32_t(mDistanceModel));
+  }
+
+  void SetPosition(double aX, double aY, double aZ, ErrorResult& aRv);
+  void SetOrientation(double aX, double aY, double aZ, ErrorResult& aRv);
+
+  double RefDistance() const { return mRefDistance; }
+  void SetRefDistance(double aRefDistance, ErrorResult& aRv) {
+    if (WebAudioUtils::FuzzyEqual(mRefDistance, aRefDistance)) {
+      return;
+    }
+
+    if (aRefDistance < 0) {
+      aRv.ThrowRangeError(
+          "The refDistance value passed to PannerNode must not be negative.");
+      return;
+    }
+
+    mRefDistance = aRefDistance;
+    SendDoubleParameterToTrack(REF_DISTANCE, mRefDistance);
+  }
+
+  double MaxDistance() const { return mMaxDistance; }
+  void SetMaxDistance(double aMaxDistance, ErrorResult& aRv) {
+    if (WebAudioUtils::FuzzyEqual(mMaxDistance, aMaxDistance)) {
+      return;
+    }
+
+    if (aMaxDistance <= 0) {
+      aRv.ThrowRangeError(
+          "The maxDistance value passed to PannerNode must be positive.");
+      return;
+    }
+
+    mMaxDistance = aMaxDistance;
+    SendDoubleParameterToTrack(MAX_DISTANCE, mMaxDistance);
+  }
+
+  double RolloffFactor() const { return mRolloffFactor; }
+  void SetRolloffFactor(double aRolloffFactor, ErrorResult& aRv) {
+    if (WebAudioUtils::FuzzyEqual(mRolloffFactor, aRolloffFactor)) {
+      return;
+    }
+
+    if (aRolloffFactor < 0) {
+      aRv.ThrowRangeError(
+          "The rolloffFactor value passed to PannerNode must not be negative.");
+      return;
+    }
+
+    mRolloffFactor = aRolloffFactor;
+    SendDoubleParameterToTrack(ROLLOFF_FACTOR, mRolloffFactor);
+  }
+
+  double ConeInnerAngle() const { return mConeInnerAngle; }
+  void SetConeInnerAngle(double aConeInnerAngle) {
+    if (WebAudioUtils::FuzzyEqual(mConeInnerAngle, aConeInnerAngle)) {
+      return;
+    }
+    mConeInnerAngle = aConeInnerAngle;
+    SendDoubleParameterToTrack(CONE_INNER_ANGLE, mConeInnerAngle);
+  }
+
+  double ConeOuterAngle() const { return mConeOuterAngle; }
+  void SetConeOuterAngle(double aConeOuterAngle) {
+    if (WebAudioUtils::FuzzyEqual(mConeOuterAngle, aConeOuterAngle)) {
+      return;
+    }
+    mConeOuterAngle = aConeOuterAngle;
+    SendDoubleParameterToTrack(CONE_OUTER_ANGLE, mConeOuterAngle);
+  }
+
+  double ConeOuterGain() const { return mConeOuterGain; }
+  void SetConeOuterGain(double aConeOuterGain, ErrorResult& aRv) {
+    if (WebAudioUtils::FuzzyEqual(mConeOuterGain, aConeOuterGain)) {
+      return;
+    }
+
+    if (aConeOuterGain < 0 || aConeOuterGain > 1) {
+      aRv.ThrowInvalidStateError(
+          nsPrintfCString("%g is not in the range [0, 1]", aConeOuterGain));
+      return;
+    }
+
+    mConeOuterGain = aConeOuterGain;
+    SendDoubleParameterToTrack(CONE_OUTER_GAIN, mConeOuterGain);
+  }
+
+  AudioParam* PositionX() { return mPositionX; }
+
+  AudioParam* PositionY() { return mPositionY; }
+
+  AudioParam* PositionZ() { return mPositionZ; }
+
+  AudioParam* OrientationX() { return mOrientationX; }
+
+  AudioParam* OrientationY() { return mOrientationY; }
+
+  AudioParam* OrientationZ() { return mOrientationZ; }
+
+  const char* NodeType() const override { return "PannerNode"; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ private:
+  explicit PannerNode(AudioContext* aContext);
+  ~PannerNode() = default;
+
+  friend class AudioListener;
+  friend class PannerNodeEngine;
+  enum EngineParameters {
+    PANNING_MODEL,
+    DISTANCE_MODEL,
+    POSITION,
+    POSITIONX,
+    POSITIONY,
+    POSITIONZ,
+    ORIENTATION,  // unit length or zero
+    ORIENTATIONX,
+    ORIENTATIONY,
+    ORIENTATIONZ,
+    REF_DISTANCE,
+    MAX_DISTANCE,
+    ROLLOFF_FACTOR,
+    CONE_INNER_ANGLE,
+    CONE_OUTER_ANGLE,
+    CONE_OUTER_GAIN
+  };
+
+  ThreeDPoint ConvertAudioParamTo3DP(RefPtr<AudioParam> aX,
+                                     RefPtr<AudioParam> aY,
+                                     RefPtr<AudioParam> aZ) {
+    return ThreeDPoint(aX->GetValue(), aY->GetValue(), aZ->GetValue());
+  }
+
+  PanningModelType mPanningModel;
+  DistanceModelType mDistanceModel;
+  RefPtr<AudioParam> mPositionX;
+  RefPtr<AudioParam> mPositionY;
+  RefPtr<AudioParam> mPositionZ;
+  RefPtr<AudioParam> mOrientationX;
+  RefPtr<AudioParam> mOrientationY;
+  RefPtr<AudioParam> mOrientationZ;
+
+  double mRefDistance;
+  double mMaxDistance;
+  double mRolloffFactor;
+  double mConeInnerAngle;
+  double mConeOuterAngle;
+  double mConeOuterGain;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/PanningUtils.h b/dom/media/webaudio/PanningUtils.h
new file mode 100644
index 0000000000..eef8ae78c9
--- /dev/null
+++ b/dom/media/webaudio/PanningUtils.h
@@ -0,0 +1,62 @@
+/* -*- 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/. */
+
+#ifndef PANNING_UTILS_H
+#define PANNING_UTILS_H
+
+#include "AudioSegment.h"
+#include "AudioNodeEngine.h"
+
+namespace mozilla::dom {
+
+template <typename T>
+void GainMonoToStereo(const AudioBlock& aInput, AudioBlock* aOutput, T aGainL,
+                      T aGainR) {
+  float* outputL = aOutput->ChannelFloatsForWrite(0);
+  float* outputR = aOutput->ChannelFloatsForWrite(1);
+  const float* input = static_cast<const float*>(aInput.mChannelData[0]);
+
+  MOZ_ASSERT(aInput.ChannelCount() == 1);
+  MOZ_ASSERT(aOutput->ChannelCount() == 2);
+
+  AudioBlockPanMonoToStereo(input, aGainL, aGainR, outputL, outputR);
+}
+
+// T can be float or an array of float, and  U can be bool or an array of bool,
+// depending if the value of the parameters are constant for this block.
+template <typename T, typename U>
+void GainStereoToStereo(const AudioBlock& aInput, AudioBlock* aOutput, T aGainL,
+                        T aGainR, U aOnLeft) {
+  float* outputL = aOutput->ChannelFloatsForWrite(0);
+  float* outputR = aOutput->ChannelFloatsForWrite(1);
+  const float* inputL = static_cast<const float*>(aInput.mChannelData[0]);
+  const float* inputR = static_cast<const float*>(aInput.mChannelData[1]);
+
+  MOZ_ASSERT(aInput.ChannelCount() == 2);
+  MOZ_ASSERT(aOutput->ChannelCount() == 2);
+
+  AudioBlockPanStereoToStereo(inputL, inputR, aGainL, aGainR, aOnLeft, outputL,
+                              outputR);
+}
+
+// T can be float or an array of float, and  U can be bool or an array of bool,
+// depending if the value of the parameters are constant for this block.
+template <typename T, typename U>
+void ApplyStereoPanning(const AudioBlock& aInput, AudioBlock* aOutput, T aGainL,
+                        T aGainR, U aOnLeft) {
+  aOutput->AllocateChannels(2);
+
+  if (aInput.ChannelCount() == 1) {
+    GainMonoToStereo(aInput, aOutput, aGainL, aGainR);
+  } else {
+    GainStereoToStereo(aInput, aOutput, aGainL, aGainR, aOnLeft);
+  }
+  aOutput->mVolume = aInput.mVolume;
+}
+
+}  // namespace mozilla::dom
+
+#endif  // PANNING_UTILS_H
diff --git a/dom/media/webaudio/PeriodicWave.cpp b/dom/media/webaudio/PeriodicWave.cpp
new file mode 100644
index 0000000000..e036161724
--- /dev/null
+++ b/dom/media/webaudio/PeriodicWave.cpp
@@ -0,0 +1,146 @@
+/* -*- 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 "PeriodicWave.h"
+#include "AudioContext.h"
+#include "mozilla/dom/PeriodicWaveBinding.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_WRAPPERCACHE(PeriodicWave, mContext)
+
+PeriodicWave::PeriodicWave(AudioContext* aContext, const float* aRealData,
+                           const uint32_t aRealSize, const float* aImagData,
+                           const uint32_t aImagSize,
+                           const bool aDisableNormalization, ErrorResult& aRv)
+    : mContext(aContext), mDisableNormalization(aDisableNormalization) {
+  if (aRealData && aImagData && aRealSize != aImagSize) {
+    aRv.ThrowIndexSizeError("\"real\" and \"imag\" are different in length");
+    return;
+  }
+
+  uint32_t length = 0;
+  if (aRealData) {
+    length = aRealSize;
+  } else if (aImagData) {
+    length = aImagSize;
+  } else {
+    // If nothing has been passed, this PeriodicWave will be a sine wave: 2
+    // elements for each array, the second imaginary component set to 1.0.
+    length = 2;
+  }
+
+  if (length < 2) {
+    aRv.ThrowIndexSizeError(
+        "\"real\" and \"imag\" must have a length of "
+        "at least 2");
+    return;
+  }
+
+  MOZ_ASSERT(aContext);
+  MOZ_ASSERT((aRealData || aImagData) || length == 2);
+
+  // Caller should have checked this and thrown.
+  MOZ_ASSERT(length >= 2);
+  mCoefficients.mDuration = length;
+
+  // Copy coefficient data.
+  // The SharedBuffer and two arrays share a single allocation.
+  CheckedInt<size_t> bufferSize(sizeof(float));
+  bufferSize *= length;
+  bufferSize *= 2;
+  RefPtr<SharedBuffer> buffer = SharedBuffer::Create(bufferSize, fallible);
+  if (!buffer) {
+    aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+    return;
+  }
+
+  auto data = static_cast<float*>(buffer->Data());
+  mCoefficients.mBuffer = std::move(buffer);
+
+  if (!aRealData && !aImagData) {
+    PodZero(data, length);
+    mCoefficients.mChannelData.AppendElement(data);
+    data += length;
+    data[0] = 0.0f;
+    data[1] = 1.0f;
+    mCoefficients.mChannelData.AppendElement(data);
+  } else {
+    if (aRealData) {
+      PodCopy(data, aRealData, length);
+    } else {
+      PodZero(data, length);
+    }
+    mCoefficients.mChannelData.AppendElement(data);
+
+    data += length;
+    if (aImagData) {
+      PodCopy(data, aImagData, length);
+    } else {
+      PodZero(data, length);
+    }
+    mCoefficients.mChannelData.AppendElement(data);
+  }
+  mCoefficients.mVolume = 1.0f;
+  mCoefficients.mBufferFormat = AUDIO_FORMAT_FLOAT32;
+}
+
+/* static */
+already_AddRefed<PeriodicWave> PeriodicWave::Constructor(
+    const GlobalObject& aGlobal, AudioContext& aAudioContext,
+    const PeriodicWaveOptions& aOptions, ErrorResult& aRv) {
+  const float* realData;
+  const float* imagData;
+  uint32_t realSize;
+  uint32_t imagSize;
+
+  if (aOptions.mReal.WasPassed()) {
+    realData = aOptions.mReal.Value().Elements();
+    realSize = aOptions.mReal.Value().Length();
+  } else {
+    realData = nullptr;
+    realSize = 0;
+  }
+
+  if (aOptions.mImag.WasPassed()) {
+    imagData = aOptions.mImag.Value().Elements();
+    imagSize = aOptions.mImag.Value().Length();
+  } else {
+    imagData = nullptr;
+    imagSize = 0;
+  }
+
+  RefPtr<PeriodicWave> wave =
+      new PeriodicWave(&aAudioContext, realData, realSize, imagData, imagSize,
+                       aOptions.mDisableNormalization, aRv);
+  if (aRv.Failed()) {
+    return nullptr;
+  }
+
+  return wave.forget();
+}
+
+size_t PeriodicWave::SizeOfExcludingThisIfNotShared(
+    MallocSizeOf aMallocSizeOf) const {
+  // Not owned:
+  // - mContext
+  size_t amount = 0;
+  amount += mCoefficients.SizeOfExcludingThisIfUnshared(aMallocSizeOf);
+
+  return amount;
+}
+
+size_t PeriodicWave::SizeOfIncludingThisIfNotShared(
+    MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThisIfNotShared(aMallocSizeOf);
+}
+
+JSObject* PeriodicWave::WrapObject(JSContext* aCx,
+                                   JS::Handle<JSObject*> aGivenProto) {
+  return PeriodicWave_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/PeriodicWave.h b/dom/media/webaudio/PeriodicWave.h
new file mode 100644
index 0000000000..18408789a9
--- /dev/null
+++ b/dom/media/webaudio/PeriodicWave.h
@@ -0,0 +1,59 @@
+/* -*- 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/. */
+
+#ifndef PeriodicWave_h_
+#define PeriodicWave_h_
+
+#include "nsWrapperCache.h"
+#include "nsCycleCollectionParticipant.h"
+#include "mozilla/Attributes.h"
+#include "AudioContext.h"
+#include "AudioNodeEngine.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct PeriodicWaveOptions;
+
+class PeriodicWave final : public nsWrapperCache {
+ public:
+  PeriodicWave(AudioContext* aContext, const float* aRealData,
+               const uint32_t aRealSize, const float* aImagData,
+               const uint32_t aImagSize, const bool aDisableNormalization,
+               ErrorResult& aRv);
+
+  NS_INLINE_DECL_CYCLE_COLLECTING_NATIVE_REFCOUNTING(PeriodicWave)
+  NS_DECL_CYCLE_COLLECTION_NATIVE_WRAPPERCACHE_CLASS(PeriodicWave)
+
+  static already_AddRefed<PeriodicWave> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const PeriodicWaveOptions& aOptions, ErrorResult& aRv);
+
+  AudioContext* GetParentObject() const { return mContext; }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  uint32_t DataLength() const { return mCoefficients.mDuration; }
+
+  bool DisableNormalization() const { return mDisableNormalization; }
+
+  const AudioChunk& GetThreadSharedBuffer() const { return mCoefficients; }
+
+  size_t SizeOfExcludingThisIfNotShared(MallocSizeOf aMallocSizeOf) const;
+  size_t SizeOfIncludingThisIfNotShared(MallocSizeOf aMallocSizeOf) const;
+
+ private:
+  ~PeriodicWave() = default;
+
+  AudioChunk mCoefficients;
+  RefPtr<AudioContext> mContext;
+  bool mDisableNormalization;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/PlayingRefChangeHandler.h b/dom/media/webaudio/PlayingRefChangeHandler.h
new file mode 100644
index 0000000000..cda01fdcd3
--- /dev/null
+++ b/dom/media/webaudio/PlayingRefChangeHandler.h
@@ -0,0 +1,42 @@
+/* -*- 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/. */
+
+#ifndef PlayingRefChangeHandler_h__
+#define PlayingRefChangeHandler_h__
+
+#include "nsThreadUtils.h"
+#include "AudioNodeTrack.h"
+
+namespace mozilla::dom {
+
+class PlayingRefChangeHandler final : public Runnable {
+ public:
+  enum ChangeType { ADDREF, RELEASE };
+  PlayingRefChangeHandler(AudioNodeTrack* aTrack, ChangeType aChange)
+      : Runnable("dom::PlayingRefChangeHandler"),
+        mTrack(aTrack),
+        mChange(aChange) {}
+
+  NS_IMETHOD Run() override {
+    RefPtr<AudioNode> node = mTrack->Engine()->NodeMainThread();
+    if (node) {
+      if (mChange == ADDREF) {
+        node->MarkActive();
+      } else if (mChange == RELEASE) {
+        node->MarkInactive();
+      }
+    }
+    return NS_OK;
+  }
+
+ private:
+  RefPtr<AudioNodeTrack> mTrack;
+  ChangeType mChange;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/ReportDecodeResultTask.h b/dom/media/webaudio/ReportDecodeResultTask.h
new file mode 100644
index 0000000000..49b7e39e04
--- /dev/null
+++ b/dom/media/webaudio/ReportDecodeResultTask.h
@@ -0,0 +1,37 @@
+/* -*- 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/. */
+
+#ifndef ReportDecodeResultTask_h_
+#define ReportDecodeResultTask_h_
+
+#include "mozilla/Attributes.h"
+#include "MediaBufferDecoder.h"
+
+namespace mozilla {
+
+class ReportDecodeResultTask final : public Runnable {
+ public:
+  ReportDecodeResultTask(DecodeJob& aDecodeJob, DecodeJob::ResultFn aFunction)
+      : mDecodeJob(aDecodeJob), mFunction(aFunction) {
+    MOZ_ASSERT(aFunction);
+  }
+
+  NS_IMETHOD Run() override {
+    MOZ_ASSERT(NS_IsMainThread());
+
+    (mDecodeJob.*mFunction)();
+
+    return NS_OK;
+  }
+
+ private:
+  DecodeJob& mDecodeJob;
+  DecodeJob::ResultFn mFunction;
+};
+
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/ScriptProcessorNode.cpp b/dom/media/webaudio/ScriptProcessorNode.cpp
new file mode 100644
index 0000000000..1e645a2b42
--- /dev/null
+++ b/dom/media/webaudio/ScriptProcessorNode.cpp
@@ -0,0 +1,549 @@
+/* -*- 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 "ScriptProcessorNode.h"
+#include "mozilla/dom/ScriptProcessorNodeBinding.h"
+#include "AudioBuffer.h"
+#include "AudioDestinationNode.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "AudioProcessingEvent.h"
+#include "WebAudioUtils.h"
+#include "mozilla/dom/ScriptSettings.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/PodOperations.h"
+#include <deque>
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+// The maximum latency, in seconds, that we can live with before dropping
+// buffers.
+static const float MAX_LATENCY_S = 0.5;
+
+// This class manages a queue of output buffers shared between
+// the main thread and the Media Track Graph thread.
+class SharedBuffers final {
+ private:
+  class OutputQueue final {
+   public:
+    explicit OutputQueue(const char* aName) : mMutex(aName) {}
+
+    size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const
+        MOZ_REQUIRES(mMutex) {
+      mMutex.AssertCurrentThreadOwns();
+
+      size_t amount = 0;
+      for (size_t i = 0; i < mBufferList.size(); i++) {
+        amount += mBufferList[i].SizeOfExcludingThis(aMallocSizeOf, false);
+      }
+
+      return amount;
+    }
+
+    Mutex& Lock() const MOZ_RETURN_CAPABILITY(mMutex) {
+      return const_cast<OutputQueue*>(this)->mMutex;
+    }
+
+    size_t ReadyToConsume() const MOZ_REQUIRES(mMutex) {
+      // Accessed on both main thread and media graph thread.
+      mMutex.AssertCurrentThreadOwns();
+      return mBufferList.size();
+    }
+
+    // Produce one buffer
+    AudioChunk& Produce() MOZ_REQUIRES(mMutex) {
+      mMutex.AssertCurrentThreadOwns();
+      MOZ_ASSERT(NS_IsMainThread());
+      mBufferList.push_back(AudioChunk());
+      return mBufferList.back();
+    }
+
+    // Consumes one buffer.
+    AudioChunk Consume() MOZ_REQUIRES(mMutex) {
+      mMutex.AssertCurrentThreadOwns();
+      MOZ_ASSERT(!NS_IsMainThread());
+      MOZ_ASSERT(ReadyToConsume() > 0);
+      AudioChunk front = mBufferList.front();
+      mBufferList.pop_front();
+      return front;
+    }
+
+    // Empties the buffer queue.
+    void Clear() MOZ_REQUIRES(mMutex) {
+      mMutex.AssertCurrentThreadOwns();
+      mBufferList.clear();
+    }
+
+   private:
+    typedef std::deque<AudioChunk> BufferList;
+
+    // Synchronizes access to mBufferList.  Note that it's the responsibility
+    // of the callers to perform the required locking, and we assert that every
+    // time we access mBufferList.
+    Mutex mMutex MOZ_UNANNOTATED;
+    // The list representing the queue.
+    BufferList mBufferList;
+  };
+
+ public:
+  explicit SharedBuffers(float aSampleRate)
+      : mOutputQueue("SharedBuffers::outputQueue"),
+        mDelaySoFar(TRACK_TIME_MAX),
+        mSampleRate(aSampleRate),
+        mLatency(0.0),
+        mDroppingBuffers(false) {}
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+    size_t amount = aMallocSizeOf(this);
+
+    {
+      MutexAutoLock lock(mOutputQueue.Lock());
+      amount += mOutputQueue.SizeOfExcludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  // main thread
+
+  // NotifyNodeIsConnected() may be called even when the state has not
+  // changed.
+  void NotifyNodeIsConnected(bool aIsConnected) {
+    MOZ_ASSERT(NS_IsMainThread());
+    if (!aIsConnected) {
+      // Reset main thread state for FinishProducingOutputBuffer().
+      mLatency = 0.0f;
+      mLastEventTime = TimeStamp();
+      mDroppingBuffers = false;
+      // Don't flush the output buffer here because the graph thread may be
+      // using it now.  The graph thread will flush when it knows it is
+      // disconnected.
+    }
+    mNodeIsConnected = aIsConnected;
+  }
+
+  void FinishProducingOutputBuffer(const AudioChunk& aBuffer) {
+    MOZ_ASSERT(NS_IsMainThread());
+
+    if (!mNodeIsConnected) {
+      // The output buffer is not used, and mLastEventTime will not be
+      // initialized until the node is re-connected.
+      return;
+    }
+
+    TimeStamp now = TimeStamp::Now();
+
+    if (mLastEventTime.IsNull()) {
+      mLastEventTime = now;
+    } else {
+      // When main thread blocking has built up enough so
+      // |mLatency > MAX_LATENCY_S|, frame dropping starts. It continues until
+      // the output buffer is completely empty, at which point the accumulated
+      // latency is also reset to 0.
+      // It could happen that the output queue becomes empty before the input
+      // node has fully caught up. In this case there will be events where
+      // |(now - mLastEventTime)| is very short, making mLatency negative.
+      // As this happens and the size of |mLatency| becomes greater than
+      // MAX_LATENCY_S, frame dropping starts again to maintain an as short
+      // output queue as possible.
+      float latency = (now - mLastEventTime).ToSeconds();
+      float bufferDuration = aBuffer.mDuration / mSampleRate;
+      mLatency += latency - bufferDuration;
+      mLastEventTime = now;
+      if (fabs(mLatency) > MAX_LATENCY_S) {
+        mDroppingBuffers = true;
+      }
+    }
+
+    MutexAutoLock lock(mOutputQueue.Lock());
+    if (mDroppingBuffers) {
+      if (mOutputQueue.ReadyToConsume()) {
+        return;
+      }
+      mDroppingBuffers = false;
+      mLatency = 0;
+    }
+
+    for (uint32_t offset = 0; offset < aBuffer.mDuration;
+         offset += WEBAUDIO_BLOCK_SIZE) {
+      AudioChunk& chunk = mOutputQueue.Produce();
+      chunk = aBuffer;
+      chunk.SliceTo(offset, offset + WEBAUDIO_BLOCK_SIZE);
+    }
+  }
+
+  // graph thread
+
+  AudioChunk GetOutputBuffer() {
+    MOZ_ASSERT(!NS_IsMainThread());
+    AudioChunk buffer;
+
+    {
+      MutexAutoLock lock(mOutputQueue.Lock());
+      if (mOutputQueue.ReadyToConsume() > 0) {
+        if (mDelaySoFar == TRACK_TIME_MAX) {
+          mDelaySoFar = 0;
+        }
+        buffer = mOutputQueue.Consume();
+      } else {
+        // If we're out of buffers to consume, just output silence
+        buffer.SetNull(WEBAUDIO_BLOCK_SIZE);
+        if (mDelaySoFar != TRACK_TIME_MAX) {
+          // Remember the delay that we just hit
+          mDelaySoFar += WEBAUDIO_BLOCK_SIZE;
+        }
+      }
+    }
+
+    return buffer;
+  }
+
+  TrackTime DelaySoFar() const {
+    MOZ_ASSERT(!NS_IsMainThread());
+    return mDelaySoFar == TRACK_TIME_MAX ? 0 : mDelaySoFar;
+  }
+
+  void Flush() {
+    MOZ_ASSERT(!NS_IsMainThread());
+    mDelaySoFar = TRACK_TIME_MAX;
+    {
+      MutexAutoLock lock(mOutputQueue.Lock());
+      mOutputQueue.Clear();
+    }
+  }
+
+ private:
+  OutputQueue mOutputQueue;
+  // How much delay we've seen so far.  This measures the amount of delay
+  // caused by the main thread lagging behind in producing output buffers.
+  // TRACK_TIME_MAX means that we have not received our first buffer yet.
+  // Graph thread only.
+  TrackTime mDelaySoFar;
+  // The samplerate of the context.
+  const float mSampleRate;
+  // The remaining members are main thread only.
+  // This is the latency caused by the buffering. If this grows too high, we
+  // will drop buffers until it is acceptable.
+  float mLatency;
+  // This is the time at which we last produced a buffer, to detect if the main
+  // thread has been blocked.
+  TimeStamp mLastEventTime;
+  // True if we should be dropping buffers.
+  bool mDroppingBuffers;
+  // True iff the AudioNode has at least one input or output connected.
+  bool mNodeIsConnected;
+};
+
+class ScriptProcessorNodeEngine final : public AudioNodeEngine {
+ public:
+  ScriptProcessorNodeEngine(ScriptProcessorNode* aNode,
+                            AudioDestinationNode* aDestination,
+                            uint32_t aBufferSize,
+                            uint32_t aNumberOfInputChannels)
+      : AudioNodeEngine(aNode),
+        mDestination(aDestination->Track()),
+        mSharedBuffers(new SharedBuffers(mDestination->mSampleRate)),
+        mBufferSize(aBufferSize),
+        mInputChannelCount(aNumberOfInputChannels),
+        mInputWriteIndex(0) {}
+
+  SharedBuffers* GetSharedBuffers() const { return mSharedBuffers.get(); }
+
+  enum {
+    IS_CONNECTED,
+  };
+
+  void SetInt32Parameter(uint32_t aIndex, int32_t aParam) override {
+    switch (aIndex) {
+      case IS_CONNECTED:
+        mIsConnected = aParam;
+        break;
+      default:
+        NS_ERROR("Bad Int32Parameter");
+    }  // End index switch.
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    TRACE("ScriptProcessorNodeEngine::ProcessBlock");
+
+    // This node is not connected to anything. Per spec, we don't fire the
+    // onaudioprocess event. We also want to clear out the input and output
+    // buffer queue, and output a null buffer.
+    if (!mIsConnected) {
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+      mSharedBuffers->Flush();
+      mInputWriteIndex = 0;
+      return;
+    }
+
+    // The input buffer is allocated lazily when non-null input is received.
+    if (!aInput.IsNull() && !mInputBuffer) {
+      mInputBuffer = ThreadSharedFloatArrayBufferList::Create(
+          mInputChannelCount, mBufferSize, fallible);
+      if (mInputBuffer && mInputWriteIndex) {
+        // Zero leading for null chunks that were skipped.
+        for (uint32_t i = 0; i < mInputChannelCount; ++i) {
+          float* channelData = mInputBuffer->GetDataForWrite(i);
+          PodZero(channelData, mInputWriteIndex);
+        }
+      }
+    }
+
+    // First, record our input buffer, if its allocation succeeded.
+    uint32_t inputChannelCount = mInputBuffer ? mInputBuffer->GetChannels() : 0;
+    for (uint32_t i = 0; i < inputChannelCount; ++i) {
+      float* writeData = mInputBuffer->GetDataForWrite(i) + mInputWriteIndex;
+      if (aInput.IsNull()) {
+        PodZero(writeData, aInput.GetDuration());
+      } else {
+        MOZ_ASSERT(aInput.GetDuration() == WEBAUDIO_BLOCK_SIZE, "sanity check");
+        MOZ_ASSERT(aInput.ChannelCount() == inputChannelCount);
+        AudioBlockCopyChannelWithScale(
+            static_cast<const float*>(aInput.mChannelData[i]), aInput.mVolume,
+            writeData);
+      }
+    }
+    mInputWriteIndex += aInput.GetDuration();
+
+    // Now, see if we have data to output
+    // Note that we need to do this before sending the buffer to the main
+    // thread so that our delay time is updated.
+    *aOutput = mSharedBuffers->GetOutputBuffer();
+
+    if (mInputWriteIndex >= mBufferSize) {
+      SendBuffersToMainThread(aTrack, aFrom);
+      mInputWriteIndex -= mBufferSize;
+    }
+  }
+
+  bool IsActive() const override {
+    // Could return false when !mIsConnected after all output chunks produced
+    // by main thread events calling
+    // SharedBuffers::FinishProducingOutputBuffer() have been processed.
+    return true;
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    // Not owned:
+    // - mDestination (probably)
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    amount += mSharedBuffers->SizeOfIncludingThis(aMallocSizeOf);
+    if (mInputBuffer) {
+      amount += mInputBuffer->SizeOfIncludingThis(aMallocSizeOf);
+    }
+
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  void SendBuffersToMainThread(AudioNodeTrack* aTrack, GraphTime aFrom) {
+    MOZ_ASSERT(!NS_IsMainThread());
+
+    // we now have a full input buffer ready to be sent to the main thread.
+    TrackTime playbackTick = mDestination->GraphTimeToTrackTime(aFrom);
+    // Add the duration of the current sample
+    playbackTick += WEBAUDIO_BLOCK_SIZE;
+    // Add the delay caused by the main thread
+    playbackTick += mSharedBuffers->DelaySoFar();
+    // Compute the playback time in the coordinate system of the destination
+    double playbackTime = mDestination->TrackTimeToSeconds(playbackTick);
+
+    class Command final : public Runnable {
+     public:
+      Command(AudioNodeTrack* aTrack,
+              already_AddRefed<ThreadSharedFloatArrayBufferList> aInputBuffer,
+              double aPlaybackTime)
+          : mozilla::Runnable("Command"),
+            mTrack(aTrack),
+            mInputBuffer(aInputBuffer),
+            mPlaybackTime(aPlaybackTime) {}
+
+      NS_IMETHOD Run() override {
+        auto engine = static_cast<ScriptProcessorNodeEngine*>(mTrack->Engine());
+        AudioChunk output;
+        output.SetNull(engine->mBufferSize);
+        {
+          auto node =
+              static_cast<ScriptProcessorNode*>(engine->NodeMainThread());
+          if (!node) {
+            return NS_OK;
+          }
+
+          if (node->HasListenersFor(nsGkAtoms::onaudioprocess)) {
+            DispatchAudioProcessEvent(node, &output);
+          }
+          // The node may have been destroyed during event dispatch.
+        }
+
+        // Append it to our output buffer queue
+        engine->GetSharedBuffers()->FinishProducingOutputBuffer(output);
+
+        return NS_OK;
+      }
+
+      // Sets up |output| iff buffers are set in event handlers.
+      void DispatchAudioProcessEvent(ScriptProcessorNode* aNode,
+                                     AudioChunk* aOutput) {
+        AudioContext* context = aNode->Context();
+        if (!context) {
+          return;
+        }
+
+        AutoJSAPI jsapi;
+        if (NS_WARN_IF(!jsapi.Init(aNode->GetOwner()))) {
+          return;
+        }
+        JSContext* cx = jsapi.cx();
+        uint32_t inputChannelCount = aNode->ChannelCount();
+
+        // Create the input buffer
+        RefPtr<AudioBuffer> inputBuffer;
+        if (mInputBuffer) {
+          ErrorResult rv;
+          inputBuffer = AudioBuffer::Create(
+              context->GetOwner(), inputChannelCount, aNode->BufferSize(),
+              context->SampleRate(), mInputBuffer.forget(), rv);
+          if (rv.Failed()) {
+            rv.SuppressException();
+            return;
+          }
+        }
+
+        // Ask content to produce data in the output buffer
+        // Note that we always avoid creating the output buffer here, and we try
+        // to avoid creating the input buffer as well.  The AudioProcessingEvent
+        // class knows how to lazily create them if needed once the script tries
+        // to access them.  Otherwise, we may be able to get away without
+        // creating them!
+        RefPtr<AudioProcessingEvent> event =
+            new AudioProcessingEvent(aNode, nullptr, nullptr);
+        event->InitEvent(inputBuffer, inputChannelCount, mPlaybackTime);
+        aNode->DispatchTrustedEvent(event);
+
+        // Steal the output buffers if they have been set.
+        // Don't create a buffer if it hasn't been used to return output;
+        // FinishProducingOutputBuffer() will optimize output = null.
+        // GetThreadSharedChannelsForRate() may also return null after OOM.
+        if (event->HasOutputBuffer()) {
+          ErrorResult rv;
+          AudioBuffer* buffer = event->GetOutputBuffer(rv);
+          // HasOutputBuffer() returning true means that GetOutputBuffer()
+          // will not fail.
+          MOZ_ASSERT(!rv.Failed());
+          *aOutput = buffer->GetThreadSharedChannelsForRate(cx);
+          MOZ_ASSERT(aOutput->IsNull() ||
+                         aOutput->mBufferFormat == AUDIO_FORMAT_FLOAT32,
+                     "AudioBuffers initialized from JS have float data");
+        }
+      }
+
+     private:
+      RefPtr<AudioNodeTrack> mTrack;
+      RefPtr<ThreadSharedFloatArrayBufferList> mInputBuffer;
+      double mPlaybackTime;
+    };
+
+    RefPtr<Command> command =
+        new Command(aTrack, mInputBuffer.forget(), playbackTime);
+    AbstractThread::MainThread()->Dispatch(command.forget());
+  }
+
+  friend class ScriptProcessorNode;
+
+  RefPtr<AudioNodeTrack> mDestination;
+  UniquePtr<SharedBuffers> mSharedBuffers;
+  RefPtr<ThreadSharedFloatArrayBufferList> mInputBuffer;
+  const uint32_t mBufferSize;
+  const uint32_t mInputChannelCount;
+  // The write index into the current input buffer
+  uint32_t mInputWriteIndex;
+  bool mIsConnected = false;
+};
+
+ScriptProcessorNode::ScriptProcessorNode(AudioContext* aContext,
+                                         uint32_t aBufferSize,
+                                         uint32_t aNumberOfInputChannels,
+                                         uint32_t aNumberOfOutputChannels)
+    : AudioNode(aContext, aNumberOfInputChannels,
+                mozilla::dom::ChannelCountMode::Explicit,
+                mozilla::dom::ChannelInterpretation::Speakers),
+      mBufferSize(aBufferSize ? aBufferSize
+                              :  // respect what the web developer requested
+                      4096)      // choose our own buffer size -- 4KB for now
+      ,
+      mNumberOfOutputChannels(aNumberOfOutputChannels) {
+  MOZ_ASSERT(BufferSize() % WEBAUDIO_BLOCK_SIZE == 0, "Invalid buffer size");
+  ScriptProcessorNodeEngine* engine = new ScriptProcessorNodeEngine(
+      this, aContext->Destination(), BufferSize(), aNumberOfInputChannels);
+  mTrack = AudioNodeTrack::Create(
+      aContext, engine, AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+}
+
+ScriptProcessorNode::~ScriptProcessorNode() = default;
+
+size_t ScriptProcessorNode::SizeOfExcludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t ScriptProcessorNode::SizeOfIncludingThis(
+    MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+void ScriptProcessorNode::EventListenerAdded(nsAtom* aType) {
+  AudioNode::EventListenerAdded(aType);
+  if (aType == nsGkAtoms::onaudioprocess) {
+    UpdateConnectedStatus();
+  }
+}
+
+void ScriptProcessorNode::EventListenerRemoved(nsAtom* aType) {
+  AudioNode::EventListenerRemoved(aType);
+  if (aType == nsGkAtoms::onaudioprocess && mTrack) {
+    UpdateConnectedStatus();
+  }
+}
+
+JSObject* ScriptProcessorNode::WrapObject(JSContext* aCx,
+                                          JS::Handle<JSObject*> aGivenProto) {
+  return ScriptProcessorNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void ScriptProcessorNode::UpdateConnectedStatus() {
+  bool isConnected =
+      mHasPhantomInput || !(OutputNodes().IsEmpty() &&
+                            OutputParams().IsEmpty() && InputNodes().IsEmpty());
+
+  // Events are queued even when there is no listener because a listener
+  // may be added while events are in the queue.
+  SendInt32ParameterToTrack(ScriptProcessorNodeEngine::IS_CONNECTED,
+                            isConnected);
+
+  if (isConnected && HasListenersFor(nsGkAtoms::onaudioprocess)) {
+    MarkActive();
+  } else {
+    MarkInactive();
+  }
+
+  // MarkInactive above might have released this node, check if it has a track.
+  if (!mTrack) {
+    return;
+  }
+
+  auto engine = static_cast<ScriptProcessorNodeEngine*>(mTrack->Engine());
+  engine->GetSharedBuffers()->NotifyNodeIsConnected(isConnected);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/ScriptProcessorNode.h b/dom/media/webaudio/ScriptProcessorNode.h
new file mode 100644
index 0000000000..c8f023cb17
--- /dev/null
+++ b/dom/media/webaudio/ScriptProcessorNode.h
@@ -0,0 +1,131 @@
+/* -*- 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/. */
+
+#ifndef ScriptProcessorNode_h_
+#define ScriptProcessorNode_h_
+
+#include "AudioNode.h"
+#include "mozilla/ErrorResult.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+class SharedBuffers;
+
+class ScriptProcessorNode final : public AudioNode {
+ public:
+  ScriptProcessorNode(AudioContext* aContext, uint32_t aBufferSize,
+                      uint32_t aNumberOfInputChannels,
+                      uint32_t aNumberOfOutputChannels);
+
+  NS_INLINE_DECL_REFCOUNTING_INHERITED(ScriptProcessorNode, AudioNode)
+
+  IMPL_EVENT_HANDLER(audioprocess)
+
+  using EventTarget::EventListenerAdded;
+  void EventListenerAdded(nsAtom* aType) override;
+
+  using EventTarget::EventListenerRemoved;
+  void EventListenerRemoved(nsAtom* aType) override;
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  AudioNode* Connect(AudioNode& aDestination, uint32_t aOutput, uint32_t aInput,
+                     ErrorResult& aRv) override {
+    AudioNode* node = AudioNode::Connect(aDestination, aOutput, aInput, aRv);
+    if (!aRv.Failed()) {
+      UpdateConnectedStatus();
+    }
+    return node;
+  }
+
+  void Connect(AudioParam& aDestination, uint32_t aOutput,
+               ErrorResult& aRv) override {
+    AudioNode::Connect(aDestination, aOutput, aRv);
+    if (!aRv.Failed()) {
+      UpdateConnectedStatus();
+    }
+  }
+  void Disconnect(ErrorResult& aRv) override {
+    AudioNode::Disconnect(aRv);
+    UpdateConnectedStatus();
+  }
+  void Disconnect(uint32_t aOutput, ErrorResult& aRv) override {
+    AudioNode::Disconnect(aOutput, aRv);
+    UpdateConnectedStatus();
+  }
+  void NotifyInputsChanged() override { UpdateConnectedStatus(); }
+  void NotifyHasPhantomInput() override {
+    mHasPhantomInput = true;
+    // No need to UpdateConnectedStatus() because there was previously an
+    // input in InputNodes().
+  }
+  void Disconnect(AudioNode& aDestination, ErrorResult& aRv) override {
+    AudioNode::Disconnect(aDestination, aRv);
+    UpdateConnectedStatus();
+  }
+  void Disconnect(AudioNode& aDestination, uint32_t aOutput,
+                  ErrorResult& aRv) override {
+    AudioNode::Disconnect(aDestination, aOutput, aRv);
+    UpdateConnectedStatus();
+  }
+  void Disconnect(AudioNode& aDestination, uint32_t aOutput, uint32_t aInput,
+                  ErrorResult& aRv) override {
+    AudioNode::Disconnect(aDestination, aOutput, aInput, aRv);
+    UpdateConnectedStatus();
+  }
+  void Disconnect(AudioParam& aDestination, ErrorResult& aRv) override {
+    AudioNode::Disconnect(aDestination, aRv);
+    UpdateConnectedStatus();
+  }
+  void Disconnect(AudioParam& aDestination, uint32_t aOutput,
+                  ErrorResult& aRv) override {
+    AudioNode::Disconnect(aDestination, aOutput, aRv);
+    UpdateConnectedStatus();
+  }
+  void SetChannelCount(uint32_t aChannelCount, ErrorResult& aRv) override {
+    if (aChannelCount != ChannelCount()) {
+      // Spec says to throw InvalidStateError, but see
+      // https://github.com/WebAudio/web-audio-api/issues/2153
+      aRv.ThrowNotSupportedError(
+          "Cannot change channel count of ScriptProcessorNode");
+    }
+  }
+  void SetChannelCountModeValue(ChannelCountMode aMode,
+                                ErrorResult& aRv) override {
+    if (aMode != ChannelCountMode::Explicit) {
+      // Spec says to throw InvalidStateError, but see
+      // https://github.com/WebAudio/web-audio-api/issues/2154
+      aRv.ThrowNotSupportedError(
+          "Cannot change channel count mode of ScriptProcessorNode");
+    }
+  }
+
+  uint32_t BufferSize() const { return mBufferSize; }
+
+  uint32_t NumberOfOutputChannels() const { return mNumberOfOutputChannels; }
+
+  using DOMEventTargetHelper::DispatchTrustedEvent;
+
+  const char* NodeType() const override { return "ScriptProcessorNode"; }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ private:
+  virtual ~ScriptProcessorNode();
+
+  void UpdateConnectedStatus();
+
+  const uint32_t mBufferSize;
+  const uint32_t mNumberOfOutputChannels;
+  bool mHasPhantomInput = false;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/StereoPannerNode.cpp b/dom/media/webaudio/StereoPannerNode.cpp
new file mode 100644
index 0000000000..758a92511a
--- /dev/null
+++ b/dom/media/webaudio/StereoPannerNode.cpp
@@ -0,0 +1,197 @@
+/* -*- 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 "StereoPannerNode.h"
+#include "mozilla/dom/StereoPannerNodeBinding.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "AudioDestinationNode.h"
+#include "AlignmentUtils.h"
+#include "WebAudioUtils.h"
+#include "PanningUtils.h"
+#include "AudioParamTimeline.h"
+#include "AudioParam.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_INHERITED(StereoPannerNode, AudioNode, mPan)
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(StereoPannerNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(StereoPannerNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(StereoPannerNode, AudioNode)
+
+class StereoPannerNodeEngine final : public AudioNodeEngine {
+ public:
+  StereoPannerNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination)
+      : AudioNodeEngine(aNode),
+        mDestination(aDestination->Track())
+        // Keep the default value in sync with the default value in
+        // StereoPannerNode::StereoPannerNode.
+        ,
+        mPan(0.f) {}
+
+  enum Parameters { PAN };
+  void RecvTimelineEvent(uint32_t aIndex, AudioParamEvent& aEvent) override {
+    MOZ_ASSERT(mDestination);
+    aEvent.ConvertToTicks(mDestination);
+
+    switch (aIndex) {
+      case PAN:
+        mPan.InsertEvent<int64_t>(aEvent);
+        break;
+      default:
+        NS_ERROR("Bad StereoPannerNode TimelineParameter");
+    }
+  }
+
+  void GetGainValuesForPanning(float aPanning, bool aMonoToStereo,
+                               float& aLeftGain, float& aRightGain) {
+    // Clamp and normalize the panning in [0; 1]
+    aPanning = std::min(std::max(aPanning, -1.f), 1.f);
+
+    if (aMonoToStereo) {
+      aPanning += 1;
+      aPanning /= 2;
+    } else if (aPanning <= 0) {
+      aPanning += 1;
+    }
+
+    aLeftGain = fdlibm_cos(0.5 * M_PI * aPanning);
+    aRightGain = fdlibm_sin(0.5 * M_PI * aPanning);
+  }
+
+  void SetToSilentStereoBlock(AudioBlock* aChunk) {
+    for (uint32_t channel = 0; channel < 2; channel++) {
+      float* samples = aChunk->ChannelFloatsForWrite(channel);
+      for (uint32_t i = 0; i < WEBAUDIO_BLOCK_SIZE; i++) {
+        samples[i] = 0.f;
+      }
+    }
+  }
+
+  void UpmixToStereoIfNeeded(const AudioBlock& aInput, AudioBlock* aOutput) {
+    if (aInput.ChannelCount() == 2) {
+      *aOutput = aInput;
+    } else {
+      MOZ_ASSERT(aInput.ChannelCount() == 1);
+      aOutput->SetBuffer(aInput.GetBuffer());
+      aOutput->mChannelData.SetLength(2);
+      for (uint32_t i = 0; i < 2; ++i) {
+        aOutput->mChannelData[i] = aInput.ChannelData<float>()[0];
+      }
+      // 1/sqrt(2) multiplier is because StereoPanner up-mixing differs from
+      // input up-mixing.
+      aOutput->mVolume = M_SQRT1_2 * aInput.mVolume;
+      aOutput->mBufferFormat = AUDIO_FORMAT_FLOAT32;
+    }
+  }
+
+  virtual void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                            const AudioBlock& aInput, AudioBlock* aOutput,
+                            bool* aFinished) override {
+    // The output of this node is always stereo, no matter what the inputs are.
+    MOZ_ASSERT(aInput.ChannelCount() <= 2);
+    TRACE("StereoPannerNodeEngine::ProcessBlock");
+
+    bool monoToStereo = aInput.ChannelCount() == 1;
+
+    if (aInput.IsNull()) {
+      // If input is silent, so is the output
+      aOutput->SetNull(WEBAUDIO_BLOCK_SIZE);
+    } else if (mPan.HasSimpleValue()) {
+      float panning = mPan.GetValue();
+      // If the panning is 0.0, we can simply copy the input to the
+      // output with gain applied, up-mixing to stereo if needed.
+      if (panning == 0.0f) {
+        UpmixToStereoIfNeeded(aInput, aOutput);
+      } else {
+        // Optimize the case where the panning is constant for this processing
+        // block: we can just apply a constant gain on the left and right
+        // channel
+        float gainL, gainR;
+
+        GetGainValuesForPanning(panning, monoToStereo, gainL, gainR);
+        ApplyStereoPanning(aInput, aOutput, gainL, gainR, panning <= 0);
+      }
+    } else {
+      float computedGain[2 * WEBAUDIO_BLOCK_SIZE + 4];
+      bool onLeft[WEBAUDIO_BLOCK_SIZE];
+
+      float values[WEBAUDIO_BLOCK_SIZE];
+      TrackTime tick = mDestination->GraphTimeToTrackTime(aFrom);
+      mPan.GetValuesAtTime(tick, values, WEBAUDIO_BLOCK_SIZE);
+
+      float* alignedComputedGain = ALIGNED16(computedGain);
+      ASSERT_ALIGNED16(alignedComputedGain);
+      for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
+        float left, right;
+        GetGainValuesForPanning(values[counter], monoToStereo, left, right);
+
+        alignedComputedGain[counter] = left;
+        alignedComputedGain[WEBAUDIO_BLOCK_SIZE + counter] = right;
+        onLeft[counter] = values[counter] <= 0;
+      }
+
+      // Apply the gain to the output buffer
+      ApplyStereoPanning(aInput, aOutput, alignedComputedGain,
+                         &alignedComputedGain[WEBAUDIO_BLOCK_SIZE], onLeft);
+    }
+  }
+
+  virtual size_t SizeOfIncludingThis(
+      MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  RefPtr<AudioNodeTrack> mDestination;
+  AudioParamTimeline mPan;
+};
+
+StereoPannerNode::StereoPannerNode(AudioContext* aContext)
+    : AudioNode(aContext, 2, ChannelCountMode::Clamped_max,
+                ChannelInterpretation::Speakers) {
+  mPan =
+      CreateAudioParam(StereoPannerNodeEngine::PAN, u"pan"_ns, 0.f, -1.f, 1.f);
+  StereoPannerNodeEngine* engine =
+      new StereoPannerNodeEngine(this, aContext->Destination());
+  mTrack = AudioNodeTrack::Create(
+      aContext, engine, AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+}
+
+/* static */
+already_AddRefed<StereoPannerNode> StereoPannerNode::Create(
+    AudioContext& aAudioContext, const StereoPannerOptions& aOptions,
+    ErrorResult& aRv) {
+  RefPtr<StereoPannerNode> audioNode = new StereoPannerNode(&aAudioContext);
+
+  audioNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  audioNode->Pan()->SetInitialValue(aOptions.mPan);
+  return audioNode.forget();
+}
+
+size_t StereoPannerNode::SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+  size_t amount = AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  amount += mPan->SizeOfIncludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t StereoPannerNode::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+}
+
+JSObject* StereoPannerNode::WrapObject(JSContext* aCx,
+                                       JS::Handle<JSObject*> aGivenProto) {
+  return StereoPannerNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/StereoPannerNode.h b/dom/media/webaudio/StereoPannerNode.h
new file mode 100644
index 0000000000..9dc7370fd1
--- /dev/null
+++ b/dom/media/webaudio/StereoPannerNode.h
@@ -0,0 +1,74 @@
+/* -*- 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/. */
+
+#ifndef StereoPannerNode_h_
+#define StereoPannerNode_h_
+
+#include "AudioNode.h"
+#include "nsPrintfCString.h"
+#include "mozilla/RefCounted.h"
+#include "mozilla/dom/StereoPannerNodeBinding.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct StereoPannerOptions;
+
+class StereoPannerNode final : public AudioNode {
+ public:
+  static already_AddRefed<StereoPannerNode> Create(
+      AudioContext& aAudioContext, const StereoPannerOptions& aOptions,
+      ErrorResult& aRv);
+
+  MOZ_DECLARE_REFCOUNTED_TYPENAME(StereoPannerNode)
+
+  static already_AddRefed<StereoPannerNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const StereoPannerOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  virtual JSObject* WrapObject(JSContext* aCx,
+                               JS::Handle<JSObject*> aGivenProto) override;
+
+  virtual void SetChannelCount(uint32_t aChannelCount,
+                               ErrorResult& aRv) override {
+    if (aChannelCount > 2) {
+      aRv.ThrowNotSupportedError(
+          nsPrintfCString("%u is greater than 2", aChannelCount));
+      return;
+    }
+    AudioNode::SetChannelCount(aChannelCount, aRv);
+  }
+  virtual void SetChannelCountModeValue(ChannelCountMode aMode,
+                                        ErrorResult& aRv) override {
+    if (aMode == ChannelCountMode::Max) {
+      aRv.ThrowNotSupportedError("Cannot set channel count mode to \"max\"");
+      return;
+    }
+    AudioNode::SetChannelCountModeValue(aMode, aRv);
+  }
+
+  AudioParam* Pan() const { return mPan; }
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_CLASS_INHERITED(StereoPannerNode, AudioNode)
+
+  virtual const char* NodeType() const override { return "StereoPannerNode"; }
+
+  virtual size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override;
+  virtual size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override;
+
+ private:
+  explicit StereoPannerNode(AudioContext* aContext);
+  ~StereoPannerNode() = default;
+
+  RefPtr<AudioParam> mPan;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/ThreeDPoint.cpp b/dom/media/webaudio/ThreeDPoint.cpp
new file mode 100644
index 0000000000..d54d03e24c
--- /dev/null
+++ b/dom/media/webaudio/ThreeDPoint.cpp
@@ -0,0 +1,38 @@
+/* -*- 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/. */
+
+/**
+ * Other similar methods can be added if needed.
+ */
+
+#include "ThreeDPoint.h"
+#include "WebAudioUtils.h"
+
+namespace mozilla::dom {
+
+bool ThreeDPoint::FuzzyEqual(const ThreeDPoint& other) {
+  return WebAudioUtils::FuzzyEqual(x, other.x) &&
+         WebAudioUtils::FuzzyEqual(y, other.y) &&
+         WebAudioUtils::FuzzyEqual(z, other.z);
+}
+
+ThreeDPoint operator-(const ThreeDPoint& lhs, const ThreeDPoint& rhs) {
+  return ThreeDPoint(lhs.x - rhs.x, lhs.y - rhs.y, lhs.z - rhs.z);
+}
+
+ThreeDPoint operator*(const ThreeDPoint& lhs, const ThreeDPoint& rhs) {
+  return ThreeDPoint(lhs.x * rhs.x, lhs.y * rhs.y, lhs.z * rhs.z);
+}
+
+ThreeDPoint operator*(const ThreeDPoint& lhs, const double rhs) {
+  return ThreeDPoint(lhs.x * rhs, lhs.y * rhs, lhs.z * rhs);
+}
+
+bool operator==(const ThreeDPoint& lhs, const ThreeDPoint& rhs) {
+  return lhs.x == rhs.x && lhs.y == rhs.y && lhs.z == rhs.z;
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/ThreeDPoint.h b/dom/media/webaudio/ThreeDPoint.h
new file mode 100644
index 0000000000..4e2c7358f8
--- /dev/null
+++ b/dom/media/webaudio/ThreeDPoint.h
@@ -0,0 +1,68 @@
+/* -*- 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/. */
+
+#ifndef ThreeDPoint_h_
+#define ThreeDPoint_h_
+
+#include <cmath>
+#include <algorithm>
+
+namespace mozilla::dom {
+
+struct ThreeDPoint final {
+  ThreeDPoint() : x(0.), y(0.), z(0.) {}
+  ThreeDPoint(double aX, double aY, double aZ) : x(aX), y(aY), z(aZ) {}
+
+  double Magnitude() const { return sqrt(x * x + y * y + z * z); }
+
+  void Normalize() {
+    // Zero vectors cannot be normalized. For our purpose, normalizing a zero
+    // vector results in a zero vector.
+    if (IsZero()) {
+      return;
+    }
+    // Normalize with the maximum norm first to avoid overflow and underflow.
+    double invMax = 1 / MaxNorm();
+    x *= invMax;
+    y *= invMax;
+    z *= invMax;
+
+    double invDistance = 1 / Magnitude();
+    x *= invDistance;
+    y *= invDistance;
+    z *= invDistance;
+  }
+
+  ThreeDPoint CrossProduct(const ThreeDPoint& rhs) const {
+    return ThreeDPoint(y * rhs.z - z * rhs.y, z * rhs.x - x * rhs.z,
+                       x * rhs.y - y * rhs.x);
+  }
+
+  double DotProduct(const ThreeDPoint& rhs) {
+    return x * rhs.x + y * rhs.y + z * rhs.z;
+  }
+
+  bool IsZero() const { return x == 0 && y == 0 && z == 0; }
+
+  // For comparing two vectors of close to unit magnitude.
+  bool FuzzyEqual(const ThreeDPoint& other);
+
+  double x, y, z;
+
+ private:
+  double MaxNorm() const {
+    return std::max(fabs(x), std::max(fabs(y), fabs(z)));
+  }
+};
+
+ThreeDPoint operator-(const ThreeDPoint& lhs, const ThreeDPoint& rhs);
+ThreeDPoint operator*(const ThreeDPoint& lhs, const ThreeDPoint& rhs);
+ThreeDPoint operator*(const ThreeDPoint& lhs, const double rhs);
+bool operator==(const ThreeDPoint& lhs, const ThreeDPoint& rhs);
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/WaveShaperNode.cpp b/dom/media/webaudio/WaveShaperNode.cpp
new file mode 100644
index 0000000000..f01b613039
--- /dev/null
+++ b/dom/media/webaudio/WaveShaperNode.cpp
@@ -0,0 +1,389 @@
+/* -*- 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 "WaveShaperNode.h"
+#include "mozilla/dom/WaveShaperNodeBinding.h"
+#include "AlignmentUtils.h"
+#include "AudioNode.h"
+#include "AudioNodeEngine.h"
+#include "AudioNodeTrack.h"
+#include "mozilla/PodOperations.h"
+#include "Tracing.h"
+
+namespace mozilla::dom {
+
+NS_IMPL_CYCLE_COLLECTION_CLASS(WaveShaperNode)
+
+NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(WaveShaperNode, AudioNode)
+  NS_IMPL_CYCLE_COLLECTION_UNLINK_PRESERVED_WRAPPER
+NS_IMPL_CYCLE_COLLECTION_UNLINK_END
+
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(WaveShaperNode, AudioNode)
+NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
+
+NS_IMPL_CYCLE_COLLECTION_TRACE_BEGIN(WaveShaperNode)
+  NS_IMPL_CYCLE_COLLECTION_TRACE_PRESERVED_WRAPPER
+NS_IMPL_CYCLE_COLLECTION_TRACE_END
+
+NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(WaveShaperNode)
+NS_INTERFACE_MAP_END_INHERITING(AudioNode)
+
+NS_IMPL_ADDREF_INHERITED(WaveShaperNode, AudioNode)
+NS_IMPL_RELEASE_INHERITED(WaveShaperNode, AudioNode)
+
+static uint32_t ValueOf(OverSampleType aType) {
+  switch (aType) {
+    case OverSampleType::None:
+      return 1;
+    case OverSampleType::_2x:
+      return 2;
+    case OverSampleType::_4x:
+      return 4;
+    default:
+      MOZ_ASSERT_UNREACHABLE("We should never reach here");
+      return 1;
+  }
+}
+
+class Resampler final {
+ public:
+  Resampler()
+      : mType(OverSampleType::None),
+        mUpSampler(nullptr),
+        mDownSampler(nullptr),
+        mChannels(0),
+        mSampleRate(0) {}
+
+  ~Resampler() { Destroy(); }
+
+  void Reset(uint32_t aChannels, TrackRate aSampleRate, OverSampleType aType) {
+    if (aChannels == mChannels && aSampleRate == mSampleRate &&
+        aType == mType) {
+      return;
+    }
+
+    mChannels = aChannels;
+    mSampleRate = aSampleRate;
+    mType = aType;
+
+    Destroy();
+
+    if (aType == OverSampleType::None) {
+      mBuffer.Clear();
+      return;
+    }
+
+    mUpSampler = speex_resampler_init(aChannels, aSampleRate,
+                                      aSampleRate * ValueOf(aType),
+                                      SPEEX_RESAMPLER_QUALITY_MIN, nullptr);
+    mDownSampler =
+        speex_resampler_init(aChannels, aSampleRate * ValueOf(aType),
+                             aSampleRate, SPEEX_RESAMPLER_QUALITY_MIN, nullptr);
+    mBuffer.SetLength(WEBAUDIO_BLOCK_SIZE * ValueOf(aType));
+  }
+
+  float* UpSample(uint32_t aChannel, const float* aInputData,
+                  uint32_t aBlocks) {
+    uint32_t inSamples = WEBAUDIO_BLOCK_SIZE;
+    uint32_t outSamples = WEBAUDIO_BLOCK_SIZE * aBlocks;
+    float* outputData = mBuffer.Elements();
+
+    MOZ_ASSERT(mBuffer.Length() == outSamples);
+
+    WebAudioUtils::SpeexResamplerProcess(mUpSampler, aChannel, aInputData,
+                                         &inSamples, outputData, &outSamples);
+
+    MOZ_ASSERT(inSamples == WEBAUDIO_BLOCK_SIZE &&
+               outSamples == WEBAUDIO_BLOCK_SIZE * aBlocks);
+
+    return outputData;
+  }
+
+  void DownSample(uint32_t aChannel, float* aOutputData, uint32_t aBlocks) {
+    uint32_t inSamples = WEBAUDIO_BLOCK_SIZE * aBlocks;
+    uint32_t outSamples = WEBAUDIO_BLOCK_SIZE;
+    const float* inputData = mBuffer.Elements();
+
+    MOZ_ASSERT(mBuffer.Length() == inSamples);
+
+    WebAudioUtils::SpeexResamplerProcess(mDownSampler, aChannel, inputData,
+                                         &inSamples, aOutputData, &outSamples);
+
+    MOZ_ASSERT(inSamples == WEBAUDIO_BLOCK_SIZE * aBlocks &&
+               outSamples == WEBAUDIO_BLOCK_SIZE);
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const {
+    size_t amount = 0;
+    // Future: properly measure speex memory
+    amount += aMallocSizeOf(mUpSampler);
+    amount += aMallocSizeOf(mDownSampler);
+    amount += mBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+ private:
+  void Destroy() {
+    if (mUpSampler) {
+      speex_resampler_destroy(mUpSampler);
+      mUpSampler = nullptr;
+    }
+    if (mDownSampler) {
+      speex_resampler_destroy(mDownSampler);
+      mDownSampler = nullptr;
+    }
+  }
+
+ private:
+  OverSampleType mType;
+  SpeexResamplerState* mUpSampler;
+  SpeexResamplerState* mDownSampler;
+  uint32_t mChannels;
+  TrackRate mSampleRate;
+  nsTArray<float> mBuffer;
+};
+
+class WaveShaperNodeEngine final : public AudioNodeEngine {
+ public:
+  explicit WaveShaperNodeEngine(AudioNode* aNode)
+      : AudioNodeEngine(aNode), mType(OverSampleType::None) {}
+
+  enum Parameters { TYPE };
+
+  void SetRawArrayData(nsTArray<float>&& aCurve) override {
+    mCurve = std::move(aCurve);
+  }
+
+  void SetInt32Parameter(uint32_t aIndex, int32_t aValue) override {
+    switch (aIndex) {
+      case TYPE:
+        mType = static_cast<OverSampleType>(aValue);
+        break;
+      default:
+        NS_ERROR("Bad WaveShaperNode Int32Parameter");
+    }
+  }
+
+  template <uint32_t blocks>
+  void ProcessCurve(const float* aInputBuffer, float* aOutputBuffer) {
+    for (uint32_t j = 0; j < WEBAUDIO_BLOCK_SIZE * blocks; ++j) {
+      // Index into the curve array based on the amplitude of the
+      // incoming signal by using an amplitude range of [-1, 1] and
+      // performing a linear interpolation of the neighbor values.
+      float index = (mCurve.Length() - 1) * (aInputBuffer[j] + 1.0f) / 2.0f;
+      if (index < 0.0f) {
+        aOutputBuffer[j] = mCurve[0];
+      } else {
+        int32_t indexLower = index;
+        if (static_cast<uint32_t>(indexLower) >= mCurve.Length() - 1) {
+          aOutputBuffer[j] = mCurve[mCurve.Length() - 1];
+        } else {
+          uint32_t indexHigher = indexLower + 1;
+          float interpolationFactor = index - indexLower;
+          aOutputBuffer[j] = (1.0f - interpolationFactor) * mCurve[indexLower] +
+                             interpolationFactor * mCurve[indexHigher];
+        }
+      }
+    }
+  }
+
+  void ProcessBlock(AudioNodeTrack* aTrack, GraphTime aFrom,
+                    const AudioBlock& aInput, AudioBlock* aOutput,
+                    bool* aFinished) override {
+    TRACE("WaveShaperNodeEngine::ProcessBlock");
+
+    uint32_t channelCount = aInput.ChannelCount();
+    if (!mCurve.Length()) {
+      // Optimize the case where we don't have a curve buffer
+      *aOutput = aInput;
+      return;
+    }
+
+    // If the input is null, check to see if non-null output will be produced
+    bool nullInput = false;
+    if (channelCount == 0) {
+      float index = (mCurve.Length() - 1) * 0.5;
+      uint32_t indexLower = index;
+      uint32_t indexHigher = indexLower + 1;
+      float interpolationFactor = index - indexLower;
+      if ((1.0f - interpolationFactor) * mCurve[indexLower] +
+              interpolationFactor * mCurve[indexHigher] ==
+          0.0) {
+        *aOutput = aInput;
+        return;
+      }
+      nullInput = true;
+      channelCount = 1;
+    }
+
+    aOutput->AllocateChannels(channelCount);
+    for (uint32_t i = 0; i < channelCount; ++i) {
+      const float* inputSamples;
+      float scaledInput[WEBAUDIO_BLOCK_SIZE + 4];
+      float* alignedScaledInput = ALIGNED16(scaledInput);
+      ASSERT_ALIGNED16(alignedScaledInput);
+      if (!nullInput) {
+        if (aInput.mVolume != 1.0f) {
+          AudioBlockCopyChannelWithScale(
+              static_cast<const float*>(aInput.mChannelData[i]), aInput.mVolume,
+              alignedScaledInput);
+          inputSamples = alignedScaledInput;
+        } else {
+          inputSamples = static_cast<const float*>(aInput.mChannelData[i]);
+        }
+      } else {
+        PodZero(alignedScaledInput, WEBAUDIO_BLOCK_SIZE);
+        inputSamples = alignedScaledInput;
+      }
+      float* outputBuffer = aOutput->ChannelFloatsForWrite(i);
+      float* sampleBuffer;
+
+      switch (mType) {
+        case OverSampleType::None:
+          mResampler.Reset(channelCount, aTrack->mSampleRate,
+                           OverSampleType::None);
+          ProcessCurve<1>(inputSamples, outputBuffer);
+          break;
+        case OverSampleType::_2x:
+          mResampler.Reset(channelCount, aTrack->mSampleRate,
+                           OverSampleType::_2x);
+          sampleBuffer = mResampler.UpSample(i, inputSamples, 2);
+          ProcessCurve<2>(sampleBuffer, sampleBuffer);
+          mResampler.DownSample(i, outputBuffer, 2);
+          break;
+        case OverSampleType::_4x:
+          mResampler.Reset(channelCount, aTrack->mSampleRate,
+                           OverSampleType::_4x);
+          sampleBuffer = mResampler.UpSample(i, inputSamples, 4);
+          ProcessCurve<4>(sampleBuffer, sampleBuffer);
+          mResampler.DownSample(i, outputBuffer, 4);
+          break;
+        default:
+          MOZ_ASSERT_UNREACHABLE("We should never reach here");
+      }
+    }
+  }
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    size_t amount = AudioNodeEngine::SizeOfExcludingThis(aMallocSizeOf);
+    amount += mCurve.ShallowSizeOfExcludingThis(aMallocSizeOf);
+    amount += mResampler.SizeOfExcludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  nsTArray<float> mCurve;
+  OverSampleType mType;
+  Resampler mResampler;
+};
+
+WaveShaperNode::WaveShaperNode(AudioContext* aContext)
+    : AudioNode(aContext, 2, ChannelCountMode::Max,
+                ChannelInterpretation::Speakers),
+      mType(OverSampleType::None) {
+  WaveShaperNodeEngine* engine = new WaveShaperNodeEngine(this);
+  mTrack = AudioNodeTrack::Create(
+      aContext, engine, AudioNodeTrack::NO_TRACK_FLAGS, aContext->Graph());
+}
+
+/* static */
+already_AddRefed<WaveShaperNode> WaveShaperNode::Create(
+    AudioContext& aAudioContext, const WaveShaperOptions& aOptions,
+    ErrorResult& aRv) {
+  RefPtr<WaveShaperNode> audioNode = new WaveShaperNode(&aAudioContext);
+
+  audioNode->Initialize(aOptions, aRv);
+  if (NS_WARN_IF(aRv.Failed())) {
+    return nullptr;
+  }
+
+  if (aOptions.mCurve.WasPassed()) {
+    audioNode->SetCurveInternal(aOptions.mCurve.Value(), aRv);
+    if (NS_WARN_IF(aRv.Failed())) {
+      return nullptr;
+    }
+  }
+
+  audioNode->SetOversample(aOptions.mOversample);
+  return audioNode.forget();
+}
+
+JSObject* WaveShaperNode::WrapObject(JSContext* aCx,
+                                     JS::Handle<JSObject*> aGivenProto) {
+  return WaveShaperNode_Binding::Wrap(aCx, this, aGivenProto);
+}
+
+void WaveShaperNode::SetCurve(const Nullable<Float32Array>& aCurve,
+                              ErrorResult& aRv) {
+  // Let's purge the cached value for the curve attribute.
+  WaveShaperNode_Binding::ClearCachedCurveValue(this);
+
+  if (aCurve.IsNull()) {
+    CleanCurveInternal();
+    return;
+  }
+
+  nsTArray<float> curve;
+  if (!aCurve.Value().AppendDataTo(curve)) {
+    aRv.Throw(NS_ERROR_OUT_OF_MEMORY);
+    return;
+  }
+
+  SetCurveInternal(curve, aRv);
+}
+
+void WaveShaperNode::SetCurveInternal(const nsTArray<float>& aCurve,
+                                      ErrorResult& aRv) {
+  if (aCurve.Length() < 2) {
+    aRv.ThrowInvalidStateError("Must have at least two entries");
+    return;
+  }
+
+  mCurve = aCurve.Clone();
+  SendCurveToTrack();
+}
+
+void WaveShaperNode::CleanCurveInternal() {
+  mCurve.Clear();
+  SendCurveToTrack();
+}
+
+void WaveShaperNode::SendCurveToTrack() {
+  AudioNodeTrack* ns = mTrack;
+  MOZ_ASSERT(ns, "Why don't we have a track here?");
+
+  nsTArray<float> copyCurve(mCurve.Clone());
+  ns->SetRawArrayData(std::move(copyCurve));
+}
+
+void WaveShaperNode::GetCurve(JSContext* aCx,
+                              JS::MutableHandle<JSObject*> aRetval,
+                              ErrorResult& aError) {
+  // Let's return a null value if the list is empty.
+  if (mCurve.IsEmpty()) {
+    aRetval.set(nullptr);
+    return;
+  }
+
+  MOZ_ASSERT(mCurve.Length() >= 2);
+  JSObject* curve = Float32Array::Create(aCx, this, mCurve, aError);
+  if (aError.Failed()) {
+    return;
+  }
+  aRetval.set(curve);
+}
+
+void WaveShaperNode::SetOversample(OverSampleType aType) {
+  mType = aType;
+  SendInt32ParameterToTrack(WaveShaperNodeEngine::TYPE,
+                            static_cast<int32_t>(aType));
+}
+
+}  // namespace mozilla::dom
diff --git a/dom/media/webaudio/WaveShaperNode.h b/dom/media/webaudio/WaveShaperNode.h
new file mode 100644
index 0000000000..5c91960af9
--- /dev/null
+++ b/dom/media/webaudio/WaveShaperNode.h
@@ -0,0 +1,72 @@
+/* -*- 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/. */
+
+#ifndef WaveShaperNode_h_
+#define WaveShaperNode_h_
+
+#include "AudioNode.h"
+#include "mozilla/dom/WaveShaperNodeBinding.h"
+#include "mozilla/dom/TypedArray.h"
+
+namespace mozilla::dom {
+
+class AudioContext;
+struct WaveShaperOptions;
+
+class WaveShaperNode final : public AudioNode {
+ public:
+  static already_AddRefed<WaveShaperNode> Create(
+      AudioContext& aAudioContext, const WaveShaperOptions& aOptions,
+      ErrorResult& aRv);
+
+  NS_DECL_ISUPPORTS_INHERITED
+  NS_DECL_CYCLE_COLLECTION_SCRIPT_HOLDER_CLASS_INHERITED(WaveShaperNode,
+                                                         AudioNode)
+
+  static already_AddRefed<WaveShaperNode> Constructor(
+      const GlobalObject& aGlobal, AudioContext& aAudioContext,
+      const WaveShaperOptions& aOptions, ErrorResult& aRv) {
+    return Create(aAudioContext, aOptions, aRv);
+  }
+
+  JSObject* WrapObject(JSContext* aCx,
+                       JS::Handle<JSObject*> aGivenProto) override;
+
+  void GetCurve(JSContext* aCx, JS::MutableHandle<JSObject*> aRetval,
+                ErrorResult& aRv);
+  void SetCurve(const Nullable<Float32Array>& aData, ErrorResult& aRv);
+
+  OverSampleType Oversample() const { return mType; }
+  void SetOversample(OverSampleType aType);
+
+  size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const override {
+    // Possibly track in the future:
+    // - mCurve
+    return AudioNode::SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+  const char* NodeType() const override { return "WaveShaperNode"; }
+
+  size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const override {
+    return aMallocSizeOf(this) + SizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  explicit WaveShaperNode(AudioContext* aContext);
+  ~WaveShaperNode() = default;
+
+  void SetCurveInternal(const nsTArray<float>& aCurve, ErrorResult& aRv);
+  void CleanCurveInternal();
+
+  void SendCurveToTrack();
+
+  nsTArray<float> mCurve;
+  OverSampleType mType;
+};
+
+}  // namespace mozilla::dom
+
+#endif
diff --git a/dom/media/webaudio/WebAudioUtils.cpp b/dom/media/webaudio/WebAudioUtils.cpp
new file mode 100644
index 0000000000..a1338650e0
--- /dev/null
+++ b/dom/media/webaudio/WebAudioUtils.cpp
@@ -0,0 +1,114 @@
+/* -*- 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 "WebAudioUtils.h"
+#include "blink/HRTFDatabaseLoader.h"
+
+#include "nsComponentManagerUtils.h"
+#include "nsContentUtils.h"
+#include "nsIConsoleService.h"
+#include "nsIScriptError.h"
+#include "nsJSUtils.h"
+#include "nsServiceManagerUtils.h"
+
+#include "mozilla/SchedulerGroup.h"
+
+namespace mozilla {
+
+LazyLogModule gWebAudioAPILog("WebAudioAPI");
+
+namespace dom {
+
+void WebAudioUtils::Shutdown() { WebCore::HRTFDatabaseLoader::shutdown(); }
+
+int WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
+                                         uint32_t aChannel, const float* aIn,
+                                         uint32_t* aInLen, float* aOut,
+                                         uint32_t* aOutLen) {
+  return speex_resampler_process_float(aResampler, aChannel, aIn, aInLen, aOut,
+                                       aOutLen);
+}
+
+int WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
+                                         uint32_t aChannel, const int16_t* aIn,
+                                         uint32_t* aInLen, float* aOut,
+                                         uint32_t* aOutLen) {
+  AutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE * 4> tmp;
+  tmp.SetLength(*aInLen);
+  ConvertAudioSamples(aIn, tmp.Elements(), *aInLen);
+  int result = speex_resampler_process_float(
+      aResampler, aChannel, tmp.Elements(), aInLen, aOut, aOutLen);
+  return result;
+}
+
+int WebAudioUtils::SpeexResamplerProcess(SpeexResamplerState* aResampler,
+                                         uint32_t aChannel, const int16_t* aIn,
+                                         uint32_t* aInLen, int16_t* aOut,
+                                         uint32_t* aOutLen) {
+  AutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE * 4> tmp1;
+  AutoTArray<AudioDataValue, WEBAUDIO_BLOCK_SIZE * 4> tmp2;
+  tmp1.SetLength(*aInLen);
+  tmp2.SetLength(*aOutLen);
+  ConvertAudioSamples(aIn, tmp1.Elements(), *aInLen);
+  int result = speex_resampler_process_float(
+      aResampler, aChannel, tmp1.Elements(), aInLen, tmp2.Elements(), aOutLen);
+  ConvertAudioSamples(tmp2.Elements(), aOut, *aOutLen);
+  return result;
+}
+
+void WebAudioUtils::LogToDeveloperConsole(uint64_t aWindowID,
+                                          const char* aKey) {
+  // This implementation is derived from dom/media/VideoUtils.cpp, but we
+  // use a windowID so that the message is delivered to the developer console.
+  // It is similar to ContentUtils::ReportToConsole, but also works off main
+  // thread.
+  if (!NS_IsMainThread()) {
+    nsCOMPtr<nsIRunnable> task = NS_NewRunnableFunction(
+        "dom::WebAudioUtils::LogToDeveloperConsole",
+        [aWindowID, aKey] { LogToDeveloperConsole(aWindowID, aKey); });
+    SchedulerGroup::Dispatch(task.forget());
+    return;
+  }
+
+  nsCOMPtr<nsIConsoleService> console(
+      do_GetService("@mozilla.org/consoleservice;1"));
+  if (!console) {
+    NS_WARNING("Failed to log message to console.");
+    return;
+  }
+
+  nsAutoString spec;
+  uint32_t aLineNumber = 0, aColumnNumber = 1;
+  JSContext* cx = nsContentUtils::GetCurrentJSContext();
+  if (cx) {
+    nsJSUtils::GetCallingLocation(cx, spec, &aLineNumber, &aColumnNumber);
+  }
+
+  nsresult rv;
+  nsCOMPtr<nsIScriptError> errorObject =
+      do_CreateInstance(NS_SCRIPTERROR_CONTRACTID, &rv);
+  if (!errorObject) {
+    NS_WARNING("Failed to log message to console.");
+    return;
+  }
+
+  nsAutoString result;
+  rv = nsContentUtils::GetLocalizedString(nsContentUtils::eDOM_PROPERTIES, aKey,
+                                          result);
+
+  if (NS_FAILED(rv)) {
+    NS_WARNING("Failed to log message to console.");
+    return;
+  }
+
+  errorObject->InitWithWindowID(result, spec, u""_ns, aLineNumber,
+                                aColumnNumber, nsIScriptError::warningFlag,
+                                "Web Audio", aWindowID);
+  console->LogMessage(errorObject);
+}
+
+}  // namespace dom
+}  // namespace mozilla
diff --git a/dom/media/webaudio/WebAudioUtils.h b/dom/media/webaudio/WebAudioUtils.h
new file mode 100644
index 0000000000..08e4974771
--- /dev/null
+++ b/dom/media/webaudio/WebAudioUtils.h
@@ -0,0 +1,184 @@
+/* -*- 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/. */
+
+#ifndef WebAudioUtils_h_
+#define WebAudioUtils_h_
+
+#include <cmath>
+#include <limits>
+#include <type_traits>
+#include "fdlibm.h"
+#include "mozilla/FloatingPoint.h"
+#include "mozilla/Logging.h"
+#include "MediaSegment.h"
+
+// Forward declaration
+typedef struct SpeexResamplerState_ SpeexResamplerState;
+
+namespace mozilla {
+
+extern LazyLogModule gWebAudioAPILog;
+#define WEB_AUDIO_API_LOG(...) \
+  MOZ_LOG(gWebAudioAPILog, LogLevel::Debug, (__VA_ARGS__))
+
+namespace dom::WebAudioUtils {
+
+// 32 is the minimum required by the spec for createBuffer() and
+// createScriptProcessor() and matches what is used by Blink.  The limit
+// protects against large memory allocations.
+const size_t MaxChannelCount = 32;
+// AudioContext::CreateBuffer() "must support sample-rates in at least the
+// range 22050 to 96000."
+const uint32_t MinSampleRate = 8000;
+const uint32_t MaxSampleRate = 192000;
+
+inline bool FuzzyEqual(float v1, float v2) { return fabsf(v1 - v2) < 1e-7f; }
+inline bool FuzzyEqual(double v1, double v2) { return fabs(v1 - v2) < 1e-7; }
+
+/**
+ * Converts a linear value to decibels.  Returns aMinDecibels if the linear
+ * value is 0.
+ */
+inline float ConvertLinearToDecibels(float aLinearValue, float aMinDecibels) {
+  return aLinearValue ? 20.0f * fdlibm_log10f(aLinearValue) : aMinDecibels;
+}
+
+/**
+ * Converts a decibel value to a linear value.
+ */
+inline float ConvertDecibelsToLinear(float aDecibels) {
+  return fdlibm_powf(10.0f, 0.05f * aDecibels);
+}
+
+inline void FixNaN(double& aDouble) {
+  if (std::isnan(aDouble) || std::isinf(aDouble)) {
+    aDouble = 0.0;
+  }
+}
+
+inline double DiscreteTimeConstantForSampleRate(double timeConstant,
+                                                double sampleRate) {
+  return 1.0 - fdlibm_exp(-1.0 / (sampleRate * timeConstant));
+}
+
+inline bool IsTimeValid(double aTime) {
+  return aTime >= 0 && aTime <= (MEDIA_TIME_MAX >> TRACK_RATE_MAX_BITS);
+}
+
+/**
+ * Converts a floating point value to an integral type in a safe and
+ * platform agnostic way.  The following program demonstrates the kinds
+ * of ways things can go wrong depending on the CPU architecture you're
+ * compiling for:
+ *
+ * #include <stdio.h>
+ * volatile float r;
+ * int main()
+ * {
+ *   unsigned int q;
+ *   r = 1e100;
+ *   q = r;
+ *   printf("%f %d\n", r, q);
+ *   r = -1e100;
+ *   q = r;
+ *   printf("%f %d\n", r, q);
+ *   r = 1e15;
+ *   q = r;
+ *   printf("%f %x\n", r, q);
+ *   r = 0/0.;
+ *   q = r;
+ *   printf("%f %d\n", r, q);
+ * }
+ *
+ * This program, when compiled for unsigned int, generates the following
+ * results depending on the architecture:
+ *
+ * x86 and x86-64
+ * ---
+ *  inf 0
+ *  -inf 0
+ *  999999995904.000000 -727384064 d4a50000
+ *  nan 0
+ *
+ * ARM
+ * ---
+ *  inf -1
+ *  -inf 0
+ *  999999995904.000000 -1
+ *  nan 0
+ *
+ * When compiled for int, this program generates the following results:
+ *
+ * x86 and x86-64
+ * ---
+ *  inf -2147483648
+ *  -inf -2147483648
+ *  999999995904.000000 -2147483648
+ *  nan -2147483648
+ *
+ * ARM
+ * ---
+ *  inf 2147483647
+ *  -inf -2147483648
+ *  999999995904.000000 2147483647
+ *  nan 0
+ *
+ * Note that the caller is responsible to make sure that the value
+ * passed to this function is not a NaN.  This function will abort if
+ * it sees a NaN.
+ */
+template <typename IntType, typename FloatType>
+IntType TruncateFloatToInt(FloatType f) {
+  using std::numeric_limits;
+  static_assert(std::is_integral_v<IntType> == true,
+                "IntType must be an integral type");
+  static_assert(std::is_floating_point_v<FloatType> == true,
+                "FloatType must be a floating point type");
+
+  if (std::isnan(f)) {
+    // It is the responsibility of the caller to deal with NaN values.
+    // If we ever get to this point, we have a serious bug to fix.
+    MOZ_CRASH("We should never see a NaN here");
+  }
+
+  // If the floating point value is outside of the range of maximum
+  // integral value for this type, just clamp to the maximum value.
+  // The equality case must also return max() due to loss of precision when
+  // converting max() to float.
+  if (f >= FloatType(numeric_limits<IntType>::max())) {
+    return numeric_limits<IntType>::max();
+  }
+
+  if (f <= FloatType(numeric_limits<IntType>::min())) {
+    // If the floating point value is outside of the range of minimum
+    // integral value for this type, just clamp to the minimum value.
+    return numeric_limits<IntType>::min();
+  }
+
+  // Otherwise, this conversion must be well defined.
+  return IntType(f);
+}
+
+void Shutdown();
+
+int SpeexResamplerProcess(SpeexResamplerState* aResampler, uint32_t aChannel,
+                          const float* aIn, uint32_t* aInLen, float* aOut,
+                          uint32_t* aOutLen);
+
+int SpeexResamplerProcess(SpeexResamplerState* aResampler, uint32_t aChannel,
+                          const int16_t* aIn, uint32_t* aInLen, float* aOut,
+                          uint32_t* aOutLen);
+
+int SpeexResamplerProcess(SpeexResamplerState* aResampler, uint32_t aChannel,
+                          const int16_t* aIn, uint32_t* aInLen, int16_t* aOut,
+                          uint32_t* aOutLen);
+
+void LogToDeveloperConsole(uint64_t aWindowID, const char* aKey);
+
+}  // namespace dom::WebAudioUtils
+}  // namespace mozilla
+
+#endif
diff --git a/dom/media/webaudio/blink/Biquad.cpp b/dom/media/webaudio/blink/Biquad.cpp
new file mode 100644
index 0000000000..f8ce08f9bb
--- /dev/null
+++ b/dom/media/webaudio/blink/Biquad.cpp
@@ -0,0 +1,442 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "Biquad.h"
+
+#include "DenormalDisabler.h"
+
+#include <float.h>
+#include <algorithm>
+#include <math.h>
+
+#include "fdlibm.h"
+
+namespace WebCore {
+
+Biquad::Biquad() {
+  // Initialize as pass-thru (straight-wire, no filter effect)
+  setNormalizedCoefficients(1, 0, 0, 1, 0, 0);
+
+  reset();  // clear filter memory
+}
+
+Biquad::~Biquad() = default;
+
+void Biquad::process(const float* sourceP, float* destP,
+                     size_t framesToProcess) {
+  // Create local copies of member variables
+  double x1 = m_x1;
+  double x2 = m_x2;
+  double y1 = m_y1;
+  double y2 = m_y2;
+
+  double b0 = m_b0;
+  double b1 = m_b1;
+  double b2 = m_b2;
+  double a1 = m_a1;
+  double a2 = m_a2;
+
+  for (size_t i = 0; i < framesToProcess; ++i) {
+    // FIXME: this can be optimized by pipelining the multiply adds...
+    double x = sourceP[i];
+    double y = b0 * x + b1 * x1 + b2 * x2 - a1 * y1 - a2 * y2;
+
+    destP[i] = y;
+
+    // Update state variables
+    x2 = x1;
+    x1 = x;
+    y2 = y1;
+    y1 = y;
+  }
+
+  // Avoid introducing a stream of subnormals when input is silent and the
+  // tail approaches zero.
+  if (x1 == 0.0 && x2 == 0.0 && (y1 != 0.0 || y2 != 0.0) &&
+      fabs(y1) < FLT_MIN && fabs(y2) < FLT_MIN) {
+    // Flush future values to zero (until there is new input).
+    y1 = y2 = 0.0;
+// Flush calculated values.
+#ifndef HAVE_DENORMAL
+    for (int i = framesToProcess; i-- && fabsf(destP[i]) < FLT_MIN;) {
+      destP[i] = 0.0f;
+    }
+#endif
+  }
+  // Local variables back to member.
+  m_x1 = x1;
+  m_x2 = x2;
+  m_y1 = y1;
+  m_y2 = y2;
+}
+
+void Biquad::reset() { m_x1 = m_x2 = m_y1 = m_y2 = 0; }
+
+void Biquad::setLowpassParams(double cutoff, double resonance) {
+  // Limit cutoff to 0 to 1.
+  cutoff = std::max(0.0, std::min(cutoff, 1.0));
+
+  if (cutoff == 1) {
+    // When cutoff is 1, the z-transform is 1.
+    setNormalizedCoefficients(1, 0, 0, 1, 0, 0);
+  } else if (cutoff > 0) {
+    // Compute biquad coefficients for lowpass filter
+    double g = fdlibm_pow(10.0, -0.05 * resonance);
+    double w0 = M_PI * cutoff;
+    double cos_w0 = fdlibm_cos(w0);
+    double alpha = 0.5 * fdlibm_sin(w0) * g;
+
+    double b1 = 1.0 - cos_w0;
+    double b0 = 0.5 * b1;
+    double b2 = b0;
+    double a0 = 1.0 + alpha;
+    double a1 = -2.0 * cos_w0;
+    double a2 = 1.0 - alpha;
+
+    setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+  } else {
+    // When cutoff is zero, nothing gets through the filter, so set
+    // coefficients up correctly.
+    setNormalizedCoefficients(0, 0, 0, 1, 0, 0);
+  }
+}
+
+void Biquad::setHighpassParams(double cutoff, double resonance) {
+  // Limit cutoff to 0 to 1.
+  cutoff = std::max(0.0, std::min(cutoff, 1.0));
+
+  if (cutoff == 1) {
+    // The z-transform is 0.
+    setNormalizedCoefficients(0, 0, 0, 1, 0, 0);
+  } else if (cutoff > 0) {
+    // Compute biquad coefficients for highpass filter
+    double g = fdlibm_pow(10.0, -0.05 * resonance);
+    double w0 = M_PI * cutoff;
+    double cos_w0 = fdlibm_cos(w0);
+    double alpha = 0.5 * fdlibm_sin(w0) * g;
+
+    double b1 = -1.0 - cos_w0;
+    double b0 = -0.5 * b1;
+    double b2 = b0;
+    double a0 = 1.0 + alpha;
+    double a1 = -2.0 * cos_w0;
+    double a2 = 1.0 - alpha;
+
+    setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+  } else {
+    // When cutoff is zero, we need to be careful because the above
+    // gives a quadratic divided by the same quadratic, with poles
+    // and zeros on the unit circle in the same place. When cutoff
+    // is zero, the z-transform is 1.
+    setNormalizedCoefficients(1, 0, 0, 1, 0, 0);
+  }
+}
+
+void Biquad::setNormalizedCoefficients(double b0, double b1, double b2,
+                                       double a0, double a1, double a2) {
+  double a0Inverse = 1 / a0;
+
+  m_b0 = b0 * a0Inverse;
+  m_b1 = b1 * a0Inverse;
+  m_b2 = b2 * a0Inverse;
+  m_a1 = a1 * a0Inverse;
+  m_a2 = a2 * a0Inverse;
+}
+
+void Biquad::setLowShelfParams(double frequency, double dbGain) {
+  // Clip frequencies to between 0 and 1, inclusive.
+  frequency = std::max(0.0, std::min(frequency, 1.0));
+
+  double A = fdlibm_pow(10.0, dbGain / 40);
+
+  if (frequency == 1) {
+    // The z-transform is a constant gain.
+    setNormalizedCoefficients(A * A, 0, 0, 1, 0, 0);
+  } else if (frequency > 0) {
+    double w0 = M_PI * frequency;
+    double S = 1;  // filter slope (1 is max value)
+    double alpha = 0.5 * fdlibm_sin(w0) * sqrt((A + 1 / A) * (1 / S - 1) + 2);
+    double k = fdlibm_cos(w0);
+    double k2 = 2 * sqrt(A) * alpha;
+    double aPlusOne = A + 1;
+    double aMinusOne = A - 1;
+
+    double b0 = A * (aPlusOne - aMinusOne * k + k2);
+    double b1 = 2 * A * (aMinusOne - aPlusOne * k);
+    double b2 = A * (aPlusOne - aMinusOne * k - k2);
+    double a0 = aPlusOne + aMinusOne * k + k2;
+    double a1 = -2 * (aMinusOne + aPlusOne * k);
+    double a2 = aPlusOne + aMinusOne * k - k2;
+
+    setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+  } else {
+    // When frequency is 0, the z-transform is 1.
+    setNormalizedCoefficients(1, 0, 0, 1, 0, 0);
+  }
+}
+
+void Biquad::setHighShelfParams(double frequency, double dbGain) {
+  // Clip frequencies to between 0 and 1, inclusive.
+  frequency = std::max(0.0, std::min(frequency, 1.0));
+
+  double A = fdlibm_pow(10.0, dbGain / 40);
+
+  if (frequency == 1) {
+    // The z-transform is 1.
+    setNormalizedCoefficients(1, 0, 0, 1, 0, 0);
+  } else if (frequency > 0) {
+    double w0 = M_PI * frequency;
+    double S = 1;  // filter slope (1 is max value)
+    double alpha = 0.5 * fdlibm_sin(w0) * sqrt((A + 1 / A) * (1 / S - 1) + 2);
+    double k = fdlibm_cos(w0);
+    double k2 = 2 * sqrt(A) * alpha;
+    double aPlusOne = A + 1;
+    double aMinusOne = A - 1;
+
+    double b0 = A * (aPlusOne + aMinusOne * k + k2);
+    double b1 = -2 * A * (aMinusOne + aPlusOne * k);
+    double b2 = A * (aPlusOne + aMinusOne * k - k2);
+    double a0 = aPlusOne - aMinusOne * k + k2;
+    double a1 = 2 * (aMinusOne - aPlusOne * k);
+    double a2 = aPlusOne - aMinusOne * k - k2;
+
+    setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+  } else {
+    // When frequency = 0, the filter is just a gain, A^2.
+    setNormalizedCoefficients(A * A, 0, 0, 1, 0, 0);
+  }
+}
+
+void Biquad::setPeakingParams(double frequency, double Q, double dbGain) {
+  // Clip frequencies to between 0 and 1, inclusive.
+  frequency = std::max(0.0, std::min(frequency, 1.0));
+
+  // Don't let Q go negative, which causes an unstable filter.
+  Q = std::max(0.0, Q);
+
+  double A = fdlibm_pow(10.0, dbGain / 40);
+
+  if (frequency > 0 && frequency < 1) {
+    if (Q > 0) {
+      double w0 = M_PI * frequency;
+      double alpha = fdlibm_sin(w0) / (2 * Q);
+      double k = fdlibm_cos(w0);
+
+      double b0 = 1 + alpha * A;
+      double b1 = -2 * k;
+      double b2 = 1 - alpha * A;
+      double a0 = 1 + alpha / A;
+      double a1 = -2 * k;
+      double a2 = 1 - alpha / A;
+
+      setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+    } else {
+      // When Q = 0, the above formulas have problems. If we look at
+      // the z-transform, we can see that the limit as Q->0 is A^2, so
+      // set the filter that way.
+      setNormalizedCoefficients(A * A, 0, 0, 1, 0, 0);
+    }
+  } else {
+    // When frequency is 0 or 1, the z-transform is 1.
+    setNormalizedCoefficients(1, 0, 0, 1, 0, 0);
+  }
+}
+
+void Biquad::setAllpassParams(double frequency, double Q) {
+  // Clip frequencies to between 0 and 1, inclusive.
+  frequency = std::max(0.0, std::min(frequency, 1.0));
+
+  // Don't let Q go negative, which causes an unstable filter.
+  Q = std::max(0.0, Q);
+
+  if (frequency > 0 && frequency < 1) {
+    if (Q > 0) {
+      double w0 = M_PI * frequency;
+      double alpha = fdlibm_sin(w0) / (2 * Q);
+      double k = fdlibm_cos(w0);
+
+      double b0 = 1 - alpha;
+      double b1 = -2 * k;
+      double b2 = 1 + alpha;
+      double a0 = 1 + alpha;
+      double a1 = -2 * k;
+      double a2 = 1 - alpha;
+
+      setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+    } else {
+      // When Q = 0, the above formulas have problems. If we look at
+      // the z-transform, we can see that the limit as Q->0 is -1, so
+      // set the filter that way.
+      setNormalizedCoefficients(-1, 0, 0, 1, 0, 0);
+    }
+  } else {
+    // When frequency is 0 or 1, the z-transform is 1.
+    setNormalizedCoefficients(1, 0, 0, 1, 0, 0);
+  }
+}
+
+void Biquad::setNotchParams(double frequency, double Q) {
+  // Clip frequencies to between 0 and 1, inclusive.
+  frequency = std::max(0.0, std::min(frequency, 1.0));
+
+  // Don't let Q go negative, which causes an unstable filter.
+  Q = std::max(0.0, Q);
+
+  if (frequency > 0 && frequency < 1) {
+    if (Q > 0) {
+      double w0 = M_PI * frequency;
+      double alpha = fdlibm_sin(w0) / (2 * Q);
+      double k = fdlibm_cos(w0);
+
+      double b0 = 1;
+      double b1 = -2 * k;
+      double b2 = 1;
+      double a0 = 1 + alpha;
+      double a1 = -2 * k;
+      double a2 = 1 - alpha;
+
+      setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+    } else {
+      // When Q = 0, the above formulas have problems. If we look at
+      // the z-transform, we can see that the limit as Q->0 is 0, so
+      // set the filter that way.
+      setNormalizedCoefficients(0, 0, 0, 1, 0, 0);
+    }
+  } else {
+    // When frequency is 0 or 1, the z-transform is 1.
+    setNormalizedCoefficients(1, 0, 0, 1, 0, 0);
+  }
+}
+
+void Biquad::setBandpassParams(double frequency, double Q) {
+  // No negative frequencies allowed.
+  frequency = std::max(0.0, frequency);
+
+  // Don't let Q go negative, which causes an unstable filter.
+  Q = std::max(0.0, Q);
+
+  if (frequency > 0 && frequency < 1) {
+    double w0 = M_PI * frequency;
+    if (Q > 0) {
+      double alpha = fdlibm_sin(w0) / (2 * Q);
+      double k = fdlibm_cos(w0);
+
+      double b0 = alpha;
+      double b1 = 0;
+      double b2 = -alpha;
+      double a0 = 1 + alpha;
+      double a1 = -2 * k;
+      double a2 = 1 - alpha;
+
+      setNormalizedCoefficients(b0, b1, b2, a0, a1, a2);
+    } else {
+      // When Q = 0, the above formulas have problems. If we look at
+      // the z-transform, we can see that the limit as Q->0 is 1, so
+      // set the filter that way.
+      setNormalizedCoefficients(1, 0, 0, 1, 0, 0);
+    }
+  } else {
+    // When the cutoff is zero, the z-transform approaches 0, if Q
+    // > 0. When both Q and cutoff are zero, the z-transform is
+    // pretty much undefined. What should we do in this case?
+    // For now, just make the filter 0. When the cutoff is 1, the
+    // z-transform also approaches 0.
+    setNormalizedCoefficients(0, 0, 0, 1, 0, 0);
+  }
+}
+
+void Biquad::setZeroPolePairs(const Complex& zero, const Complex& pole) {
+  double b0 = 1;
+  double b1 = -2 * zero.real();
+
+  double zeroMag = abs(zero);
+  double b2 = zeroMag * zeroMag;
+
+  double a1 = -2 * pole.real();
+
+  double poleMag = abs(pole);
+  double a2 = poleMag * poleMag;
+  setNormalizedCoefficients(b0, b1, b2, 1, a1, a2);
+}
+
+void Biquad::setAllpassPole(const Complex& pole) {
+  Complex zero = Complex(1, 0) / pole;
+  setZeroPolePairs(zero, pole);
+}
+
+void Biquad::getFrequencyResponse(int nFrequencies, const float* frequency,
+                                  float* magResponse, float* phaseResponse) {
+  // Evaluate the Z-transform of the filter at given normalized
+  // frequency from 0 to 1.  (1 corresponds to the Nyquist
+  // frequency.)
+  //
+  // The z-transform of the filter is
+  //
+  // H(z) = (b0 + b1*z^(-1) + b2*z^(-2))/(1 + a1*z^(-1) + a2*z^(-2))
+  //
+  // Evaluate as
+  //
+  // b0 + (b1 + b2*z1)*z1
+  // --------------------
+  // 1 + (a1 + a2*z1)*z1
+  //
+  // with z1 = 1/z and z = exp(j*pi*frequency). Hence z1 = exp(-j*pi*frequency)
+
+  // Make local copies of the coefficients as a micro-optimization.
+  double b0 = m_b0;
+  double b1 = m_b1;
+  double b2 = m_b2;
+  double a1 = m_a1;
+  double a2 = m_a2;
+
+  for (int k = 0; k < nFrequencies; ++k) {
+    double omega = -M_PI * frequency[k];
+    Complex z = Complex(fdlibm_cos(omega), fdlibm_sin(omega));
+    Complex numerator = b0 + (b1 + b2 * z) * z;
+    Complex denominator = Complex(1, 0) + (a1 + a2 * z) * z;
+    // Strangely enough, using complex division:
+    // e.g. Complex response = numerator / denominator;
+    // fails on our test machines, yielding infinities and NaNs, so we do
+    // things the long way here.
+    double n = norm(denominator);
+    double r = (real(numerator) * real(denominator) +
+                imag(numerator) * imag(denominator)) /
+               n;
+    double i = (imag(numerator) * real(denominator) -
+                real(numerator) * imag(denominator)) /
+               n;
+    std::complex<double> response = std::complex<double>(r, i);
+
+    magResponse[k] =
+        static_cast<float>(fdlibm_hypot(real(response), imag(response)));
+    phaseResponse[k] =
+        static_cast<float>(fdlibm_atan2(imag(response), real(response)));
+  }
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/Biquad.h b/dom/media/webaudio/blink/Biquad.h
new file mode 100644
index 0000000000..d64ed72b50
--- /dev/null
+++ b/dom/media/webaudio/blink/Biquad.h
@@ -0,0 +1,108 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef Biquad_h
+#define Biquad_h
+
+#include <complex>
+
+namespace WebCore {
+
+typedef std::complex<double> Complex;
+
+// A basic biquad (two-zero / two-pole digital filter)
+//
+// It can be configured to a number of common and very useful filters:
+//    lowpass, highpass, shelving, parameteric, notch, allpass, ...
+
+class Biquad {
+ public:
+  Biquad();
+  ~Biquad();
+
+  void process(const float* sourceP, float* destP, size_t framesToProcess);
+
+  // frequency is 0 - 1 normalized, resonance and dbGain are in decibels.
+  // Q is a unitless quality factor.
+  void setLowpassParams(double frequency, double resonance);
+  void setHighpassParams(double frequency, double resonance);
+  void setBandpassParams(double frequency, double Q);
+  void setLowShelfParams(double frequency, double dbGain);
+  void setHighShelfParams(double frequency, double dbGain);
+  void setPeakingParams(double frequency, double Q, double dbGain);
+  void setAllpassParams(double frequency, double Q);
+  void setNotchParams(double frequency, double Q);
+
+  // Set the biquad coefficients given a single zero (other zero will be
+  // conjugate) and a single pole (other pole will be conjugate)
+  void setZeroPolePairs(const Complex& zero, const Complex& pole);
+
+  // Set the biquad coefficients given a single pole (other pole will be
+  // conjugate) (The zeroes will be the inverse of the poles)
+  void setAllpassPole(const Complex& pole);
+
+  // Return true iff the next output block will contain sound even with
+  // silent input.
+  bool hasTail() const { return m_y1 || m_y2 || m_x1 || m_x2; }
+
+  // Resets filter state
+  void reset();
+
+  // Filter response at a set of n frequencies. The magnitude and
+  // phase response are returned in magResponse and phaseResponse.
+  // The phase response is in radians.
+  void getFrequencyResponse(int nFrequencies, const float* frequency,
+                            float* magResponse, float* phaseResponse);
+
+ private:
+  void setNormalizedCoefficients(double b0, double b1, double b2, double a0,
+                                 double a1, double a2);
+
+  // Filter coefficients. The filter is defined as
+  //
+  // y[n] + m_a1*y[n-1] + m_a2*y[n-2] = m_b0*x[n] + m_b1*x[n-1] + m_b2*x[n-2].
+  double m_b0;
+  double m_b1;
+  double m_b2;
+  double m_a1;
+  double m_a2;
+
+  // Filter memory
+  //
+  // Double precision for the output values is valuable because errors can
+  // accumulate.  Input values are also stored as double so they need not be
+  // converted again for computation.
+  double m_x1;  // input delayed by 1 sample
+  double m_x2;  // input delayed by 2 samples
+  double m_y1;  // output delayed by 1 sample
+  double m_y2;  // output delayed by 2 samples
+};
+
+}  // namespace WebCore
+
+#endif  // Biquad_h
diff --git a/dom/media/webaudio/blink/DenormalDisabler.h b/dom/media/webaudio/blink/DenormalDisabler.h
new file mode 100644
index 0000000000..646482b74f
--- /dev/null
+++ b/dom/media/webaudio/blink/DenormalDisabler.h
@@ -0,0 +1,170 @@
+/*
+ * Copyright (C) 2011, Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1.  Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+ * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef DenormalDisabler_h
+#define DenormalDisabler_h
+
+#include <cmath>
+#include <cinttypes>
+#include <cstring>
+#include <float.h>
+
+namespace WebCore {
+
+// Deal with denormals. They can very seriously impact performance on x86.
+
+// Define HAVE_DENORMAL if we support flushing denormals to zero.
+
+#if defined(XP_WIN) && defined(_MSC_VER)
+// Windows compiled using MSVC with SSE2
+#  define HAVE_DENORMAL 1
+#endif
+
+#if defined(__GNUC__) && defined(__SSE__)
+#  define HAVE_DENORMAL 1
+#endif
+
+#if defined(__arm__) || defined(__aarch64__)
+#  define HAVE_DENORMAL 1
+#endif
+
+#ifdef HAVE_DENORMAL
+class DenormalDisabler {
+ public:
+  DenormalDisabler() : m_savedCSR(0) { disableDenormals(); }
+
+  ~DenormalDisabler() { restoreState(); }
+
+  // This is a nop if we can flush denormals to zero in hardware.
+  static inline float flushDenormalFloatToZero(float f) { return f; }
+
+ private:
+  unsigned m_savedCSR;
+
+#  if defined(__GNUC__) && defined(__SSE__)
+  static inline bool isDAZSupported() {
+#    if defined(__x86_64__)
+    return true;
+#    else
+    static bool s_isInited = false;
+    static bool s_isSupported = false;
+    if (s_isInited) {
+      return s_isSupported;
+    }
+
+    struct fxsaveResult {
+      uint8_t before[28];
+      uint32_t CSRMask;
+      uint8_t after[480];
+    } __attribute__((aligned(16)));
+
+    fxsaveResult registerData;
+    memset(&registerData, 0, sizeof(fxsaveResult));
+    asm volatile("fxsave %0" : "=m"(registerData));
+    s_isSupported = registerData.CSRMask & 0x0040;
+    s_isInited = true;
+    return s_isSupported;
+#    endif
+  }
+
+  inline void disableDenormals() {
+    m_savedCSR = getCSR();
+    setCSR(m_savedCSR | (isDAZSupported() ? 0x8040 : 0x8000));
+  }
+
+  inline void restoreState() { setCSR(m_savedCSR); }
+
+  inline int getCSR() {
+    int result;
+    asm volatile("stmxcsr %0" : "=m"(result));
+    return result;
+  }
+
+  inline void setCSR(int a) {
+    int temp = a;
+    asm volatile("ldmxcsr %0" : : "m"(temp));
+  }
+
+#  elif defined(XP_WIN) && defined(_MSC_VER)
+  inline void disableDenormals() {
+    // Save the current state, and set mode to flush denormals.
+    //
+    // http://stackoverflow.com/questions/637175/possible-bug-in-controlfp-s-may-not-restore-control-word-correctly
+    _controlfp_s(&m_savedCSR, 0, 0);
+    unsigned unused;
+    _controlfp_s(&unused, _DN_FLUSH, _MCW_DN);
+  }
+
+  inline void restoreState() {
+    unsigned unused;
+    _controlfp_s(&unused, m_savedCSR, _MCW_DN);
+  }
+#  elif defined(__arm__) || defined(__aarch64__)
+  inline void disableDenormals() {
+    m_savedCSR = getStatusWord();
+    // Bit 24 is the flush-to-zero mode control bit. Setting it to 1 flushes
+    // denormals to 0.
+    setStatusWord(m_savedCSR | (1 << 24));
+  }
+
+  inline void restoreState() { setStatusWord(m_savedCSR); }
+
+  inline int getStatusWord() {
+    int result;
+#    if defined(__aarch64__)
+    asm volatile("mrs %x[result], FPCR" : [result] "=r"(result));
+#    else
+    asm volatile("vmrs %[result], FPSCR" : [result] "=r"(result));
+#    endif
+    return result;
+  }
+
+  inline void setStatusWord(int a) {
+#    if defined(__aarch64__)
+    asm volatile("msr FPCR, %x[src]" : : [src] "r"(a));
+#    else
+    asm volatile("vmsr FPSCR, %[src]" : : [src] "r"(a));
+#    endif
+  }
+
+#  endif
+};
+
+#else
+// FIXME: add implementations for other architectures and compilers
+class DenormalDisabler {
+ public:
+  DenormalDisabler() {}
+
+  // Assume the worst case that other architectures and compilers
+  // need to flush denormals to zero manually.
+  static inline float flushDenormalFloatToZero(float f) {
+    return (fabs(f) < FLT_MIN) ? 0.0f : f;
+  }
+};
+
+#endif
+
+}  // namespace WebCore
+#endif  // DenormalDisabler_h
diff --git a/dom/media/webaudio/blink/DynamicsCompressor.cpp b/dom/media/webaudio/blink/DynamicsCompressor.cpp
new file mode 100644
index 0000000000..0ee9426692
--- /dev/null
+++ b/dom/media/webaudio/blink/DynamicsCompressor.cpp
@@ -0,0 +1,315 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "DynamicsCompressor.h"
+#include "AlignmentUtils.h"
+#include "AudioBlock.h"
+
+#include <cmath>
+#include "AudioNodeEngine.h"
+#include "nsDebug.h"
+
+using mozilla::AudioBlockCopyChannelWithScale;
+using mozilla::WEBAUDIO_BLOCK_SIZE;
+
+namespace WebCore {
+
+DynamicsCompressor::DynamicsCompressor(float sampleRate,
+                                       unsigned numberOfChannels)
+    : m_numberOfChannels(numberOfChannels),
+      m_sampleRate(sampleRate),
+      m_compressor(sampleRate, numberOfChannels) {
+  // Uninitialized state - for parameter recalculation.
+  m_lastFilterStageRatio = -1;
+  m_lastAnchor = -1;
+  m_lastFilterStageGain = -1;
+
+  setNumberOfChannels(numberOfChannels);
+  initializeParameters();
+}
+
+size_t DynamicsCompressor::sizeOfIncludingThis(
+    mozilla::MallocSizeOf aMallocSizeOf) const {
+  size_t amount = aMallocSizeOf(this);
+  amount += m_preFilterPacks.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < m_preFilterPacks.Length(); i++) {
+    if (m_preFilterPacks[i]) {
+      amount += m_preFilterPacks[i]->sizeOfIncludingThis(aMallocSizeOf);
+    }
+  }
+
+  amount += m_postFilterPacks.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < m_postFilterPacks.Length(); i++) {
+    if (m_postFilterPacks[i]) {
+      amount += m_postFilterPacks[i]->sizeOfIncludingThis(aMallocSizeOf);
+    }
+  }
+
+  amount += aMallocSizeOf(m_sourceChannels.get());
+  amount += aMallocSizeOf(m_destinationChannels.get());
+  amount += m_compressor.sizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+void DynamicsCompressor::setParameterValue(unsigned parameterID, float value) {
+  MOZ_ASSERT(parameterID < ParamLast);
+  if (parameterID < ParamLast) m_parameters[parameterID] = value;
+}
+
+void DynamicsCompressor::initializeParameters() {
+  // Initializes compressor to default values.
+
+  m_parameters[ParamThreshold] = -24;    // dB
+  m_parameters[ParamKnee] = 30;          // dB
+  m_parameters[ParamRatio] = 12;         // unit-less
+  m_parameters[ParamAttack] = 0.003f;    // seconds
+  m_parameters[ParamRelease] = 0.250f;   // seconds
+  m_parameters[ParamPreDelay] = 0.006f;  // seconds
+
+  // Release zone values 0 -> 1.
+  m_parameters[ParamReleaseZone1] = 0.09f;
+  m_parameters[ParamReleaseZone2] = 0.16f;
+  m_parameters[ParamReleaseZone3] = 0.42f;
+  m_parameters[ParamReleaseZone4] = 0.98f;
+
+  m_parameters[ParamFilterStageGain] = 4.4f;  // dB
+  m_parameters[ParamFilterStageRatio] = 2;
+  m_parameters[ParamFilterAnchor] = 15000 / nyquist();
+
+  m_parameters[ParamPostGain] = 0;   // dB
+  m_parameters[ParamReduction] = 0;  // dB
+
+  // Linear crossfade (0 -> 1).
+  m_parameters[ParamEffectBlend] = 1;
+}
+
+float DynamicsCompressor::parameterValue(unsigned parameterID) {
+  MOZ_ASSERT(parameterID < ParamLast);
+  return m_parameters[parameterID];
+}
+
+void DynamicsCompressor::setEmphasisStageParameters(
+    unsigned stageIndex, float gain, float normalizedFrequency /* 0 -> 1 */) {
+  float gk = 1 - gain / 20;
+  float f1 = normalizedFrequency * gk;
+  float f2 = normalizedFrequency / gk;
+  float r1 = fdlibm_expf(-f1 * M_PI);
+  float r2 = fdlibm_expf(-f2 * M_PI);
+
+  MOZ_ASSERT(m_numberOfChannels == m_preFilterPacks.Length());
+
+  for (unsigned i = 0; i < m_numberOfChannels; ++i) {
+    // Set pre-filter zero and pole to create an emphasis filter.
+    ZeroPole& preFilter = m_preFilterPacks[i]->filters[stageIndex];
+    preFilter.setZero(r1);
+    preFilter.setPole(r2);
+
+    // Set post-filter with zero and pole reversed to create the de-emphasis
+    // filter. If there were no compressor kernel in between, they would cancel
+    // each other out (allpass filter).
+    ZeroPole& postFilter = m_postFilterPacks[i]->filters[stageIndex];
+    postFilter.setZero(r2);
+    postFilter.setPole(r1);
+  }
+}
+
+void DynamicsCompressor::setEmphasisParameters(float gain, float anchorFreq,
+                                               float filterStageRatio) {
+  setEmphasisStageParameters(0, gain, anchorFreq);
+  setEmphasisStageParameters(1, gain, anchorFreq / filterStageRatio);
+  setEmphasisStageParameters(
+      2, gain, anchorFreq / (filterStageRatio * filterStageRatio));
+  setEmphasisStageParameters(
+      3, gain,
+      anchorFreq / (filterStageRatio * filterStageRatio * filterStageRatio));
+}
+
+void DynamicsCompressor::process(const AudioBlock* sourceChunk,
+                                 AudioBlock* destinationChunk,
+                                 unsigned framesToProcess) {
+  // Though numberOfChannels is retrived from destinationBus, we still name it
+  // numberOfChannels instead of numberOfDestinationChannels. It's because we
+  // internally match sourceChannels's size to destinationBus by channel up/down
+  // mix. Thus we need numberOfChannels to do the loop work for both
+  // m_sourceChannels and m_destinationChannels.
+
+  unsigned numberOfChannels = destinationChunk->ChannelCount();
+  unsigned numberOfSourceChannels = sourceChunk->ChannelCount();
+
+  MOZ_ASSERT(numberOfChannels == m_numberOfChannels && numberOfSourceChannels);
+
+  if (numberOfChannels != m_numberOfChannels || !numberOfSourceChannels) {
+    destinationChunk->SetNull(WEBAUDIO_BLOCK_SIZE);
+    return;
+  }
+
+  switch (numberOfChannels) {
+    case 2:  // stereo
+      m_sourceChannels[0] =
+          static_cast<const float*>(sourceChunk->mChannelData[0]);
+
+      if (numberOfSourceChannels > 1)
+        m_sourceChannels[1] =
+            static_cast<const float*>(sourceChunk->mChannelData[1]);
+      else
+        // Simply duplicate mono channel input data to right channel for stereo
+        // processing.
+        m_sourceChannels[1] = m_sourceChannels[0];
+
+      break;
+    case 1:
+      m_sourceChannels[0] =
+          static_cast<const float*>(sourceChunk->mChannelData[0]);
+      break;
+    default:
+      MOZ_CRASH("not supported.");
+  }
+
+  for (unsigned i = 0; i < numberOfChannels; ++i)
+    m_destinationChannels[i] = const_cast<float*>(
+        static_cast<const float*>(destinationChunk->mChannelData[i]));
+
+  float filterStageGain = parameterValue(ParamFilterStageGain);
+  float filterStageRatio = parameterValue(ParamFilterStageRatio);
+  float anchor = parameterValue(ParamFilterAnchor);
+
+  if (filterStageGain != m_lastFilterStageGain ||
+      filterStageRatio != m_lastFilterStageRatio || anchor != m_lastAnchor) {
+    m_lastFilterStageGain = filterStageGain;
+    m_lastFilterStageRatio = filterStageRatio;
+    m_lastAnchor = anchor;
+
+    setEmphasisParameters(filterStageGain, anchor, filterStageRatio);
+  }
+
+  float sourceWithVolume[WEBAUDIO_BLOCK_SIZE + 4];
+  float* alignedSourceWithVolume = ALIGNED16(sourceWithVolume);
+  ASSERT_ALIGNED16(alignedSourceWithVolume);
+
+  // Apply pre-emphasis filter.
+  // Note that the final three stages are computed in-place in the destination
+  // buffer.
+  for (unsigned i = 0; i < numberOfChannels; ++i) {
+    const float* sourceData;
+    if (sourceChunk->mVolume == 1.0f) {
+      // Fast path, the volume scale doesn't need to get taken into account
+      sourceData = m_sourceChannels[i];
+    } else {
+      AudioBlockCopyChannelWithScale(m_sourceChannels[i], sourceChunk->mVolume,
+                                     alignedSourceWithVolume);
+      sourceData = alignedSourceWithVolume;
+    }
+
+    float* destinationData = m_destinationChannels[i];
+    ZeroPole* preFilters = m_preFilterPacks[i]->filters;
+
+    preFilters[0].process(sourceData, destinationData, framesToProcess);
+    preFilters[1].process(destinationData, destinationData, framesToProcess);
+    preFilters[2].process(destinationData, destinationData, framesToProcess);
+    preFilters[3].process(destinationData, destinationData, framesToProcess);
+  }
+
+  float dbThreshold = parameterValue(ParamThreshold);
+  float dbKnee = parameterValue(ParamKnee);
+  float ratio = parameterValue(ParamRatio);
+  float attackTime = parameterValue(ParamAttack);
+  float releaseTime = parameterValue(ParamRelease);
+  float preDelayTime = parameterValue(ParamPreDelay);
+
+  // This is effectively a master volume on the compressed signal
+  // (pre-blending).
+  float dbPostGain = parameterValue(ParamPostGain);
+
+  // Linear blending value from dry to completely processed (0 -> 1)
+  // 0 means the signal is completely unprocessed.
+  // 1 mixes in only the compressed signal.
+  float effectBlend = parameterValue(ParamEffectBlend);
+
+  float releaseZone1 = parameterValue(ParamReleaseZone1);
+  float releaseZone2 = parameterValue(ParamReleaseZone2);
+  float releaseZone3 = parameterValue(ParamReleaseZone3);
+  float releaseZone4 = parameterValue(ParamReleaseZone4);
+
+  // Apply compression to the pre-filtered signal.
+  // The processing is performed in place.
+  m_compressor.process(m_destinationChannels.get(), m_destinationChannels.get(),
+                       numberOfChannels, framesToProcess,
+
+                       dbThreshold, dbKnee, ratio, attackTime, releaseTime,
+                       preDelayTime, dbPostGain, effectBlend,
+
+                       releaseZone1, releaseZone2, releaseZone3, releaseZone4);
+
+  // Update the compression amount.
+  setParameterValue(ParamReduction, m_compressor.meteringGain());
+
+  // Apply de-emphasis filter.
+  for (unsigned i = 0; i < numberOfChannels; ++i) {
+    float* destinationData = m_destinationChannels[i];
+    ZeroPole* postFilters = m_postFilterPacks[i]->filters;
+
+    postFilters[0].process(destinationData, destinationData, framesToProcess);
+    postFilters[1].process(destinationData, destinationData, framesToProcess);
+    postFilters[2].process(destinationData, destinationData, framesToProcess);
+    postFilters[3].process(destinationData, destinationData, framesToProcess);
+  }
+}
+
+void DynamicsCompressor::reset() {
+  m_lastFilterStageRatio = -1;  // for recalc
+  m_lastAnchor = -1;
+  m_lastFilterStageGain = -1;
+
+  for (unsigned channel = 0; channel < m_numberOfChannels; ++channel) {
+    for (unsigned stageIndex = 0; stageIndex < 4; ++stageIndex) {
+      m_preFilterPacks[channel]->filters[stageIndex].reset();
+      m_postFilterPacks[channel]->filters[stageIndex].reset();
+    }
+  }
+
+  m_compressor.reset();
+}
+
+void DynamicsCompressor::setNumberOfChannels(unsigned numberOfChannels) {
+  if (m_preFilterPacks.Length() == numberOfChannels) return;
+
+  m_preFilterPacks.Clear();
+  m_postFilterPacks.Clear();
+  for (unsigned i = 0; i < numberOfChannels; ++i) {
+    m_preFilterPacks.AppendElement(new ZeroPoleFilterPack4());
+    m_postFilterPacks.AppendElement(new ZeroPoleFilterPack4());
+  }
+
+  m_sourceChannels = mozilla::MakeUnique<const float*[]>(numberOfChannels);
+  m_destinationChannels = mozilla::MakeUnique<float*[]>(numberOfChannels);
+
+  m_compressor.setNumberOfChannels(numberOfChannels);
+  m_numberOfChannels = numberOfChannels;
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/DynamicsCompressor.h b/dom/media/webaudio/blink/DynamicsCompressor.h
new file mode 100644
index 0000000000..1838e268fe
--- /dev/null
+++ b/dom/media/webaudio/blink/DynamicsCompressor.h
@@ -0,0 +1,135 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef DynamicsCompressor_h
+#define DynamicsCompressor_h
+
+#include "DynamicsCompressorKernel.h"
+#include "ZeroPole.h"
+
+#include "nsTArray.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/UniquePtr.h"
+
+namespace mozilla {
+class AudioBlock;
+}  // namespace mozilla
+
+namespace WebCore {
+
+using mozilla::AudioBlock;
+using mozilla::UniquePtr;
+
+// DynamicsCompressor implements a flexible audio dynamics compression effect
+// such as is commonly used in musical production and game audio. It lowers the
+// volume of the loudest parts of the signal and raises the volume of the
+// softest parts, making the sound richer, fuller, and more controlled.
+
+class DynamicsCompressor {
+ public:
+  enum {
+    ParamThreshold,
+    ParamKnee,
+    ParamRatio,
+    ParamAttack,
+    ParamRelease,
+    ParamPreDelay,
+    ParamReleaseZone1,
+    ParamReleaseZone2,
+    ParamReleaseZone3,
+    ParamReleaseZone4,
+    ParamPostGain,
+    ParamFilterStageGain,
+    ParamFilterStageRatio,
+    ParamFilterAnchor,
+    ParamEffectBlend,
+    ParamReduction,
+    ParamLast
+  };
+
+  DynamicsCompressor(float sampleRate, unsigned numberOfChannels);
+
+  void process(const AudioBlock* sourceChunk, AudioBlock* destinationChunk,
+               unsigned framesToProcess);
+  void reset();
+  void setNumberOfChannels(unsigned);
+  unsigned numberOfChannels() const { return m_numberOfChannels; }
+
+  void setParameterValue(unsigned parameterID, float value);
+  float parameterValue(unsigned parameterID);
+
+  float sampleRate() const { return m_sampleRate; }
+  float nyquist() const { return m_sampleRate / 2; }
+
+  double tailTime() const { return 0; }
+  double latencyTime() const {
+    return m_compressor.latencyFrames() / static_cast<double>(sampleRate());
+  }
+
+  size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+ protected:
+  unsigned m_numberOfChannels;
+
+  // m_parameters holds the tweakable compressor parameters.
+  float m_parameters[ParamLast];
+  void initializeParameters();
+
+  float m_sampleRate;
+
+  // Emphasis filter controls.
+  float m_lastFilterStageRatio;
+  float m_lastAnchor;
+  float m_lastFilterStageGain;
+
+  struct ZeroPoleFilterPack4 {
+    ZeroPole filters[4];
+    size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const {
+      return aMallocSizeOf(this);
+    }
+  };
+
+  // Per-channel emphasis filters.
+  nsTArray<UniquePtr<ZeroPoleFilterPack4> > m_preFilterPacks;
+  nsTArray<UniquePtr<ZeroPoleFilterPack4> > m_postFilterPacks;
+
+  mozilla::UniquePtr<const float*[]> m_sourceChannels;
+  mozilla::UniquePtr<float*[]> m_destinationChannels;
+
+  void setEmphasisStageParameters(unsigned stageIndex, float gain,
+                                  float normalizedFrequency /* 0 -> 1 */);
+  void setEmphasisParameters(float gain, float anchorFreq,
+                             float filterStageRatio);
+
+  // The core compressor.
+  DynamicsCompressorKernel m_compressor;
+};
+
+}  // namespace WebCore
+
+#endif  // DynamicsCompressor_h
diff --git a/dom/media/webaudio/blink/DynamicsCompressorKernel.cpp b/dom/media/webaudio/blink/DynamicsCompressorKernel.cpp
new file mode 100644
index 0000000000..7fac8b78a3
--- /dev/null
+++ b/dom/media/webaudio/blink/DynamicsCompressorKernel.cpp
@@ -0,0 +1,496 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "DynamicsCompressorKernel.h"
+
+#include "DenormalDisabler.h"
+#include <algorithm>
+#include <cmath>
+
+#include "mozilla/FloatingPoint.h"
+#include "WebAudioUtils.h"
+
+using namespace mozilla::dom;  // for WebAudioUtils
+using mozilla::MakeUnique;
+using mozilla::PositiveInfinity;
+
+namespace WebCore {
+
+// Metering hits peaks instantly, but releases this fast (in seconds).
+const float meteringReleaseTimeConstant = 0.325f;
+
+const float uninitializedValue = -1;
+
+DynamicsCompressorKernel::DynamicsCompressorKernel(float sampleRate,
+                                                   unsigned numberOfChannels)
+    : m_sampleRate(sampleRate),
+      m_lastPreDelayFrames(DefaultPreDelayFrames),
+      m_preDelayReadIndex(0),
+      m_preDelayWriteIndex(DefaultPreDelayFrames),
+      m_ratio(uninitializedValue),
+      m_slope(uninitializedValue),
+      m_linearThreshold(uninitializedValue),
+      m_dbThreshold(uninitializedValue),
+      m_dbKnee(uninitializedValue),
+      m_kneeThreshold(uninitializedValue),
+      m_kneeThresholdDb(uninitializedValue),
+      m_ykneeThresholdDb(uninitializedValue),
+      m_K(uninitializedValue) {
+  setNumberOfChannels(numberOfChannels);
+
+  // Initializes most member variables
+  reset();
+
+  m_meteringReleaseK =
+      static_cast<float>(WebAudioUtils::DiscreteTimeConstantForSampleRate(
+          meteringReleaseTimeConstant, sampleRate));
+}
+
+size_t DynamicsCompressorKernel::sizeOfExcludingThis(
+    mozilla::MallocSizeOf aMallocSizeOf) const {
+  size_t amount = 0;
+  amount += m_preDelayBuffers.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < m_preDelayBuffers.Length(); i++) {
+    amount += aMallocSizeOf(m_preDelayBuffers[i].get());
+  }
+
+  return amount;
+}
+
+void DynamicsCompressorKernel::setNumberOfChannels(unsigned numberOfChannels) {
+  if (m_preDelayBuffers.Length() == numberOfChannels) return;
+
+  m_preDelayBuffers.Clear();
+  for (unsigned i = 0; i < numberOfChannels; ++i)
+    m_preDelayBuffers.AppendElement(MakeUnique<float[]>(MaxPreDelayFrames));
+}
+
+void DynamicsCompressorKernel::setPreDelayTime(float preDelayTime) {
+  // Re-configure look-ahead section pre-delay if delay time has changed.
+  unsigned preDelayFrames = preDelayTime * sampleRate();
+  if (preDelayFrames > MaxPreDelayFrames - 1)
+    preDelayFrames = MaxPreDelayFrames - 1;
+
+  if (m_lastPreDelayFrames != preDelayFrames) {
+    m_lastPreDelayFrames = preDelayFrames;
+    for (unsigned i = 0; i < m_preDelayBuffers.Length(); ++i)
+      memset(m_preDelayBuffers[i].get(), 0, sizeof(float) * MaxPreDelayFrames);
+
+    m_preDelayReadIndex = 0;
+    m_preDelayWriteIndex = preDelayFrames;
+  }
+}
+
+// Exponential curve for the knee.
+// It is 1st derivative matched at m_linearThreshold and asymptotically
+// approaches the value m_linearThreshold + 1 / k.
+float DynamicsCompressorKernel::kneeCurve(float x, float k) {
+  // Linear up to threshold.
+  if (x < m_linearThreshold) return x;
+
+  return m_linearThreshold +
+         (1 - fdlibm_expf(-k * (x - m_linearThreshold))) / k;
+}
+
+// Full compression curve with constant ratio after knee.
+float DynamicsCompressorKernel::saturate(float x, float k) {
+  float y;
+
+  if (x < m_kneeThreshold)
+    y = kneeCurve(x, k);
+  else {
+    // Constant ratio after knee.
+    float xDb = WebAudioUtils::ConvertLinearToDecibels(x, -1000.0f);
+    float yDb = m_ykneeThresholdDb + m_slope * (xDb - m_kneeThresholdDb);
+
+    y = WebAudioUtils::ConvertDecibelsToLinear(yDb);
+  }
+
+  return y;
+}
+
+// Approximate 1st derivative with input and output expressed in dB.
+// This slope is equal to the inverse of the compression "ratio".
+// In other words, a compression ratio of 20 would be a slope of 1/20.
+float DynamicsCompressorKernel::slopeAt(float x, float k) {
+  if (x < m_linearThreshold) return 1;
+
+  float x2 = x * 1.001;
+
+  float xDb = WebAudioUtils::ConvertLinearToDecibels(x, -1000.0f);
+  float x2Db = WebAudioUtils::ConvertLinearToDecibels(x2, -1000.0f);
+
+  float yDb = WebAudioUtils::ConvertLinearToDecibels(kneeCurve(x, k), -1000.0f);
+  float y2Db =
+      WebAudioUtils::ConvertLinearToDecibels(kneeCurve(x2, k), -1000.0f);
+
+  float m = (y2Db - yDb) / (x2Db - xDb);
+
+  return m;
+}
+
+float DynamicsCompressorKernel::kAtSlope(float desiredSlope) {
+  float xDb = m_dbThreshold + m_dbKnee;
+  float x = WebAudioUtils::ConvertDecibelsToLinear(xDb);
+
+  // Approximate k given initial values.
+  float minK = 0.1f;
+  float maxK = 10000;
+  float k = 5;
+
+  for (int i = 0; i < 15; ++i) {
+    // A high value for k will more quickly asymptotically approach a slope of
+    // 0.
+    float slope = slopeAt(x, k);
+
+    if (slope < desiredSlope) {
+      // k is too high.
+      maxK = k;
+    } else {
+      // k is too low.
+      minK = k;
+    }
+
+    // Re-calculate based on geometric mean.
+    k = sqrtf(minK * maxK);
+  }
+
+  return k;
+}
+
+float DynamicsCompressorKernel::updateStaticCurveParameters(float dbThreshold,
+                                                            float dbKnee,
+                                                            float ratio) {
+  if (dbThreshold != m_dbThreshold || dbKnee != m_dbKnee || ratio != m_ratio) {
+    // Threshold and knee.
+    m_dbThreshold = dbThreshold;
+    m_linearThreshold = WebAudioUtils::ConvertDecibelsToLinear(dbThreshold);
+    m_dbKnee = dbKnee;
+
+    // Compute knee parameters.
+    m_ratio = ratio;
+    m_slope = 1 / m_ratio;
+
+    float k = kAtSlope(1 / m_ratio);
+
+    m_kneeThresholdDb = dbThreshold + dbKnee;
+    m_kneeThreshold = WebAudioUtils::ConvertDecibelsToLinear(m_kneeThresholdDb);
+
+    m_ykneeThresholdDb = WebAudioUtils::ConvertLinearToDecibels(
+        kneeCurve(m_kneeThreshold, k), -1000.0f);
+
+    m_K = k;
+  }
+  return m_K;
+}
+
+void DynamicsCompressorKernel::process(
+    float* sourceChannels[], float* destinationChannels[],
+    unsigned numberOfChannels, unsigned framesToProcess,
+
+    float dbThreshold, float dbKnee, float ratio, float attackTime,
+    float releaseTime, float preDelayTime, float dbPostGain,
+    float effectBlend, /* equal power crossfade */
+
+    float releaseZone1, float releaseZone2, float releaseZone3,
+    float releaseZone4) {
+  MOZ_ASSERT(m_preDelayBuffers.Length() == numberOfChannels);
+
+  float sampleRate = this->sampleRate();
+
+  float dryMix = 1 - effectBlend;
+  float wetMix = effectBlend;
+
+  float k = updateStaticCurveParameters(dbThreshold, dbKnee, ratio);
+
+  // Makeup gain.
+  float fullRangeGain = saturate(1, k);
+  float fullRangeMakeupGain = 1 / fullRangeGain;
+
+  // Empirical/perceptual tuning.
+  fullRangeMakeupGain = fdlibm_powf(fullRangeMakeupGain, 0.6f);
+
+  float masterLinearGain =
+      WebAudioUtils::ConvertDecibelsToLinear(dbPostGain) * fullRangeMakeupGain;
+
+  // Attack parameters.
+  attackTime = std::max(0.001f, attackTime);
+  float attackFrames = attackTime * sampleRate;
+
+  // Release parameters.
+  float releaseFrames = sampleRate * releaseTime;
+
+  // Detector release time.
+  float satReleaseTime = 0.0025f;
+  float satReleaseFrames = satReleaseTime * sampleRate;
+
+  // Create a smooth function which passes through four points.
+
+  // Polynomial of the form
+  // y = a + b*x + c*x^2 + d*x^3 + e*x^4;
+
+  float y1 = releaseFrames * releaseZone1;
+  float y2 = releaseFrames * releaseZone2;
+  float y3 = releaseFrames * releaseZone3;
+  float y4 = releaseFrames * releaseZone4;
+
+  // All of these coefficients were derived for 4th order polynomial curve
+  // fitting where the y values match the evenly spaced x values as follows:
+  //  (y1 : x == 0, y2 : x == 1, y3 : x == 2, y4 : x == 3)
+  float kA = 0.9999999999999998f * y1 + 1.8432219684323923e-16f * y2 -
+             1.9373394351676423e-16f * y3 + 8.824516011816245e-18f * y4;
+  float kB = -1.5788320352845888f * y1 + 2.3305837032074286f * y2 -
+             0.9141194204840429f * y3 + 0.1623677525612032f * y4;
+  float kC = 0.5334142869106424f * y1 - 1.272736789213631f * y2 +
+             0.9258856042207512f * y3 - 0.18656310191776226f * y4;
+  float kD = 0.08783463138207234f * y1 - 0.1694162967925622f * y2 +
+             0.08588057951595272f * y3 - 0.00429891410546283f * y4;
+  float kE = -0.042416883008123074f * y1 + 0.1115693827987602f * y2 -
+             0.09764676325265872f * y3 + 0.028494263462021576f * y4;
+
+  // x ranges from 0 -> 3       0    1    2   3
+  //                           -15  -10  -5   0db
+
+  // y calculates adaptive release frames depending on the amount of
+  // compression.
+
+  setPreDelayTime(preDelayTime);
+
+  const int nDivisionFrames = 32;
+
+  const int nDivisions = framesToProcess / nDivisionFrames;
+
+  unsigned frameIndex = 0;
+  for (int i = 0; i < nDivisions; ++i) {
+    // ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+    // Calculate desired gain
+    // ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+    // Fix gremlins.
+    if (std::isnan(m_detectorAverage)) m_detectorAverage = 1;
+    if (std::isinf(m_detectorAverage)) m_detectorAverage = 1;
+
+    float desiredGain = m_detectorAverage;
+
+    // Pre-warp so we get desiredGain after sin() warp below.
+    float scaledDesiredGain = fdlibm_asinf(desiredGain) / (0.5f * M_PI);
+
+    // ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+    // Deal with envelopes
+    // ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+    // envelopeRate is the rate we slew from current compressor level to the
+    // desired level. The exact rate depends on if we're attacking or releasing
+    // and by how much.
+    float envelopeRate;
+
+    bool isReleasing = scaledDesiredGain > m_compressorGain;
+
+    // compressionDiffDb is the difference between current compression level and
+    // the desired level.
+    float compressionDiffDb;
+    if (scaledDesiredGain == 0.0) {
+      compressionDiffDb = PositiveInfinity<float>();
+    } else {
+      compressionDiffDb = WebAudioUtils::ConvertLinearToDecibels(
+          m_compressorGain / scaledDesiredGain, -1000.0f);
+    }
+
+    if (isReleasing) {
+      // Release mode - compressionDiffDb should be negative dB
+      m_maxAttackCompressionDiffDb = -1;
+
+      // Fix gremlins.
+      if (std::isnan(compressionDiffDb)) compressionDiffDb = -1;
+      if (std::isinf(compressionDiffDb)) compressionDiffDb = -1;
+
+      // Adaptive release - higher compression (lower compressionDiffDb)
+      // releases faster.
+
+      // Contain within range: -12 -> 0 then scale to go from 0 -> 3
+      float x = compressionDiffDb;
+      x = std::max(-12.0f, x);
+      x = std::min(0.0f, x);
+      x = 0.25f * (x + 12);
+
+      // Compute adaptive release curve using 4th order polynomial.
+      // Normal values for the polynomial coefficients would create a
+      // monotonically increasing function.
+      float x2 = x * x;
+      float x3 = x2 * x;
+      float x4 = x2 * x2;
+      float releaseFrames = kA + kB * x + kC * x2 + kD * x3 + kE * x4;
+
+#define kSpacingDb 5
+      float dbPerFrame = kSpacingDb / releaseFrames;
+
+      envelopeRate = WebAudioUtils::ConvertDecibelsToLinear(dbPerFrame);
+    } else {
+      // Attack mode - compressionDiffDb should be positive dB
+
+      // Fix gremlins.
+      if (std::isnan(compressionDiffDb)) compressionDiffDb = 1;
+      if (std::isinf(compressionDiffDb)) compressionDiffDb = 1;
+
+      // As long as we're still in attack mode, use a rate based off
+      // the largest compressionDiffDb we've encountered so far.
+      if (m_maxAttackCompressionDiffDb == -1 ||
+          m_maxAttackCompressionDiffDb < compressionDiffDb)
+        m_maxAttackCompressionDiffDb = compressionDiffDb;
+
+      float effAttenDiffDb = std::max(0.5f, m_maxAttackCompressionDiffDb);
+
+      float x = 0.25f / effAttenDiffDb;
+      envelopeRate = 1 - fdlibm_powf(x, 1 / attackFrames);
+    }
+
+    // ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+    // Inner loop - calculate shaped power average - apply compression.
+    // ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+    {
+      int preDelayReadIndex = m_preDelayReadIndex;
+      int preDelayWriteIndex = m_preDelayWriteIndex;
+      float detectorAverage = m_detectorAverage;
+      float compressorGain = m_compressorGain;
+
+      int loopFrames = nDivisionFrames;
+      while (loopFrames--) {
+        float compressorInput = 0;
+
+        // Predelay signal, computing compression amount from un-delayed
+        // version.
+        for (unsigned i = 0; i < numberOfChannels; ++i) {
+          float* delayBuffer = m_preDelayBuffers[i].get();
+          float undelayedSource = sourceChannels[i][frameIndex];
+          delayBuffer[preDelayWriteIndex] = undelayedSource;
+
+          float absUndelayedSource =
+              undelayedSource > 0 ? undelayedSource : -undelayedSource;
+          if (compressorInput < absUndelayedSource)
+            compressorInput = absUndelayedSource;
+        }
+
+        // Calculate shaped power on undelayed input.
+
+        float scaledInput = compressorInput;
+        float absInput = scaledInput > 0 ? scaledInput : -scaledInput;
+
+        // Put through shaping curve.
+        // This is linear up to the threshold, then enters a "knee" portion
+        // followed by the "ratio" portion. The transition from the threshold to
+        // the knee is smooth (1st derivative matched). The transition from the
+        // knee to the ratio portion is smooth (1st derivative matched).
+        float shapedInput = saturate(absInput, k);
+
+        float attenuation = absInput <= 0.0001f ? 1 : shapedInput / absInput;
+
+        float attenuationDb =
+            -WebAudioUtils::ConvertLinearToDecibels(attenuation, -1000.0f);
+        attenuationDb = std::max(2.0f, attenuationDb);
+
+        float dbPerFrame = attenuationDb / satReleaseFrames;
+
+        float satReleaseRate =
+            WebAudioUtils::ConvertDecibelsToLinear(dbPerFrame) - 1;
+
+        bool isRelease = (attenuation > detectorAverage);
+        float rate = isRelease ? satReleaseRate : 1;
+
+        detectorAverage += (attenuation - detectorAverage) * rate;
+        detectorAverage = std::min(1.0f, detectorAverage);
+
+        // Fix gremlins.
+        if (std::isnan(detectorAverage)) detectorAverage = 1;
+        if (std::isinf(detectorAverage)) detectorAverage = 1;
+
+        // Exponential approach to desired gain.
+        if (envelopeRate < 1) {
+          // Attack - reduce gain to desired.
+          compressorGain += (scaledDesiredGain - compressorGain) * envelopeRate;
+        } else {
+          // Release - exponentially increase gain to 1.0
+          compressorGain *= envelopeRate;
+          compressorGain = std::min(1.0f, compressorGain);
+        }
+
+        // Warp pre-compression gain to smooth out sharp exponential transition
+        // points.
+        float postWarpCompressorGain =
+            fdlibm_sinf(0.5f * M_PI * compressorGain);
+
+        // Calculate total gain using master gain and effect blend.
+        float totalGain =
+            dryMix + wetMix * masterLinearGain * postWarpCompressorGain;
+
+        // Calculate metering.
+        float dbRealGain = 20 * fdlibm_log10f(postWarpCompressorGain);
+        if (dbRealGain < m_meteringGain)
+          m_meteringGain = dbRealGain;
+        else
+          m_meteringGain += (dbRealGain - m_meteringGain) * m_meteringReleaseK;
+
+        // Apply final gain.
+        for (unsigned i = 0; i < numberOfChannels; ++i) {
+          float* delayBuffer = m_preDelayBuffers[i].get();
+          destinationChannels[i][frameIndex] =
+              delayBuffer[preDelayReadIndex] * totalGain;
+        }
+
+        frameIndex++;
+        preDelayReadIndex = (preDelayReadIndex + 1) & MaxPreDelayFramesMask;
+        preDelayWriteIndex = (preDelayWriteIndex + 1) & MaxPreDelayFramesMask;
+      }
+
+      // Locals back to member variables.
+      m_preDelayReadIndex = preDelayReadIndex;
+      m_preDelayWriteIndex = preDelayWriteIndex;
+      m_detectorAverage =
+          DenormalDisabler::flushDenormalFloatToZero(detectorAverage);
+      m_compressorGain =
+          DenormalDisabler::flushDenormalFloatToZero(compressorGain);
+    }
+  }
+}
+
+void DynamicsCompressorKernel::reset() {
+  m_detectorAverage = 0;
+  m_compressorGain = 1;
+  m_meteringGain = 1;
+
+  // Predelay section.
+  for (unsigned i = 0; i < m_preDelayBuffers.Length(); ++i)
+    memset(m_preDelayBuffers[i].get(), 0, sizeof(float) * MaxPreDelayFrames);
+
+  m_preDelayReadIndex = 0;
+  m_preDelayWriteIndex = DefaultPreDelayFrames;
+
+  m_maxAttackCompressionDiffDb = -1;  // uninitialized state
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/DynamicsCompressorKernel.h b/dom/media/webaudio/blink/DynamicsCompressorKernel.h
new file mode 100644
index 0000000000..bb7defbd7e
--- /dev/null
+++ b/dom/media/webaudio/blink/DynamicsCompressorKernel.h
@@ -0,0 +1,125 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef DynamicsCompressorKernel_h
+#define DynamicsCompressorKernel_h
+
+#include "nsTArray.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/UniquePtr.h"
+
+namespace WebCore {
+
+class DynamicsCompressorKernel {
+ public:
+  DynamicsCompressorKernel(float sampleRate, unsigned numberOfChannels);
+
+  void setNumberOfChannels(unsigned);
+
+  // Performs stereo-linked compression.
+  void process(float* sourceChannels[], float* destinationChannels[],
+               unsigned numberOfChannels, unsigned framesToProcess,
+
+               float dbThreshold, float dbKnee, float ratio, float attackTime,
+               float releaseTime, float preDelayTime, float dbPostGain,
+               float effectBlend,
+
+               float releaseZone1, float releaseZone2, float releaseZone3,
+               float releaseZone4);
+
+  void reset();
+
+  unsigned latencyFrames() const { return m_lastPreDelayFrames; }
+
+  float sampleRate() const { return m_sampleRate; }
+
+  float meteringGain() const { return m_meteringGain; }
+
+  size_t sizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+ protected:
+  float m_sampleRate;
+
+  float m_detectorAverage;
+  float m_compressorGain;
+
+  // Metering
+  float m_meteringReleaseK;
+  float m_meteringGain;
+
+  // Lookahead section.
+  enum { MaxPreDelayFrames = 1024 };
+  enum { MaxPreDelayFramesMask = MaxPreDelayFrames - 1 };
+  enum {
+    DefaultPreDelayFrames = 256
+  };  // setPreDelayTime() will override this initial value
+  unsigned m_lastPreDelayFrames;
+  void setPreDelayTime(float);
+
+  nsTArray<mozilla::UniquePtr<float[]>> m_preDelayBuffers;
+  int m_preDelayReadIndex;
+  int m_preDelayWriteIndex;
+
+  float m_maxAttackCompressionDiffDb;
+
+  // Static compression curve.
+  float kneeCurve(float x, float k);
+  float saturate(float x, float k);
+  float slopeAt(float x, float k);
+  float kAtSlope(float desiredSlope);
+
+  float updateStaticCurveParameters(float dbThreshold, float dbKnee,
+                                    float ratio);
+
+  // Amount of input change in dB required for 1 dB of output change.
+  // This applies to the portion of the curve above m_kneeThresholdDb (see
+  // below).
+  float m_ratio;
+  float m_slope;  // Inverse ratio.
+
+  // The input to output change below the threshold is linear 1:1.
+  float m_linearThreshold;
+  float m_dbThreshold;
+
+  // m_dbKnee is the number of dB above the threshold before we enter the
+  // "ratio" portion of the curve. m_kneeThresholdDb = m_dbThreshold + m_dbKnee
+  // The portion between m_dbThreshold and m_kneeThresholdDb is the "soft knee"
+  // portion of the curve which transitions smoothly from the linear portion to
+  // the ratio portion.
+  float m_dbKnee;
+  float m_kneeThreshold;
+  float m_kneeThresholdDb;
+  float m_ykneeThresholdDb;
+
+  // Internal parameter for the knee portion of the curve.
+  float m_K;
+};
+
+}  // namespace WebCore
+
+#endif  // DynamicsCompressorKernel_h
diff --git a/dom/media/webaudio/blink/FFTConvolver.cpp b/dom/media/webaudio/blink/FFTConvolver.cpp
new file mode 100644
index 0000000000..2ade9031ce
--- /dev/null
+++ b/dom/media/webaudio/blink/FFTConvolver.cpp
@@ -0,0 +1,118 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "FFTConvolver.h"
+#include "mozilla/PodOperations.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+FFTConvolver::FFTConvolver(size_t fftSize, size_t renderPhase)
+    : m_frame(fftSize), m_readWriteIndex(renderPhase % (fftSize / 2)) {
+  MOZ_ASSERT(fftSize >= 2 * WEBAUDIO_BLOCK_SIZE);
+  m_inputBuffer.SetLength(fftSize);
+  PodZero(m_inputBuffer.Elements(), fftSize);
+  m_outputBuffer.SetLength(fftSize);
+  PodZero(m_outputBuffer.Elements(), fftSize);
+  m_lastOverlapBuffer.SetLength(fftSize / 2);
+  PodZero(m_lastOverlapBuffer.Elements(), fftSize / 2);
+}
+
+size_t FFTConvolver::sizeOfExcludingThis(
+    mozilla::MallocSizeOf aMallocSizeOf) const {
+  size_t amount = 0;
+  amount += m_frame.SizeOfExcludingThis(aMallocSizeOf);
+  amount += m_inputBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  amount += m_outputBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  amount += m_lastOverlapBuffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  return amount;
+}
+
+size_t FFTConvolver::sizeOfIncludingThis(
+    mozilla::MallocSizeOf aMallocSizeOf) const {
+  return aMallocSizeOf(this) + sizeOfExcludingThis(aMallocSizeOf);
+}
+
+const float* FFTConvolver::process(FFTBlock* fftKernel, const float* sourceP) {
+  size_t halfSize = fftSize() / 2;
+
+  // WEBAUDIO_BLOCK_SIZE must be an exact multiple of halfSize,
+  // halfSize must be a multiple of WEBAUDIO_BLOCK_SIZE
+  // and > WEBAUDIO_BLOCK_SIZE.
+  MOZ_ASSERT(halfSize % WEBAUDIO_BLOCK_SIZE == 0 &&
+             WEBAUDIO_BLOCK_SIZE <= halfSize);
+
+  // Copy samples to input buffer (note contraint above!)
+  float* inputP = m_inputBuffer.Elements();
+
+  MOZ_ASSERT(sourceP && inputP &&
+             m_readWriteIndex + WEBAUDIO_BLOCK_SIZE <= m_inputBuffer.Length());
+
+  memcpy(inputP + m_readWriteIndex, sourceP,
+         sizeof(float) * WEBAUDIO_BLOCK_SIZE);
+
+  float* outputP = m_outputBuffer.Elements();
+  m_readWriteIndex += WEBAUDIO_BLOCK_SIZE;
+
+  // Check if it's time to perform the next FFT
+  if (m_readWriteIndex == halfSize) {
+    // The input buffer is now filled (get frequency-domain version)
+    m_frame.PerformFFT(m_inputBuffer.Elements());
+    m_frame.Multiply(*fftKernel);
+    m_frame.GetInverseWithoutScaling(m_outputBuffer.Elements());
+
+    // Overlap-add 1st half from previous time
+    AudioBufferAddWithScale(m_lastOverlapBuffer.Elements(), 1.0f,
+                            m_outputBuffer.Elements(), halfSize);
+
+    // Finally, save 2nd half of result
+    MOZ_ASSERT(m_outputBuffer.Length() == 2 * halfSize &&
+               m_lastOverlapBuffer.Length() == halfSize);
+
+    memcpy(m_lastOverlapBuffer.Elements(), m_outputBuffer.Elements() + halfSize,
+           sizeof(float) * halfSize);
+
+    // Reset index back to start for next time
+    m_readWriteIndex = 0;
+  }
+
+  return outputP + m_readWriteIndex;
+}
+
+void FFTConvolver::reset() {
+  PodZero(m_lastOverlapBuffer.Elements(), m_lastOverlapBuffer.Length());
+  m_readWriteIndex = 0;
+}
+
+size_t FFTConvolver::latencyFrames() const {
+  return std::max<size_t>(fftSize() / 2, WEBAUDIO_BLOCK_SIZE) -
+         WEBAUDIO_BLOCK_SIZE;
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/FFTConvolver.h b/dom/media/webaudio/blink/FFTConvolver.h
new file mode 100644
index 0000000000..d54b07f7a4
--- /dev/null
+++ b/dom/media/webaudio/blink/FFTConvolver.h
@@ -0,0 +1,87 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FFTConvolver_h
+#define FFTConvolver_h
+
+#include "nsTArray.h"
+#include "mozilla/FFTBlock.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace WebCore {
+
+typedef AlignedTArray<float> AlignedAudioFloatArray;
+using mozilla::FFTBlock;
+
+class FFTConvolver {
+ public:
+  // |fftSize| must be a power of two.
+  //
+  // |renderPhase| is the initial offset in the initially zero input buffer.
+  // It is coordinated with the other stages, so they don't all do their
+  // FFTs at the same time.
+  explicit FFTConvolver(size_t fftSize, size_t renderPhase = 0);
+
+  // Process WEBAUDIO_BLOCK_SIZE elements of array |sourceP| and return a
+  // pointer to an output array of the same size.
+  //
+  // |fftKernel| must be pre-scaled for FFTBlock::GetInverseWithoutScaling().
+  //
+  // FIXME: Later, we can do more sophisticated buffering to relax this
+  // requirement...
+  const float* process(FFTBlock* fftKernel, const float* sourceP);
+
+  void reset();
+
+  size_t fftSize() const { return m_frame.FFTSize(); }
+
+  // The input to output latency is up to fftSize / 2, but alignment of the
+  // FFTs with the blocks reduces this by one block.
+  size_t latencyFrames() const;
+
+  size_t sizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+  size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+ private:
+  FFTBlock m_frame;
+
+  // Buffer input until we get fftSize / 2 samples then do an FFT
+  size_t m_readWriteIndex;
+  AlignedAudioFloatArray m_inputBuffer;
+
+  // Stores output which we read a little at a time
+  AlignedAudioFloatArray m_outputBuffer;
+
+  // Saves the 2nd half of the FFT buffer, so we can do an overlap-add with the
+  // 1st half of the next one
+  AlignedAudioFloatArray m_lastOverlapBuffer;
+};
+
+}  // namespace WebCore
+
+#endif  // FFTConvolver_h
diff --git a/dom/media/webaudio/blink/HRTFDatabase.cpp b/dom/media/webaudio/blink/HRTFDatabase.cpp
new file mode 100644
index 0000000000..54be5ea958
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFDatabase.cpp
@@ -0,0 +1,135 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "HRTFDatabase.h"
+
+#include "HRTFElevation.h"
+
+namespace WebCore {
+
+const int HRTFDatabase::MinElevation = -45;
+const int HRTFDatabase::MaxElevation = 90;
+const unsigned HRTFDatabase::RawElevationAngleSpacing = 15;
+const unsigned HRTFDatabase::NumberOfRawElevations =
+    10;  // -45 -> +90 (each 15 degrees)
+const unsigned HRTFDatabase::InterpolationFactor = 1;
+const unsigned HRTFDatabase::NumberOfTotalElevations =
+    NumberOfRawElevations * InterpolationFactor;
+
+nsReturnRef<HRTFDatabase> HRTFDatabase::create(float sampleRate) {
+  return nsReturnRef<HRTFDatabase>(new HRTFDatabase(sampleRate));
+}
+
+HRTFDatabase::HRTFDatabase(float sampleRate) : m_sampleRate(sampleRate) {
+  m_elevations.SetLength(NumberOfTotalElevations);
+
+  unsigned elevationIndex = 0;
+  for (int elevation = MinElevation; elevation <= MaxElevation;
+       elevation += RawElevationAngleSpacing) {
+    nsAutoRef<HRTFElevation> hrtfElevation(
+        HRTFElevation::createBuiltin(elevation, sampleRate));
+    MOZ_ASSERT(hrtfElevation.get());
+    if (!hrtfElevation.get()) return;
+
+    m_elevations[elevationIndex] = hrtfElevation.out();
+    elevationIndex += InterpolationFactor;
+  }
+
+  // Now, go back and interpolate elevations.
+  if (InterpolationFactor > 1) {
+    for (unsigned i = 0; i < NumberOfTotalElevations;
+         i += InterpolationFactor) {
+      unsigned j = (i + InterpolationFactor);
+      if (j >= NumberOfTotalElevations)
+        j = i;  // for last elevation interpolate with itself
+
+      // Create the interpolated convolution kernels and delays.
+      for (unsigned jj = 1; jj < InterpolationFactor; ++jj) {
+        float x =
+            static_cast<float>(jj) / static_cast<float>(InterpolationFactor);
+        m_elevations[i + jj] = HRTFElevation::createByInterpolatingSlices(
+            m_elevations[i].get(), m_elevations[j].get(), x, sampleRate);
+        MOZ_ASSERT(m_elevations[i + jj].get());
+      }
+    }
+  }
+}
+
+size_t HRTFDatabase::sizeOfIncludingThis(
+    mozilla::MallocSizeOf aMallocSizeOf) const {
+  size_t amount = aMallocSizeOf(this);
+  amount += m_elevations.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < m_elevations.Length(); i++) {
+    amount += m_elevations[i]->sizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  return amount;
+}
+
+void HRTFDatabase::getKernelsFromAzimuthElevation(
+    double azimuthBlend, unsigned azimuthIndex, double elevationAngle,
+    HRTFKernel*& kernelL, HRTFKernel*& kernelR, double& frameDelayL,
+    double& frameDelayR) {
+  unsigned elevationIndex = indexFromElevationAngle(elevationAngle);
+  MOZ_ASSERT(elevationIndex < m_elevations.Length() &&
+             m_elevations.Length() > 0);
+
+  if (!m_elevations.Length()) {
+    kernelL = 0;
+    kernelR = 0;
+    return;
+  }
+
+  if (elevationIndex > m_elevations.Length() - 1)
+    elevationIndex = m_elevations.Length() - 1;
+
+  HRTFElevation* hrtfElevation = m_elevations[elevationIndex].get();
+  MOZ_ASSERT(hrtfElevation);
+  if (!hrtfElevation) {
+    kernelL = 0;
+    kernelR = 0;
+    return;
+  }
+
+  hrtfElevation->getKernelsFromAzimuth(azimuthBlend, azimuthIndex, kernelL,
+                                       kernelR, frameDelayL, frameDelayR);
+}
+
+unsigned HRTFDatabase::indexFromElevationAngle(double elevationAngle) {
+  // Clamp to allowed range.
+  elevationAngle =
+      mozilla::clamped(elevationAngle, static_cast<double>(MinElevation),
+                       static_cast<double>(MaxElevation));
+
+  unsigned elevationIndex =
+      static_cast<int>(InterpolationFactor * (elevationAngle - MinElevation) /
+                       RawElevationAngleSpacing);
+  return elevationIndex;
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/HRTFDatabase.h b/dom/media/webaudio/blink/HRTFDatabase.h
new file mode 100644
index 0000000000..50cf2958bd
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFDatabase.h
@@ -0,0 +1,104 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef HRTFDatabase_h
+#define HRTFDatabase_h
+
+#include "HRTFElevation.h"
+#include "nsAutoRef.h"
+#include "nsTArray.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace WebCore {
+
+class HRTFKernel;
+
+class HRTFDatabase {
+ public:
+  static nsReturnRef<HRTFDatabase> create(float sampleRate);
+
+  // clang-format off
+    // getKernelsFromAzimuthElevation() returns a left and right ear kernel, and an interpolated left and right frame delay for the given azimuth and elevation.
+    // azimuthBlend must be in the range 0 -> 1.
+    // Valid values for azimuthIndex are 0 -> HRTFElevation::NumberOfTotalAzimuths - 1 (corresponding to angles of 0 -> 360).
+    // Valid values for elevationAngle are MinElevation -> MaxElevation.
+  // clang-format on
+  void getKernelsFromAzimuthElevation(double azimuthBlend,
+                                      unsigned azimuthIndex,
+                                      double elevationAngle,
+                                      HRTFKernel*& kernelL,
+                                      HRTFKernel*& kernelR, double& frameDelayL,
+                                      double& frameDelayR);
+
+  // Returns the number of different azimuth angles.
+  static unsigned numberOfAzimuths() {
+    return HRTFElevation::NumberOfTotalAzimuths;
+  }
+
+  float sampleRate() const { return m_sampleRate; }
+
+  // Number of elevations loaded from resource.
+  static const unsigned NumberOfRawElevations;
+
+  size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+ private:
+  HRTFDatabase(const HRTFDatabase& other) = delete;
+  void operator=(const HRTFDatabase& other) = delete;
+
+  explicit HRTFDatabase(float sampleRate);
+
+  // Minimum and maximum elevation angles (inclusive) for a HRTFDatabase.
+  static const int MinElevation;
+  static const int MaxElevation;
+  static const unsigned RawElevationAngleSpacing;
+
+  // Interpolates by this factor to get the total number of elevations from
+  // every elevation loaded from resource.
+  static const unsigned InterpolationFactor;
+
+  // Total number of elevations after interpolation.
+  static const unsigned NumberOfTotalElevations;
+
+  // Returns the index for the correct HRTFElevation given the elevation angle.
+  static unsigned indexFromElevationAngle(double);
+
+  nsTArray<nsAutoRef<HRTFElevation> > m_elevations;
+  float m_sampleRate;
+};
+
+}  // namespace WebCore
+
+template <>
+class nsAutoRefTraits<WebCore::HRTFDatabase>
+    : public nsPointerRefTraits<WebCore::HRTFDatabase> {
+ public:
+  static void Release(WebCore::HRTFDatabase* ptr) { delete (ptr); }
+};
+
+#endif  // HRTFDatabase_h
diff --git a/dom/media/webaudio/blink/HRTFDatabaseLoader.cpp b/dom/media/webaudio/blink/HRTFDatabaseLoader.cpp
new file mode 100644
index 0000000000..3db455545e
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFDatabaseLoader.cpp
@@ -0,0 +1,215 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "HRTFDatabaseLoader.h"
+#include "HRTFDatabase.h"
+#include "GeckoProfiler.h"
+#include "nsThreadUtils.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+// Singleton
+nsTHashtable<HRTFDatabaseLoader::LoaderByRateEntry>*
+    HRTFDatabaseLoader::s_loaderMap = nullptr;
+
+size_t HRTFDatabaseLoader::sizeOfLoaders(mozilla::MallocSizeOf aMallocSizeOf) {
+  return s_loaderMap ? s_loaderMap->SizeOfIncludingThis(aMallocSizeOf) : 0;
+}
+
+already_AddRefed<HRTFDatabaseLoader>
+HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(float sampleRate) {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  RefPtr<HRTFDatabaseLoader> loader;
+
+  if (!s_loaderMap) {
+    s_loaderMap = new nsTHashtable<LoaderByRateEntry>();
+  }
+
+  LoaderByRateEntry* entry = s_loaderMap->PutEntry(sampleRate);
+  loader = entry->mLoader;
+  if (loader) {  // existing entry
+    MOZ_ASSERT(sampleRate == loader->databaseSampleRate());
+    return loader.forget();
+  }
+
+  loader = new HRTFDatabaseLoader(sampleRate);
+  entry->mLoader = loader;
+
+  loader->loadAsynchronously();
+
+  return loader.forget();
+}
+
+HRTFDatabaseLoader::HRTFDatabaseLoader(float sampleRate)
+    : m_refCnt(0),
+      m_threadLock("HRTFDatabaseLoader"),
+      m_databaseLoaderThread(nullptr),
+      m_databaseSampleRate(sampleRate),
+      m_databaseLoaded(false) {
+  MOZ_ASSERT(NS_IsMainThread());
+}
+
+HRTFDatabaseLoader::~HRTFDatabaseLoader() {
+  MOZ_ASSERT(NS_IsMainThread());
+
+  waitForLoaderThreadCompletion();
+  m_hrtfDatabase.reset();
+
+  if (s_loaderMap) {
+    // Remove ourself from the map.
+    s_loaderMap->RemoveEntry(m_databaseSampleRate);
+    if (s_loaderMap->Count() == 0) {
+      delete s_loaderMap;
+      s_loaderMap = nullptr;
+    }
+  }
+}
+
+size_t HRTFDatabaseLoader::sizeOfIncludingThis(
+    mozilla::MallocSizeOf aMallocSizeOf) const {
+  size_t amount = aMallocSizeOf(this);
+
+  // NB: Need to make sure we're not competing with the loader thread.
+  const_cast<HRTFDatabaseLoader*>(this)->waitForLoaderThreadCompletion();
+
+  if (m_hrtfDatabase) {
+    amount += m_hrtfDatabase->sizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  return amount;
+}
+
+class HRTFDatabaseLoader::ProxyReleaseEvent final : public Runnable {
+ public:
+  explicit ProxyReleaseEvent(HRTFDatabaseLoader* loader)
+      : mozilla::Runnable("WebCore::HRTFDatabaseLoader::ProxyReleaseEvent"),
+        mLoader(loader) {}
+  NS_IMETHOD Run() override {
+    mLoader->MainThreadRelease();
+    return NS_OK;
+  }
+
+ private:
+  // Ownership transferred by ProxyRelease
+  HRTFDatabaseLoader* MOZ_OWNING_REF mLoader;
+};
+
+void HRTFDatabaseLoader::ProxyRelease() {
+  nsCOMPtr<nsIEventTarget> mainTarget = GetMainThreadSerialEventTarget();
+  if (MOZ_LIKELY(mainTarget)) {
+    RefPtr<ProxyReleaseEvent> event = new ProxyReleaseEvent(this);
+    DebugOnly<nsresult> rv = mainTarget->Dispatch(event, NS_DISPATCH_NORMAL);
+    MOZ_ASSERT(NS_SUCCEEDED(rv), "Failed to dispatch release event");
+  } else {
+    // Should be in XPCOM shutdown.
+    MOZ_ASSERT(NS_IsMainThread(), "Main thread is not available for dispatch.");
+    MainThreadRelease();
+  }
+}
+
+void HRTFDatabaseLoader::MainThreadRelease() {
+  MOZ_ASSERT(NS_IsMainThread());
+  int count = --m_refCnt;
+  MOZ_ASSERT(count >= 0, "extra release");
+  NS_LOG_RELEASE(this, count, "HRTFDatabaseLoader");
+  if (count == 0) {
+    // It is safe to delete here as the first reference can only be added
+    // on this (main) thread.
+    delete this;
+  }
+}
+
+// Asynchronously load the database in this thread.
+static void databaseLoaderEntry(void* threadData) {
+  AUTO_PROFILER_REGISTER_THREAD("HRTFDatabaseLdr");
+  NS_SetCurrentThreadName("HRTFDatabaseLdr");
+
+  HRTFDatabaseLoader* loader =
+      reinterpret_cast<HRTFDatabaseLoader*>(threadData);
+  MOZ_ASSERT(loader);
+  loader->load();
+}
+
+void HRTFDatabaseLoader::load() {
+  MOZ_ASSERT(!NS_IsMainThread());
+  MOZ_ASSERT(!m_hrtfDatabase.get(), "Called twice");
+  // Load the default HRTF database.
+  m_hrtfDatabase = HRTFDatabase::create(m_databaseSampleRate);
+  m_databaseLoaded = true;
+  // Notifies the main thread of completion.  See loadAsynchronously().
+  Release();
+}
+
+void HRTFDatabaseLoader::loadAsynchronously() {
+  MOZ_ASSERT(NS_IsMainThread());
+  MOZ_ASSERT(m_refCnt, "Must not be called before a reference is added");
+
+  // Add a reference so that the destructor won't run and wait for the
+  // loader thread, until load() has completed.
+  AddRef();
+
+  MutexAutoLock locker(m_threadLock);
+
+  MOZ_ASSERT(!m_hrtfDatabase.get() && !m_databaseLoaderThread, "Called twice");
+  // Start the asynchronous database loading process.
+  m_databaseLoaderThread = PR_CreateThread(
+      PR_USER_THREAD, databaseLoaderEntry, this, PR_PRIORITY_NORMAL,
+      PR_GLOBAL_THREAD, PR_JOINABLE_THREAD, 0);
+}
+
+bool HRTFDatabaseLoader::isLoaded() const { return m_hrtfDatabase.get(); }
+
+void HRTFDatabaseLoader::waitForLoaderThreadCompletion() {
+  MutexAutoLock locker(m_threadLock);
+
+  // waitForThreadCompletion() should not be called twice for the same thread.
+  if (m_databaseLoaderThread) {
+    DebugOnly<PRStatus> status = PR_JoinThread(m_databaseLoaderThread);
+    MOZ_ASSERT(status == PR_SUCCESS, "PR_JoinThread failed");
+  }
+  m_databaseLoaderThread = nullptr;
+}
+
+void HRTFDatabaseLoader::shutdown() {
+  MOZ_ASSERT(NS_IsMainThread());
+  if (s_loaderMap) {
+    // Set s_loaderMap to nullptr so that the hashtable is not modified on
+    // reference release during enumeration.
+    nsTHashtable<LoaderByRateEntry>* loaderMap = s_loaderMap;
+    s_loaderMap = nullptr;
+    for (auto iter = loaderMap->Iter(); !iter.Done(); iter.Next()) {
+      iter.Get()->mLoader->waitForLoaderThreadCompletion();
+    }
+    delete loaderMap;
+  }
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/HRTFDatabaseLoader.h b/dom/media/webaudio/blink/HRTFDatabaseLoader.h
new file mode 100644
index 0000000000..5d4548d247
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFDatabaseLoader.h
@@ -0,0 +1,155 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef HRTFDatabaseLoader_h
+#define HRTFDatabaseLoader_h
+
+#include "nsHashKeys.h"
+#include "mozilla/RefPtr.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/Mutex.h"
+#include "HRTFDatabase.h"
+
+template <class EntryType>
+class nsTHashtable;
+template <class T>
+class nsAutoRef;
+
+namespace WebCore {
+
+// HRTFDatabaseLoader will asynchronously load the default HRTFDatabase in a new
+// thread.
+
+class HRTFDatabaseLoader {
+ public:
+  // Lazily creates a HRTFDatabaseLoader (if not already created) for the given
+  // sample-rate and starts loading asynchronously (when created the first
+  // time). Returns the HRTFDatabaseLoader. Must be called from the main thread.
+  static already_AddRefed<HRTFDatabaseLoader>
+  createAndLoadAsynchronouslyIfNecessary(float sampleRate);
+
+  // AddRef and Release may be called from any thread.
+  void AddRef() {
+#if defined(DEBUG) || defined(NS_BUILD_REFCNT_LOGGING)
+    int count =
+#endif
+        ++m_refCnt;
+    MOZ_ASSERT(count > 0, "invalid ref count");
+    NS_LOG_ADDREF(this, count, "HRTFDatabaseLoader", sizeof(*this));
+  }
+
+  void Release() {
+    // The last reference can't be removed on a non-main thread because
+    // the object can be accessed on the main thread from the hash
+    // table via createAndLoadAsynchronouslyIfNecessary().
+    int count = m_refCnt;
+    MOZ_ASSERT(count > 0, "extra release");
+    // Optimization attempt to possibly skip proxying the release to the
+    // main thread.
+    if (count != 1 && m_refCnt.compareExchange(count, count - 1)) {
+      NS_LOG_RELEASE(this, count - 1, "HRTFDatabaseLoader");
+      return;
+    }
+
+    ProxyRelease();
+  }
+
+  // Returns true once the default database has been completely loaded.
+  bool isLoaded() const;
+
+  // waitForLoaderThreadCompletion() may be called more than once,
+  // on any thread except m_databaseLoaderThread.
+  void waitForLoaderThreadCompletion();
+
+  HRTFDatabase* database() {
+    if (!m_databaseLoaded) {
+      return nullptr;
+    }
+    return m_hrtfDatabase.get();
+  }
+
+  float databaseSampleRate() const { return m_databaseSampleRate; }
+
+  static void shutdown();
+
+  // Called in asynchronous loading thread.
+  void load();
+
+  // Sums the size of all cached database loaders.
+  static size_t sizeOfLoaders(mozilla::MallocSizeOf aMallocSizeOf);
+
+ private:
+  // Both constructor and destructor must be called from the main thread.
+  explicit HRTFDatabaseLoader(float sampleRate);
+  ~HRTFDatabaseLoader();
+
+  size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+  void ProxyRelease();       // any thread
+  void MainThreadRelease();  // main thread only
+  class ProxyReleaseEvent;
+
+  // If it hasn't already been loaded, creates a new thread and initiates
+  // asynchronous loading of the default database. This must be called from the
+  // main thread.
+  void loadAsynchronously();
+
+  // Map from sample-rate to loader.
+  class LoaderByRateEntry : public nsFloatHashKey {
+   public:
+    explicit LoaderByRateEntry(KeyTypePointer aKey)
+        : nsFloatHashKey(aKey),
+          mLoader()  // so PutEntry() will zero-initialize
+    {}
+
+    size_t SizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const {
+      return mLoader ? mLoader->sizeOfIncludingThis(aMallocSizeOf) : 0;
+    }
+
+    // The HRTFDatabaseLoader removes itself from s_loaderMap on destruction.
+    HRTFDatabaseLoader* MOZ_NON_OWNING_REF mLoader;
+  };
+
+  // Keeps track of loaders on a per-sample-rate basis.
+  static nsTHashtable<LoaderByRateEntry>* s_loaderMap;  // singleton
+
+  mozilla::Atomic<int> m_refCnt;
+
+  nsAutoRef<HRTFDatabase> m_hrtfDatabase;
+
+  // Holding a m_threadLock is required when accessing m_databaseLoaderThread.
+  mozilla::Mutex m_threadLock;
+  PRThread* m_databaseLoaderThread MOZ_GUARDED_BY(m_threadLock);
+
+  float m_databaseSampleRate;
+  mozilla::Atomic<bool> m_databaseLoaded;
+};
+
+}  // namespace WebCore
+
+#endif  // HRTFDatabaseLoader_h
diff --git a/dom/media/webaudio/blink/HRTFElevation.cpp b/dom/media/webaudio/blink/HRTFElevation.cpp
new file mode 100644
index 0000000000..38ac0f8614
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFElevation.cpp
@@ -0,0 +1,317 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "HRTFElevation.h"
+
+#include <speex/speex_resampler.h>
+#include "mozilla/PodOperations.h"
+#include "AudioSampleFormat.h"
+
+#include "IRC_Composite_C_R0195-incl.cpp"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+const int elevationSpacing = irc_composite_c_r0195_elevation_interval;
+const int firstElevation = irc_composite_c_r0195_first_elevation;
+const int numberOfElevations = MOZ_ARRAY_LENGTH(irc_composite_c_r0195);
+
+const unsigned HRTFElevation::NumberOfTotalAzimuths = 360 / 15 * 8;
+
+const int rawSampleRate = irc_composite_c_r0195_sample_rate;
+
+// Number of frames in an individual impulse response.
+const size_t ResponseFrameSize = 256;
+
+size_t HRTFElevation::sizeOfIncludingThis(
+    mozilla::MallocSizeOf aMallocSizeOf) const {
+  size_t amount = aMallocSizeOf(this);
+
+  amount += m_kernelListL.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < m_kernelListL.Length(); i++) {
+    amount += m_kernelListL[i]->sizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  return amount;
+}
+
+size_t HRTFElevation::fftSizeForSampleRate(float sampleRate) {
+  // The IRCAM HRTF impulse responses were 512 sample-frames @44.1KHz,
+  // but these have been truncated to 256 samples.
+  // An FFT-size of twice impulse response size is used (for convolution).
+  // So for sample rates of 44.1KHz an FFT size of 512 is good.
+  // We double the FFT-size only for sample rates at least double this.
+  // If the FFT size is too large then the impulse response will be padded
+  // with zeros without the fade-out provided by HRTFKernel.
+  MOZ_ASSERT(sampleRate > 1.0 && sampleRate < 1048576.0);
+
+  // This is the size if we were to use all raw response samples.
+  unsigned resampledLength =
+      floorf(ResponseFrameSize * sampleRate / rawSampleRate);
+  // Keep things semi-sane, with max FFT size of 1024.
+  unsigned size = std::min(resampledLength, 1023U);
+  // Ensure a minimum of 2 * WEBAUDIO_BLOCK_SIZE (with the size++ below) for
+  // FFTConvolver and set the 8 least significant bits for rounding up to
+  // the next power of 2 below.
+  size |= 2 * WEBAUDIO_BLOCK_SIZE - 1;
+  // Round up to the next power of 2, making the FFT size no more than twice
+  // the impulse response length.  This doubles size for values that are
+  // already powers of 2.  This works by filling in alls bit to right of the
+  // most significant bit.  The most significant bit is no greater than
+  // 1 << 9, and the least significant 8 bits were already set above, so
+  // there is at most one bit to add.
+  size |= (size >> 1);
+  size++;
+  MOZ_ASSERT((size & (size - 1)) == 0);
+
+  return size;
+}
+
+nsReturnRef<HRTFKernel> HRTFElevation::calculateKernelForAzimuthElevation(
+    int azimuth, int elevation, SpeexResamplerState* resampler,
+    float sampleRate) {
+  int elevationIndex = (elevation - firstElevation) / elevationSpacing;
+  MOZ_ASSERT(elevationIndex >= 0 && elevationIndex <= numberOfElevations);
+
+  int numberOfAzimuths = irc_composite_c_r0195[elevationIndex].count;
+  int azimuthSpacing = 360 / numberOfAzimuths;
+  MOZ_ASSERT(numberOfAzimuths * azimuthSpacing == 360);
+
+  int azimuthIndex = azimuth / azimuthSpacing;
+  MOZ_ASSERT(azimuthIndex * azimuthSpacing == azimuth);
+
+  const int16_t(&impulse_response_data)[ResponseFrameSize] =
+      irc_composite_c_r0195[elevationIndex].azimuths[azimuthIndex];
+
+  float response[ResponseFrameSize];
+  ConvertAudioSamples(impulse_response_data, response, ResponseFrameSize);
+  float* resampledResponse;
+
+  // Note that depending on the fftSize returned by the panner, we may be
+  // truncating the impulse response.
+  const size_t resampledResponseLength = fftSizeForSampleRate(sampleRate) / 2;
+
+  AutoTArray<AudioDataValue, 2 * ResponseFrameSize> resampled;
+  if (sampleRate == rawSampleRate) {
+    resampledResponse = response;
+    MOZ_ASSERT(resampledResponseLength == ResponseFrameSize);
+  } else {
+    resampled.SetLength(resampledResponseLength);
+    resampledResponse = resampled.Elements();
+    speex_resampler_skip_zeros(resampler);
+
+    // Feed the input buffer into the resampler.
+    spx_uint32_t in_len = ResponseFrameSize;
+    spx_uint32_t out_len = resampled.Length();
+    speex_resampler_process_float(resampler, 0, response, &in_len,
+                                  resampled.Elements(), &out_len);
+
+    if (out_len < resampled.Length()) {
+      // The input should have all been processed.
+      MOZ_ASSERT(in_len == ResponseFrameSize);
+      // Feed in zeros get the data remaining in the resampler.
+      spx_uint32_t out_index = out_len;
+      in_len = speex_resampler_get_input_latency(resampler);
+      out_len = resampled.Length() - out_index;
+      speex_resampler_process_float(resampler, 0, nullptr, &in_len,
+                                    resampled.Elements() + out_index, &out_len);
+      out_index += out_len;
+      // There may be some uninitialized samples remaining for very low
+      // sample rates.
+      PodZero(resampled.Elements() + out_index, resampled.Length() - out_index);
+    }
+
+    speex_resampler_reset_mem(resampler);
+  }
+
+  return HRTFKernel::create(resampledResponse, resampledResponseLength,
+                            sampleRate);
+}
+
+// The range of elevations for the IRCAM impulse responses varies depending on
+// azimuth, but the minimum elevation appears to always be -45.
+//
+// Here's how it goes:
+static int maxElevations[] = {
+    //  Azimuth
+    //
+    90,  // 0
+    45,  // 15
+    60,  // 30
+    45,  // 45
+    75,  // 60
+    45,  // 75
+    60,  // 90
+    45,  // 105
+    75,  // 120
+    45,  // 135
+    60,  // 150
+    45,  // 165
+    75,  // 180
+    45,  // 195
+    60,  // 210
+    45,  // 225
+    75,  // 240
+    45,  // 255
+    60,  // 270
+    45,  // 285
+    75,  // 300
+    45,  // 315
+    60,  // 330
+    45   //  345
+};
+
+nsReturnRef<HRTFElevation> HRTFElevation::createBuiltin(int elevation,
+                                                        float sampleRate) {
+  if (elevation < firstElevation ||
+      elevation > firstElevation + numberOfElevations * elevationSpacing ||
+      (elevation / elevationSpacing) * elevationSpacing != elevation)
+    return nsReturnRef<HRTFElevation>();
+
+  // Spacing, in degrees, between every azimuth loaded from resource.
+  // Some elevations do not have data for all these intervals.
+  // See maxElevations.
+  static const unsigned AzimuthSpacing = 15;
+  static const unsigned NumberOfRawAzimuths = 360 / AzimuthSpacing;
+  static_assert(AzimuthSpacing * NumberOfRawAzimuths == 360, "Not a multiple");
+  static const unsigned InterpolationFactor =
+      NumberOfTotalAzimuths / NumberOfRawAzimuths;
+  static_assert(
+      NumberOfTotalAzimuths == NumberOfRawAzimuths * InterpolationFactor,
+      "Not a multiple");
+
+  HRTFKernelList kernelListL;
+  kernelListL.SetLength(NumberOfTotalAzimuths);
+
+  SpeexResamplerState* resampler =
+      sampleRate == rawSampleRate
+          ? nullptr
+          : speex_resampler_init(1, rawSampleRate, sampleRate,
+                                 SPEEX_RESAMPLER_QUALITY_MIN, nullptr);
+
+  // Load convolution kernels from HRTF files.
+  int interpolatedIndex = 0;
+  for (unsigned rawIndex = 0; rawIndex < NumberOfRawAzimuths; ++rawIndex) {
+    // Don't let elevation exceed maximum for this azimuth.
+    int maxElevation = maxElevations[rawIndex];
+    int actualElevation = std::min(elevation, maxElevation);
+
+    kernelListL[interpolatedIndex] = calculateKernelForAzimuthElevation(
+        rawIndex * AzimuthSpacing, actualElevation, resampler, sampleRate);
+
+    interpolatedIndex += InterpolationFactor;
+  }
+
+  if (resampler) speex_resampler_destroy(resampler);
+
+  // Now go back and interpolate intermediate azimuth values.
+  for (unsigned i = 0; i < NumberOfTotalAzimuths; i += InterpolationFactor) {
+    int j = (i + InterpolationFactor) % NumberOfTotalAzimuths;
+
+    // Create the interpolated convolution kernels and delays.
+    for (unsigned jj = 1; jj < InterpolationFactor; ++jj) {
+      float x =
+          float(jj) / float(InterpolationFactor);  // interpolate from 0 -> 1
+
+      kernelListL[i + jj] = HRTFKernel::createInterpolatedKernel(
+          kernelListL[i], kernelListL[j], x);
+    }
+  }
+
+  return nsReturnRef<HRTFElevation>(
+      new HRTFElevation(std::move(kernelListL), elevation, sampleRate));
+}
+
+nsReturnRef<HRTFElevation> HRTFElevation::createByInterpolatingSlices(
+    HRTFElevation* hrtfElevation1, HRTFElevation* hrtfElevation2, float x,
+    float sampleRate) {
+  MOZ_ASSERT(hrtfElevation1 && hrtfElevation2);
+  if (!hrtfElevation1 || !hrtfElevation2) return nsReturnRef<HRTFElevation>();
+
+  MOZ_ASSERT(x >= 0.0 && x < 1.0);
+
+  HRTFKernelList kernelListL;
+  kernelListL.SetLength(NumberOfTotalAzimuths);
+
+  const HRTFKernelList& kernelListL1 = hrtfElevation1->kernelListL();
+  const HRTFKernelList& kernelListL2 = hrtfElevation2->kernelListL();
+
+  // Interpolate kernels of corresponding azimuths of the two elevations.
+  for (unsigned i = 0; i < NumberOfTotalAzimuths; ++i) {
+    kernelListL[i] = HRTFKernel::createInterpolatedKernel(kernelListL1[i],
+                                                          kernelListL2[i], x);
+  }
+
+  // Interpolate elevation angle.
+  double angle = (1.0 - x) * hrtfElevation1->elevationAngle() +
+                 x * hrtfElevation2->elevationAngle();
+
+  return nsReturnRef<HRTFElevation>(new HRTFElevation(
+      std::move(kernelListL), static_cast<int>(angle), sampleRate));
+}
+
+void HRTFElevation::getKernelsFromAzimuth(
+    double azimuthBlend, unsigned azimuthIndex, HRTFKernel*& kernelL,
+    HRTFKernel*& kernelR, double& frameDelayL, double& frameDelayR) {
+  bool checkAzimuthBlend = azimuthBlend >= 0.0 && azimuthBlend < 1.0;
+  MOZ_ASSERT(checkAzimuthBlend);
+  if (!checkAzimuthBlend) azimuthBlend = 0.0;
+
+  unsigned numKernels = m_kernelListL.Length();
+
+  bool isIndexGood = azimuthIndex < numKernels;
+  MOZ_ASSERT(isIndexGood);
+  if (!isIndexGood) {
+    kernelL = 0;
+    kernelR = 0;
+    return;
+  }
+
+  // Return the left and right kernels,
+  // using symmetry to produce the right kernel.
+  kernelL = m_kernelListL[azimuthIndex];
+  int azimuthIndexR = (numKernels - azimuthIndex) % numKernels;
+  kernelR = m_kernelListL[azimuthIndexR];
+
+  frameDelayL = kernelL->frameDelay();
+  frameDelayR = kernelR->frameDelay();
+
+  int azimuthIndex2L = (azimuthIndex + 1) % numKernels;
+  double frameDelay2L = m_kernelListL[azimuthIndex2L]->frameDelay();
+  int azimuthIndex2R = (numKernels - azimuthIndex2L) % numKernels;
+  double frameDelay2R = m_kernelListL[azimuthIndex2R]->frameDelay();
+
+  // Linearly interpolate delays.
+  frameDelayL =
+      (1.0 - azimuthBlend) * frameDelayL + azimuthBlend * frameDelay2L;
+  frameDelayR =
+      (1.0 - azimuthBlend) * frameDelayR + azimuthBlend * frameDelay2R;
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/HRTFElevation.h b/dom/media/webaudio/blink/HRTFElevation.h
new file mode 100644
index 0000000000..cd54bbc04a
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFElevation.h
@@ -0,0 +1,114 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef HRTFElevation_h
+#define HRTFElevation_h
+
+#include "HRTFKernel.h"
+#include "nsAutoRef.h"
+#include "mozilla/MemoryReporting.h"
+
+struct SpeexResamplerState_;
+typedef struct SpeexResamplerState_ SpeexResamplerState;
+
+namespace WebCore {
+
+// HRTFElevation contains all of the HRTFKernels (one left ear and one right ear
+// per azimuth angle) for a particular elevation.
+
+class HRTFElevation {
+ public:
+  // Loads and returns an HRTFElevation with the given HRTF database subject
+  // name and elevation from browser (or WebKit.framework) resources. Normally,
+  // there will only be a single HRTF database set, but this API supports the
+  // possibility of multiple ones with different names. Interpolated azimuths
+  // will be generated based on InterpolationFactor. Valid values for elevation
+  // are -45 -> +90 in 15 degree increments.
+  static nsReturnRef<HRTFElevation> createBuiltin(int elevation,
+                                                  float sampleRate);
+
+  // Given two HRTFElevations, and an interpolation factor x: 0 -> 1, returns an
+  // interpolated HRTFElevation.
+  static nsReturnRef<HRTFElevation> createByInterpolatingSlices(
+      HRTFElevation* hrtfElevation1, HRTFElevation* hrtfElevation2, float x,
+      float sampleRate);
+
+  double elevationAngle() const { return m_elevationAngle; }
+  unsigned numberOfAzimuths() const { return NumberOfTotalAzimuths; }
+  float sampleRate() const { return m_sampleRate; }
+
+  // Returns the left and right kernels for the given azimuth index.
+  // The interpolated delays based on azimuthBlend: 0 -> 1 are returned in
+  // frameDelayL and frameDelayR.
+  void getKernelsFromAzimuth(double azimuthBlend, unsigned azimuthIndex,
+                             HRTFKernel*& kernelL, HRTFKernel*& kernelR,
+                             double& frameDelayL, double& frameDelayR);
+
+  // Total number of azimuths after interpolation.
+  static const unsigned NumberOfTotalAzimuths;
+
+  static size_t fftSizeForSampleRate(float sampleRate);
+
+  size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+ private:
+  HRTFElevation(const HRTFElevation& other) = delete;
+  void operator=(const HRTFElevation& other) = delete;
+
+  HRTFElevation(HRTFKernelList&& kernelListL, int elevation, float sampleRate)
+      : m_kernelListL(std::move(kernelListL)),
+        m_elevationAngle(elevation),
+        m_sampleRate(sampleRate) {}
+
+  // Returns the list of left ear HRTFKernels for all the azimuths going from 0
+  // to 360 degrees.
+  const HRTFKernelList& kernelListL() { return m_kernelListL; }
+
+  // Given a specific azimuth and elevation angle, returns the left HRTFKernel.
+  // Values for azimuth must be multiples of 15 in 0 -> 345,
+  // but not all azimuths are available for elevations > +45.
+  // Valid values for elevation are -45 -> +90 in 15 degree increments.
+  static nsReturnRef<HRTFKernel> calculateKernelForAzimuthElevation(
+      int azimuth, int elevation, SpeexResamplerState* resampler,
+      float sampleRate);
+
+  HRTFKernelList m_kernelListL;
+  double m_elevationAngle;
+  float m_sampleRate;
+};
+
+}  // namespace WebCore
+
+template <>
+class nsAutoRefTraits<WebCore::HRTFElevation>
+    : public nsPointerRefTraits<WebCore::HRTFElevation> {
+ public:
+  static void Release(WebCore::HRTFElevation* ptr) { delete (ptr); }
+};
+
+#endif  // HRTFElevation_h
diff --git a/dom/media/webaudio/blink/HRTFKernel.cpp b/dom/media/webaudio/blink/HRTFKernel.cpp
new file mode 100644
index 0000000000..ecaa846a66
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFKernel.cpp
@@ -0,0 +1,109 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "HRTFKernel.h"
+namespace WebCore {
+
+// Takes the input audio channel |impulseP| as an input impulse response and
+// calculates the average group delay. This represents the initial delay before
+// the most energetic part of the impulse response. The sample-frame delay is
+// removed from the |impulseP| impulse response, and this value  is returned.
+// The |length| of the passed in |impulseP| must be must be a power of 2.
+static float extractAverageGroupDelay(float* impulseP, size_t length) {
+  // Check for power-of-2.
+  MOZ_ASSERT(length && (length & (length - 1)) == 0);
+
+  FFTBlock estimationFrame(length);
+  estimationFrame.PerformFFT(impulseP);
+
+  float frameDelay =
+      static_cast<float>(estimationFrame.ExtractAverageGroupDelay());
+  estimationFrame.GetInverse(impulseP);
+
+  return frameDelay;
+}
+
+HRTFKernel::HRTFKernel(float* impulseResponse, size_t length, float sampleRate)
+    : m_frameDelay(0), m_sampleRate(sampleRate) {
+  AlignedTArray<float> buffer;
+  // copy to a 32-byte aligned buffer
+  if (((uintptr_t)impulseResponse & 31) != 0) {
+    buffer.SetLength(length);
+    mozilla::PodCopy(buffer.Elements(), impulseResponse, length);
+    impulseResponse = buffer.Elements();
+  }
+
+  // Determine the leading delay (average group delay) for the response.
+  m_frameDelay = extractAverageGroupDelay(impulseResponse, length);
+
+  // The FFT size (with zero padding) needs to be twice the response length
+  // in order to do proper convolution.
+  unsigned fftSize = 2 * length;
+
+  // Quick fade-out (apply window) at truncation point
+  // because the impulse response has been truncated.
+  unsigned numberOfFadeOutFrames = static_cast<unsigned>(
+      sampleRate / 4410);  // 10 sample-frames @44.1KHz sample-rate
+  MOZ_ASSERT(numberOfFadeOutFrames < length);
+  if (numberOfFadeOutFrames < length) {
+    for (unsigned i = length - numberOfFadeOutFrames; i < length; ++i) {
+      float x =
+          1.0f - static_cast<float>(i - (length - numberOfFadeOutFrames)) /
+                     numberOfFadeOutFrames;
+      impulseResponse[i] *= x;
+    }
+  }
+
+  m_fftFrame = mozilla::MakeUnique<FFTBlock>(fftSize);
+  m_fftFrame->PadAndMakeScaledDFT(impulseResponse, length);
+}
+
+// Interpolates two kernels with x: 0 -> 1 and returns the result.
+nsReturnRef<HRTFKernel> HRTFKernel::createInterpolatedKernel(
+    HRTFKernel* kernel1, HRTFKernel* kernel2, float x) {
+  MOZ_ASSERT(kernel1 && kernel2);
+  if (!kernel1 || !kernel2) return nsReturnRef<HRTFKernel>();
+
+  MOZ_ASSERT(x >= 0.0 && x < 1.0);
+  x = mozilla::clamped(x, 0.0f, 1.0f);
+
+  float sampleRate1 = kernel1->sampleRate();
+  float sampleRate2 = kernel2->sampleRate();
+  MOZ_ASSERT(sampleRate1 == sampleRate2);
+  if (sampleRate1 != sampleRate2) return nsReturnRef<HRTFKernel>();
+
+  float frameDelay =
+      (1 - x) * kernel1->frameDelay() + x * kernel2->frameDelay();
+
+  UniquePtr<FFTBlock> interpolatedFrame(FFTBlock::CreateInterpolatedBlock(
+      *kernel1->fftFrame(), *kernel2->fftFrame(), x));
+  return HRTFKernel::create(std::move(interpolatedFrame), frameDelay,
+                            sampleRate1);
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/HRTFKernel.h b/dom/media/webaudio/blink/HRTFKernel.h
new file mode 100644
index 0000000000..34479f6301
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFKernel.h
@@ -0,0 +1,129 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef HRTFKernel_h
+#define HRTFKernel_h
+
+#include "nsAutoRef.h"
+#include "nsTArray.h"
+#include "mozilla/FFTBlock.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/UniquePtr.h"
+
+namespace WebCore {
+
+using mozilla::FFTBlock;
+using mozilla::UniquePtr;
+
+// HRTF stands for Head-Related Transfer Function.
+// HRTFKernel is a frequency-domain representation of an impulse-response used
+// as part of the spatialized panning system. For a given azimuth / elevation
+// angle there will be one HRTFKernel for the left ear transfer function, and
+// one for the right ear. The leading delay (average group delay) for each
+// impulse response is extracted:
+//      m_fftFrame is the frequency-domain representation of the impulse
+//                 response with the delay removed
+//      m_frameDelay is the leading delay of the original impulse response.
+class HRTFKernel {
+ public:
+  // Note: this is destructive on the passed in |impulseResponse|.
+  // The |length| of |impulseResponse| must be a power of two.
+  // The size of the DFT will be |2 * length|.
+  static nsReturnRef<HRTFKernel> create(float* impulseResponse, size_t length,
+                                        float sampleRate);
+
+  static nsReturnRef<HRTFKernel> create(UniquePtr<FFTBlock> fftFrame,
+                                        float frameDelay, float sampleRate);
+
+  // Given two HRTFKernels, and an interpolation factor x: 0 -> 1, returns an
+  // interpolated HRTFKernel.
+  static nsReturnRef<HRTFKernel> createInterpolatedKernel(HRTFKernel* kernel1,
+                                                          HRTFKernel* kernel2,
+                                                          float x);
+
+  FFTBlock* fftFrame() { return m_fftFrame.get(); }
+
+  size_t fftSize() const { return m_fftFrame->FFTSize(); }
+  float frameDelay() const { return m_frameDelay; }
+
+  float sampleRate() const { return m_sampleRate; }
+  double nyquist() const { return 0.5 * sampleRate(); }
+
+  size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const {
+    size_t amount = aMallocSizeOf(this);
+    amount += m_fftFrame->SizeOfIncludingThis(aMallocSizeOf);
+    return amount;
+  }
+
+ private:
+  HRTFKernel(const HRTFKernel& other) = delete;
+  void operator=(const HRTFKernel& other) = delete;
+
+  // Note: this is destructive on the passed in |impulseResponse|.
+  HRTFKernel(float* impulseResponse, size_t fftSize, float sampleRate);
+
+  HRTFKernel(UniquePtr<FFTBlock> fftFrame, float frameDelay, float sampleRate)
+      : m_fftFrame(std::move(fftFrame)),
+        m_frameDelay(frameDelay),
+        m_sampleRate(sampleRate) {}
+
+  UniquePtr<FFTBlock> m_fftFrame;
+  float m_frameDelay;
+  float m_sampleRate;
+};
+
+typedef nsTArray<nsAutoRef<HRTFKernel> > HRTFKernelList;
+
+}  // namespace WebCore
+
+template <>
+class nsAutoRefTraits<WebCore::HRTFKernel>
+    : public nsPointerRefTraits<WebCore::HRTFKernel> {
+ public:
+  static void Release(WebCore::HRTFKernel* ptr) { delete (ptr); }
+};
+
+namespace WebCore {
+
+inline nsReturnRef<HRTFKernel> HRTFKernel::create(float* impulseResponse,
+                                                  size_t length,
+                                                  float sampleRate) {
+  return nsReturnRef<HRTFKernel>(
+      new HRTFKernel(impulseResponse, length, sampleRate));
+}
+
+inline nsReturnRef<HRTFKernel> HRTFKernel::create(UniquePtr<FFTBlock> fftFrame,
+                                                  float frameDelay,
+                                                  float sampleRate) {
+  return nsReturnRef<HRTFKernel>(
+      new HRTFKernel(std::move(fftFrame), frameDelay, sampleRate));
+}
+
+}  // namespace WebCore
+
+#endif  // HRTFKernel_h
diff --git a/dom/media/webaudio/blink/HRTFPanner.cpp b/dom/media/webaudio/blink/HRTFPanner.cpp
new file mode 100644
index 0000000000..3d01fb5b39
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFPanner.cpp
@@ -0,0 +1,328 @@
+/*
+ * Copyright (C) 2010, Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1.  Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+ * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "HRTFPanner.h"
+#include "HRTFDatabaseLoader.h"
+
+#include "FFTConvolver.h"
+#include "HRTFDatabase.h"
+#include "AudioBlock.h"
+
+using namespace mozilla;
+using dom::ChannelInterpretation;
+
+namespace WebCore {
+
+// The value of 2 milliseconds is larger than the largest delay which exists in
+// any HRTFKernel from the default HRTFDatabase (0.0136 seconds). We ASSERT the
+// delay values used in process() with this value.
+const float MaxDelayTimeSeconds = 0.002f;
+
+const int UninitializedAzimuth = -1;
+
+HRTFPanner::HRTFPanner(float sampleRate,
+                       already_AddRefed<HRTFDatabaseLoader> databaseLoader)
+    : m_databaseLoader(databaseLoader),
+      m_sampleRate(sampleRate),
+      m_crossfadeSelection(CrossfadeSelection1),
+      m_azimuthIndex1(UninitializedAzimuth),
+      m_azimuthIndex2(UninitializedAzimuth)
+      // m_elevation1 and m_elevation2 are initialized in pan()
+      ,
+      m_crossfadeX(0),
+      m_crossfadeIncr(0),
+      m_convolverL1(HRTFElevation::fftSizeForSampleRate(sampleRate)),
+      m_convolverR1(m_convolverL1.fftSize()),
+      m_convolverL2(m_convolverL1.fftSize()),
+      m_convolverR2(m_convolverL1.fftSize()),
+      m_delayLine(MaxDelayTimeSeconds * sampleRate) {
+  MOZ_ASSERT(m_databaseLoader);
+  MOZ_COUNT_CTOR(HRTFPanner);
+}
+
+HRTFPanner::~HRTFPanner() { MOZ_COUNT_DTOR(HRTFPanner); }
+
+size_t HRTFPanner::sizeOfIncludingThis(
+    mozilla::MallocSizeOf aMallocSizeOf) const {
+  size_t amount = aMallocSizeOf(this);
+
+  // NB: m_databaseLoader can be shared, so it is not measured here
+  amount += m_convolverL1.sizeOfExcludingThis(aMallocSizeOf);
+  amount += m_convolverR1.sizeOfExcludingThis(aMallocSizeOf);
+  amount += m_convolverL2.sizeOfExcludingThis(aMallocSizeOf);
+  amount += m_convolverR2.sizeOfExcludingThis(aMallocSizeOf);
+  amount += m_delayLine.SizeOfExcludingThis(aMallocSizeOf);
+
+  return amount;
+}
+
+void HRTFPanner::reset() {
+  m_azimuthIndex1 = UninitializedAzimuth;
+  m_azimuthIndex2 = UninitializedAzimuth;
+  // m_elevation1 and m_elevation2 are initialized in pan()
+  m_crossfadeSelection = CrossfadeSelection1;
+  m_crossfadeX = 0.0f;
+  m_crossfadeIncr = 0.0f;
+  m_convolverL1.reset();
+  m_convolverR1.reset();
+  m_convolverL2.reset();
+  m_convolverR2.reset();
+  m_delayLine.Reset();
+}
+
+int HRTFPanner::calculateDesiredAzimuthIndexAndBlend(double azimuth,
+                                                     double& azimuthBlend) {
+  // Convert the azimuth angle from the range -180 -> +180 into the range 0 ->
+  // 360. The azimuth index may then be calculated from this positive value.
+  if (azimuth < 0) azimuth += 360.0;
+
+  int numberOfAzimuths = HRTFDatabase::numberOfAzimuths();
+  const double angleBetweenAzimuths = 360.0 / numberOfAzimuths;
+
+  // Calculate the azimuth index and the blend (0 -> 1) for interpolation.
+  double desiredAzimuthIndexFloat = azimuth / angleBetweenAzimuths;
+  int desiredAzimuthIndex = static_cast<int>(desiredAzimuthIndexFloat);
+  azimuthBlend =
+      desiredAzimuthIndexFloat - static_cast<double>(desiredAzimuthIndex);
+
+  // We don't immediately start using this azimuth index, but instead approach
+  // this index from the last index we rendered at. This minimizes the clicks
+  // and graininess for moving sources which occur otherwise.
+  desiredAzimuthIndex = std::max(0, desiredAzimuthIndex);
+  desiredAzimuthIndex = std::min(numberOfAzimuths - 1, desiredAzimuthIndex);
+  return desiredAzimuthIndex;
+}
+
+void HRTFPanner::pan(double desiredAzimuth, double elevation,
+                     const AudioBlock* inputBus, AudioBlock* outputBus) {
+#ifdef DEBUG
+  unsigned numInputChannels = inputBus->IsNull() ? 0 : inputBus->ChannelCount();
+
+  MOZ_ASSERT(numInputChannels <= 2);
+  MOZ_ASSERT(inputBus->GetDuration() == WEBAUDIO_BLOCK_SIZE);
+#endif
+
+  bool isOutputGood = outputBus && outputBus->ChannelCount() == 2 &&
+                      outputBus->GetDuration() == WEBAUDIO_BLOCK_SIZE;
+  MOZ_ASSERT(isOutputGood);
+
+  if (!isOutputGood) {
+    if (outputBus) outputBus->SetNull(outputBus->GetDuration());
+    return;
+  }
+
+  HRTFDatabase* database = m_databaseLoader->database();
+  if (!database) {  // not yet loaded
+    outputBus->SetNull(outputBus->GetDuration());
+    return;
+  }
+
+  // IRCAM HRTF azimuths values from the loaded database is reversed from the
+  // panner's notion of azimuth.
+  double azimuth = -desiredAzimuth;
+
+  bool isAzimuthGood = azimuth >= -180.0 && azimuth <= 180.0;
+  MOZ_ASSERT(isAzimuthGood);
+  if (!isAzimuthGood) {
+    outputBus->SetNull(outputBus->GetDuration());
+    return;
+  }
+
+  // Normally, we'll just be dealing with mono sources.
+  // If we have a stereo input, implement stereo panning with left source
+  // processed by left HRTF, and right source by right HRTF.
+
+  // Get destination pointers.
+  float* destinationL =
+      static_cast<float*>(const_cast<void*>(outputBus->mChannelData[0]));
+  float* destinationR =
+      static_cast<float*>(const_cast<void*>(outputBus->mChannelData[1]));
+
+  double azimuthBlend;
+  int desiredAzimuthIndex =
+      calculateDesiredAzimuthIndexAndBlend(azimuth, azimuthBlend);
+
+  // Initially snap azimuth and elevation values to first values encountered.
+  if (m_azimuthIndex1 == UninitializedAzimuth) {
+    m_azimuthIndex1 = desiredAzimuthIndex;
+    m_elevation1 = elevation;
+  }
+  if (m_azimuthIndex2 == UninitializedAzimuth) {
+    m_azimuthIndex2 = desiredAzimuthIndex;
+    m_elevation2 = elevation;
+  }
+
+  // Cross-fade / transition over a period of around 45 milliseconds.
+  // This is an empirical value tuned to be a reasonable trade-off between
+  // smoothness and speed.
+  const double fadeFrames = sampleRate() <= 48000 ? 2048 : 4096;
+
+  // Check for azimuth and elevation changes, initiating a cross-fade if needed.
+  if (!m_crossfadeX && m_crossfadeSelection == CrossfadeSelection1) {
+    if (desiredAzimuthIndex != m_azimuthIndex1 || elevation != m_elevation1) {
+      // Cross-fade from 1 -> 2
+      m_crossfadeIncr = 1 / fadeFrames;
+      m_azimuthIndex2 = desiredAzimuthIndex;
+      m_elevation2 = elevation;
+    }
+  }
+  if (m_crossfadeX == 1 && m_crossfadeSelection == CrossfadeSelection2) {
+    if (desiredAzimuthIndex != m_azimuthIndex2 || elevation != m_elevation2) {
+      // Cross-fade from 2 -> 1
+      m_crossfadeIncr = -1 / fadeFrames;
+      m_azimuthIndex1 = desiredAzimuthIndex;
+      m_elevation1 = elevation;
+    }
+  }
+
+  // Get the HRTFKernels and interpolated delays.
+  HRTFKernel* kernelL1;
+  HRTFKernel* kernelR1;
+  HRTFKernel* kernelL2;
+  HRTFKernel* kernelR2;
+  double frameDelayL1;
+  double frameDelayR1;
+  double frameDelayL2;
+  double frameDelayR2;
+  database->getKernelsFromAzimuthElevation(azimuthBlend, m_azimuthIndex1,
+                                           m_elevation1, kernelL1, kernelR1,
+                                           frameDelayL1, frameDelayR1);
+  database->getKernelsFromAzimuthElevation(azimuthBlend, m_azimuthIndex2,
+                                           m_elevation2, kernelL2, kernelR2,
+                                           frameDelayL2, frameDelayR2);
+
+  bool areKernelsGood = kernelL1 && kernelR1 && kernelL2 && kernelR2;
+  MOZ_ASSERT(areKernelsGood);
+  if (!areKernelsGood) {
+    outputBus->SetNull(outputBus->GetDuration());
+    return;
+  }
+
+  MOZ_ASSERT(frameDelayL1 / sampleRate() < MaxDelayTimeSeconds &&
+             frameDelayR1 / sampleRate() < MaxDelayTimeSeconds);
+  MOZ_ASSERT(frameDelayL2 / sampleRate() < MaxDelayTimeSeconds &&
+             frameDelayR2 / sampleRate() < MaxDelayTimeSeconds);
+
+  // Crossfade inter-aural delays based on transitions.
+  float frameDelaysL[WEBAUDIO_BLOCK_SIZE];
+  float frameDelaysR[WEBAUDIO_BLOCK_SIZE];
+  {
+    float x = m_crossfadeX;
+    float incr = m_crossfadeIncr;
+    for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+      frameDelaysL[i] = (1 - x) * frameDelayL1 + x * frameDelayL2;
+      frameDelaysR[i] = (1 - x) * frameDelayR1 + x * frameDelayR2;
+      x += incr;
+    }
+  }
+
+  // First run through delay lines for inter-aural time difference.
+  m_delayLine.Write(*inputBus);
+  // "Speakers" means a mono input is read into both outputs (with possibly
+  // different delays).
+  m_delayLine.ReadChannel(frameDelaysL, outputBus, 0,
+                          ChannelInterpretation::Speakers);
+  m_delayLine.ReadChannel(frameDelaysR, outputBus, 1,
+                          ChannelInterpretation::Speakers);
+  m_delayLine.NextBlock();
+
+  bool needsCrossfading = m_crossfadeIncr;
+
+  const float* convolutionDestinationL1;
+  const float* convolutionDestinationR1;
+  const float* convolutionDestinationL2;
+  const float* convolutionDestinationR2;
+
+  // Now do the convolutions.
+  // Note that we avoid doing convolutions on both sets of convolvers if we're
+  // not currently cross-fading.
+
+  if (m_crossfadeSelection == CrossfadeSelection1 || needsCrossfading) {
+    convolutionDestinationL1 =
+        m_convolverL1.process(kernelL1->fftFrame(), destinationL);
+    convolutionDestinationR1 =
+        m_convolverR1.process(kernelR1->fftFrame(), destinationR);
+  }
+
+  if (m_crossfadeSelection == CrossfadeSelection2 || needsCrossfading) {
+    convolutionDestinationL2 =
+        m_convolverL2.process(kernelL2->fftFrame(), destinationL);
+    convolutionDestinationR2 =
+        m_convolverR2.process(kernelR2->fftFrame(), destinationR);
+  }
+
+  if (needsCrossfading) {
+    // Apply linear cross-fade.
+    float x = m_crossfadeX;
+    float incr = m_crossfadeIncr;
+    for (unsigned i = 0; i < WEBAUDIO_BLOCK_SIZE; ++i) {
+      destinationL[i] = (1 - x) * convolutionDestinationL1[i] +
+                        x * convolutionDestinationL2[i];
+      destinationR[i] = (1 - x) * convolutionDestinationR1[i] +
+                        x * convolutionDestinationR2[i];
+      x += incr;
+    }
+    // Update cross-fade value from local.
+    m_crossfadeX = x;
+
+    if (m_crossfadeIncr > 0 && fabs(m_crossfadeX - 1) < m_crossfadeIncr) {
+      // We've fully made the crossfade transition from 1 -> 2.
+      m_crossfadeSelection = CrossfadeSelection2;
+      m_crossfadeX = 1;
+      m_crossfadeIncr = 0;
+    } else if (m_crossfadeIncr < 0 && fabs(m_crossfadeX) < -m_crossfadeIncr) {
+      // We've fully made the crossfade transition from 2 -> 1.
+      m_crossfadeSelection = CrossfadeSelection1;
+      m_crossfadeX = 0;
+      m_crossfadeIncr = 0;
+    }
+  } else {
+    const float* sourceL;
+    const float* sourceR;
+    if (m_crossfadeSelection == CrossfadeSelection1) {
+      sourceL = convolutionDestinationL1;
+      sourceR = convolutionDestinationR1;
+    } else {
+      sourceL = convolutionDestinationL2;
+      sourceR = convolutionDestinationR2;
+    }
+    PodCopy(destinationL, sourceL, WEBAUDIO_BLOCK_SIZE);
+    PodCopy(destinationR, sourceR, WEBAUDIO_BLOCK_SIZE);
+  }
+}
+
+int HRTFPanner::maxTailFrames() const {
+  // Although the ideal tail time would be the length of the impulse
+  // response, there is additional tail time from the approximations in the
+  // implementation.  Because HRTFPanner is implemented with a DelayKernel
+  // and a FFTConvolver, the tailTime of the HRTFPanner is the sum of the
+  // tailTime of the DelayKernel and the tailTime of the FFTConvolver.  The
+  // FFTs of the convolver are fftSize(), half of which is latency, but this
+  // is aligned with blocks and so is reduced by the one block which is
+  // processed immediately.
+  return m_delayLine.MaxDelayTicks() + m_convolverL1.fftSize() / 2 +
+         m_convolverL1.latencyFrames();
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/HRTFPanner.h b/dom/media/webaudio/blink/HRTFPanner.h
new file mode 100644
index 0000000000..b30689557b
--- /dev/null
+++ b/dom/media/webaudio/blink/HRTFPanner.h
@@ -0,0 +1,120 @@
+/*
+ * Copyright (C) 2010, Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1.  Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR
+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
+ * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef HRTFPanner_h
+#define HRTFPanner_h
+
+#include "FFTConvolver.h"
+#include "DelayBuffer.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace mozilla {
+class AudioBlock;
+}  // namespace mozilla
+
+namespace WebCore {
+
+typedef nsTArray<float> AudioFloatArray;
+
+class HRTFDatabaseLoader;
+
+using mozilla::AudioBlock;
+
+class HRTFPanner {
+ public:
+  HRTFPanner(float sampleRate,
+             already_AddRefed<HRTFDatabaseLoader> databaseLoader);
+  ~HRTFPanner();
+
+  // chunk durations must be 128
+  void pan(double azimuth, double elevation, const AudioBlock* inputBus,
+           AudioBlock* outputBus);
+  void reset();
+
+  size_t fftSize() const { return m_convolverL1.fftSize(); }
+
+  float sampleRate() const { return m_sampleRate; }
+
+  int maxTailFrames() const;
+
+  size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+ private:
+  // Given an azimuth angle in the range -180 -> +180, returns the corresponding
+  // azimuth index for the database, and azimuthBlend which is an interpolation
+  // value from 0 -> 1.
+  int calculateDesiredAzimuthIndexAndBlend(double azimuth,
+                                           double& azimuthBlend);
+
+  RefPtr<HRTFDatabaseLoader> m_databaseLoader;
+
+  float m_sampleRate;
+
+  // We maintain two sets of convolvers for smooth cross-faded interpolations
+  // when then azimuth and elevation are dynamically changing. When the azimuth
+  // and elevation are not changing, we simply process with one of the two sets.
+  // Initially we use CrossfadeSelection1 corresponding to m_convolverL1 and
+  // m_convolverR1. Whenever the azimuth or elevation changes, a crossfade is
+  // initiated to transition to the new position. So if we're currently
+  // processing with CrossfadeSelection1, then we transition to
+  // CrossfadeSelection2 (and vice versa). If we're in the middle of a
+  // transition, then we wait until it is complete before initiating a new
+  // transition.
+
+  // Selects either the convolver set (m_convolverL1, m_convolverR1) or
+  // (m_convolverL2, m_convolverR2).
+  enum CrossfadeSelection { CrossfadeSelection1, CrossfadeSelection2 };
+
+  CrossfadeSelection m_crossfadeSelection;
+
+  // azimuth/elevation for CrossfadeSelection1.
+  int m_azimuthIndex1;
+  double m_elevation1;
+
+  // azimuth/elevation for CrossfadeSelection2.
+  int m_azimuthIndex2;
+  double m_elevation2;
+
+  // A crossfade value 0 <= m_crossfadeX <= 1.
+  float m_crossfadeX;
+
+  // Per-sample-frame crossfade value increment.
+  float m_crossfadeIncr;
+
+  FFTConvolver m_convolverL1;
+  FFTConvolver m_convolverR1;
+  FFTConvolver m_convolverL2;
+  FFTConvolver m_convolverR2;
+
+  mozilla::DelayBuffer m_delayLine;
+
+  AudioFloatArray m_tempL1;
+  AudioFloatArray m_tempR1;
+  AudioFloatArray m_tempL2;
+  AudioFloatArray m_tempR2;
+};
+
+}  // namespace WebCore
+
+#endif  // HRTFPanner_h
diff --git a/dom/media/webaudio/blink/IIRFilter.cpp b/dom/media/webaudio/blink/IIRFilter.cpp
new file mode 100644
index 0000000000..5ea027e83a
--- /dev/null
+++ b/dom/media/webaudio/blink/IIRFilter.cpp
@@ -0,0 +1,181 @@
+// Copyright 2016 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "IIRFilter.h"
+
+#include "DenormalDisabler.h"
+#include "fdlibm.h"
+#include "mozilla/FloatingPoint.h"
+
+#include <mozilla/Assertions.h>
+
+#include <complex>
+
+namespace blink {
+
+// The length of the memory buffers for the IIR filter.  This MUST be a power of
+// two and must be greater than the possible length of the filter coefficients.
+const int kBufferLength = 32;
+static_assert(kBufferLength >= IIRFilter::kMaxOrder + 1,
+              "Internal IIR buffer length must be greater than maximum IIR "
+              "Filter order.");
+
+IIRFilter::IIRFilter(const AudioDoubleArray* feedforwardCoef,
+                     const AudioDoubleArray* feedbackCoef)
+    : m_bufferIndex(0),
+      m_feedback(feedbackCoef),
+      m_feedforward(feedforwardCoef) {
+  m_xBuffer.SetLength(kBufferLength);
+  m_yBuffer.SetLength(kBufferLength);
+  reset();
+}
+
+IIRFilter::~IIRFilter() = default;
+
+void IIRFilter::reset() {
+  memset(m_xBuffer.Elements(), 0, m_xBuffer.Length() * sizeof(double));
+  memset(m_yBuffer.Elements(), 0, m_yBuffer.Length() * sizeof(double));
+}
+
+static std::complex<double> evaluatePolynomial(const double* coef,
+                                               std::complex<double> z,
+                                               int order) {
+  // Use Horner's method to evaluate the polynomial P(z) = sum(coef[k]*z^k, k,
+  // 0, order);
+  std::complex<double> result = 0;
+
+  for (int k = order; k >= 0; --k)
+    result = result * z + std::complex<double>(coef[k]);
+
+  return result;
+}
+
+void IIRFilter::process(const float* sourceP, float* destP,
+                        size_t framesToProcess) {
+  // Compute
+  //
+  //   y[n] = sum(b[k] * x[n - k], k = 0, M) - sum(a[k] * y[n - k], k = 1, N)
+  //
+  // where b[k] are the feedforward coefficients and a[k] are the feedback
+  // coefficients of the filter.
+
+  // This is a Direct Form I implementation of an IIR Filter.  Should we
+  // consider doing a different implementation such as Transposed Direct Form
+  // II?
+  const double* feedback = m_feedback->Elements();
+  const double* feedforward = m_feedforward->Elements();
+
+  MOZ_ASSERT(feedback);
+  MOZ_ASSERT(feedforward);
+
+  // Sanity check to see if the feedback coefficients have been scaled
+  // appropriately. It must be EXACTLY 1!
+  MOZ_ASSERT(feedback[0] == 1);
+
+  int feedbackLength = m_feedback->Length();
+  int feedforwardLength = m_feedforward->Length();
+  int minLength = std::min(feedbackLength, feedforwardLength);
+
+  double* xBuffer = m_xBuffer.Elements();
+  double* yBuffer = m_yBuffer.Elements();
+
+  for (size_t n = 0; n < framesToProcess; ++n) {
+    // To help minimize roundoff, we compute using double's, even though the
+    // filter coefficients only have single precision values.
+    double yn = feedforward[0] * sourceP[n];
+
+    // Run both the feedforward and feedback terms together, when possible.
+    for (int k = 1; k < minLength; ++k) {
+      int n = (m_bufferIndex - k) & (kBufferLength - 1);
+      yn += feedforward[k] * xBuffer[n];
+      yn -= feedback[k] * yBuffer[n];
+    }
+
+    // Handle any remaining feedforward or feedback terms.
+    for (int k = minLength; k < feedforwardLength; ++k)
+      yn += feedforward[k] * xBuffer[(m_bufferIndex - k) & (kBufferLength - 1)];
+
+    for (int k = minLength; k < feedbackLength; ++k)
+      yn -= feedback[k] * yBuffer[(m_bufferIndex - k) & (kBufferLength - 1)];
+
+    // Save the current input and output values in the memory buffers for the
+    // next output.
+    m_xBuffer[m_bufferIndex] = sourceP[n];
+    m_yBuffer[m_bufferIndex] = yn;
+
+    m_bufferIndex = (m_bufferIndex + 1) & (kBufferLength - 1);
+
+    // Avoid introducing a stream of subnormals
+    destP[n] = WebCore::DenormalDisabler::flushDenormalFloatToZero(yn);
+    MOZ_ASSERT(destP[n] == 0.0 || std::fabs(destP[n]) > FLT_MIN ||
+                   std::isnan(destP[n]),
+               "output should not be subnormal, but can be NaN");
+  }
+}
+
+void IIRFilter::getFrequencyResponse(int nFrequencies, const float* frequency,
+                                     float* magResponse, float* phaseResponse) {
+  // Evaluate the z-transform of the filter at the given normalized frequencies
+  // from 0 to 1. (One corresponds to the Nyquist frequency.)
+  //
+  // The z-tranform of the filter is
+  //
+  // H(z) = sum(b[k]*z^(-k), k, 0, M) / sum(a[k]*z^(-k), k, 0, N);
+  //
+  // The desired frequency response is H(exp(j*omega)), where omega is in
+  // [0, 1).
+  //
+  // Let P(x) = sum(c[k]*x^k, k, 0, P) be a polynomial of order P.  Then each of
+  // the sums in H(z) is equivalent to evaluating a polynomial at the point 1/z.
+
+  for (int k = 0; k < nFrequencies; ++k) {
+    // zRecip = 1/z = exp(-j*frequency)
+    double omega = -M_PI * frequency[k];
+    std::complex<double> zRecip =
+        std::complex<double>(fdlibm_cos(omega), fdlibm_sin(omega));
+
+    std::complex<double> numerator = evaluatePolynomial(
+        m_feedforward->Elements(), zRecip, m_feedforward->Length() - 1);
+    std::complex<double> denominator = evaluatePolynomial(
+        m_feedback->Elements(), zRecip, m_feedback->Length() - 1);
+    // Strangely enough, using complex division:
+    // e.g. Complex response = numerator / denominator;
+    // fails on our test machines, yielding infinities and NaNs, so we do
+    // things the long way here.
+    double n = norm(denominator);
+    double r = (real(numerator) * real(denominator) +
+                imag(numerator) * imag(denominator)) /
+               n;
+    double i = (imag(numerator) * real(denominator) -
+                real(numerator) * imag(denominator)) /
+               n;
+    std::complex<double> response = std::complex<double>(r, i);
+
+    magResponse[k] =
+        static_cast<float>(fdlibm_hypot(real(response), imag(response)));
+    phaseResponse[k] =
+        static_cast<float>(fdlibm_atan2(imag(response), real(response)));
+  }
+}
+
+bool IIRFilter::buffersAreZero() {
+  double* xBuffer = m_xBuffer.Elements();
+  double* yBuffer = m_yBuffer.Elements();
+
+  for (size_t k = 0; k < m_feedforward->Length(); ++k) {
+    if (xBuffer[(m_bufferIndex - k) & (kBufferLength - 1)] != 0.0) {
+      return false;
+    }
+  }
+
+  for (size_t k = 0; k < m_feedback->Length(); ++k) {
+    if (fabs(yBuffer[(m_bufferIndex - k) & (kBufferLength - 1)]) >= FLT_MIN) {
+      return false;
+    }
+  }
+
+  return true;
+}
+
+}  // namespace blink
diff --git a/dom/media/webaudio/blink/IIRFilter.h b/dom/media/webaudio/blink/IIRFilter.h
new file mode 100644
index 0000000000..87522add06
--- /dev/null
+++ b/dom/media/webaudio/blink/IIRFilter.h
@@ -0,0 +1,62 @@
+// Copyright 2016 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#ifndef IIRFilter_h
+#define IIRFilter_h
+
+#include "nsTArray.h"
+
+typedef nsTArray<double> AudioDoubleArray;
+
+namespace blink {
+
+class IIRFilter final {
+ public:
+  // The maximum IIR filter order.  This also limits the number of feedforward
+  // coefficients.  The maximum number of coefficients is 20 according to the
+  // spec.
+  const static size_t kMaxOrder = 19;
+  IIRFilter(const AudioDoubleArray* feedforwardCoef,
+            const AudioDoubleArray* feedbackCoef);
+  ~IIRFilter();
+
+  void process(const float* sourceP, float* destP, size_t framesToProcess);
+
+  void reset();
+
+  void getFrequencyResponse(int nFrequencies, const float* frequency,
+                            float* magResponse, float* phaseResponse);
+
+  bool buffersAreZero();
+
+ private:
+  // Filter memory
+  //
+  // For simplicity, we assume |m_xBuffer| and |m_yBuffer| have the same length,
+  // and the length is a power of two.  Since the number of coefficients has a
+  // fixed upper length, the size of xBuffer and yBuffer is fixed. |m_xBuffer|
+  // holds the old input values and |m_yBuffer| holds the old output values
+  // needed to compute the new output value.
+  //
+  // m_yBuffer[m_bufferIndex] holds the most recent output value, say, y[n].
+  // Then m_yBuffer[m_bufferIndex - k] is y[n - k].  Similarly for m_xBuffer.
+  //
+  // To minimize roundoff, these arrays are double's instead of floats.
+  AudioDoubleArray m_xBuffer;
+  AudioDoubleArray m_yBuffer;
+
+  // Index into the xBuffer and yBuffer arrays where the most current x and y
+  // values should be stored.  xBuffer[bufferIndex] corresponds to x[n], the
+  // current x input value and yBuffer[bufferIndex] is where y[n], the current
+  // output value.
+  int m_bufferIndex;
+
+  // Coefficients of the IIR filter.
+  const AudioDoubleArray* m_feedback;
+  const AudioDoubleArray* m_feedforward;
+};
+
+}  // namespace blink
+
+#endif  // IIRFilter_h
diff --git a/dom/media/webaudio/blink/IRC_Composite_C_R0195-incl.cpp b/dom/media/webaudio/blink/IRC_Composite_C_R0195-incl.cpp
new file mode 100644
index 0000000000..37d731d3a2
--- /dev/null
+++ b/dom/media/webaudio/blink/IRC_Composite_C_R0195-incl.cpp
@@ -0,0 +1,4571 @@
+/* -*- 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/. */
+
+/**
+ * The sample data in the arrays here was derived for Webkit by Chris Rogers
+ * through averaging of impulse responses from the IRCAM Listen HRTF Database.
+ * The responses here are half the length of the Listen responses.  This
+ * sample data has been granted to the Public Domain.
+ *
+ * This file is intended to be included in compilation of a single
+ * implementation file.
+ *
+ * Each elevation (p) contains impulse responses at a varying number of
+ * equally spaced azimuths for the left ear, ordered clockwise from in front
+ * the listener.
+ */
+
+#include "mozilla/ArrayUtils.h"
+
+using mozilla::ArrayLength;
+
+const int16_t irc_composite_c_r0195_p315[][256] =
+    {/* IRC_Composite_C_R0195_T000_P315.wav */
+     {-37,   37,   -38,  39,    -39,   40,    -41,  42,   -42,   43,    -43,
+      44,    -44,  44,   -44,   44,    -43,   42,   -39,  36,    -31,   23,
+      -10,   -10,  5,    -655,  -410,  -552,  -353, -474, -1525, -758,  656,
+      -263,  70,   1414, -1528, -731,  -1811, 1646, 1312, -1501, -407,  8893,
+      -1543, 7696, 8084, 2629,  -2452, -234,  3799, 1676, 177,   -1077, -474,
+      -1325, 3527, 985,  265,   -884,  373,   971,  1024, -412,  507,   -173,
+      259,   -799, 635,  -628,  507,   -344,  394,  -359, 178,   -276,  349,
+      -201,  137,  -249, 377,   -311,  263,   -404, 284,  -244,  173,   -243,
+      330,   -320, 112,  -150,  164,   -142,  174,  -300, 158,   -197,  13,
+      -141,  85,   -190, 64,    -122,  41,    -122, 60,   -195,  125,   -163,
+      10,    -67,  -6,   -122,  77,    -133,  26,   -71,  -42,   -156,  48,
+      -152,  -12,  -89,  -120,  -104,  -37,   -154, -57,  -139,  -80,   -165,
+      -95,   -242, -81,  -146,  -111,  -178,  -109, -208, -48,   -178,  -131,
+      -163,  -68,  -169, -94,   -190,  -139,  -190, -118, -204,  -160,  -220,
+      -140,  -204, -171, -238,  -126,  -203,  -114, -209, -138,  -177,  -124,
+      -184,  -130, -175, -170,  -185,  -180,  -231, -189, -233,  -210,  -236,
+      -245,  -288, -208, -329,  -246,  -274,  -199, -273, -189,  -267,  -208,
+      -215,  -199, -187, -209,  -206,  -210,  -123, -197, -156,  -173,  -142,
+      -97,   -123, -97,  -107,  -73,   -84,   -39,  -50,  -66,   -11,   -50,
+      -12,   -51,  8,    -27,   19,    -48,   -9,   -18,  5,     -42,   -15,
+      -35,   -31,  -27,  -27,   -64,   -33,   -54,  -1,   -98,   -47,   -56,
+      -7,    -76,  -47,  -70,   -42,   -54,   -65,  -76,  -43,   -57,   -9,
+      -61,   -39,  -58,  33,    -39,   3,     -34,  20,   -19,   4,     -71,
+      61,    -22,  10},
+     /* IRC_Composite_C_R0195_T015_P315.wav */
+     {81,    -82,   83,    -84,  84,   -85,   86,   -87,   87,    -87,  87,
+      -87,   87,    -86,   84,   -82,  78,    -72,  63,    -49,   23,   33,
+      -344,  -481,  -563,  -443, -265, -1527, -713, 251,   -1453, 939,  2510,
+      -1221, -1282, -1307, 806,  585,  -990,  -82,  9029,  -4621, 9096, 11230,
+      4611,  -3051, 400,   2105, 749,  1644,  165,  -1556, -1521, 3009, 1430,
+      723,   -902,  933,   187,  28,   480,   951,  -214,  122,   -730, -95,
+      -137,  573,   -593,  558,  -692, -62,   28,   16,    -505,  426,  -283,
+      227,   -320,  210,   -374, 303,  -435,  127,  -128,  76,    -349, 106,
+      -364,  139,   -348,  184,  -425, -36,   -441, 91,    -413,  93,   -444,
+      33,    -257,  -74,   -414, 93,   -379,  19,   -327,  -43,   -366, 79,
+      -293,  20,    -199,  -76,  -207, 149,   -239, 68,    -247,  66,   -219,
+      61,    -232,  38,    -212, 36,   -209,  -27,  -209,  -58,   -227, -6,
+      -309,  -12,   -225,  -60,  -199, -10,   -277, 19,    -207,  -69,  -211,
+      -30,   -261,  -24,   -233, -102, -217,  -131, -247,  -144,  -229, -111,
+      -256,  -104,  -254,  -130, -241, -66,   -249, -88,   -223,  -144, -199,
+      -148,  -229,  -101,  -242, -189, -240,  -163, -308,  -147,  -285, -217,
+      -239,  -224,  -291,  -173, -269, -220,  -209, -208,  -240,  -180, -212,
+      -201,  -217,  -169,  -184, -174, -178,  -146, -209,  -108,  -186, -96,
+      -108,  -90,   -120,  -44,  -101, -49,   -39,  -22,   -44,   1,    -58,
+      3,     -19,   -21,   -8,   -6,   -22,   -9,   18,    -15,   -7,   -23,
+      -12,   -24,   -11,   -43,  -33,  -33,   -31,  -57,   -5,    -64,  -22,
+      -45,   -16,   -87,   -8,   -40,  -45,   -57,  -14,   -53,   -5,   -47,
+      -21,   -45,   -8,    -33,  13,   -16,   3,    -29,   6,     -18,  23,
+      -31,   24,    -41},
+     /* IRC_Composite_C_R0195_T030_P315.wav */
+     {9,     -9,    8,    -8,   8,     -8,    8,     -7,   7,    -6,    5,
+      -4,    3,     -2,   0,    2,     -5,    9,     -14,  23,   -127,  -380,
+      -922,  -742,  -548, 257,  -2362, 972,   -1653, 1079, 1193, 1855,  -423,
+      -3573, -1945, 3974, 274,  -2796, 11656, -988,  5085, 9362, 10293, -750,
+      -3105, -1864, 2111, 2283, -318,  -221,  -1158, -110, 2923, 410,   -981,
+      900,   122,   -303, 833,  532,   -152,  20,    -462, 17,   -48,   336,
+      235,   63,    -647, 577,  -362,  -19,   -259,  102,  168,  -103,  -338,
+      -333,  24,    -9,   -458, 287,   -470,  -273,  -216, 46,   -484,  13,
+      -220,  -143,  -392, -290, -340,  -198,  -301,  -237, -241, -176,  -489,
+      -201,  -271,  -252, -335, -60,   -395,  -225,  -412, -191, -198,  -164,
+      -259,  -64,   -380, -75,  -287,  -9,    -159,  -126, -100, -116,  -155,
+      -96,   -113,  -23,  -100, -117,  -115,  -164,  -180, 8,    -173,  -57,
+      -208,  -89,   -129, -73,  -188,  -102,  -184,  -93,  -204, -81,   -217,
+      -128,  -180,  -66,  -279, -127,  -244,  -206,  -146, -178, -190,  -135,
+      -228,  -154,  -205, -100, -196,  -143,  -185,  -140, -235, -135,  -209,
+      -144,  -158,  -250, -202, -234,  -227,  -157,  -214, -201, -260,  -180,
+      -244,  -184,  -234, -151, -192,  -198,  -223,  -199, -199, -202,  -189,
+      -164,  -233,  -143, -220, -175,  -186,  -107,  -156, -84,  -181,  -106,
+      -116,  -72,   -63,  -29,  -76,   -11,   -83,   15,   -65,  30,    -2,
+      -4,    -41,   4,    -49,  38,    -44,   57,    -2,   9,    -61,   11,
+      -29,   35,    -27,  2,    -50,   -23,   -17,   1,    -57,  2,     -26,
+      -14,   -67,   -39,  -9,   -30,   -30,   -21,   -44,  -39,  -75,   -12,
+      -27,   -13,   -32,  13,   -22,   -16,   5,     -2,   23,   -25,   48,
+      -24,   -32,   -30},
+     /* IRC_Composite_C_R0195_T045_P315.wav */
+     {52,   -54,   57,   -59,   62,    -64,  67,   -71,   74,    -78,   82,
+      -86,  91,    -95,  100,   -104,  107,  -105, 80,    -502,  -332,  -993,
+      -162, -1907, 1505, -2021, -314,  756,  -842, 5057,  -1148, -2263, -3462,
+      3639, -1435, 4503, -272,  9367,  494,  8006, 14353, 933,   -5033, -2453,
+      2315, 1073,  2918, -1236, -3010, 2416, 371,  1956,  -531,  -1294, -177,
+      691,  1104,  -312, 402,   -1300, 693,  -586, 315,   -26,   538,   -594,
+      294,  -267,  -119, -580,  378,   28,   261,  -1010, 385,   -342,  8,
+      -224, 217,   -399, 43,    -603,  -1,   -295, -252,  167,   -340,  -419,
+      -328, -348,  -86,  -497,  -57,   -294, -381, -499,  -201,  -454,  -232,
+      -518, -17,   -717, -310,  -542,  -84,  -458, -96,   -313,  -154,  -496,
+      -112, -226,  -165, -242,  -163,  -234, -306, -270,  -70,   -238,  -108,
+      -139, -101,  -251, -218,  -175,  -17,  -215, -23,   -205,  -109,  -221,
+      -12,  -131,  -27,  -205,  -45,   -264, -158, -151,  -4,    -169,  -113,
+      -251, -102,  -247, -100,  -151,  -158, -212, -170,  -190,  -217,  -220,
+      -105, -182,  -191, -216,  -210,  -208, -188, -144,  -93,   -288,  -132,
+      -248, -124,  -281, -112,  -217,  -92,  -321, -80,   -343,  -94,   -222,
+      -100, -231,  -229, -216,  -139,  -209, -164, -138,  -158,  -204,  -180,
+      -171, -145,  -125, -68,   -150,  -139, -133, -97,   -43,   -133,  -83,
+      -49,  -109,  -53,  -101,  -20,   -84,  23,   -64,   -10,   -116,  38,
+      -50,  15,    -23,  -2,    -3,    -27,  -28,  3,     14,    36,    -13,
+      -21,  -18,   -38,  28,    -26,   9,    -12,  -28,   -46,   -29,   -8,
+      8,    18,    -21,  -12,   -47,   16,   -23,  -21,   -22,   -28,   -21,
+      -46,  -32,   37,   -14,   -1,    -31,  11,   -4,    -17,   51,    -42,
+      23,   -30,   55},
+     /* IRC_Composite_C_R0195_T060_P315.wav */
+     {-9,   10,    -10,  10,    -11,   11,    -11,   12,   -12,   13,    -13,
+      14,   -14,   14,   -15,   15,    -14,   14,    -237, -853,  -211,  -839,
+      -537, -1171, 1035, -1099, 1039,  294,   -1596, 6549, -2739, -2660, -4050,
+      4749, 2134,  7195, 4024,  6882,  -1377, 13010, 8996, -6905, -3319, -3088,
+      4606, 2892,  2461, -4423, -1310, 1787,  1273,  1672, -1868, -79,   1190,
+      70,   -141,  -131, 222,   -677,  570,   -820,  675,  -811,  128,   -382,
+      165,  -353,  -183, -560,  68,    -440,  -382,  -163, -67,   -467,  -152,
+      -570, 177,   -472, -46,   -374,  -58,   -324,  -179, -380,  -114,  -308,
+      -223, -231,  -266, -228,  -188,  -382,  -93,   -468, -172,  -439,  -277,
+      -467, -233,  -484, -158,  -431,  -177,  -375,  -153, -360,  -208,  -377,
+      -90,  -359,  -252, -321,  -261,  -288,  -236,  -352, -117,  -397,  -126,
+      -318, -103,  -229, -203,  -302,  -132,  -152,  -113, -159,  -188,  -103,
+      -160, -141,  -121, -237,  -158,  -186,  -215,  -126, -180,  -203,  -129,
+      -258, -131,  -232, -90,   -192,  -139,  -222,  -58,  -238,  -81,   -214,
+      -111, -210,  -119, -232,  -97,   -257,  -79,   -254, -124,  -244,  -169,
+      -224, -170,  -218, -187,  -194,  -208,  -142,  -212, -136,  -216,  -224,
+      -142, -210,  -181, -238,  -144,  -243,  -81,   -233, -107,  -193,  -82,
+      -124, -113,  -114, -86,   -61,   -72,   -88,   -74,  -42,   -62,   -68,
+      -57,  -28,   -20,  -32,   -42,   -4,    -55,   -25,  11,    -34,   -17,
+      -21,  12,    -38,  -17,   -4,    -21,   -19,   -51,  -19,   -40,   -1,
+      -58,  -23,   -65,  -46,   -5,    -23,   -26,   -24,  -22,   36,    -42,
+      4,    -20,   73,   -50,   19,    -61,   15,    -46,  1,     -36,   -37,
+      -6,   -9,    13,   -34,   18,    -10,   58,    -13,  38,    32,    29,
+      4,    16,    -6},
+     /* IRC_Composite_C_R0195_T075_P315.wav */
+     {-13,  13,    -14,   14,    -14,   14,   -14,   14,    -14,   13,    -13,
+      11,   -9,    5,     2,     -16,   50,   -741,  214,   -932,  242,   -1432,
+      -236, -662,  1347,  -571,  -1196, 4105, -1805, 3633,  -3512, -1059, -3340,
+      6144, 8904,  2949,  -3056, 13761, 4639, 6581,  1016,  -7994, -537,  2069,
+      9498, -3772, -2314, -3272, 3945,  434,  437,   -1200, -83,   923,   138,
+      258,  258,   -7,    455,   -141,  284,  -478,  398,   -1090, 528,   -789,
+      -29,  -665,  -287,  -554,  -73,   -808, -317,  -229,  -409,  -754,  -201,
+      -562, 103,   -767,  -233,  -393,  90,   -725,  -54,   -341,  -112,  -375,
+      -320, -304,  -39,   -501,  -232,  -150, -220,  -432,  -164,  -401,  -81,
+      -462, -293,  -278,  -139,  -468,  -172, -335,  -180,  -318,  -233,  -226,
+      -103, -361,  -214,  -194,  -168,  -399, -129,  -254,  -92,   -285,  -156,
+      -134, -28,   -353,  -136,  -218,  -68,  -245,  -164,  -257,  -128,  -294,
+      -188, -259,  -264,  -283,  -216,  -266, -192,  -249,  -243,  -221,  -193,
+      -297, -159,  -234,  -247,  -176,  -224, -169,  -180,  -199,  -220,  -117,
+      -192, -112,  -197,  -146,  -176,  -92,  -190,  -108,  -218,  -108,  -210,
+      -116, -237,  -105,  -215,  -104,  -222, -112,  -224,  -93,   -273,  -105,
+      -266, -128,  -293,  -88,   -283,  -37,  -279,  -49,   -211,  -28,   -227,
+      20,   -197,  2,     -189,  32,    -205, 58,    -176,  59,    -144,  57,
+      -167, 132,   -119,  93,    -104,  104,  -78,   87,    -109,  101,   -75,
+      67,   -98,   89,    -87,   61,    -85,  56,    -78,   56,    -103,  66,
+      -107, 43,    -101,  53,    -122,  26,   -89,   31,    -77,   5,     -71,
+      25,   -79,   9,     -93,   21,    -74,  23,    -83,   51,    -92,   20,
+      -72,  56,    -59,   56,    -36,   38,   19,    39,    4,     62,    -17,
+      56,   -23,   68},
+     /* IRC_Composite_C_R0195_T090_P315.wav */
+     {87,   -92,   97,    -103, 109,  -116,  123,   -131,  140,   -150, 162,
+      -174, 189,   -205,  224,  -244, 266,   -286,  51,    -859,  535,  -1364,
+      139,  -1195, 1589,  -390, 911,  -1048, 4305,  473,   -5108, 558,  766,
+      2725, 10227, -95,   2100, 7357, 10939, 4034,  -9033, 1273,  -876, 7923,
+      448,  -413,  -5710, 1509, 967,  2067,  -1395, -1318, 853,   624,  -242,
+      51,   284,   401,   299,  341,  -114,  602,   -538,  101,   -460, -168,
+      -515, -276,  -591,  -395, -537, -566,  -206,  -633,  -470,  -595, -421,
+      -387, -497,  -528,  -554, -277, -415,  -451,  -399,  -376,  -198, -441,
+      -287, -406,  -190,  -420, -196, -316,  -218,  -371,  -230,  -316, -251,
+      -337, -304,  -180,  -217, -314, -203,  -175,  -197,  -285,  -220, -238,
+      -109, -305,  -167,  -234, -146, -316,  -92,   -192,  -70,   -236, -136,
+      -80,  -80,   -227,  -175, -115, -77,   -157,  -162,  -216,  -61,  -249,
+      -168, -273,  -182,  -316, -207, -288,  -178,  -282,  -287,  -286, -206,
+      -333, -257,  -356,  -226, -357, -253,  -372,  -144,  -341,  -201, -306,
+      -120, -225,  -145,  -256, -69,  -151,  -103,  -191,  -78,   -159, -72,
+      -231, -86,   -209,  -102, -254, -75,   -225,  -131,  -246,  -122, -189,
+      -129, -238,  -123,  -178, -85,  -185,  -80,   -172,  -78,   -165, -42,
+      -131, -62,   -158,  -35,  -137, -15,   -123,  -23,   -84,   14,   -68,
+      23,   -29,   8,     12,   -7,   21,    17,    -7,    6,     -12,  18,
+      -60,  33,    -67,   51,   -69,  41,    -34,   41,    -45,   33,   -37,
+      31,   -56,   11,    -29,  -2,   -76,   4,     -43,   17,    -73,  -1,
+      -63,  19,    -62,   -23,  -71,  -23,   -50,   -30,   -62,   24,   -59,
+      23,   -75,   63,    -53,  35,   -32,   38,    -9,    16,    4,    53,
+      -2,   34,    -15},
+     /* IRC_Composite_C_R0195_T105_P315.wav */
+     {-4,   4,     -4,    4,     -4,   4,    -4,    4,     -5,    5,     -5,
+      6,    -7,    9,     -11,   17,   -28,  67,    21,    -444,  -141,  -224,
+      -503, -492,  56,    581,   659,  -254, 970,   3243,  -1085, -4435, 2584,
+      328,  6386,  5564,  -809,  5990, 6762, 10583, -5626, -2592, 332,   5216,
+      4379, -2326, -2369, -2841, 2364, 794,  414,   -2336, 841,   434,   185,
+      -245, 467,   54,    226,   66,   291,  109,   -14,   -250,  -37,   -434,
+      -55,  -468,  -301,  -501,  -528, -390, -260,  -361,  -561,  -460,  -370,
+      -387, -354,  -670,  -414,  -508, -325, -671,  -240,  -470,  -287,  -539,
+      -267, -403,  -309,  -523,  -407, -423, -444,  -437,  -347,  -265,  -394,
+      -345, -296,  -231,  -221,  -203, -113, -208,  -173,  -209,  -71,   -127,
+      -120, -168,  -119,  -150,  -164, -160, -171,  -155,  -129,  -132,  -133,
+      -114, -84,   -152,  -145,  -150, -50,  -140,  -100,  -199,  -121,  -188,
+      -145, -209,  -221,  -240,  -266, -216, -273,  -254,  -311,  -241,  -309,
+      -303, -306,  -288,  -304,  -324, -325, -280,  -286,  -254,  -285,  -213,
+      -282, -199,  -235,  -168,  -198, -151, -152,  -157,  -138,  -157,  -155,
+      -189, -176,  -212,  -179,  -192, -145, -184,  -171,  -174,  -110,  -150,
+      -113, -170,  -84,   -115,  -76,  -135, -71,   -161,  -62,   -153,  -49,
+      -144, -73,   -126,  -80,   -100, -82,  -63,   -82,   -57,   -74,   14,
+      -40,  -10,   -30,   25,    -21,  31,   -22,   45,    -26,   43,    -15,
+      48,   -18,   20,    -19,   34,   -32,  26,    -30,   33,    -39,   33,
+      -38,  25,    -62,   21,    -45,  -12,  -65,   -6,    -49,   -13,   -61,
+      8,    -71,   9,     -80,   15,   -87,  37,    -62,   49,    -58,   33,
+      -30,  20,    -31,   -30,   -19,  -18,  -22,   12,    -19,   18,    -18,
+      62,   -6,    19},
+     /* IRC_Composite_C_R0195_T120_P315.wav */
+     {-8,   8,    -9,    10,    -10,   11,   -12,  13,    -14,   15,    -17,
+      18,   -20,  22,    -24,   27,    -30,  33,   -36,   -6,    -353,  -241,
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+      -194, -12,  -168,  -14,   -161,  -15,  -200, -2,    -184,  -29,   -168,
+      -39,  -203, -35,   -143,  -59,   -138, -57,  -120,  -39,   -78,   -38,
+      -71,  4,    -85,   25,    -41,   42,   -65,  60,    -35,   71,    -28,
+      56,   -41,  66,    -43,   40,    -26,  -6,   -23,   -11,   -1,    -28,
+      3,    -16,  -8,    -25,   -4,    -23,  -19,  -21,   -43,   6,     -38,
+      -12,  -49,  4,     -21,   -1,    -32,  9,    -8,    22,    -23,   -1,
+      -11,  -14,  -32,   -12,   -40,   -9,   -31,  23,    -39,   25,    -13,
+      20,   -8,   9},
+     /* IRC_Composite_C_R0195_T135_P315.wav */
+     {-23,   24,    -24,  24,    -24,   24,    -24,  24,   -24,  24,    -24,
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+      -56,   -23,   -79,  7,     -88,   34,    -80,  43,   -77,  63,    -50,
+      59,    -68,   43,   -51,   41,    -54,   60,   -92,  29,   -102,  48,
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+      47,    -47,   53,   -38,   54,    -42,   32,   -33,  27,   -15,   13,
+      -24,   -26,   -34,  -6,    -41,   23,    -23,  1,    -31,  23,    -1,
+      22,    -48,   -14},
+     /* IRC_Composite_C_R0195_T150_P315.wav */
+     {-14,   14,    -14,   14,    -14,  14,    -14,   14,    -14,  14,   -15,
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+      35,    -28,   34,    -23,   25,   -31,   33,    -25,   -2,   -7,   10,
+      -26,   -60,   -40},
+     /* IRC_Composite_C_R0195_T165_P315.wav */
+     {-17,  17,   -18,  19,    -19,  20,    -21,  21,    -22,   23,    -24,
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+      -8,   24,   11,   13,    -8,   11,    55,   13,    32,    16,    31,
+      9,    -51,  -42},
+     /* IRC_Composite_C_R0195_T180_P315.wav */
+     {24,   -25,  26,    -27,  28,    -30,  31,    -33,  34,   -36,  38,   -40,
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+      12,   -27,  25,    -7,   7,     29,   -15,   24,   -9,   67,   32,   52,
+      -7,   19,   -10,   -34},
+     /* IRC_Composite_C_R0195_T195_P315.wav */
+     {-3,   3,    -4,   4,    -4,   4,    -4,   4,    -4,   4,    -4,   4,
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+      -27,  0,    -4,   51,   21,   29,   16,   43,   17,   21,   38,   16,
+      70,   -26,  47,   -23},
+     /* IRC_Composite_C_R0195_T210_P315.wav */
+     {6,    -6,   6,    -6,   6,    -6,   6,    -6,   6,    -6,   6,    -7,
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+      -6,   -21,  -10,  -12,  -6,   14,   -8,   19,   1,    57,   22,   38,
+      15,   27,   32,   33},
+     /* IRC_Composite_C_R0195_T225_P315.wav */
+     {-7,   7,    -8,   8,    -8,   8,    -8,   8,    -8,   8,    -8,   8,
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+      -86,  -58,  -73,  -44,  -52,  -66,  -55,  -76,  -50,  -92,  -33,  -91,
+      -19,  -106, -44,  -97,  -52,  -76,  -62,  -65,  -52,  -53,  -56,  -64,
+      -38,  -57,  -27,  -67,  -32,  -61,  -26,  -56,  -44,  -40,  -36,  -27,
+      -22,  -5,   -13,  -6,   12,   -15,  8,    -1,   18,   1,    -3,   6,
+      -13,  24,   6,    33,   17,   44,   11,   63,   31,   57,   47,   73,
+      42,   34,   14,   5},
+     /* IRC_Composite_C_R0195_T240_P315.wav */
+     {0,    0,    0,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,
+      1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,
+      1,    -1,   2,    58,   78,   16,   22,   51,   114,  84,   41,   96,
+      70,   113,  111,  119,  154,  190,  332,  340,  437,  381,  393,  351,
+      291,  690,  869,  1203, 1249, 1386, 1308, 1507, 1378, 1185, 595,  424,
+      365,  626,  420,  458,  252,  382,  637,  446,  207,  255,  488,  305,
+      321,  49,   221,  53,   63,   79,   -40,  108,  -82,  101,  -101, -13,
+      -203, -152, -221, -220, -235, -289, -282, -281, -296, -224, -344, -241,
+      -368, -252, -366, -268, -359, -308, -318, -288, -298, -317, -264, -285,
+      -261, -303, -269, -298, -270, -320, -331, -336, -351, -327, -351, -348,
+      -331, -347, -276, -351, -246, -372, -243, -370, -280, -365, -300, -410,
+      -295, -413, -330, -447, -345, -440, -333, -430, -333, -415, -326, -403,
+      -318, -379, -309, -372, -303, -357, -286, -300, -274, -299, -260, -274,
+      -222, -258, -207, -249, -178, -228, -179, -202, -181, -164, -194, -132,
+      -181, -100, -149, -105, -124, -85,  -117, -92,  -115, -104, -129, -96,
+      -119, -72,  -137, -65,  -104, -51,  -87,  -75,  -80,  -56,  -63,  -66,
+      -73,  -81,  -74,  -88,  -75,  -87,  -74,  -68,  -80,  -63,  -77,  -50,
+      -74,  -36,  -76,  -42,  -77,  -46,  -59,  -47,  -48,  -54,  -32,  -62,
+      -21,  -50,  -24,  -43,  -33,  -23,  -40,  -10,  -7,   -13,  -16,  -4,
+      2,    7,    17,   -10,  37,   -7,   35,   11,   27,   -4,   32,   -1,
+      15,   -21,  -1,   -28,  7,    -9,   16,   21,   30,   38,   52,   37,
+      48,   38,   34,   3},
+     /* IRC_Composite_C_R0195_T255_P315.wav */
+     {-2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,
+      -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,
+      -3,   3,    -3,   3,    -3,   4,    -6,   74,   75,   79,   61,   128,
+      72,   79,   70,   94,   189,  182,  258,  204,  285,  366,  406,  480,
+      409,  503,  603,  776,  875,  1073, 1366, 1703, 1601, 1585, 1344, 1253,
+      1054, 708,  209,  184,  245,  471,  220,  82,   328,  234,  143,  -43,
+      229,  63,   146,  52,   46,   143,  -26,  108,  -162, -81,  -337, -290,
+      -357, -257, -274, -266, -282, -295, -253, -242, -177, -217, -241, -242,
+      -180, -187, -261, -282, -286, -285, -278, -318, -290, -350, -241, -329,
+      -216, -356, -250, -344, -253, -301, -333, -307, -381, -310, -406, -310,
+      -398, -327, -389, -328, -383, -321, -397, -364, -416, -386, -447, -400,
+      -469, -437, -477, -423, -470, -435, -472, -439, -441, -374, -396, -361,
+      -359, -340, -300, -318, -278, -293, -236, -257, -203, -214, -171, -197,
+      -164, -178, -159, -160, -150, -162, -167, -187, -166, -177, -176, -193,
+      -156, -159, -125, -174, -110, -158, -92,  -129, -96,  -113, -107, -94,
+      -103, -101, -73,  -98,  -45,  -93,  -27,  -108, -14,  -122, -36,  -134,
+      -52,  -131, -66,  -121, -83,  -114, -73,  -107, -51,  -91,  -32,  -79,
+      2,    -74,  2,    -68,  4,    -68,  20,   -45,  9,    -63,  -6,   -50,
+      8,    -15,  -2,   -22,  -9,   6,    4,    25,   -11,  21,   -12,  18,
+      -6,   -1,   -16,  -16,  -15,  -8,   7,    8,    -10,  14,   -14,  21,
+      -9,   18,   -34,  -11,  0,    0,    38,   9,    28,   24,   33,   41,
+      32,   56,   10,   45},
+     /* IRC_Composite_C_R0195_T270_P315.wav */
+     {5,    -5,   5,    -5,   5,    -5,   5,    -6,   6,    -6,   6,    -6,
+      6,    -6,   7,    -7,   7,    -7,   8,    -8,   8,    -9,   9,    -10,
+      10,   -11,  12,   -13,  14,   -16,  19,   -24,  35,   -112, -87,  -172,
+      -147, -119, -127, -152, -113, -103, -89,  -113, -68,  -116, -46,  22,
+      42,   151,  321,  392,  548,  700,  898,  1192, 1367, 1289, 1342, 1432,
+      1348, 1134, 1055, 823,  716,  745,  988,  819,  628,  418,  468,  438,
+      382,  415,  357,  462,  457,  454,  331,  457,  286,  349,  199,  216,
+      110,  272,  103,  141,  96,   107,  157,  39,   151,  90,   143,  187,
+      135,  145,  132,  119,  90,   29,   67,   9,    33,   -99,  -4,   -78,
+      -31,  -119, -118, -125, -133, -39,  -116, 6,    -53,  -9,   -2,   -28,
+      -8,   -68,  -109, -175, -231, -243, -320, -296, -388, -312, -439, -323,
+      -431, -308, -403, -341, -379, -320, -333, -304, -317, -322, -321, -302,
+      -340, -277, -367, -233, -348, -216, -344, -197, -304, -195, -255, -199,
+      -226, -173, -179, -167, -168, -175, -177, -169, -197, -168, -238, -156,
+      -231, -167, -240, -168, -219, -192, -218, -202, -205, -205, -186, -197,
+      -157, -172, -143, -163, -152, -143, -150, -149, -177, -148, -159, -149,
+      -150, -163, -154, -183, -160, -189, -132, -190, -124, -156, -96,  -125,
+      -92,  -99,  -99,  -74,  -85,  -61,  -83,  -67,  -68,  -74,  -62,  -91,
+      -54,  -91,  -46,  -108, -22,  -86,  -15,  -70,  -6,   -63,  -3,   -43,
+      -35,  -76,  -52,  -91,  -99,  -89,  -83,  -72,  -67,  -70,  -42,  -53,
+      -31,  -59,  -32,  -49,  -39,  -55,  -65,  -43,  -43,  -36,  -50,  -25,
+      -39,  -34,  -26,  -31},
+     /* IRC_Composite_C_R0195_T285_P315.wav */
+     {0,    0,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,
+      -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   2,    -2,   2,
+      -2,   3,    -3,   0,    -53,  -54,  -56,  -85,  -47,  -114, -135, -181,
+      -153, -193, -267, -244, -55,  -96,  1,    65,   -104, -149, -27,  77,
+      318,  544,  620,  776,  1096, 1388, 1457, 1232, 925,  941,  734,  587,
+      918,  901,  616,  762,  669,  803,  637,  587,  517,  641,  585,  459,
+      366,  454,  473,  472,  377,  381,  369,  477,  434,  303,  311,  258,
+      424,  217,  330,  203,  348,  239,  206,  163,  147,  190,  93,   131,
+      3,    88,   29,   61,   25,   -4,   30,   -29,  -32,  -79,  -68,  -48,
+      -73,  -75,  -119, -13,  -30,  17,   -48,  5,    -35,  5,    -39,  -80,
+      -142, -105, -167, -166, -226, -200, -281, -233, -317, -276, -334, -290,
+      -353, -338, -366, -316, -342, -297, -343, -306, -273, -252, -266, -278,
+      -281, -276, -239, -265, -258, -270, -226, -250, -191, -215, -181, -217,
+      -149, -202, -131, -195, -119, -225, -148, -263, -146, -220, -148, -221,
+      -170, -215, -156, -215, -188, -262, -202, -269, -194, -235, -202, -241,
+      -188, -209, -153, -204, -154, -203, -167, -217, -151, -214, -172, -225,
+      -158, -182, -145, -170, -150, -164, -121, -136, -87,  -134, -90,  -102,
+      -41,  -87,  -63,  -73,  -53,  -73,  -60,  -73,  -67,  -76,  -98,  -68,
+      -64,  -72,  -69,  -72,  -35,  -71,  -32,  -55,  -37,  -56,  -47,  -39,
+      -69,  -58,  -90,  -53,  -99,  -57,  -83,  -34,  -50,  -55,  -58,  -67,
+      -57,  -63,  -70,  -62,  -74,  -53,  -70,  -30,  -49,  -56,  -37,  -35,
+      -21,  -10,  -19,  -29},
+     /* IRC_Composite_C_R0195_T300_P315.wav */
+     {2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,
+      2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,
+      2,    -2,   2,    -3,   4,    -10,  -133, -133, -170, -242, -249, -433,
+      -299, -139, -159, 192,  133,  -70,  22,   -303, 18,   -19,  635,  218,
+      706,  1507, 1430, 1383, 2387, 1796, 1075, 392,  368,  909,  611,  309,
+      -168, 353,  2,    566,  420,  570,  464,  600,  533,  643,  640,  611,
+      592,  536,  352,  641,  435,  676,  299,  466,  377,  487,  301,  349,
+      330,  262,  326,  234,  228,  152,  89,   163,  56,   54,   -10,  7,
+      -78,  -45,  -66,  -85,  -68,  -104, -18,  4,    -26,  41,   2,    31,
+      0,    79,   5,    36,   26,   46,   69,   42,   -19,  -67,  -82,  -80,
+      -133, -166, -222, -198, -222, -228, -274, -302, -279, -261, -261, -282,
+      -253, -280, -237, -241, -240, -230, -249, -229, -231, -221, -230, -219,
+      -245, -228, -234, -188, -202, -201, -222, -167, -182, -137, -203, -165,
+      -172, -150, -180, -163, -208, -169, -194, -164, -229, -193, -245, -204,
+      -252, -223, -246, -236, -257, -266, -272, -274, -271, -280, -267, -251,
+      -243, -225, -207, -200, -174, -184, -165, -180, -160, -164, -160, -158,
+      -167, -156, -169, -151, -138, -125, -125, -108, -104, -80,  -94,  -71,
+      -82,  -75,  -85,  -57,  -68,  -57,  -49,  -58,  -52,  -65,  -51,  -59,
+      -53,  -74,  -44,  -57,  -44,  -43,  -49,  -55,  -42,  -57,  -55,  -65,
+      -60,  -62,  -53,  -54,  -78,  -70,  -64,  -55,  -66,  -73,  -66,  -48,
+      -68,  -87,  -49,  -57,  -32,  -77,  -35,  -66,  -33,  -71,  -35,  -70,
+      15,   -17,  -11,  -33},
+     /* IRC_Composite_C_R0195_T315_P315.wav */
+     {-3,   3,    -4,   4,    -4,   4,    -4,   4,    -4,   4,    -4,   5,
+      -5,   5,    -5,   5,    -5,   5,    -6,   6,    -6,   6,    -6,   6,
+      -6,   6,    -5,   3,    2,    -152, -177, -483, -235, -110, -653, -348,
+      -150, -38,  208,  331,  33,   -385, -315, -27,  364,  811,  -94,  2504,
+      888,  2086, 3574, 3328, 531,  314,  789,  729,  404,  -337, 56,   36,
+      119,  405,  418,  656,  613,  503,  327,  557,  477,  459,  703,  343,
+      212,  405,  546,  432,  317,  280,  459,  304,  364,  271,  334,  99,
+      204,  199,  155,  38,   34,   17,   94,   27,   75,   -28,  51,   -48,
+      69,   -29,  44,   -56,  45,   -30,  20,   5,    41,   -28,  41,   31,
+      56,   13,   68,   19,   90,   -37,  31,   -65,  21,   -87,  -51,  -142,
+      -103, -155, -98,  -191, -137, -221, -116, -207, -139, -209, -136, -203,
+      -147, -207, -168, -209, -164, -226, -176, -243, -198, -258, -194, -267,
+      -197, -246, -196, -229, -191, -208, -190, -182, -197, -201, -171, -193,
+      -170, -221, -151, -220, -141, -245, -167, -251, -184, -255, -210, -246,
+      -238, -245, -223, -258, -247, -284, -254, -278, -249, -311, -255, -297,
+      -217, -263, -193, -236, -176, -198, -176, -178, -155, -172, -140, -181,
+      -132, -195, -91,  -184, -92,  -165, -90,  -121, -68,  -117, -88,  -97,
+      -47,  -73,  -55,  -70,  -16,  -46,  -13,  -61,  -26,  -38,  -50,  -48,
+      -41,  -43,  -40,  -60,  -49,  -55,  -58,  -57,  -47,  -68,  -57,  -81,
+      -49,  -74,  -61,  -74,  -29,  -70,  -35,  -79,  -39,  -87,  -35,  -94,
+      -49,  -86,  -55,  -65,  -49,  -58,  -43,  -30,  -32,  -23,  -33,  -5,
+      -23,  19,   -13,  -17},
+     /* IRC_Composite_C_R0195_T330_P315.wav */
+     {-7,   7,    -7,   7,    -7,   7,    -7,   7,    -7,   7,    -7,   7,
+      -7,   7,    -7,   7,    -7,   8,    -8,   8,    -8,   9,    -9,   9,
+      -10,  11,   -11,  -55,  -296, -386, -360, -162, -486, -689, -594, 323,
+      -279, 489,  253,  -225, -799, -208, -290, 1653, -875, 1421, 3185, 831,
+      3972, 4758, 3089, -835, 674,  981,  1685, -351, -114, -17,  248,  390,
+      777,  359,  517,  535,  503,  290,  712,  58,   681,  346,  142,  -203,
+      198,  199,  331,  -307, 34,   132,  139,  70,   114,  91,   13,   123,
+      184,  191,  4,    23,   205,  163,  125,  72,   141,  -40,  14,   124,
+      -28,  33,   -51,  40,   -5,   35,   -58,  73,   -52,  -16,  56,   78,
+      -40,  -37,  45,   32,   27,   55,   -7,   -15,  -18,  29,   -29,  22,
+      -117, 6,    -57,  -69,  -79,  -66,  -162, -105, -133, -121, -164, -165,
+      -210, -152, -236, -167, -217, -215, -247, -204, -244, -201, -239, -194,
+      -273, -180, -249, -181, -223, -192, -224, -139, -212, -188, -198, -169,
+      -226, -134, -219, -153, -208, -148, -204, -149, -219, -168, -214, -162,
+      -245, -160, -257, -219, -229, -208, -279, -218, -292, -266, -282, -268,
+      -275, -227, -298, -216, -236, -164, -238, -151, -226, -142, -217, -147,
+      -203, -145, -177, -146, -138, -122, -131, -73,  -151, -51,  -110, -31,
+      -107, -6,   -86,  -9,   -40,  -10,  -36,  -1,   -49,  -9,   -42,  -8,
+      -74,  2,    -73,  3,    -80,  -26,  -79,  6,    -86,  -19,  -76,  -29,
+      -75,  -32,  -78,  -38,  -65,  -46,  -77,  -46,  -88,  -55,  -75,  -74,
+      -82,  -34,  -77,  -58,  -74,  -44,  -61,  -16,  -21,  -21,  -1,   -32,
+      25,   -5,   -9,   -4},
+     /* IRC_Composite_C_R0195_T345_P315.wav */
+     {-3,   3,    -3,    3,    -3,    3,    -3,   3,    -3,    3,    -2,   2,
+      -2,   2,    -1,    1,    0,     0,    1,    -2,   3,     -4,   6,    -9,
+      13,   -23,  -205,  -456, -570,  -178, -609, -516, -1049, -349, 515,  157,
+      -280, 1141, -1675, -135, -1096, 1502, 703,  -567, 2409,  4580, 1615, 5569,
+      6660, -925, 461,   21,   3064,  -94,  275,  -939, 176,   137,  1125, 718,
+      -145, 338,  911,   169,  330,   365,  552,  148,  369,   -393, 179,  -127,
+      204,  80,   -84,   -339, 8,     81,   -65,  -288, 145,   50,   -22,  -24,
+      61,   62,   45,    -71,  194,   159,  54,   -19,  226,   -56,  24,   48,
+      99,   -31,  -91,   -8,   53,    24,   -79,  12,   113,   -69,  -3,   43,
+      52,   -53,  25,    22,   42,    -13,  -5,   -1,   64,    -132, -15,  -42,
+      -12,  -176, -19,   -103, -113,  -146, -76,  -99,  -119,  -177, -139, -96,
+      -193, -229, -107,  -210, -181,  -214, -106, -219, -211,  -156, -157, -151,
+      -246, -153, -144,  -151, -254,  -173, -147, -195, -202,  -155, -196, -154,
+      -199, -151, -187,  -147, -192,  -144, -157, -216, -170,  -139, -200, -196,
+      -193, -158, -269,  -172, -256,  -190, -293, -226, -252,  -229, -318, -238,
+      -250, -265, -306,  -166, -254,  -191, -300, -137, -229,  -129, -286, -136,
+      -186, -156, -221,  -108, -162,  -109, -143, -76,  -137,  -51,  -129, -13,
+      -99,  -24,  -56,   17,   -62,   -20,  -41,  40,   -86,   44,   -52,  39,
+      -123, 42,   -65,   39,   -115,  -16,  -46,  0,    -101,  -28,  -69,  -10,
+      -115, -27,  -101,  20,   -128,  3,    -110, 13,   -133,  -3,   -101, -28,
+      -103, -3,   -79,   -10,  -82,   10,   -56,  27,   -77,   25,   -50,  57,
+      -33,  20,   -49,   39}};
+
+const int16_t irc_composite_c_r0195_p330[][256] =
+    {/* IRC_Composite_C_R0195_T000_P330.wav */
+     {-59,   61,    -63,  65,   -67,   69,    -71,   73,   -76,  78,    -81,
+      83,    -86,   89,   -92,  95,    -98,   100,   -102, 103,  -103,  99,
+      -90,   66,    27,   -206, -447,  -812,  -1040, -532, -532, -221,  -2282,
+      2737,  -1254, 634,  -40,  577,   -3131, -820,  -289, 5598, -4456, 4514,
+      11627, -957,  5804, 7393, 108,   -5260, 1410,  3318, 1621, -1983, 2287,
+      284,   1686,  1238, 2060, -1196, 399,   489,   1551, 51,   439,   -136,
+      620,   290,   -51,  26,   548,   -418,  535,   -429, 119,  -693,  256,
+      -317,  191,   -412, 18,   -258,  -48,   -331,  42,   -245, -151,  -84,
+      -244,  -168,  -112, -248, 56,    -243,  -130,  -346, 3,    -311,  -140,
+      -201,  -205,  -208, -164, -257,  -153,  -216,  -142, -212, -32,   -240,
+      -169,  -200,  -80,  -259, -56,   -266,  -49,   -259, -153, -247,  -131,
+      -186,  -180,  -157, -174, -170,  -168,  -133,  -130, -144, -118,  -156,
+      -112,  -157,  -115, -125, -52,   -108,  -95,   -100, -113, -69,   -143,
+      -89,   -121,  -105, -124, -163,  -161,  -185,  -102, -218, -93,   -213,
+      -136,  -158,  -144, -182, -122,  -122,  -161,  -147, -153, -151,  -116,
+      -162,  -100,  -155, -94,  -176,  -98,   -138,  -94,  -127, -119,  -122,
+      -158,  -125,  -151, -130, -178,  -168,  -144,  -200, -186, -199,  -181,
+      -157,  -192,  -201, -216, -176,  -230,  -170,  -197, -219, -207,  -194,
+      -203,  -187,  -195, -174, -204,  -175,  -205,  -170, -207, -182,  -206,
+      -156,  -195,  -149, -193, -165,  -200,  -125,  -166, -120, -182,  -142,
+      -113,  -73,   -137, -86,  -97,   -75,   -88,   -72,  -83,  -37,   -82,
+      -24,   -72,   -40,  -53,  24,    -38,   -23,   -48,  32,   10,    7,
+      -19,   18,    7,    43,   7,     34,    32,    65,   53,   56,    21,
+      94,    55,    42},
+     /* IRC_Composite_C_R0195_T015_P330.wav */
+     {-33,   32,    -32,   31,    -31,  30,    -28,   27,    -25,   22,    -19,
+      15,    -10,   4,     5,     -16,  31,    -53,   84,    -136,  240,   -673,
+      177,   -174,  -1030, -1321, -76,  -1250, -372,  289,   -2348, 2418,  1442,
+      -314,  -2260, 351,   -1076, 96,   -4201, 12653, -2793, 350,   16253, 7416,
+      -1734, 373,   1945,  -1722, 32,   2367,  1990,  -2906, 2200,  2573,  955,
+      -397,  1580,  -424,  684,   371,  1473,  401,   8,     -1088, 1156,  -110,
+      164,   -526,  361,   -233,  -337, -706,  160,   -330,  -24,   -130,  132,
+      -810,  170,   -491,  242,   -434, 33,    -622,  355,   -505,  -308,  -68,
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+      -282,  9,     -234,  -87,   -198, -101,  -238,  -154,  -207,  -159,  -166,
+      -106,  -164,  -136,  -199,  -109, -121,  -73,   -175,  -65,   -149,  -73,
+      -162,  -108,  -135,  -78,   -99,  -124,  -126,  -179,  -91,   -133,  -84,
+      -188,  -122,  -167,  -148,  -155, -182,  -183,  -160,  -158,  -219,  -207,
+      -224,  -181,  -161,  -192,  -206, -207,  -174,  -243,  -140,  -189,  -173,
+      -194,  -138,  -203,  -173,  -192, -143,  -155,  -154,  -216,  -158,  -170,
+      -138,  -202,  -110,  -182,  -132, -154,  -83,   -201,  -104,  -130,  -44,
+      -129,  -80,   -139,  -19,   -72,  -80,   -88,   -16,   -110,  -20,   -46,
+      -35,   -88,   15,    -32,   6,    -67,   -9,    -16,   52,    -18,   27,
+      27,    26,    -2,    60,    51,   76,    28,    98,    40,    77,    33,
+      94,    11,    62},
+     /* IRC_Composite_C_R0195_T030_P330.wav */
+     {15,    -16,   18,    -19,   21,   -23,   26,   -29,   32,    -36,
+      41,    -47,   54,    -62,   73,   -86,   104,  -126,  155,   -102,
+      601,   -1085, -197,  -1634, 218,  -1892, 836,  -1472, 627,   332,
+      2248,  1679,  -4388, -1076, 2778, -5228, 3611, 10385, -5976, 10952,
+      11266, 9110,  -4453, 651,   -313, -354,  33,   2898,  -224,  -1072,
+      2571,  1824,  -115,  539,   940,  -84,   401,  1263,  337,   3,
+      -171,  -426,  685,   -69,   -27,  -94,   -916, -1001, 131,   -752,
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+      -69,   -237,  -35,   -248,  -105, -213,  -38,  -207,  -108,  -284,
+      -91,   -256,  -64,   -223,  -92,  -227,  -52,  -205,  -48,   -178,
+      -94,   -174,  -26,   -226,  -71,  -202,  -71,  -184,  -34,   -207,
+      -88,   -154,  -79,   -180,  -41,  -222,  -49,  -215,  -56,   -245,
+      -90,   -251,  -51,   -260,  -109, -246,  -150, -229,  -109,  -239,
+      -125,  -253,  -146,  -228,  -85,  -219,  -106, -252,  -82,   -250,
+      -58,   -194,  -116,  -232,  -74,  -193,  -81,  -184,  -110,  -195,
+      -65,   -162,  -92,   -193,  -65,  -199,  -29,  -203,  0,     -193,
+      28,    -151,  -16,   -159,  -4,   -109,  -9,   -90,   -21,   -99,
+      36,    -92,   27,    -82,   30,   -74,   71,   -74,   64,    -43,
+      102,   -28,   93,    18,    88,   22,    98,   29,    116,   21,
+      126,   -20,   133,   -40,   143,  -66},
+     /* IRC_Composite_C_R0195_T045_P330.wav */
+     {60,    -60,  60,    -60,   60,    -60,   59,    -57,  55,    -51,   47,
+      -40,   30,   -15,   -9,    50,    -138,  582,   -538, -783,  -136,  149,
+      -1756, -848, -253,  -449,  1379,  -797,  1611,  1075, 2352,  -3846, -4473,
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+      -112,  3354, 2968,  -606,  -2642, 2599,  370,   2356, -155,  634,   -118,
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+      -168,  -84,  -181,  -86,   -207,  -54,   -197,  -48,  -239,  -19,   -220,
+      -19,   -245, -20,   -221,  -34,   -247,  -31,   -246, -93,   -181,  -96,
+      -211,  -94,  -210,  -85,   -235,  -118,  -230,  -101, -229,  -107,  -245,
+      -109,  -215, -112,  -220,  -113,  -204,  -100,  -188, -108,  -185,  -89,
+      -182,  -62,  -197,  -34,   -195,  -28,   -194,  -18,  -182,  -55,   -157,
+      -81,   -141, -90,   -141,  -91,   -129,  -74,   -112, -85,   -107,  -64,
+      -103,  -64,  -86,   -50,   -93,   -43,   -62,   -35,  -57,   -57,   -38,
+      -24,   -28,  -39,   -24,   -20,   -16,   22,    2,    35,    17,    56,
+      35,    75,   47,    56,    59,    88,    45,    61,   91,    80,    67,
+      62,    68,   59},
+     /* IRC_Composite_C_R0195_T060_P330.wav */
+     {-16,  18,    -21,   24,    -27,   30,    -34,  39,    -45,   51,    -59,
+      68,   -79,   93,    -109,  120,   -488,  -209, -843,  -446,  581,   -2538,
+      150,  -1048, 1674,  -298,  921,   -422,  4633, -1379, -3456, -5235, 5897,
+      -232, 13802, 4771,  -1871, 13790, 9150,  509,  -7321, -2062, 426,   4376,
+      2825, -66,   -4120, 2160,  1014,  1965,  -182, 10,    -66,   1012,  -1092,
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+      -109, -99,   -40,   -137,  -112,  -65,   -123, -140,  -104,  -208,  -112,
+      -191, -189,  -188,  -164,  -205,  -119,  -255, -149,  -161,  -150,  -183,
+      -173, -162,  -174,  -126,  -195,  -97,   -145, -155,  -118,  -129,  -86,
+      -145, -98,   -141,  -61,   -119,  -118,  -63,  -133,  -28,   -155,  2,
+      -161, -3,    -128,  -39,   -100,  -78,   -65,  -82,   -72,   -98,   -47,
+      -85,  -70,   -46,   -75,   -12,   -101,  21,   -108,  22,    -100,  40,
+      -92,  45,    -75,   34,    -63,   31,    -40,  44,    -41,   58,    -23,
+      74,   -14,   74,    -19,   131,   -21,   113,  30,    101,   49,    110,
+      57,   108,   51},
+     /* IRC_Composite_C_R0195_T075_P330.wav */
+     {19,    -20,   21,   -23,   25,   -26,   28,    -31,   33,    -36,  39,
+      -42,   45,    -49,  49,    -225, -118,  -411,  -201,  -767,  -125, -1198,
+      -1086, 1587,  699,  -114,  366,  4108,  -2207, 1045,  -7292, 822,  4055,
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+      -127,  -183,  -140, -187,  -127, -176,  -160,  -172,  -135,  -190, -179,
+      -159,  -144,  -201, -142,  -175, -140,  -206,  -117,  -148,  -146, -179,
+      -121,  -175,  -129, -135,  -92,  -156,  -126,  -159,  -68,   -176, -108,
+      -129,  -91,   -133, -75,   -91,  -67,   -116,  -63,   -52,   -31,  -122,
+      -11,   -115,  3,    -129,  15,   -110,  -22,   -119,  -11,   -83,  -62,
+      -34,   -39,   -48,  -27,   -43,  -19,   -79,   23,    -48,   -1,   -86,
+      40,    -64,   16,   -37,   23,   -6,    -6,    9,     18,    18,   23,
+      24,    45,    17,   82,    40,   87,    28,    105,   40,    116,  77,
+      81,    80,    85},
+     /* IRC_Composite_C_R0195_T090_P330.wav */
+     {6,     -4,    3,     -1,    0,     3,     -5,    8,    -11,   16,
+      -21,   29,    -39,   57,    -101,  -332,  89,    -835, -16,   8,
+      -1022, -629,  -474,  1050,  924,   -984,  2683,  3640, -2207, -5863,
+      2261,  -3476, 15168, 2953,  -3266, 10727, 14390, 3174, -8455, -1966,
+      1449,  7129,  3075,  -3700, -2692, 1997,  423,   1234, -1087, 216,
+      -471,  1061,  -1107, 458,   -1402, 142,   319,   -88,  -425,  76,
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+      -227,  -156,  -224,  -178,  -249,  -190,  -199,  -133, -154,  -164,
+      -183,  -191,  -143,  -185,  -101,  -190,  -166,  -201, -98,   -150,
+      -133,  -119,  -165,  -142,  -178,  -120,  -115,  -180, -149,  -197,
+      -152,  -187,  -75,   -218,  -143,  -264,  -128,  -205, -139,  -221,
+      -148,  -269,  -159,  -215,  -126,  -214,  -123,  -256, -105,  -233,
+      -71,   -203,  -97,   -215,  -79,   -182,  -53,   -120, -82,   -122,
+      -49,   -119,  20,    -120,  24,    -141,  6,     -133, 24,    -115,
+      14,    -116,  -39,   -71,   -8,    -71,   6,     -93,  -11,   -108,
+      40,    -59,   43,    -147,  22,    -98,   28,    -64,  17,    -87,
+      -13,   -67,   63,    -36,   44,    -53,   62,    -77,  123,   -52,
+      145,   -81,   82,    -33,   141,   -4,    116,   8,    98,    14,
+      137,   56,    139,   -7,    138,   4},
+     /* IRC_Composite_C_R0195_T105_P330.wav */
+     {61,    -61,  61,   -61,   62,    -62,   62,    -62,    63,   -63,  63,
+      -64,   64,   -65,  67,    -301,  294,   -741,  -246,   -201, 457,  -1576,
+      -144,  -73,  2001, -1275, 1763,  1153,  6179,  -10039, 184,  1999, 2771,
+      10917, -378, 522,  13686, 11704, -7376, -5609, 1748,   7382, 4547, -1426,
+      -5565, -24,  2068, 630,   -441,  -1191, 186,   450,    -243, 220,  -909,
+      -111,  -78,  770,  -727,  124,   -899,  -156,  -79,    202,  -685, -336,
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+      -208,  -154, -172, -145,  -172,  -242,  -211,  -169,   -70,  -176, -103,
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+      -220,  -229, -254, -179,  -191,  -192,  -230,  -211,   -235, -161, -158,
+      -161,  -230, -195, -206,  -174,  -165,  -144,  -216,   -190, -173, -127,
+      -136,  -156, -157, -179,  -150,  -128,  -122,  -138,   -124, -152, -83,
+      -109,  -30,  -73,  -66,   -103,  -55,   -77,   -32,    -75,  -113, -125,
+      -86,   -47,  -60,  -39,   -89,   -29,   -60,   20,     -18,  -21,  -32,
+      -8,    -50,  24,   -43,   -1,    -69,   -6,    -57,    11,   -53,  21,
+      -58,   -2,   -13,  35,    -10,   30,    5,     44,     12,   60,   31,
+      79,    -4,   79,   46,    97,    47,    97,    36,     128,  43,   132,
+      30,    81,   38},
+     /* IRC_Composite_C_R0195_T120_P330.wav */
+     {1,     4,     -8,    14,   -20,   27,    -35,   44,    -54,   66,   -79,
+      95,    -114,  135,   -159, 177,   -87,   615,   -830,  163,   -909, 1304,
+      -1626, -325,  297,   1206, 69,    -817,  4405,  1017,  -2560, -616, -3770,
+      4387,  14092, -5772, 3858, 10124, 9882,  -2074, -2959, -3337, 7410, 7437,
+      -1780, -6636, -2212, 2639, 1700,  -1078, -1371, -262,  1153,  -566, 417,
+      -320,  27,    -39,   -111, 485,   -441,  -792,  -288,  -146,  -108, -616,
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+      -353,  -222,  -328,  -290, -351,  -239,  -329,  -117,  -69,   -109, -182,
+      -35,   -73,   -135,  -239, -228,  -363,  -234,  -203,  -243,  -333, -218,
+      -233,  -231,  -229,  -116, -190,  -172,  -177,  -158,  -144,  -144, -172,
+      -173,  -215,  -124,  -180, -106,  -203,  -150,  -185,  -121,  -123, -149,
+      -106,  -93,   -132,  -84,  -116,  -33,   -138,  -72,   -186,  -81,  -187,
+      -113,  -205,  -194,  -255, -235,  -238,  -195,  -263,  -202,  -264, -198,
+      -268,  -158,  -241,  -175, -267,  -183,  -249,  -160,  -199,  -167, -203,
+      -153,  -184,  -118,  -137, -101,  -164,  -93,   -143,  -66,   -128, -30,
+      -127,  -42,   -113,  -31,  -78,   -23,   -59,   -78,   -78,   -48,  -79,
+      -53,   -70,   -40,   -106, -22,   -46,   -9,    -51,   -12,   -37,  -31,
+      -54,   -23,   -49,   -43,  -68,   -43,   -55,   -39,   -14,   -29,  -16,
+      -13,   -8,    32,    0,    28,    5,     31,    33,    68,    33,   69,
+      42,    71,    38,    89,   64,    82,    64,    81,    94,    57,   103,
+      62,    51,    87},
+     /* IRC_Composite_C_R0195_T135_P330.wav */
+     {13,    -11,   8,     -5,    2,     2,    -7,   12,    -18,  26,    -34,
+      45,    -58,   75,    -96,   126,   -173, 263,  -756,  239,  -382,  327,
+      -504,  374,   -988,  304,   -762,  1963, 32,   1243,  612,  772,   3408,
+      -6680, -2100, 10509, 4890,  -1044, 6633, 2426, 10000, 1888, -3554, -3952,
+      7154,  5846,  -3202, -5792, -2133, 1379, 2382, -1044, -683, -846,  1056,
+      -24,   511,   18,    303,   -710,  -25,  -39,  -85,   -861, -206,  -321,
+      -34,   -428,  -245,  -428,  -30,   -639, -166, -459,  -198, -463,  -354,
+      -339,  -255,  -329,  -348,  -224,  -286, -255, -287,  -204, -181,  -232,
+      -210,  -296,  -199,  -313,  -262,  -347, -268, -316,  -307, -365,  -310,
+      -360,  -308,  -316,  -324,  -335,  -371, -355, -377,  -361, -310,  -401,
+      -283,  -402,  -244,  -368,  -238,  -331, -274, -247,  -165, -162,  -178,
+      -185,  -197,  -205,  -193,  -250,  -203, -288, -187,  -284, -187,  -262,
+      -183,  -219,  -206,  -232,  -196,  -178, -177, -216,  -242, -166,  -214,
+      -147,  -219,  -161,  -195,  -148,  -178, -168, -190,  -170, -141,  -148,
+      -164,  -119,  -116,  -35,   -121,  -54,  -95,  -56,   -120, -107,  -112,
+      -149,  -152,  -172,  -185,  -195,  -227, -175, -253,  -211, -280,  -223,
+      -250,  -252,  -243,  -251,  -235,  -239, -213, -184,  -218, -176,  -225,
+      -139,  -205,  -123,  -184,  -114,  -154, -113, -98,   -100, -89,   -89,
+      -91,   -67,   -61,   -32,   -65,   -41,  -68,  -21,   -70,  -18,   -87,
+      -13,   -97,   -10,   -79,   -5,    -41,  -42,  -23,   -67,  8,     -74,
+      15,    -75,   -43,   -63,   -41,   -18,  -81,  -7,    -69,  6,     -49,
+      2,     -9,    -7,    21,    -8,    24,   -11,  44,    32,   44,    38,
+      36,    66,    51,    91,    52,    103,  44,   90,    55,   87,    41,
+      60,    28,    49},
+     /* IRC_Composite_C_R0195_T150_P330.wav */
+     {15,    -18,   20,    -22,   25,    -28,   31,    -35,  38,   -42,  47,
+      -52,   57,    -63,   70,    -76,   84,    -91,   97,   -95,  51,   -395,
+      348,   -546,  384,   -515,  230,   -347,  78,    797,  314,  1274, 475,
+      2177,  1227,  -4216, -2348, 10076, 258,   3084,  3609, 6149, 5472, 3566,
+      -4049, -1019, 5212,  3739,  -5321, -3637, -1580, 2270, 1066, -633, -1395,
+      -136,  1340,  -32,   75,    -243,  507,   -628,  158,  108,  -226, -702,
+      -291,  -112,  -126,  -404,  -330,  -254,  -212,  -362, -151, -436, -14,
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+      -207,  -368,  -232,  -411,  -224,  -349,  -159,  -358, -152, -366, -101,
+      -313,  -105,  -307,  -196,  -327,  -268,  -341,  -250, -319, -245, -350,
+      -228,  -325,  -151,  -267,  -108,  -294,  -109,  -251, -84,  -251, -104,
+      -207,  -91,   -253,  -103,  -244,  -91,   -252,  -85,  -272, -130, -311,
+      -129,  -241,  -94,   -173,  -90,   -147,  -80,   -140, -35,  -126, -79,
+      -186,  -118,  -197,  -120,  -210,  -155,  -272,  -176, -269, -156, -268,
+      -151,  -277,  -170,  -275,  -153,  -283,  -181,  -265, -186, -257, -194,
+      -217,  -154,  -209,  -128,  -189,  -106,  -201,  -80,  -156, -34,  -154,
+      -17,   -151,  12,    -97,   27,    -70,   9,     -39,  -39,  -3,   -46,
+      -29,   -50,   -61,   -28,   -93,   -9,    -89,   -8,   -103, 1,    -74,
+      -1,    -70,   -9,    -35,   -44,   -51,   -51,   -29,  -31,  -30,  -13,
+      -35,   10,    8,     7,     -4,    11,    13,    35,   12,   38,   37,
+      49,    36,    56,    47,    69,    81,    52,    69,   15,   90,   11,
+      52,    -17,   56},
+     /* IRC_Composite_C_R0195_T165_P330.wav */
+     {-38,  38,    -39,  39,    -39,  40,    -40,  41,    -42,  43,   -44,
+      46,   -47,   49,   -52,   55,   -59,   64,   -70,   79,   -90,  108,
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+      5399, 3026,  943,  -4554, 1848, 1742,  -912, -4176, -875, -267, 1882,
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+      -381, -194,  -377, -183,  -352, -126,  -339, -126,  -286, -92,  -262,
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+      -219, -79,   -217, -73,   -199, -48,   -168, -54,   -223, -113, -224,
+      -121, -228,  -134, -243,  -180, -267,  -148, -265,  -176, -290, -166,
+      -274, -160,  -279, -151,  -277, -159,  -262, -146,  -233, -125, -202,
+      -129, -198,  -90,  -162,  -74,  -165,  -69,  -153,  -33,  -141, -13,
+      -136, 2,     -114, 40,    -123, 47,    -102, 43,    -95,  48,   -108,
+      50,   -112,  44,   -100,  32,   -92,   18,   -95,   22,   -82,  17,
+      -82,  23,    -102, 19,    -84,  22,    -128, 24,    -87,  44,   -69,
+      35,   -66,   63,   -29,   79,   -26,   54,   -19,   109,  -9,   110,
+      -5,   137,   -8,   111,   4,    120,   34,   92,    -13,  96,   -16,
+      93,   -54,   92},
+     /* IRC_Composite_C_R0195_T180_P330.wav */
+     {-26,   26,   -26,  25,   -25,  25,   -24,   23,    -23,   22,    -21,
+      20,    -19,  17,   -16,  14,   -11,  8,     -4,    -1,    7,     -15,
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+      526,   177,  1579, 111,  1304, -624, 3003,  -1570, -1854, 2371,  5215,
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+      -109,  -323, 76,   15,   16,   -276, 4,     -8,    -1,    -123,  -108,
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+      -184,  -146, -121, -121, -86,  -121, -59,   -86,   -47,   -78,   -63,
+      -71,   -90,  -66,  -82,  -53,  -82,  -82,   -54,   -72,   -16,   -90,
+      -24,   -88,  -44,  -31,  -45,  -32,  -70,   -43,   -25,   -59,   -33,
+      -76,   -24,  -59,  -46,  -55,  -45,  -56,   -56,   -64,   -32,   -54,
+      -6,    -57,  -1,   -22,  23,   2,    9,     -2,    35,    54,    34,
+      72,    22,   84,   15,   94,   10,   75,    34,    67,    47,    21,
+      57,    14,   36},
+     /* IRC_Composite_C_R0195_T195_P330.wav */
+     {19,   -19,  19,   -20,  20,    -21,  21,    -22,  23,   -23,  24,    -24,
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+      1166, 711,  738,  -60,  1508,  162,  166,   165,  449,  1532, 5355,  3392,
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+      -70,  -61,  -76,  -83,  -71,   -84,  -52,   -102, -40,  -87,  -65,   -74,
+      -59,  -59,  -44,  -53,  -35,   -32,  -38,   -24,  -30,  -33,  -41,   -31,
+      -32,  -32,  -53,  -10,  -53,   7,    -39,   3,    11,   -3,   30,    29,
+      44,   32,   63,   46,   79,    39,   81,    48,   51,   38,   73,    36,
+      50,   22,   77,   -11},
+     /* IRC_Composite_C_R0195_T210_P330.wav */
+     {4,    -4,   4,    -4,   4,    -4,   4,    -4,   4,    -4,   4,     -4,
+      4,    -4,   4,    -4,   4,    -4,   4,    -3,   3,    -3,   2,     -2,
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+      91,   -96,  90,   124,  958,  676,  187,  107,  -283, 310,  1299,  1265,
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+      355,  37,   155,  -122, 149,  -90,  17,   -126, 0,    -119, -101,  -98,
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+      -258, -201, -269, -174, -249, -165, -249, -157, -236, -157, -222,  -150,
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+      -248, -203, -236, -188, -226, -202, -206, -202, -200, -198, -175,  -174,
+      -139, -154, -129, -144, -119, -116, -115, -116, -101, -118, -88,   -110,
+      -115, -124, -128, -104, -139, -117, -161, -110, -128, -96,  -116,  -100,
+      -91,  -78,  -46,  -54,  -38,  -54,  -28,  -28,  -31,  -22,  -47,   -22,
+      -45,  -45,  -34,  -47,  -26,  -63,  -19,  -69,  15,   -46,  17,    -34,
+      36,   -7,   28,   11,   12,   21,   18,   27,   33,   12,   47,    14,
+      27,   2,    38,   -2},
+     /* IRC_Composite_C_R0195_T225_P330.wav */
+     {-1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,
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+      -160, -123, -166, -123, -164, -100, -174, -109, -180, -112, -164, -109,
+      -168, -131, -171, -106, -158, -104, -162, -121, -153, -114, -132, -112,
+      -125, -95,  -82,  -74,  -52,  -73,  -50,  -54,  -44,  -32,  -38,  -20,
+      -57,  -17,  -36,  -14,  -46,  -36,  -11,  -10,  -23,  -34,  -7,   -16,
+      7,    -20,  -30,  -21,  -13,  -19,  -3,   -32,  -10,  -25,  9,    -16,
+      17,   18,   -2,   23},
+     /* IRC_Composite_C_R0195_T240_P330.wav */
+     {6,    -6,   6,    -6,   6,    -6,   6,    -6,   7,    -7,   7,    -7,
+      7,    -7,   7,    -7,   7,    -7,   7,    -7,   8,    -8,   8,    -8,
+      8,    -8,   8,    -8,   1,    -47,  -37,  -124, -93,  -95,  -58,  -38,
+      -70,  -58,  -105, -121, -151, -76,  14,   175,  104,  51,   147,  291,
+      191,  -40,  158,  429,  1090, 1311, 1373, 1241, 1326, 1526, 1181, 419,
+      583,  828,  657,  621,  656,  466,  551,  572,  739,  747,  685,  612,
+      423,  667,  583,  601,  619,  480,  483,  323,  428,  299,  276,  192,
+      162,  105,  83,   89,   89,   51,   98,   -46,  97,   -2,   117,  -65,
+      34,   -69,  1,    -69,  -69,  -127, -170, -131, -173, -121, -145, -147,
+      -134, -204, -102, -232, -83,  -273, -123, -325, -159, -309, -183, -275,
+      -249, -249, -252, -185, -266, -178, -245, -201, -259, -208, -248, -243,
+      -270, -238, -253, -227, -254, -209, -249, -203, -269, -184, -271, -141,
+      -258, -121, -230, -113, -188, -120, -198, -167, -178, -200, -214, -242,
+      -233, -264, -244, -276, -244, -267, -267, -261, -267, -239, -274, -233,
+      -272, -226, -277, -260, -268, -247, -282, -253, -289, -226, -270, -213,
+      -255, -189, -247, -158, -192, -146, -173, -170, -171, -177, -143, -184,
+      -151, -187, -146, -147, -141, -154, -121, -150, -139, -166, -139, -155,
+      -135, -176, -141, -174, -144, -174, -160, -177, -162, -156, -152, -157,
+      -114, -126, -89,  -103, -86,  -98,  -78,  -68,  -62,  -53,  -84,  -35,
+      -53,  -7,   -51,  -15,  -38,  -6,   -34,  -13,  -34,  -15,  -32,  -28,
+      -35,  -19,  -32,  1,    -19,  7,    -27,  9,    3,    4,    -14,  3,
+      -5,   36,   2,    41},
+     /* IRC_Composite_C_R0195_T255_P330.wav */
+     {-1,   1,    -1,   1,    -1,   1,    -2,   2,    -2,   2,    -2,   2,
+      -2,   2,    -2,   2,    -2,   3,    -3,   3,    -3,   3,    -3,   4,
+      -4,   4,    -5,   5,    -6,   7,    -8,   10,   -13,  22,   -83,  -97,
+      -106, -152, -99,  -59,  -90,  -92,  -128, -74,  -117, -122, -12,  68,
+      86,   137,  156,  199,  190,  329,  533,  767,  1036, 1233, 1574, 1651,
+      1491, 1169, 954,  781,  906,  1168, 950,  370,  704,  786,  629,  605,
+      494,  653,  561,  698,  550,  738,  501,  619,  385,  441,  218,  218,
+      189,  132,  100,  39,   147,  27,   91,   -38,  77,   -24,  27,   -36,
+      -30,  -17,  -41,  -95,  -49,  -122, -68,  -169, -78,  -186, -103, -225,
+      -135, -197, -120, -153, -145, -183, -142, -167, -150, -231, -159, -244,
+      -153, -214, -165, -219, -193, -224, -173, -215, -157, -238, -162, -231,
+      -149, -238, -155, -243, -185, -203, -182, -175, -206, -184, -214, -165,
+      -184, -162, -168, -165, -138, -160, -133, -192, -158, -236, -216, -272,
+      -273, -290, -295, -306, -316, -279, -301, -261, -303, -259, -299, -287,
+      -289, -292, -273, -302, -247, -283, -207, -253, -202, -230, -201, -189,
+      -205, -172, -192, -162, -193, -165, -169, -166, -191, -174, -179, -167,
+      -155, -172, -157, -151, -142, -148, -140, -156, -161, -162, -168, -167,
+      -172, -156, -177, -164, -174, -152, -168, -156, -171, -153, -173, -152,
+      -171, -139, -162, -124, -144, -85,  -111, -53,  -108, -40,  -71,  -33,
+      -75,  -21,  -62,  -20,  -68,  0,    -50,  5,    -37,  -3,   -30,  28,
+      -23,  3,    -23,  12,   -19,  28,   -12,  24,   -5,   41,   17,   24,
+      -7,   17,   -24,  23},
+     /* IRC_Composite_C_R0195_T270_P330.wav */
+     {0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
+      0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
+      0,    0,    0,    1,    -1,   1,    -1,   1,    -1,   1,    -2,   3,
+      -127, -152, -188, -165, -160, -178, -195, -131, -72,  -37,  -53,  -57,
+      -34,  48,   106,  135,  127,  188,  385,  765,  884,  1201, 1389, 1575,
+      1756, 1541, 1264, 1282, 908,  615,  982,  1053, 498,  403,  667,  706,
+      647,  681,  663,  732,  712,  700,  422,  527,  505,  426,  401,  192,
+      156,  184,  190,  100,  26,   -34,  55,   -50,  19,   -119, 28,   -59,
+      -54,  -94,  -103, -24,  -139, -24,  -165, -13,  -154, -5,   -159, -81,
+      -104, -118, -99,  -136, -98,  -118, -106, -89,  -142, -101, -168, -112,
+      -178, -178, -148, -180, -114, -217, -134, -255, -104, -252, -140, -238,
+      -148, -235, -187, -236, -202, -242, -197, -183, -140, -137, -108, -143,
+      -129, -142, -128, -211, -166, -246, -182, -269, -212, -271, -233, -292,
+      -267, -287, -263, -282, -269, -312, -249, -304, -257, -293, -240, -287,
+      -230, -257, -242, -248, -234, -224, -236, -219, -241, -188, -227, -176,
+      -221, -206, -210, -185, -185, -166, -179, -163, -176, -164, -168, -164,
+      -186, -168, -180, -154, -159, -141, -142, -150, -142, -147, -152, -145,
+      -159, -150, -168, -167, -187, -177, -186, -176, -192, -163, -197, -167,
+      -186, -164, -170, -159, -148, -137, -112, -102, -104, -88,  -70,  -56,
+      -50,  -28,  -30,  -11,  -12,  -16,  -10,  -21,  -31,  -4,   -26,  2,
+      15,   8,    10,   -3,   -2,   5,    10,   12,   -9,   0,    -5,   9,
+      -13,  -19,  -29,  -16},
+     /* IRC_Composite_C_R0195_T285_P330.wav */
+     {0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
+      0,    0,    0,    0,    0,    0,    0,    1,    -1,   1,    -1,   1,
+      -1,   1,    -1,   2,    -2,   3,    -6,   13,   -111, -184, -215, -184,
+      -140, -139, -179, -220, -168, -247, -124, 1,    -91,  -84,  42,   -80,
+      17,   -54,  56,   218,  634,  881,  967,  1181, 1587, 1636, 1273, 1643,
+      1650, 702,  357,  915,  823,  759,  711,  555,  754,  615,  798,  685,
+      720,  707,  793,  638,  630,  510,  595,  460,  429,  211,  188,  178,
+      148,  102,  16,   162,  0,    142,  -40,  140,  -32,  103,  24,   -7,
+      -30,  11,   -24,  -1,   -87,  12,   -168, 51,   -103, 64,   -105, 39,
+      -118, 10,   -91,  -91,  -119, -125, -90,  -149, -75,  -130, -138, -153,
+      -195, -185, -254, -205, -313, -223, -303, -243, -266, -224, -223, -230,
+      -188, -216, -155, -167, -129, -140, -137, -102, -110, -74,  -146, -82,
+      -177, -89,  -192, -110, -215, -160, -231, -232, -220, -242, -237, -262,
+      -238, -271, -237, -259, -263, -234, -261, -228, -278, -222, -281, -252,
+      -268, -270, -265, -263, -228, -244, -187, -230, -176, -214, -164, -194,
+      -141, -175, -152, -173, -143, -161, -171, -179, -174, -184, -157, -194,
+      -163, -211, -148, -196, -144, -205, -158, -219, -155, -190, -162, -178,
+      -166, -174, -156, -186, -169, -200, -161, -200, -167, -208, -167, -218,
+      -158, -186, -143, -155, -119, -123, -104, -117, -86,  -101, -54,  -82,
+      -39,  -73,  -25,  -68,  -7,   -55,  -18,  -42,  23,   -14,  19,   -21,
+      7,    -24,  19,   -4,   2,    -14,  7,    -10,  12,   -14,  13,   -7,
+      26,   -23,  8,    -14},
+     /* IRC_Composite_C_R0195_T300_P330.wav */
+     {-3,   3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,
+      -3,   3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,    -3,   4,
+      -4,   4,    -4,   4,    -4,   4,    -5,   -62,  -157, -83,  -247, -284,
+      -202, -303, -294, -235, -54,  -305, 15,   156,  241,  -112, -479, -226,
+      774,  -232, 495,  1765, 1236, 1265, 2221, 2705, 1763, 850,  505,  963,
+      731,  757,  546,  519,  206,  795,  441,  708,  577,  742,  596,  545,
+      491,  526,  666,  413,  377,  307,  300,  457,  188,  260,  34,   250,
+      133,  251,  58,   87,   249,  132,  194,  -1,   200,  55,   156,  95,
+      139,  122,  50,   62,   57,   84,   -3,   -71,  -62,  -99,  -113, -176,
+      -122, -254, -180, -227, -166, -230, -202, -293, -196, -279, -248, -326,
+      -256, -320, -217, -260, -211, -244, -154, -191, -125, -171, -113, -160,
+      -81,  -135, -55,  -102, -70,  -95,  -24,  -42,  -53,  -78,  -73,  -93,
+      -89,  -149, -139, -195, -180, -235, -192, -245, -233, -268, -234, -245,
+      -224, -256, -221, -249, -225, -220, -233, -215, -250, -226, -218, -236,
+      -221, -224, -181, -198, -176, -191, -173, -185, -156, -184, -162, -178,
+      -171, -172, -156, -184, -183, -174, -192, -186, -179, -189, -180, -212,
+      -184, -191, -167, -199, -217, -182, -203, -172, -210, -178, -224, -155,
+      -220, -185, -224, -199, -194, -203, -201, -206, -192, -196, -188, -165,
+      -158, -177, -147, -145, -112, -128, -98,  -113, -74,  -77,  -54,  -65,
+      -67,  -32,  -44,  -10,  -54,  -13,  -29,  -7,   -15,  -1,   -23,  19,
+      -18,  10,   -27,  11,   -16,  8,    2,    12,   -12,  4,    -1,   37,
+      18,   19,   17,   36},
+     /* IRC_Composite_C_R0195_T315_P330.wav */
+     {0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
+      0,    0,    0,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,
+      1,    -1,   1,    -1,   2,    -3,   -227, -204, -310, -344, -179, -311,
+      -505, -298, -88,  -402, 115,  712,  -76,  -577, -211, 9,    558,  -404,
+      914,  2627, 1047, 2118, 4245, 2542, 908,  788,  1382, 865,  422,  674,
+      -11,  248,  192,  1314, 191,  817,  366,  966,  306,  467,  192,  289,
+      334,  100,  222,  7,    227,  236,  315,  254,  209,  305,  479,  300,
+      240,  335,  445,  260,  199,  112,  202,  84,   42,   63,   85,   -45,
+      -100, -80,  -73,  -157, -181, -120, -209, -167, -259, -161, -262, -234,
+      -281, -209, -243, -208, -221, -245, -173, -195, -197, -241, -151, -201,
+      -166, -240, -132, -182, -104, -150, -85,  -104, -75,  -148, -59,  -141,
+      -36,  -144, -33,  -133, -71,  -89,  -7,   -47,  -109, -70,  -108, -44,
+      -152, -113, -164, -122, -182, -216, -170, -212, -172, -263, -155, -228,
+      -178, -252, -176, -207, -200, -242, -206, -220, -204, -208, -195, -218,
+      -188, -193, -156, -217, -159, -200, -119, -211, -133, -181, -131, -177,
+      -169, -158, -166, -171, -195, -175, -181, -214, -193, -224, -184, -218,
+      -204, -219, -183, -222, -197, -222, -197, -226, -215, -204, -205, -207,
+      -216, -212, -210, -224, -222, -213, -210, -205, -202, -194, -174, -155,
+      -148, -134, -144, -126, -89,  -101, -75,  -94,  -84,  -69,  -47,  -37,
+      -69,  -67,  -26,  -35,  -13,  -29,  -17,  -23,  -3,   13,   -18,  -8,
+      -26,  22,   -30,  20,   -15,  30,   -13,  8,    -11,  23,   49,   30,
+      39,   22,   36,   16},
+     /* IRC_Composite_C_R0195_T330_P330.wav */
+     {9,    -9,   9,    -10,  10,   -10,  10,    -11,  11,   -12,  12,   -12,
+      13,   -13,  14,   -15,  15,   -16,  17,    -17,  18,   -19,  20,   -22,
+      23,   -25,  30,   -50,  -332, -388, -131,  -617, -214, -313, -733, -659,
+      255,  -155, -66,  699,  -77,  -372, -1255, -242, 2041, -707, -6,   4736,
+      2036, 2809, 3545, 4509, 22,   -72,  1207,  1973, -173, 508,  959,  302,
+      542,  540,  656,  123,  461,  665,  400,   314,  251,  504,  190,  265,
+      35,   373,  467,  91,   154,  128,  447,   203,  170,  90,   232,  80,
+      149,  193,  21,   -6,   -90,  -101, -59,   -80,  -255, -144, -154, -358,
+      -29,  -238, -96,  -237, -126, -195, 15,    -288, -107, -147, -89,  -276,
+      -120, -223, -134, -174, -181, -152, -158,  -179, -71,  -138, -116, -177,
+      -93,  -189, -60,  -206, -162, -159, -113,  -111, -150, -123, -93,  -106,
+      -83,  -101, -52,  -113, -43,  -99,  -10,   -99,  -66,  -88,  -58,  -127,
+      -64,  -152, -99,  -152, -144, -200, -127,  -259, -116, -227, -129, -245,
+      -98,  -258, -94,  -239, -152, -227, -154,  -225, -147, -242, -169, -202,
+      -172, -212, -140, -232, -141, -182, -139,  -168, -130, -205, -127, -137,
+      -140, -176, -175, -160, -166, -171, -205,  -188, -193, -212, -199, -238,
+      -185, -248, -173, -246, -186, -227, -206,  -245, -184, -218, -210, -235,
+      -210, -228, -175, -238, -206, -213, -189,  -196, -171, -200, -159, -123,
+      -146, -114, -128, -128, -81,  -88,  -83,   -80,  -52,  -124, -70,  -58,
+      -40,  -45,  -69,  -50,  -10,  -13,  -16,   -42,  -4,   -25,  50,   -37,
+      11,   -37,  37,   -4,   24,   -11,  12,    -7,   33,   10,   69,   29,
+      60,   20,   87,   -1},
+     /* IRC_Composite_C_R0195_T345_P330.wav */
+     {1,    0,    0,    0,     -1,   1,    -2,   2,     -3,   4,    -5,   6,
+      -7,   8,    -9,   11,    -13,  15,   -17,  20,    -24,  28,   -33,  40,
+      -47,  52,   -211, -235,  -688, -550, -297, -690,  -704, -444, 713,  -1568,
+      989,  470,  -39,  -1205, 323,  -714, 276,  -2380, 5748, 2929, 602,  8080,
+      5302, -453, -722, 3812,  715,  -74,  -365, 1626,  369,  1163, 1544, 781,
+      -4,   229,  668,  441,   364,  676,  214,  441,   36,   409,  174,  565,
+      -210, 423,  -90,  172,   -132, 46,   -99,  78,    -186, 159,  -181, 96,
+      -244, 13,   -280, -20,   -156, -41,  -291, -109,  -365, -55,  -322, -117,
+      -274, -136, -169, -159,  -189, -140, -191, 12,    -178, 0,    -295, -70,
+      -206, -88,  -236, -75,   -169, -78,  -191, -95,   -204, -87,  -146, -66,
+      -185, -107, -238, -113,  -249, -119, -185, -98,   -187, -131, -176, -95,
+      -137, -31,  -142, -17,   -163, -50,  -120, -40,   -114, -19,  -100, -43,
+      -142, -64,  -163, -81,   -167, -107, -184, -140,  -220, -122, -200, -146,
+      -203, -155, -165, -178,  -177, -198, -166, -175,  -153, -160, -161, -178,
+      -163, -171, -143, -138,  -136, -139, -146, -131,  -151, -132, -167, -143,
+      -153, -146, -165, -157,  -219, -159, -215, -151,  -222, -190, -225, -204,
+      -218, -208, -214, -189,  -218, -191, -221, -190,  -214, -185, -216, -196,
+      -215, -178, -200, -217,  -218, -194, -162, -182,  -170, -189, -185, -152,
+      -157, -123, -144, -122,  -111, -99,  -99,  -89,   -76,  -63,  -81,  -56,
+      -61,  -31,  -49,  -85,   -23,  -30,  -14,  -9,    -21,  -45,  -18,  -1,
+      16,   3,    5,    -5,    5,    8,    -5,   64,    39,   55,   37,   73,
+      42,   56,   31,   84}};
+
+const int16_t irc_composite_c_r0195_p345[][256] =
+    {/* IRC_Composite_C_R0195_T000_P345.wav */
+     {2,    -1,    1,    -1,    1,    -1,    1,    -1,    0,     0,     0,
+      0,    -1,    1,    -2,    2,    -3,    4,    -5,    7,     -9,    11,
+      -15,  21,    -203, -88,   -927, -196,  -604, -22,   -2141, -496,  -650,
+      1585, -548,  803,  -263,  412,  -5131, 5346, -2103, -7267, 12288, 1672,
+      1403, 10338, 7456, -3182, -874, 2680,  1965, -5296, 1356,  1865,  68,
+      -578, 2523,  1092, 1554,  2056, 2530,  1058, 1432,  379,   1062,  -82,
+      -380, -971,  369,  -376,  62,   -56,   669,  -390,  190,   379,   537,
+      98,   -46,   -35,  315,   348,  110,   183,  491,   -38,   19,    211,
+      170,  -22,   -231, -125,  -221, 138,   -512, -336,  -126,  10,    -387,
+      -311, -283,  -176, -241,  -345, -432,  -207, -330,  -489,  -386,  -392,
+      -351, -423,  -274, -457,  -352, -361,  -311, -213,  -391,  -323,  -302,
+      -143, -325,  -378, -206,  -229, -190,  -319, -334,  -196,  -236,  -126,
+      -387, -117,  -339, -105,  -264, -135,  -207, -172,  -176,  -254,  -160,
+      -183, -177,  -200, -218,  -201, -189,  -116, -205,  -120,  -224,  -113,
+      -159, -119,  -241, -118,  -127, -119,  -188, -91,   -119,  -15,   -186,
+      -30,  -132,  28,   -213,  -19,  -166,  21,   -190,  -46,   -185,  -49,
+      -159, -65,   -168, -65,   -190, -59,   -218, -46,   -217,  -24,   -231,
+      -74,  -204,  -52,  -181,  -89,  -211,  -94,  -196,  -70,   -219,  -104,
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+      -145, -7,    -123, -21,   -161, -34,   -116, -47,   -173,  -82,   -145,
+      -76,  -157,  -72,  -137,  -75,  -186,  -79,  -163,  -31,   -131,  -17,
+      -156, -51,   -99,  26,    -74,  -5,    -101, 37,    -82,   48,    -83,
+      35,   -130,  20},
+     /* IRC_Composite_C_R0195_T015_P345.wav */
+     {-8,    8,     -9,    9,     -9,    10,    -10,   11,    -11,  12,   -13,
+      14,    -15,   17,    -19,   21,    -23,   26,    -30,   34,   -31,  -453,
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+      -226,  -35,   -259,  -51,   -231,  -34,   -263,  -76,   -247, -60,  -275,
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+      -252,  -5,    -165,  49,    -233,  62,    -170,  73,    -217, 68,   -124,
+      45,    -174,  23,    -102,  4,     -167,  36,    -98,   28,   -187, 19,
+      -138,  2,     -165,  -15,   -133,  -54,   -155,  -42,   -139, -76,  -178,
+      -15,   -149,  -36,   -176,  14,    -154,  -30,   -174,  22,   -136, -10,
+      -157,  29,    -146,  8,     -148,  67,    -147,  48,    -150, 87,   -144,
+      95,    -120,  68},
+     /* IRC_Composite_C_R0195_T030_P345.wav */
+     {-5,    4,    -2,    1,     1,     -3,    5,    -8,    11,    -14,
+      18,    -22,  28,    -33,   40,    -47,   52,   -40,   -408,  -1246,
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+      -57,   -149, 39,    -73,   32,    -134,  21,   -60,   58,    -96,
+      16,    -33,  13,    -37,   -10,   -59,   0,    -62,   -11,   -108,
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+      -43,   -94,  -30,   -105,  -18,   -125,  -14,  -97,   30,    -123,
+      23,    -65,  42,    -66,   35,    -56},
+     /* IRC_Composite_C_R0195_T045_P345.wav */
+     {-8,    9,     -9,   10,    -11,   12,   -14,   15,    -17,   18,    -20,
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+      -74,   -97,   -50,  -98,   -31,   -116, 11,    -85,   32,    -106,  43,
+      -88,   79,    -70},
+     /* IRC_Composite_C_R0195_T060_P345.wav */
+     {2,     0,     -1,   2,     -4,   7,     -10,   13,   -18,   24,    -33,
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+      -16,   -79,   -18,  -107,  -18,  -114,  -3,    -27,  -1,    -36,   -19,
+      -21,   -10,   39},
+     /* IRC_Composite_C_R0195_T075_P345.wav */
+     {13,    -13,   13,    -13,   13,   -13,   13,   -14,   14,   -14,  14,
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+      -61,   -44,   -75,   -23,   -52,  -44,   10,   -43,   -23,  -47,  63,
+      -13,   31,    19},
+     /* IRC_Composite_C_R0195_T090_P345.wav */
+     {-40,   45,   -52,   60,   -69,  79,    -92,   108,   -129, 156,   -197,
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+      -30,   -55,  4,     -41,  13,   -55,   20,    -26,   32,   -22,   71,
+      -18,   17,   30},
+     /* IRC_Composite_C_R0195_T105_P345.wav */
+     {-67,   70,    -74,   77,    -81,   86,    -90,   96,    -101,   108,
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+      -10,   30,    -29,   94,    17,    67},
+     /* IRC_Composite_C_R0195_T120_P345.wav */
+     {-3,   4,     -6,    7,     -9,    12,    -14,  18,     -21,  26,    -32,
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+      -134, -170,  -168,  -181,  -178,  -179,  -104, -105,   -182, -117,  -16,
+      21,   -1,    -16,   -123,  -97,   -70,   -125, -190,   -191, -224,  -174,
+      -191, -187,  -240,  -111,  -186,  -142,  -146, -92,    -134, -84,   -176,
+      -166, -110,  -109,  -146,  -139,  -231,  -119, -73,    -70,  -193,  -83,
+      -143, -72,   -63,   -63,   -163,  -69,   -90,  -96,    -97,  -84,   -157,
+      -37,  -91,   -88,   -116,  -29,   -99,   -31,  -73,    -68,  -76,   24,
+      -65,  -8,    -90,   -71,   -32,   33,    -119, -46,    -74,  -20,   -92,
+      37,   -144,  -45,   -100,  37,    -86,   3,    -132,   2,    -42,   -29,
+      -104, -4,    -152,  -16,   -85,   -19,   -175, 14,     -94,  -9,    -106,
+      -24,  -72,   42,    -77,   -27,   -22,   2,    -33,    78,   -29,   -21,
+      21,   111,   -5},
+     /* IRC_Composite_C_R0195_T135_P345.wav */
+     {-8,    8,     -9,   9,     -9,    10,    -10,  11,    -11,  11,    -12,
+      12,    -12,   13,   -12,   12,    -14,   -199, -345,  194,  -292,  -20,
+      -363,  219,   -924, 441,   900,   798,   -558, 3720,  1651, -3208, -514,
+      -4991, 13814, 1350, -1342, 10873, 8019,  1098, -3160, 1031, 6880,  5706,
+      -3917, -4208, 1661, 708,   -1088, -1854, -201, 229,   -177, -273,  -648,
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+      -269,  -147,  -370, -183,  -325,  -201,  -333, -191,  -269, -230,  -243,
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+      -71,   -63,   -76,  -143,  -84,   -111,  -151, -139,  -218, -149,  -229,
+      -159,  -185,  -165, -192,  -165,  -156,  -158, -137,  -140, -159,  -114,
+      -167,  -123,  -164, -97,   -148,  -122,  -112, -140,  -90,  -115,  -78,
+      -111,  -104,  -86,  -114,  -49,   -142,  12,   -166,  -28,  -168,  -17,
+      -113,  -58,   -89,  -81,   -116,  -67,   -58,  -65,   -96,  -68,   -92,
+      -33,   -99,   -37,  -87,   -67,   -90,   -37,  -68,   -114, -65,   -87,
+      -50,   -124,  -30,  -83,   -1,    -96,   -4,   -55,   -19,  -87,   13,
+      -80,   -10,   -90,  -5,    -84,   -42,   -79,  4,     -67,  -63,   -38,
+      -27,   -36,   -23,  -7,    11,    -34,   14,   -18,   51,   -74,   41,
+      9,     19,    -7},
+     /* IRC_Composite_C_R0195_T150_P345.wav */
+     {9,     -10,   11,   -12,   13,    -14,   16,   -18,   20,   -22,   25,
+      -28,   32,    -36,  41,    -48,   57,    -69,  92,    -187, -93,   -65,
+      148,   -471,  61,   62,    -108,  -513,  1146, 660,   197,  907,   4643,
+      -2481, -3069, 2384, -436,  10331, -1444, 3529, 10790, 4133, -3114, 1101,
+      5334,  2231,  175,  -4639, -668,  1364,  -942, -1832, -179, 106,   -201,
+      -621,  -288,  256,  -14,   7,     -70,   10,   -62,   -163, -538,  -397,
+      250,   -662,  -393, -549,  -98,   -597,  -308, -388,  -216, -452,  -393,
+      -291,  -335,  -411, -150,  -445,  -160,  -290, -89,   -416, -142,  -299,
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+      -265,  -215,  -277, -207,  -240,  -196,  -255, -168,  -178, -155,  -144,
+      -107,  -152,  -110, -104,  -74,   -180,  -121, -180,  -122, -196,  -144,
+      -245,  -129,  -207, -127,  -182,  -107,  -225, -82,   -192, -124,  -217,
+      -97,   -227,  -156, -196,  -108,  -174,  -119, -141,  -77,  -130,  -30,
+      -126,  -49,   -162, -24,   -161,  -3,    -167, -53,   -117, -53,   -120,
+      -68,   -81,   -70,  -85,   -45,   -100,  -52,  -108,  -40,  -95,   -39,
+      -123,  -50,   -123, -64,   -117,  -58,   -114, -101,  -106, -61,   -84,
+      -90,   -69,   -81,  -96,   -40,   -52,   -52,  -64,   -50,  -43,   -17,
+      -37,   -37,   -34,  -28,   -67,   -18,   -49,  -26,   -48,  -2,    -23,
+      -34,   -24,   7,    4,     -14,   13,    9,    -1,    -21,  7,     5,
+      3,     -11,   -9},
+     /* IRC_Composite_C_R0195_T165_P345.wav */
+     {-16,  16,    -16,  16,    -16,   17,   -17,   17,   -17,  18,   -18,
+      18,   -18,   19,   -19,   19,    -19,  19,    -19,  19,   -18,  15,
+      -25,  -180,  39,   144,   -199,  84,   -114,  -19,  -188, 1254, 741,
+      544,  587,   3703, -1657, -2617, 2350, 59,    8499, 1392, 3068, 7765,
+      4121, -1476, -112, 3753,  -281,  908,  -3882, -912, -13,  232,  -1645,
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+      -91,  -166,  -97,  -138,  -90,   -161, -94,   -134, -93,  -167, -101,
+      -146, -89,   -171, -96,   -130,  -90,  -122,  -75,  -116, -77,  -102,
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+      -89,  -86,   -78,  -89,   -91,   -110, -85,   -90,  -61,  -78,  -110,
+      -68,  -88,   -12,  -123,  -18,   -120, 12,    -117, 12,   -119, 3,
+      -74,  28,    -74,  -4,    -63,   7,    -46,   0,    -67,  7,    -52,
+      21,   -52,   5,    -33,   17,    5,    -5,    -18,  -6,   3,    -14,
+      -17,  -13,   -5},
+     /* IRC_Composite_C_R0195_T180_P345.wav */
+     {14,   -14,  15,    -15,  15,    -16,  16,   -17,  18,   -18,  19,   -20,
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+      -152, -105, -161,  -118, -130,  -105, -117, -95,  -88,  -108, -66,  -84,
+      -63,  -104, -55,   -48,  -36,   -55,  -20,  -46,  -8,   -40,  5,    -55,
+      -25,  -61,  -13,   -59,  -44,   -51,  -47,  -33,  -43,  -29,  -37,  -41,
+      -40,  -27,  -26,   -28,  -32,   -23,  -2,   -7,   24,   -14,  -21,  -16,
+      -7,   -26,  -52,   -44},
+     /* IRC_Composite_C_R0195_T195_P345.wav */
+     {6,    -6,   6,    -7,   7,     -7,   7,     -7,   7,    -7,    8,    -8,
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+      -193, -139, -163, -148, -164,  -142, -130,  -130, -110, -101,  -105, -93,
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+      -15,  -43,  -40,  -44,  -53,   -61,  -41,   -51,  -45,  -65,   -44,  -54,
+      -43,  -56,  -70,  -49,  -64,   -4,   -42,   4,    -66,  5,     -19,  28,
+      -23,  -34,  -32,  -54},
+     /* IRC_Composite_C_R0195_T210_P345.wav */
+     {-2,   2,    -2,   2,    -2,   2,    -2,   1,    -1,    1,    -1,   1,
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+      7,    -9,   13,   -20,  43,   -89,  -222, 10,   103,   -107, -34,  -36,
+      142,  -400, -30,  464,  582,  998,  189,  -102, 357,   928,  -496, 3,
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+      -128, -148, -134, -149, -163, -162, -154, -157, -161,  -181, -160, -162,
+      -162, -166, -161, -175, -145, -160, -146, -183, -159,  -151, -121, -142,
+      -111, -108, -118, -82,  -70,  -76,  -70,  -81,  -45,   -58,  -32,  -55,
+      -61,  -50,  -56,  -13,  -58,  -19,  -65,  -23,  -55,   -34,  -79,  -47,
+      -53,  -75,  -77,  -59,  -49,  -49,  -45,  -23,  -65,   -15,  -50,  -1,
+      -35,  -44,  -51,  -18},
+     /* IRC_Composite_C_R0195_T225_P345.wav */
+     {4,    -4,   4,    -4,   5,    -5,   5,    -5,   5,    -5,   5,    -5,
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+      -173, -198, -185, -178, -167, -163, -174, -180, -175, -186, -163, -152,
+      -140, -147, -132, -122, -102, -97,  -72,  -79,  -75,  -66,  -54,  -46,
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+      -84,  -64,  -71,  -62,  -66,  -82,  -71,  -59,  -36,  -53,  -39,  -61,
+      -45,  -48,  -32,  -64},
+     /* IRC_Composite_C_R0195_T240_P345.wav */
+     {1,    -1,   1,    -1,   1,    -1,   1,    -2,   2,    -2,   2,    -2,
+      2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,   3,    -3,
+      3,    -3,   4,    -4,   5,    -7,   11,   -87,  -91,  -69,  -34,  -17,
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+      -250, -237, -259, -286, -308, -300, -276, -316, -306, -323, -315, -283,
+      -322, -259, -334, -262, -308, -246, -279, -237, -252, -210, -235, -204,
+      -211, -204, -216, -219, -187, -208, -184, -188, -172, -182, -147, -178,
+      -171, -209, -148, -207, -140, -214, -153, -202, -132, -178, -141, -158,
+      -129, -140, -132, -148, -104, -170, -94,  -160, -103, -179, -127, -186,
+      -148, -195, -164, -196, -178, -176, -178, -191, -206, -200, -201, -208,
+      -201, -208, -200, -195, -166, -167, -160, -164, -138, -176, -171, -170,
+      -145, -157, -125, -127, -91,  -104, -77,  -96,  -65,  -75,  -71,  -96,
+      -65,  -84,  -60,  -84,  -57,  -72,  -49,  -85,  -51,  -85,  -52,  -82,
+      -51,  -93,  -48,  -69,  -42,  -81,  -54,  -69,  -69,  -69,  -62,  -61,
+      -61,  -54,  -82,  -67},
+     /* IRC_Composite_C_R0195_T255_P345.wav */
+     {1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,
+      1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,
+      1,    -1,   2,    -2,   2,    -2,   2,    -2,   3,    -3,   5,    -29,
+      -81,  -145, -126, -102, -79,  -99,  -83,  -32,  26,   -25,  -104, -85,
+      -9,   -133, -67,  -57,  165,  465,  613,  590,  447,  814,  1074, 1050,
+      1231, 1480, 1384, 1691, 1545, 721,  693,  1027, 967,  911,  897,  961,
+      823,  595,  850,  817,  872,  522,  568,  476,  335,  222,  147,  215,
+      189,  255,  142,  248,  142,  175,  30,   4,    8,    -59,  -17,  -77,
+      -20,  -108, -60,  -97,  -148, -132, -187, -115, -138, -127, -111, -127,
+      -113, -156, -101, -162, -102, -180, -108, -179, -120, -179, -195, -199,
+      -229, -185, -270, -215, -278, -237, -258, -241, -261, -210, -244, -198,
+      -260, -213, -287, -247, -340, -255, -344, -240, -343, -221, -330, -201,
+      -299, -199, -288, -250, -262, -234, -258, -219, -245, -213, -243, -209,
+      -226, -210, -206, -229, -185, -199, -171, -204, -168, -186, -151, -175,
+      -154, -197, -147, -168, -130, -180, -149, -171, -122, -158, -129, -152,
+      -132, -156, -139, -160, -164, -170, -174, -154, -162, -126, -151, -134,
+      -162, -148, -177, -191, -199, -212, -200, -209, -220, -215, -220, -217,
+      -209, -214, -187, -188, -163, -169, -145, -176, -134, -160, -141, -157,
+      -132, -139, -118, -136, -115, -131, -104, -132, -107, -134, -96,  -102,
+      -76,  -89,  -80,  -83,  -80,  -72,  -65,  -58,  -64,  -74,  -61,  -66,
+      -46,  -79,  -66,  -94,  -77,  -85,  -74,  -86,  -77,  -80,  -67,  -80,
+      -78,  -81,  -61,  -58},
+     /* IRC_Composite_C_R0195_T270_P345.wav */
+     {1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,
+      1,    -1,   1,    -1,   2,    -2,   2,    -2,   2,    -2,   2,    -2,
+      2,    -3,   3,    -3,   3,    -3,   4,    -4,   5,    -5,   7,    -13,
+      -9,   25,   107,  159,  235,  223,  228,  197,  135,  148,  180,  104,
+      249,  332,  345,  533,  679,  812,  953,  1029, 929,  1077, 1230, 1424,
+      1718, 1696, 1725, 1631, 997,  986,  1012, 146,  -166, 111,  353,  347,
+      -6,   -43,  73,   193,  90,   62,   -93,  -92,  -338, -310, -405, -336,
+      -407, -441, -401, -511, -347, -531, -317, -432, -349, -454, -400, -327,
+      -440, -322, -442, -318, -433, -382, -424, -384, -375, -393, -396, -458,
+      -428, -480, -383, -434, -382, -464, -371, -432, -337, -394, -307, -399,
+      -335, -417, -322, -442, -324, -436, -303, -449, -304, -409, -298, -363,
+      -284, -317, -307, -308, -323, -288, -319, -293, -322, -285, -281, -248,
+      -239, -242, -269, -244, -286, -239, -275, -228, -281, -236, -270, -215,
+      -230, -183, -189, -147, -168, -142, -140, -138, -148, -145, -122, -112,
+      -122, -100, -101, -92,  -130, -111, -155, -125, -154, -111, -187, -117,
+      -190, -95,  -171, -106, -163, -109, -148, -117, -153, -132, -173, -135,
+      -163, -114, -185, -122, -191, -116, -168, -92,  -150, -108, -141, -94,
+      -122, -95,  -150, -97,  -145, -88,  -132, -65,  -125, -53,  -97,  -33,
+      -83,  -12,  -48,  -5,   -27,  7,    -11,  35,   -15,  26,   -18,  26,
+      -29,  3,    -32,  -9,   -39,  15,   11,   7,    13,   12,   6,    12,
+      11,   20,   5,    2,    4,    3,    -12,  -24,  -23,  -5,   -3,   -22,
+      4,    5,    1,    4},
+     /* IRC_Composite_C_R0195_T285_P345.wav */
+     {0,    0,    0,    0,    0,    0,    0,    0,    0,    -1,   1,    -1,
+      1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -2,   2,    -2,
+      2,    -2,   3,    -3,   3,    -4,   5,    -6,   8,    -11,  22,   -196,
+      -173, -113, -87,  -175, -179, -137, -214, -236, -209, -197, -203, -89,
+      115,  94,   63,   76,   -21,  -123, -58,  550,  837,  1139, 1631, 2133,
+      1768, 1607, 1992, 1501, 200,  482,  1172, 950,  840,  796,  772,  845,
+      1106, 940,  1271, 764,  758,  561,  601,  592,  456,  196,  274,  22,
+      269,  50,   133,  94,   119,  18,   4,    153,  30,   97,   24,   85,
+      20,   33,   14,   -38,  -26,  -152, -46,  -172, -60,  -155, -115, -148,
+      -198, -89,  -195, -112, -261, -100, -214, -212, -183, -197, -150, -219,
+      -140, -179, -164, -230, -178, -187, -230, -226, -257, -158, -281, -201,
+      -228, -189, -261, -210, -220, -259, -267, -256, -233, -276, -267, -249,
+      -227, -294, -254, -243, -276, -267, -297, -257, -292, -260, -305, -245,
+      -279, -264, -243, -230, -212, -214, -151, -148, -130, -127, -88,  -74,
+      -103, -75,  -88,  -67,  -122, -69,  -96,  -106, -123, -96,  -146, -128,
+      -172, -140, -152, -152, -190, -154, -179, -153, -191, -155, -209, -127,
+      -197, -114, -197, -134, -193, -131, -195, -130, -172, -126, -176, -158,
+      -191, -169, -206, -154, -203, -183, -230, -172, -205, -185, -204, -179,
+      -196, -157, -166, -116, -150, -115, -126, -82,  -123, -71,  -85,  -83,
+      -83,  -68,  -72,  -63,  -78,  -79,  -49,  -75,  -79,  -92,  -70,  -98,
+      -66,  -95,  -75,  -105, -104, -85,  -82,  -92,  -105, -73,  -107, -87,
+      -82,  -75,  -79,  -91},
+     /* IRC_Composite_C_R0195_T300_P345.wav */
+     {-1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   2,    -2,   2,
+      -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,   3,    -3,   3,
+      -3,   4,    -4,   5,    -6,   8,    -10,  16,   -33,  4,    109,  115,
+      258,  119,  96,   128,  286,  309,  258,  544,  795,  778,  461,  339,
+      836,  601,  406,  1682, 1975, 1154, 2276, 2834, 1223, 701,  931,  69,
+      -668, -254, 489,  635,  -539, 173,  339,  531,  396,  94,   191,  -102,
+      150,  -127, -75,  -251, -176, -409, -489, -562, -686, -623, -666, -571,
+      -644, -520, -569, -434, -437, -442, -489, -355, -376, -438, -545, -399,
+      -497, -467, -538, -479, -549, -412, -479, -420, -458, -466, -435, -392,
+      -372, -360, -342, -413, -353, -363, -348, -419, -344, -451, -370, -415,
+      -367, -468, -389, -500, -406, -491, -367, -474, -360, -481, -350, -435,
+      -296, -421, -285, -402, -267, -363, -222, -350, -221, -334, -215, -270,
+      -177, -235, -146, -214, -118, -175, -89,  -140, -81,  -133, -96,  -92,
+      -54,  -91,  -56,  -86,  -58,  -64,  -32,  -51,  -40,  -76,  -57,  -55,
+      -36,  -113, -96,  -83,  -105, -91,  -129, -152, -109, -131, -124, -142,
+      -150, -156, -119, -136, -143, -152, -125, -140, -130, -126, -129, -99,
+      -125, -119, -110, -121, -109, -118, -102, -117, -98,  -117, -101, -93,
+      -114, -105, -94,  -69,  -112, -75,  -86,  -60,  -49,  -25,  -59,  6,
+      -33,  -1,   -8,   8,    -2,   17,   23,   0,    29,   13,   17,   24,
+      18,   21,   18,   23,   -13,  11,   -8,   15,   5,    10,   -10,  8,
+      -24,  7,    1,    -11,  1,    27,   -4,   48,   5,    53,   24,   86,
+      43,   68,   50,   99},
+     /* IRC_Composite_C_R0195_T315_P345.wav */
+     {1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -2,
+      2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,
+      2,    -2,   2,    -2,   2,    -1,   0,    6,    -137, -110, -314, -483,
+      -264, -184, -467, -482, 89,   -336, 204,  277,  141,  -93,  -770, -71,
+      819,  -837, 1577, 3127, 1312, 2383, 3512, 2495, -80,  350,  890,  699,
+      71,   561,  954,  294,  1113, 1117, 1259, 495,  984,  849,  877,  553,
+      746,  524,  537,  393,  441,  428,  542,  429,  296,  210,  204,  231,
+      133,  -55,  33,   -7,   -111, -11,  -91,  34,   -190, -56,  -131, 39,
+      -14,  66,   -79,  58,   -96,  105,  -60,  169,  -64,  84,   -143, 118,
+      -100, -84,  -157, -91,  -304, -233, -224, -201, -251, -259, -309, -306,
+      -307, -286, -299, -265, -407, -357, -301, -345, -348, -313, -294, -342,
+      -334, -338, -298, -266, -288, -261, -298, -254, -268, -227, -236, -260,
+      -244, -223, -227, -260, -233, -195, -202, -171, -228, -156, -170, -114,
+      -170, -107, -146, -105, -89,  -96,  -83,  -40,  -73,  -59,  -66,  -65,
+      -71,  -53,  -107, -74,  -75,  -92,  -122, -102, -105, -87,  -130, -86,
+      -139, -115, -142, -131, -148, -147, -160, -144, -151, -139, -173, -139,
+      -186, -126, -170, -147, -194, -137, -189, -141, -184, -162, -186, -151,
+      -208, -180, -195, -172, -186, -165, -204, -165, -157, -155, -139, -132,
+      -141, -108, -126, -89,  -110, -98,  -101, -113, -94,  -111, -133, -99,
+      -126, -121, -126, -97,  -115, -103, -122, -84,  -114, -58,  -120, -106,
+      -109, -97,  -103, -75,  -128, -91,  -104, -73,  -91,  -53,  -59,  -46,
+      -63,  -33,  -22,  -14},
+     /* IRC_Composite_C_R0195_T330_P345.wav */
+     {-9,   9,    -9,   9,    -9,   9,    -9,   9,    -9,    9,    -9,   9,
+      -9,   9,    -9,   9,    -9,   9,    -9,   10,   -10,   10,   -10,  10,
+      -10,  10,   -10,  10,   -7,   -378, -458, 118,  -356,  -539, -320, -251,
+      -977, -150, 272,  171,  262,  290,  -552, -354, -2031, 2393, -54,  -667,
+      4888, 3245, 1575, 3792, 4173, -210, -907, 432,  1658,  -334, 338,  623,
+      940,  723,  1598, 1314, 1216, 764,  1038, 976,  401,   116,  250,  473,
+      239,  334,  528,  294,  340,  65,   340,  194,  348,   152,  304,  92,
+      279,  123,  93,   -24,  89,   -192, -125, -56,  -130,  68,   194,  -86,
+      77,   174,  220,  128,  98,   -234, -216, -168, -160,  -251, -283, -410,
+      -399, -240, -227, -363, -327, -361, -302, -370, -271,  -312, -309, -428,
+      -402, -318, -223, -266, -304, -253, -264, -272, -246,  -203, -278, -299,
+      -319, -254, -236, -260, -281, -188, -203, -212, -220,  -185, -220, -178,
+      -230, -207, -215, -227, -248, -182, -197, -163, -179,  -101, -143, -112,
+      -138, -84,  -115, -82,  -111, -83,  -102, -83,  -51,   -69,  -49,  -89,
+      -18,  -67,  -73,  -108, -72,  -63,  -79,  -111, -167,  -94,  -130, -99,
+      -166, -109, -181, -121, -156, -124, -154, -131, -162,  -169, -165, -170,
+      -136, -151, -159, -185, -163, -122, -139, -141, -175,  -141, -148, -148,
+      -172, -157, -130, -154, -150, -148, -142, -153, -141,  -116, -159, -97,
+      -155, -91,  -162, -92,  -141, -53,  -134, -97,  -132,  -68,  -111, -97,
+      -127, -98,  -128, -105, -137, -81,  -141, -76,  -120,  -73,  -136, -97,
+      -105, -48,  -102, -73,  -108, -56,  -97,  -41,  -75,   -56,  -70,  -20,
+      -33,  -9,   -51,  -32},
+     /* IRC_Composite_C_R0195_T345_P345.wav */
+     {1,     -2,   2,    -3,   4,    -4,    5,    -6,    7,     -8,   9,
+      -10,   11,   -13,  14,   -16,  17,    -19,  21,    -22,   24,   -25,
+      25,    -22,  13,   15,   -152, -764,  140,  -602,  18,    -900, -519,
+      -1092, -545, 489,  261,  706,  -1198, 1895, -1912, -2689, 2900, 30,
+      -3177, 8188, 3071, 2405, 4243, 6073,  615,  -3233, 507,   1858, -510,
+      -537,  1688, 110,  230,  1724, 2185,  1547, 1263,  1871,  1145, 573,
+      342,   -23,  261,  -614, -9,   -121,  805,  -137,  38,    285,  272,
+      -29,   312,  280,  184,  234,  346,   -32,  603,   80,    194,  153,
+      323,   -216, 458,  198,  10,   -275,  28,   -275,  -40,   -248, -266,
+      -416,  -305, -349, -213, -336, -338,  -460, -267,  -400,  -247, -403,
+      -303,  -379, -264, -460, -298, -347,  -227, -420,  -309,  -405, -246,
+      -238,  -250, -319, -267, -250, -266,  -221, -240,  -318,  -184, -236,
+      -175,  -305, -161, -225, -161, -213,  -190, -219,  -158,  -206, -155,
+      -198,  -165, -211, -169, -178, -141,  -180, -148,  -191,  -154, -191,
+      -120,  -222, -141, -171, -68,  -131,  -104, -142,  -95,   -72,  -93,
+      -71,   -91,  -94,  -90,  -92,  -83,   -97,  -91,   -117,  -108, -109,
+      -102,  -101, -101, -116, -95,  -82,   -119, -115,  -158,  -119, -137,
+      -108,  -151, -151, -163, -123, -125,  -148, -167,  -118,  -168, -149,
+      -196,  -146, -169, -130, -179, -147,  -164, -153,  -114,  -109, -119,
+      -128,  -121, -82,  -111, -89,  -98,   -81,  -102,  -105,  -71,  -85,
+      -85,   -116, -85,  -98,  -76,  -124,  -102, -118,  -107,  -84,  -126,
+      -141,  -114, -78,  -80,  -110, -127,  -121, -93,   -112,  -96,  -109,
+      -98,   -90,  -55,  -53,  -66,  -54,   -41,  -26,   -18,   -19,  -5,
+      -33,   -57,  -33}};
+
+const int16_t irc_composite_c_r0195_p000[][256] =
+    {/* IRC_Composite_C_R0195_T000_P000.wav */
+     {1,     -1,    1,    -1,    0,    0,     0,     0,     0,     1,     -1,
+      2,     -2,    2,    -3,    4,    -5,    6,     -7,    9,     -11,   14,
+      -20,   33,    -756, -534,  -348, 102,   -1032, -582,  -1506, 65,    -959,
+      133,   -210,  1928, 576,   -291, -5368, 5774,  -6962, -2265, 14308, -3483,
+      7090,  10599, 2076, -2731, 3149, -192,  -3412, -1652, 3866,  548,   -1226,
+      -1088, 2012,  2077, 1960,  -209, 1087,  1404,  1871,  1150,  1665,  536,
+      626,   1200,  1268, 592,   -114, -160,  -95,   133,   -37,   -246,  158,
+      -144,  -190,  -548, 421,   96,   105,   -192,  181,   129,   158,   143,
+      73,    -164,  -261, -64,   -147, -128,  -55,   -166,  -60,   98,    349,
+      -253,  126,   -172, 110,   -391, -118,  265,   -317,  256,   -97,   416,
+      -62,   274,   -244, -355,  -104, -427,  -357,  -741,  -368,  -489,  -365,
+      -418,  -498,  -150, -384,  -361, -530,  -257,  -297,  -381,  -491,  -511,
+      -393,  -429,  -294, -395,  -373, -408,  -292,  -342,  -260,  -297,  -286,
+      -427,  -324,  -291, -291,  -276, -319,  -244,  -386,  -221,  -241,  -141,
+      -260,  -273,  -231, -153,  -86,  -203,  -202,  -152,  -95,   -140,  -176,
+      -127,  -110,  -156, -237,  -148, -139,  -94,   -248,  -190,  -202,  -85,
+      -155,  -205,  -223, -169,  -158, -183,  -216,  -167,  -164,  -146,  -219,
+      -103,  -162,  -71,  -220,  -10,  -169,  7,     -134,  32,    -76,   -42,
+      -100,  -66,   -50,  -44,   -71,  -88,   -122,  -40,   -75,   21,    -78,
+      -18,   -125,  -33,  -69,   53,   -77,   -22,   -138,  -41,   -62,   -55,
+      -92,   -83,   -99,  -115,  -150, -17,   -53,   -12,   -184,  -50,   -96,
+      28,    -81,   -1,   -78,   -41,  -118,  -14,   -66,   38,    -82,   -29,
+      -104,  29,    5,    34,    -62,  14,    7,     47,    -61,   67,    -43,
+      51,    -145,  61},
+     /* IRC_Composite_C_R0195_T015_P000.wav */
+     {123,   -125,  127,   -129,  131,   -133,  135,   -137,   138,   -140,
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+      23,    -58,   69,    -65,   71,    -34,   110,   -36,    130,   -45,
+      96,    -68,   96,    -59,   38,    -71},
+     /* IRC_Composite_C_R0195_T030_P000.wav */
+     {-63,   66,    -69,   73,   -76,   80,     -84,   89,    -93,   98,
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+      -92,   9,     15,    121,  -12,   24,     -32,   74,    36,    98,
+      -9,    3,     -30,   36,   -42,   4},
+     /* IRC_Composite_C_R0195_T045_P000.wav */
+     {72,   -74,   77,    -79,   82,   -86,  91,    -96,   102,   -111,  121,
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+      176,  -85,   130,   -107,  189,  -99,  158,   -113,  138,   -127,  161,
+      -140, 111,   -221},
+     /* IRC_Composite_C_R0195_T060_P000.wav */
+     {123,   -132,  142,  -153,  166,   -180,  196,   -215, 238,   -264, 296,
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+      42,    -26,   97,   2,     68,    -13,   74,    7,    35,    -6,   34,
+      -1,    0,     -38},
+     /* IRC_Composite_C_R0195_T075_P000.wav */
+     {30,    -35,   41,    -47,  55,   -63,   72,    -84,   97,    -113,  134,
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+      35,    -34,   72,    -19,  33,   -32,   74,    -59,   39,    -25,   53,
+      1,     10,    -3},
+     /* IRC_Composite_C_R0195_T090_P000.wav */
+     {-3,    3,     -4,    5,     -6,    7,     -9,    11,    -13,   17,
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+      -36,   123,   -7,    23,    -36,   115},
+     /* IRC_Composite_C_R0195_T105_P000.wav */
+     {67,    -63,   58,    -51,  43,   -33,   19,    -1,    -24,  61,    -121,
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+      -11,   -60,   55},
+     /* IRC_Composite_C_R0195_T120_P000.wav */
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+      -45,   17,    -73,   100,  6,    64,    -7,    125,   -47,   94,
+      22,    33,    -55,   12,   -30,  7},
+     /* IRC_Composite_C_R0195_T135_P000.wav */
+     {-47,   46,    -46,   44,    -43,  40,   -37,   33,    -28,   21,    -12,
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+      -59,   33,    -26,   105,   4,    63,   13,    64,    -28,   24,    -1,
+      3,     -80,   -21},
+     /* IRC_Composite_C_R0195_T150_P000.wav */
+     {-5,    6,     -7,    7,    -8,   9,    -10,   11,   -12,   13,   -15,
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+      -16,   18,    1,     -3,   18,   72,   0,     21,   9,     42,   -23,
+      -7,    -40,   -42},
+     /* IRC_Composite_C_R0195_T165_P000.wav */
+     {-31,   31,    -31,  31,   -30,  30,   -30,  29,   -28,   27,   -25,  23,
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+      -59,   25,    -48,  -21},
+     /* IRC_Composite_C_R0195_T180_P000.wav */
+     {9,    -9,   9,     -10,  10,   -11,  11,    -12,  13,   -13,  14,    -15,
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+      28,   3,    -22,   -46},
+     /* IRC_Composite_C_R0195_T195_P000.wav */
+     {12,   -12,  12,   -12,  12,   -12,   12,    -12,  12,   -12,  12,   -12,
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+      23,   -6,   2,    -42},
+     /* IRC_Composite_C_R0195_T210_P000.wav */
+     {6,    -6,   6,    -6,   7,    -7,   7,    -7,   7,    -7,   7,    -7,
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+      24,   -9,   7,    -23},
+     /* IRC_Composite_C_R0195_T225_P000.wav */
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+      -39,  -27,  -55,  -31},
+     /* IRC_Composite_C_R0195_T240_P000.wav */
+     {0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
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+      -64,  -17,  -39,  -13},
+     /* IRC_Composite_C_R0195_T255_P000.wav */
+     {2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,
+      2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -3,   3,    -3,
+      3,    -3,   3,    -3,   3,    -3,   4,    -4,   4,    -5,   5,    -6,
+      9,    -19,  -71,  -126, -111, -131, -223, -213, -179, -73,  -136, -101,
+      -19,  30,   101,  69,   89,   95,   323,  406,  515,  721,  768,  1140,
+      1389, 925,  919,  1504, 1350, 1027, 1508, 1800, 2083, 1870, 1354, 950,
+      539,  490,  494,  392,  378,  293,  344,  461,  473,  522,  375,  344,
+      271,  237,  162,  56,   42,   11,   83,   -5,   55,   -30,  81,   -3,
+      71,   -18,  -50,  0,    -151, -66,  -240, -143, -251, -198, -256, -250,
+      -192, -239, -205, -271, -195, -240, -195, -267, -238, -257, -227, -242,
+      -241, -251, -226, -256, -185, -246, -212, -252, -224, -227, -235, -216,
+      -249, -192, -222, -170, -209, -167, -231, -184, -255, -188, -250, -181,
+      -283, -185, -279, -146, -264, -164, -273, -161, -261, -181, -298, -216,
+      -272, -226, -251, -181, -242, -222, -248, -216, -260, -249, -302, -271,
+      -289, -255, -276, -256, -246, -225, -218, -198, -202, -222, -210, -223,
+      -219, -210, -230, -209, -218, -149, -175, -141, -182, -152, -139, -146,
+      -160, -169, -152, -140, -120, -129, -130, -102, -120, -59,  -92,  -67,
+      -121, -84,  -93,  -62,  -69,  -69,  -85,  -101, -60,  -94,  -80,  -123,
+      -92,  -126, -73,  -83,  -80,  -90,  -79,  -71,  -42,  -42,  -54,  -59,
+      -77,  -88,  -83,  -86,  -100, -126, -130, -119, -130, -124, -145, -127,
+      -148, -111, -124, -92,  -85,  -75,  -85,  -70,  -68,  -66,  -66,  -59,
+      -78,  -67,  -59,  -65},
+     /* IRC_Composite_C_R0195_T270_P000.wav */
+     {-1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,
+      -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,
+      -1,   1,    -1,   1,    -1,   1,    -1,   2,    -2,   2,    -2,   3,
+      -4,   9,    -91,  -159, -163, -111, -130, -163, -90,  -102, -89,  -163,
+      -184, -188, -171, -83,  30,   36,   169,  298,  65,   214,  405,  559,
+      898,  896,  1271, 2043, 2157, 2404, 2912, 2261, 1344, 1103, 1268, 1220,
+      756,  484,  263,  283,  379,  505,  398,  413,  455,  527,  338,  482,
+      334,  406,  255,  187,  211,  119,  189,  4,    182,  -16,  145,  -56,
+      26,   -74,  -83,  -100, -148, -133, -197, -139, -191, -198, -217, -194,
+      -192, -215, -224, -197, -182, -180, -242, -199, -243, -153, -261, -180,
+      -264, -151, -250, -164, -239, -176, -235, -178, -231, -191, -258, -227,
+      -259, -218, -280, -188, -268, -159, -259, -154, -260, -137, -244, -185,
+      -245, -228, -251, -260, -250, -247, -239, -233, -209, -205, -224, -262,
+      -274, -268, -248, -262, -263, -277, -234, -255, -222, -233, -226, -262,
+      -221, -257, -211, -267, -218, -251, -186, -221, -179, -224, -195, -213,
+      -161, -226, -193, -212, -178, -219, -191, -204, -183, -190, -175, -182,
+      -165, -164, -140, -131, -97,  -107, -80,  -102, -45,  -85,  -60,  -81,
+      -66,  -74,  -59,  -82,  -82,  -91,  -88,  -102, -100, -120, -85,  -122,
+      -72,  -129, -98,  -137, -55,  -101, -62,  -69,  -41,  -72,  -56,  -83,
+      -69,  -90,  -53,  -98,  -81,  -105, -80,  -107, -83,  -112, -102, -115,
+      -97,  -125, -111, -136, -123, -145, -118, -131, -95,  -114, -62,  -70,
+      -59,  -73,  -48,  -57},
+     /* IRC_Composite_C_R0195_T285_P000.wav */
+     {-1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,
+      -1,   1,    -1,   1,    -1,   1,    -1,   2,    -2,   2,    -2,   2,
+      -2,   2,    -2,   2,    -2,   2,    -2,   2,    -3,   3,    -4,   6,
+      -9,   19,   -100, -184, -163, -233, -161, -194, -211, -190, -252, -368,
+      -121, 58,   37,   -12,  35,   111,  -96,  -223, -19,  464,  559,  1425,
+      2382, 1927, 1709, 2204, 1586, 984,  1062, 1267, 1442, 1113, 1078, 1182,
+      1130, 1148, 848,  575,  534,  584,  541,  405,  456,  300,  346,  294,
+      405,  279,  183,  240,  258,  176,  160,  100,  -27,  100,  -24,  93,
+      -139, 0,    -270, 60,   -268, 20,   -249, -123, -250, -178, -127, -185,
+      -84,  -288, -109, -237, -116, -161, -179, -137, -193, -134, -189, -164,
+      -211, -174, -211, -244, -187, -259, -227, -263, -160, -220, -206, -234,
+      -213, -214, -205, -225, -212, -266, -182, -254, -229, -238, -166, -272,
+      -223, -282, -263, -305, -287, -272, -261, -288, -239, -247, -227, -247,
+      -224, -236, -226, -259, -230, -256, -220, -217, -226, -224, -232, -214,
+      -225, -205, -261, -251, -252, -219, -235, -230, -230, -236, -236, -209,
+      -203, -226, -221, -222, -180, -160, -190, -161, -159, -154, -156, -97,
+      -111, -129, -127, -107, -66,  -95,  -69,  -92,  -64,  -55,  -64,  -43,
+      -78,  -65,  -103, -71,  -87,  -110, -101, -114, -93,  -117, -90,  -112,
+      -88,  -109, -72,  -82,  -69,  -68,  -63,  -65,  -90,  -69,  -67,  -77,
+      -62,  -56,  -47,  -91,  -59,  -73,  -34,  -101, -70,  -91,  -61,  -105,
+      -91,  -128, -154, -149, -128, -150, -140, -142, -93,  -111, -108, -78,
+      -62,  -77,  -59,  -62},
+     /* IRC_Composite_C_R0195_T300_P000.wav */
+     {4,    -4,   4,    -4,   4,    -4,   4,    -4,   4,    -5,   5,    -5,
+      5,    -5,   5,    -5,   5,    -6,   6,    -6,   6,    -6,   6,    -7,
+      7,    -7,   8,    -8,   8,    -9,   10,   -11,  13,   -16,  24,   -75,
+      -192, -243, -206, -121, -200, -413, -335, -275, -203, -85,  -38,  193,
+      -95,  -382, 283,  -596, -531, 1085, 630,  1024, 2620, 2194, 1518, 1499,
+      1392, 607,  112,  846,  1164, 950,  675,  1005, 1305, 1174, 1521, 1142,
+      1208, 859,  1074, 771,  735,  552,  498,  679,  274,  454,  160,  457,
+      127,  176,  172,  280,  101,  64,   67,   -27,  98,   11,   3,    -150,
+      -68,  -72,  -105, -163, -160, -139, -135, -141, -133, -136, -55,  -184,
+      -109, -200, -81,  -104, -78,  -213, -147, -100, -188, -136, -214, -162,
+      -281, -139, -265, -190, -228, -184, -320, -138, -205, -130, -238, -158,
+      -209, -100, -297, -175, -328, -233, -318, -199, -385, -256, -355, -254,
+      -336, -269, -310, -262, -286, -237, -272, -206, -279, -217, -262, -212,
+      -280, -260, -296, -269, -321, -249, -293, -297, -294, -263, -252, -259,
+      -248, -231, -189, -223, -204, -205, -214, -202, -172, -186, -170, -216,
+      -162, -160, -149, -167, -137, -119, -139, -138, -127, -91,  -90,  -111,
+      -123, -105, -94,  -88,  -92,  -101, -104, -94,  -87,  -104, -82,  -106,
+      -68,  -106, -75,  -113, -53,  -76,  -65,  -86,  -59,  -75,  -81,  -73,
+      -84,  -59,  -75,  -52,  -59,  -73,  -55,  -69,  -45,  -87,  -49,  -62,
+      -47,  -76,  -53,  -57,  -57,  -74,  -83,  -72,  -109, -94,  -135, -114,
+      -127, -112, -128, -125, -121, -104, -114, -94,  -97,  -102, -114, -98,
+      -73,  -61,  -46,  -72},
+     /* IRC_Composite_C_R0195_T315_P000.wav */
+     {-3,   3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,
+      -3,   3,    -3,   3,    -3,   3,    -4,   4,    -4,   4,    -4,   4,
+      -5,   5,    -5,   6,    -7,   8,    -10,  14,   -29,  -234, -230, -266,
+      -204, -305, -247, -524, -359, -150, 55,   142,  71,   572,  -570, -1004,
+      861,  -803, -374, 3189, 1120, 2120, 3144, 2428, 636,  1026, 1269, 59,
+      -187, 571,  1369, 181,  332,  532,  1375, 1240, 1554, 1033, 1045, 866,
+      1172, 1026, 1022, 606,  720,  597,  671,  349,  376,  201,  259,  47,
+      222,  146,  93,   -124, 20,   -35,  -174, -23,  6,    -91,  -108, -18,
+      6,    38,   -2,   -118, 67,   -77,  -79,  -121, -96,  -206, -171, -74,
+      -193, -141, -167, -54,  -168, -65,  -180, -137, -281, -209, -164, -172,
+      -178, -158, -127, -188, -174, -11,  -152, -142, -222, -182, -348, -203,
+      -338, -287, -380, -316, -332, -319, -347, -297, -281, -364, -362, -314,
+      -304, -376, -347, -321, -343, -330, -367, -276, -306, -278, -290, -268,
+      -295, -267, -229, -264, -248, -290, -244, -263, -201, -223, -197, -193,
+      -189, -154, -159, -137, -178, -126, -217, -115, -163, -129, -166, -130,
+      -150, -147, -107, -147, -129, -143, -117, -141, -126, -115, -135, -99,
+      -146, -90,  -112, -81,  -119, -52,  -80,  -79,  -84,  -83,  -64,  -59,
+      -53,  -89,  -91,  -74,  -101, -29,  -121, -49,  -139, -64,  -94,  -23,
+      -57,  -36,  -29,  -38,  -50,  -50,  -47,  -30,  -53,  -56,  -81,  -58,
+      -97,  -62,  -69,  -56,  -82,  -74,  -103, -94,  -111, -122, -150, -117,
+      -148, -135, -117, -104, -87,  -95,  -77,  -107, -55,  -89,  -56,  -82,
+      -65,  -70,  -36,  -13},
+     /* IRC_Composite_C_R0195_T330_P000.wav */
+     {-15,   15,   -15,  15,   -15,  15,   -15,  16,   -16,   16,   -16,   16,
+      -16,   16,   -16,  16,   -16,  16,   -16,  16,   -16,   15,   -15,   14,
+      -13,   11,   -8,   5,    0,    -7,   12,   -293, -526,  -71,  -505,  -401,
+      -1003, -83,  -138, 277,  -121, 450,  638,  -529, -2538, 1974, -1021, -928,
+      6018,  1862, 2117, 4170, 3177, -540, 80,   773,  465,   -382, 760,   707,
+      287,   -96,  895,  1260, 1490, 749,  709,  1138, 1758,  1405, 855,   490,
+      661,   833,  888,  382,  366,  310,  491,  203,  204,   155,  246,   122,
+      -32,   -122, -37,  50,   94,   -114, -242, -173, 94,    -97,  -233,  -239,
+      44,    3,    91,   -64,  -78,  -19,  148,  9,    2,     -196, 62,    -119,
+      -154,  -277, 128,  -102, -103, -182, -189, -136, -49,   51,   -260,  -285,
+      -17,   -46,  -344, -260, -264, -310, -481, -356, -385,  -299, -442,  -338,
+      -388,  -433, -413, -352, -281, -444, -379, -419, -361,  -361, -316,  -320,
+      -407,  -371, -354, -283, -325, -263, -281, -264, -300,  -242, -200,  -218,
+      -231,  -208, -283, -185, -218, -137, -254, -173, -210,  -124, -178,  -189,
+      -155,  -197, -150, -148, -83,  -183, -134, -157, -104,  -154, -134,  -133,
+      -118,  -172, -161, -120, -119, -156, -145, -96,  -143,  -172, -117,  -71,
+      -107,  -120, -113, -53,  -111, -85,  -65,  -51,  -97,   -63,  -15,   -77,
+      -94,   -67,  -18,  -45,  -126, -60,  -101, -21,  -115,  3,    -114,  -45,
+      -90,   -5,   -69,  -55,  -89,  -42,  -99,  -38,  -100,  -24,  -95,   -21,
+      -116,  -37,  -113, -13,  -85,  -64,  -143, -57,  -122,  -61,  -126,  -85,
+      -123,  -78,  -94,  -71,  -113, -55,  -72,  -37,  -93,   -41,  -42,   -10,
+      -34,   -37,  -13,  -34},
+     /* IRC_Composite_C_R0195_T345_P000.wav */
+     {-1,   1,    -1,   1,    -1,   1,    -1,   1,     -1,   1,     -1,
+      1,    -1,   1,    -1,   1,    -1,   1,    -1,    1,    -1,    1,
+      -1,   1,    -2,   2,    -3,   -196, -727, -30,   -506, -136,  -1125,
+      -612, -692, -135, 283,  619,  802,  -51,  -2221, 1398, -964,  -5308,
+      8059, 961,  2013, 8066, 4451, -858, 1438, 1808,  -955, -2183, 475,
+      1387, 742,  458,  140,  184,  1030, 1768, 1316,  965,  1001,  1155,
+      1147, 1222, 1301, 1124, 461,  289,  297,  592,   383,  273,   -220,
+      -259, -53,  323,  264,  -32,  -200, 42,   191,   421,  77,    -195,
+      -257, 18,   -154, -38,  -140, -135, -290, -47,   107,  49,    -29,
+      -98,  -94,  -50,  49,   264,  36,   153,  -148,  86,   -24,   287,
+      -156, 101,  -86,  125,  -186, -182, -278, -236,  -501, -636,  -484,
+      -401, -330, -428, -463, -554, -283, -366, -318,  -494, -441,  -375,
+      -318, -358, -465, -416, -438, -337, -400, -344,  -393, -247,  -308,
+      -313, -356, -181, -229, -204, -374, -214, -320,  -200, -307,  -179,
+      -339, -189, -252, -152, -246, -143, -210, -166,  -196, -69,   -139,
+      -172, -216, -103, -73,  -127, -214, -116, -118,  -96,  -247,  -135,
+      -187, -78,  -264, -166, -235, -95,  -173, -142,  -167, -169,  -129,
+      -117, -114, -125, -125, -103, -137, -47,  -110,  -26,  -121,  -21,
+      -72,  -15,  -71,  -68,  -34,  -53,  -38,  -72,   -96,  -96,   -90,
+      -11,  -126, -33,  -166, -22,  -92,  18,   -75,   -35,  -69,   -60,
+      -67,  -82,  -72,  -61,  -96,  -68,  -103, -26,   -100, -19,   -108,
+      0,    -110, -34,  -106, -39,  -54,  -75,  -76,   -83,  -35,   -42,
+      -69,  -51,  -59,  -38,  -55,  19,   -49,  -3,    -43,  27,    -14,
+      -5,   -46,  -24}};
+
+const int16_t irc_composite_c_r0195_p015[][256] =
+    {/* IRC_Composite_C_R0195_T000_P015.wav */
+     {4,    -4,    4,    -4,    4,     -4,    4,     -4,    4,     -4,   4,
+      -5,   5,     -5,   4,     -4,    4,     -4,    3,     -2,    1,    2,
+      -5,   -101,  -665, -424,  -897,  -170,  -507,  -460,  -1001, -828, -356,
+      47,   474,   980,  404,   2103,  -3700, -5968, 7765,  -9133, 9741, 10867,
+      71,   6925,  3456, -1455, -2458, 2718,  -170,  -1201, -1751, 1809, 507,
+      632,  -1165, 1945, 1684,  1705,  -450,  1152,  360,   1527,  269,  952,
+      644,  931,   727,  1576,  725,   835,   741,   915,   186,   162,  327,
+      384,  20,    24,   -311,  60,    225,   484,   -123,  -28,   -210, 118,
+      15,   76,    -335, -153,  -249,  49,    -120,  24,    -200,  -169, -213,
+      -175, -235,  -158, -280,  -399,  -410,  -304,  -104,  -355,  -319, -311,
+      -79,  -366,  -75,  -79,   -120,  -487,  147,   -42,   -198,  -175, 135,
+      -242, 55,    233,  123,   -314,  -58,   -179,  -96,   -352,  -26,  7,
+      11,   -105,  -129, 125,   -110,  -293,  -276,  -461,  -507,  -406, -422,
+      -497, -481,  -446, -459,  -325,  -336,  -443,  -420,  -320,  -391, -360,
+      -309, -407,  -375, -288,  -289,  -385,  -315,  -272,  -292,  -271, -256,
+      -318, -173,  -220, -185,  -329,  -142,  -239,  -76,   -219,  -117, -277,
+      -107, -204,  -117, -233,  -199,  -220,  -177,  -238,  -171,  -170, -187,
+      -285, -154,  -168, -145,  -192,  -131,  -239,  -194,  -170,  -109, -205,
+      -155, -164,  -163, -187,  -126,  -85,   -138,  -195,  -140,  -189, -84,
+      -134, -22,   -177, -51,   -128,  -26,   -105,  26,    -144,  -65,  -123,
+      -28,  -136,  19,   -69,   -43,   -112,  40,    -43,   59,    -67,  32,
+      -35,  97,    -21,  99,    -27,   90,    23,    141,   21,    38,   39,
+      61,   12,    10,   -1,    37,    -3,    7,     2,     41,    -19,  -12,
+      -12,  16,    -53},
+     /* IRC_Composite_C_R0195_T015_P015.wav */
+     {100,  -101, 102,   -102, 102,    -102,  102,   -102,  101,   -99,  96,
+      -93,  88,   -81,   71,   -58,    37,    -6,    -46,   154,   -587, -280,
+      -52,  -712, -530,  -842, -228,   -637,  -217,  -1535, -463,  274,  2688,
+      1298, -390, -2760, 4349, -16532, 13572, -2117, 2580,  22789, -850, 559,
+      1346, 2618, -4202, 397,  316,    686,   -1654, 1356,  -848,  1484, 164,
+      1873, 376,  1720,  39,   1069,   309,   1623,  748,   807,   488,  1192,
+      547,  1364, 679,   594,  48,     334,   -179,  362,   98,    -149, -587,
+      -275, -81,  371,   -430, -372,   -369,  412,   -126,  -39,   -294, 169,
+      -117, 61,   -209,  -100, -271,   -40,   -262,  -244,  -406,  12,   -243,
+      -253, -484, 54,    -243, -283,   -575,  -138,  -358,  -262,  -307, -61,
+      -567, -206, 25,    51,   -696,   -10,   -132,  -245,  -388,  187,  -132,
+      -194, -316, -92,   -71,  -291,   -374,  27,    -295,  -84,   -4,   145,
+      -576, 122,  -217,  -153, -724,   53,    -337,  -126,  -502,  -305, -401,
+      -138, -372, -296,  -502, -318,   -375,  -140,  -645,  -283,  -375, -217,
+      -529, -172, -316,  -324, -287,   -239,  -309,  -305,  -230,  -228, -310,
+      -332, -289, -168,  -302, -286,   -241,  -197,  -266,  -181,  -156, -260,
+      -272, -185, -148,  -225, -221,   -239,  -178,  -223,  -142,  -203, -226,
+      -260, -124, -148,  -200, -259,   -124,  -203,  -182,  -207,  -78,  -186,
+      -163, -136, -74,   -179, -123,   -45,   -87,   -180,  -103,  -28,  -117,
+      -135, -64,  -29,   -148, -90,    -13,   -28,   -111,  -40,   7,    -42,
+      -98,  -26,  -4,    -57,  -22,    22,    -38,   -72,   76,    67,   1,
+      -21,  80,   49,    39,   63,     66,    26,    23,    62,    97,   29,
+      39,   49,   28,    -1,   52,     74,    -6,    -4,    19,    15,   -15,
+      -12,  17,   -50},
+     /* IRC_Composite_C_R0195_T030_P015.wav */
+     {-101,  102,    -103,  105,   -105,  106,   -106,  106,  -104, 102,  -97,
+      90,    -79,    62,    -35,   -12,   109,   -431,  -649, 311,  -951, 117,
+      -1193, 10,     -1439, 943,   -1996, 57,    -1866, 3892, -160, 3817, -4940,
+      7695,  -14389, 320,   6811,  -6312, 29122, 1548,  2820, 2754, 2735, -5703,
+      -20,   838,    313,   -2387, 2394,  -1371, 1731,  262,  1937, 34,   1456,
+      -294,  1713,   221,   997,   660,   1535,  549,   1179, 928,  928,  293,
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+      -436,  -61,    -412,  -21,   -436,  -263,  -534,  -25,  -24,  127,  -328,
+      -77,   -189,   107,   -200,  -313,  -487,  -278,  -245, -413, -269, -216,
+      -100,  -434,   -149,  -238,  -270,  -527,  -94,   -144, -244, -529, -94,
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+      -225,  -263,   -470,  -227,  -238,  -276,  -329,  -339, -304, -226, -209,
+      -212,  -255,   -261,  -275,  -96,   -168,  -289,  -298, -134, -296, -334,
+      -291,  -248,   -272,  -268,  -362,  -278,  -286,  -108, -360, -215, -330,
+      -76,   -394,   -179,  -365,  -104,  -449,  -206,  -420, -128, -409, -167,
+      -434,  -175,   -416,  -105,  -318,  -125,  -408,  -157, -270, -45,  -277,
+      -160,  -326,   -56,   -220,  -114,  -280,  -69,   -199, -100, -242, -24,
+      -143,  -39,    -214,  3,     -104,  20,    -184,  20,   -140, -7,   -183,
+      31,    -135,   -21,   -164,  13,    -128,  42,    -188, 49,   -113, 71,
+      -189,  90,     -97,   102,   -87,   84,    -61,   147,  -63,  149,  -94,
+      170,   -108,   183,   -108,  208,   -150,  179,   -163, 250,  -98,  215,
+      -137,  235,    -107,  214,   -135,  246,   -126,  201,  -171, 186,  -149,
+      135,   -194,   121},
+     /* IRC_Composite_C_R0195_T045_P015.wav */
+     {-146,  153,   -161,  170,   -179,  189,   -200, 212,  -224,  238,   -252,
+      266,   -278,  280,   -241,  -262,  -1208, 369,  -877, 618,   -1865, 344,
+      -1624, 1098,  -2418, 1166,  -2204, 3338,  -287, 6351, -5670, 4490,  -9471,
+      -5595, 10415, -3658, 26422, 6163,  571,   640,  3020, -3458, 345,   262,
+      -17,   -2379, 1504,  -915,  2155,  96,    1778, -36,  1408,  800,   1533,
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+      -589,  -590,  -422,  -164,  -365,  -343,  -251, -474, 20,    -343,  -59,
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+      -458,  -263,  -403,  -259,  -462,  -188,  -423, -283, -371,  -78,   -225,
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+      -233,  -185,  -390,  -305,  -381,  -170,  -416, -344, -426,  -229,  -331,
+      -201,  -370,  -183,  -273,  -88,   -257,  -116, -206, -60,   -298,  -72,
+      -233,  -115,  -304,  -131,  -336,  -187,  -271, -44,  -375,  -190,  -355,
+      -35,   -321,  -189,  -400,  -136,  -365,  -198, -383, -200,  -395,  -244,
+      -401,  -188,  -398,  -184,  -393,  -179,  -365, -150, -314,  -118,  -327,
+      -118,  -291,  -73,   -272,  -87,   -247,  -52,  -258, -27,   -226,  25,
+      -210,  8,     -202,  51,    -136,  62,    -162, 35,   -135,  61,    -156,
+      -12,   -167,  55,    -165,  15,    -174,  47,   -131, 28,    -125,  76,
+      -91,   69,    -71,   126,   -60,   104,   -40,  141,  -51,   117,   -33,
+      141,   -58,   103,   -38,   158,   -48,   95,   -52,  149,   -45,   86,
+      -47,   120,   -68,   88,    -39,   116,   -45,  74,   -35,   89,    -68,
+      62,    -90,   53},
+     /* IRC_Composite_C_R0195_T060_P015.wav */
+     {29,    -34,   39,    -45,  52,    -59,  69,    -80,  93,     -110,  131,
+      -160,  205,   -308,  -459, -920,  -171, 388,   -674, -719,   -1064, 350,
+      -1149, 552,   -1891, 2109, -28,   5773, -3032, 5413, -11840, 194,   -1388,
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+      -189,  -226,  -188,  -154, -195,  -171, -163,  -122, -121,   -97,   -111,
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+      -97,   -100,  16,    -97,  -13,   -104, 7,     -74,  -20,    -54,   27,
+      -6,    45,    -12,   56,   35,    24,   86,    33,   48,     24,    68,
+      72,    -3,    50,    22,   52,    12,   43,    -32,  109,    -68,   106,
+      -66,   49,    -61,   102,  -74,   44,   -72,   28,   -60,    21,    -71,
+      -4,    -76,   16},
+     /* IRC_Composite_C_R0195_T075_P015.wav */
+     {19,    -16,  12,   -7,    2,     4,     -11,  18,     -27,   36,    -47,
+      59,    -68,  37,   -553,  36,    -916,  -89,  -13,    -916,  -389,  -1224,
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+      -293,  -201, -269, -218,  -264,  -188,  -203, -199,   -218,  -207,  -118,
+      -151,  -146, -184, -126,  -92,   -92,   -129, -108,   -109,  -42,   -120,
+      -57,   -144, -45,  -102,  -23,   -164,  -63,  -98,    -36,   -106,  -81,
+      -3,    -74,  -55,  -79,   -49,   -60,   -59,  -60,    -31,   -36,   -35,
+      57,    -37,  99,   -7,    135,   -26,   115,  14,     154,   -6,    78,
+      28,    119,  -3,   53,    -48,   80,    -38,  55,     -56,   13,    -65,
+      74,    -45,  19,   -79,   50,    -39,   24,   -95,    40,    -41,   15,
+      -61,   55,   -129},
+     /* IRC_Composite_C_R0195_T090_P015.wav */
+     {106,   -108,  111,  -113, 116,  -119,  121,  -123,   125,   -125,  121,
+      -102,  -150,  -386, -21,  -846, -117,  203,  -762,   -759,  -425,  -20,
+      26,    535,   183,  3678, 866,  180,   877,  -12284, 4742,  -888,  21444,
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+      -232,  -145,  -162, -100, -147, -119,  -102, -57,    -53,   -87,   -18,
+      -36,   -11,   -87,  -46,  -46,  17,    -89,  -82,    -57,   6,     -100,
+      -61,   -101,  -38,  -60,  -46,  -108,  -40,  -41,    19,    -104,  -21,
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+      92,    -42,   26,   -51,  25,   -38,   -15,  -65,    -12,   -35,   -8,
+      28,    24,    -31,  -1,   36,   92,    2,    6,      8,     52,    -33,
+      -7,    -25,   -27},
+     /* IRC_Composite_C_R0195_T105_P015.wav */
+     {16,    -19,  23,    -28,   34,   -40,   48,   -57,   68,    -82,   100,
+      -121,  33,   -823,  199,   -561, -96,   -481, 741,   -1752, -67,   -480,
+      1840,  -966, 591,   1601,  5280, -3652, 770,  -7901, 214,   10553, 2446,
+      18816, 1377, -3359, -1411, 6633, 5595,  -115, -232,  1823,  -2575, -717,
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+      32,    -190, 61,    -152,  90,   -145,  48,   -122,  71,    -96,   29,
+      -121,  6,    -57,   66,    -80,  40,    -49,  59,    -112,  -8,    -129,
+      54,    -18,  76,    -21,   97,   37,    171,  23,    59,    -61,   75,
+      -37,   21,   -127,  1,     -154, 68,    -112, -12,   -180,  86,    -58,
+      91,    -120, 128,   -28,   129,  -100,  167,  -17,   106,   -84,   94,
+      -49,   76,   -37},
+     /* IRC_Composite_C_R0195_T120_P015.wav */
+     {39,    -39,  39,   -39,   39,    -38,  38,   -37,   35,    -32,   27,
+      -19,   0,    -223, 40,    -675,  329,  -841, 457,   -348,  -274,  -1647,
+      1770,  432,  -494, 3,     4260,  289,  887,  -4454, -3947, 7446,  -2378,
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+      -275,  -161, -280, -75,   -215,  -110, -276, -33,   -130,  -19,   -261,
+      -8,    -171, 30,   -242,  -55,   -229, -65,  -254,  -74,   -218,  -134,
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+      -254,  -101, -185, -51,   -234,  -38,  -186, -6,    -168,  0,     -111,
+      -29,   -125, 29,   -96,   -23,   -153, 22,   -59,   -32,   -103,  28,
+      -118,  11,   -64,  83,    -103,  41,   -112, 28,    -109,  99,    -41,
+      39,    -86,  157,  43,    121,   -15,  157,  -11,   121,   -30,   90,
+      -113,  25,   -109, 38,    -172,  -47,  -151, 15,    -102,  21,    -97,
+      16,    -33,  104,  12,    59,    -55,  121,  -5,    63,    -116,  114,
+      -30,   115,  -85},
+     /* IRC_Composite_C_R0195_T135_P015.wav */
+     {15,    -17,  19,   -21,  24,   -28,  32,   -37,  43,    -50,   60,
+      -73,   90,   -115, 157,  -297, 361,  -673, 287,  -116,  -598,  -544,
+      1021,  -902, -263, 1113, 1563, -848, 3395, 810,  -1816, -1445, -4755,
+      10984, 3295, 5362, 6262, 2589, -529, 1444, 7698, 4758,  -2233, -2138,
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+      -265,  -144, -282, -127, -249, -66,  -222, -39,  -146,  -32,   -126,
+      -20,   -76,  5,    -92,  18,   -56,  21,   -91,  13,    -119,  -23,
+      -127,  37,   -71,  16,   -116, 10,   -105, 12,   -96,   55,    -69,
+      43,    -76,  137,  -20,  86,   -3,   123,  28,   100,   8,     49,
+      -58,   60,   -31,  -10,  -117, -15,  -44,  -19,  -89,   2,     -51,
+      -18,   -51,  62,   -39,  31,   -46,  59,   -59,  67,    -57,   112,
+      -72,   62,   -22},
+     /* IRC_Composite_C_R0195_T150_P015.wav */
+     {22,    -23,   24,    -24,  25,   -26,   27,   -27,  28,   -28,  28,
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+      -6333, 870,   5933,  2367, 6468, 4276,  4587, 970,  719,  5822, 3692,
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+      -104,  -88,   -98,   -82,  -75,  -58,   -53,  -46,  -85,  -53,  -56,
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+      37,    -14,   58,    10,   69,   35,    20,   45,   42,   27,   3,
+      3,     -29,   -5,    -10,  26,   -54,   -21,  -21,  39,   -58,  -58,
+      -23,   10,    -16,   -35,  6,    -22,   2,    -8,   54,   -13,  7,
+      -2,    51,    6},
+     /* IRC_Composite_C_R0195_T165_P015.wav */
+     {20,    -21,  21,   -22,  23,   -23,  24,   -25,  26,   -26,  27,   -28,
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+      -1,    4,    58,   -21,  25,   24,   70,   3,    30,   9,    36,   -16,
+      6,     -21,  -39,  -15},
+     /* IRC_Composite_C_R0195_T180_P015.wav */
+     {-9,   9,     -9,    9,    -10,  10,   -10,  10,   -11,  11,   -11,  11,
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+      53,   -3,    44,    26,   45,   23,   36,   66,   32,   13,   3,    39,
+      11,   -14,   8,     -12},
+     /* IRC_Composite_C_R0195_T195_P015.wav */
+     {3,    -3,   4,    -4,   4,    -5,    5,     -5,   6,    -6,   7,    -7,
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+      23,   6,    15,   15,   42,   21,    45,    37,   32,   17,   30,   46,
+      26,   13,   23,   -19},
+     /* IRC_Composite_C_R0195_T210_P015.wav */
+     {5,    -5,   6,    -6,   6,    -6,   6,    -6,   6,    -6,   7,    -7,
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+      30,   -5,   10,   -9,   35,   17,   55,   16,   50,   34,   50,   43,
+      21,   55,   18,   41},
+     /* IRC_Composite_C_R0195_T225_P015.wav */
+     {3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,    -3,
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+      62,   28,   60,   13,   34,   6,    35,   -8,   13,   -2,   6,    11,
+      8,    6,    -3,   12},
+     /* IRC_Composite_C_R0195_T240_P015.wav */
+     {0,    0,    0,    0,    1,    -1,   1,    -1,   1,    -1,   1,    -1,
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+      26,   16,   20,   -7,   0,    -5,   22,   12,   4,    3,    18,   14,
+      9,    -1,   -12,  -43},
+     /* IRC_Composite_C_R0195_T255_P015.wav */
+     {1,    -1,   2,    -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,
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+      -129, -161, -129, -137, -105, -119, -97,  -124, -84,  -109, -56,  -30,
+      29,   13,   51,   35,   66,   39,   42,   23,   8,    -3,   12,   8,
+      29,   13,   32,   3,    26,   10,   21,   -16,  -7,   -1,   7,    2,
+      -5,   -3,   -3,   -1},
+     /* IRC_Composite_C_R0195_T270_P015.wav */
+     {-1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,
+      -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,
+      -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -2,   2,
+      -2,   3,    -5,   38,   41,   81,   155,  185,  109,  43,   168,  195,
+      276,  233,  321,  381,  601,  818,  813,  1005, 694,  582,  706,  813,
+      1380, 2408, 2859, 3417, 3192, 1234, 164,  601,  379,  145,  -151, -609,
+      -543, -195, 49,   -120, -111, 136,  61,   -67,  -105, -152, -114, -346,
+      -267, -397, -310, -240, -454, -334, -493, -338, -595, -418, -579, -472,
+      -506, -514, -489, -592, -416, -592, -471, -574, -491, -531, -515, -529,
+      -526, -514, -446, -512, -519, -497, -434, -454, -430, -467, -415, -434,
+      -400, -422, -405, -387, -422, -391, -428, -343, -360, -326, -389, -310,
+      -335, -241, -318, -246, -341, -202, -264, -207, -278, -221, -234, -239,
+      -210, -224, -186, -240, -226, -220, -207, -184, -222, -220, -245, -206,
+      -189, -231, -234, -238, -195, -237, -129, -227, -188, -216, -145, -224,
+      -225, -241, -252, -274, -214, -216, -202, -212, -164, -190, -171, -187,
+      -134, -175, -141, -168, -164, -167, -150, -174, -138, -128, -101, -135,
+      -76,  -112, -70,  -116, -70,  -102, -92,  -120, -98,  -123, -112, -146,
+      -92,  -145, -88,  -135, -96,  -115, -84,  -87,  -67,  -61,  -82,  -42,
+      -48,  11,   -13,  5,    25,   35,   52,   70,   95,   98,   101,  116,
+      108,  116,  94,   137,  109,  102,  83,   89,   88,   89,   64,   59,
+      40,   43,   14,   53,   3,    16,   -3,   7,    12,   -13,  16,   -2,
+      28,   6,    37,   18},
+     /* IRC_Composite_C_R0195_T285_P015.wav */
+     {0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
+      0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
+      0,    0,    0,    0,    0,    0,    1,    -1,   1,    -1,   1,    -1,
+      2,    -3,   6,    -109, -159, -182, -252, -225, -252, -209, -220, -323,
+      -249, -123, 27,   108,  237,  39,   -227, -176, -141, 26,   983,  1563,
+      1989, 2256, 2042, 1443, 1643, 1525, 1088, 1602, 1474, 1170, 1150, 824,
+      708,  924,  1176, 1036, 269,  443,  781,  769,  425,  308,  405,  367,
+      433,  461,  448,  234,  194,  163,  218,  246,  100,  -62,  56,   -52,
+      109,  -114, 68,   -271, 33,   -231, -10,  -251, -107, -211, -189, -193,
+      -271, -173, -258, -220, -344, -246, -222, -245, -241, -310, -276, -282,
+      -207, -275, -244, -325, -317, -257, -224, -252, -269, -249, -255, -262,
+      -227, -233, -233, -271, -256, -189, -188, -164, -249, -173, -198, -150,
+      -210, -124, -172, -127, -154, -156, -239, -193, -248, -200, -220, -169,
+      -212, -123, -176, -157, -249, -249, -297, -272, -295, -303, -316, -296,
+      -263, -235, -217, -173, -234, -208, -250, -182, -255, -227, -234, -247,
+      -244, -205, -187, -219, -199, -160, -156, -146, -178, -163, -199, -166,
+      -183, -164, -214, -208, -196, -160, -168, -196, -193, -160, -204, -162,
+      -188, -175, -206, -218, -198, -177, -203, -194, -169, -169, -172, -168,
+      -160, -149, -149, -113, -128, -93,  -92,  -23,  -69,  -13,  -56,  -10,
+      -44,  8,    -28,  9,    -21,  0,    13,   41,   16,   24,   3,    43,
+      28,   54,   11,   12,   -13,  29,   -17,  4,    -27,  25,   -15,  29,
+      -5,   31,   21,   23},
+     /* IRC_Composite_C_R0195_T300_P015.wav */
+     {3,    -3,   3,    -3,   3,    -3,   3,    -3,   4,    -4,   4,    -4,
+      4,    -4,   4,    -4,   4,    -4,   5,    -5,   5,    -5,   5,    -6,
+      6,    -6,   6,    -7,   7,    -8,   9,    -10,  13,   -17,  28,   -93,
+      -234, -180, -207, -251, -293, -262, -250, -346, -376, -130, 67,   -6,
+      108,  -151, -710, -124, 223,  55,   1941, 2161, 1520, 2079, 1404, 552,
+      750,  680,  497,  384,  1277, 1177, 733,  896,  1394, 1657, 1268, 1283,
+      1309, 1208, 892,  805,  954,  655,  670,  296,  482,  320,  570,  596,
+      507,  445,  118,  348,  106,  469,  16,   25,   -157, 191,  5,    47,
+      -157, -78,  -270, -124, -129, -61,  -219, -206, -188, -118, -157, -193,
+      -216, -244, -222, -155, -190, -248, -263, -189, -260, -255, -305, -285,
+      -272, -179, -250, -320, -353, -177, -207, -158, -311, -279, -294, -136,
+      -181, -176, -231, -183, -184, -86,  -72,  -138, -172, -154, -185, -208,
+      -132, -194, -245, -197, -155, -169, -170, -257, -250, -273, -291, -308,
+      -267, -292, -322, -261, -262, -223, -282, -280, -302, -284, -249, -260,
+      -293, -276, -203, -199, -209, -219, -251, -230, -225, -236, -239, -254,
+      -234, -253, -179, -216, -170, -211, -150, -212, -180, -190, -146, -198,
+      -192, -173, -159, -174, -166, -187, -188, -211, -146, -166, -163, -201,
+      -184, -171, -129, -130, -154, -144, -145, -167, -132, -161, -149, -160,
+      -132, -127, -113, -104, -89,  -73,  -95,  -76,  -83,  -55,  -53,  -22,
+      -5,   -13,  -3,   5,    22,   0,    15,   5,    -11,  -15,  15,   -13,
+      -5,   -8,   7,    -15,  26,   37,   33,   12,   52,   50,   45,   34,
+      80,   35,   79,   31},
+     /* IRC_Composite_C_R0195_T315_P015.wav */
+     {-3,    3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,
+      -3,    3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,
+      -3,    3,    -3,   3,    -3,   2,    -2,   2,    -3,   -145, -419, -273,
+      -234,  -320, -494, -368, -332, -137, 23,   435,  141,  -274, 189,  -707,
+      -1152, 1545, 415,  1656, 4094, 1604, 1580, 2053, 473,  -525, 351,  1046,
+      431,   4,    613,  555,  843,  494,  1071, 1048, 1375, 1139, 994,  1431,
+      1283,  1617, 838,  1011, 491,  1146, 716,  701,  358,  388,  443,  310,
+      457,   32,   -5,   -93,  169,  269,  -27,  91,   -165, 86,   -189, 135,
+      -120,  -103, -270, -74,  -56,  -217, -55,  -238, -119, -288, -120, -263,
+      -264,  -223, -279, -179, -302, -173, -211, -262, -189, -255, -177, -382,
+      -172,  -263, -206, -302, -155, -222, -165, -156, -91,  -226, -129, -208,
+      -104,  -116, -7,   -182, -57,  -43,  -59,  -165, -192, -254, -334, -329,
+      -222,  -307, -218, -299, -183, -230, -153, -307, -262, -341, -352, -360,
+      -306,  -320, -349, -329, -284, -313, -262, -287, -276, -348, -332, -282,
+      -281,  -235, -311, -213, -232, -205, -232, -209, -189, -252, -237, -224,
+      -214,  -199, -229, -157, -205, -196, -185, -173, -156, -212, -157, -188,
+      -156,  -172, -131, -151, -175, -157, -139, -172, -126, -151, -157, -177,
+      -123,  -115, -119, -116, -103, -151, -157, -157, -103, -111, -83,  -143,
+      -134,  -106, -78,  -57,  -83,  -76,  -143, -96,  -110, -30,  -81,  -37,
+      -72,   -10,  -21,  -9,   18,   1,    -1,   -42,  21,   2,    14,   26,
+      52,    25,   26,   34,   53,   32,   44,   40,   74,   42,   66,   49,
+      90,    62,   69,   50},
+     /* IRC_Composite_C_R0195_T330_P015.wav */
+     {1,    -1,   1,    0,    0,    0,    0,    0,     0,    0,     0,    1,
+      -1,   1,    -2,   2,    -2,   3,    -3,   4,     -5,   6,     -7,   9,
+      -11,  14,   -18,  25,   -38,  72,   -494, -377,  -163, -458,  -490, -5,
+      -649, -505, -448, 737,  42,   720,  -5,   -1466, 655,  -1957, -272, 3638,
+      1401, 3790, 5414, 320,  1005, 1351, 716,  -1658, 736,  755,   629,  -241,
+      367,  775,  655,  1313, 522,  711,  607,  1017,  814,  1153,  1014, 727,
+      1072, 1193, 1369, 911,  877,  665,  747,  788,   602,  389,   -2,   301,
+      79,   299,  235,  78,   -171, -171, -12,  -204,  -96,  -124,  -285, -390,
+      -109, 136,  -127, -79,  -244, -125, -236, -21,   -257, -313,  -264, -266,
+      -248, -188, -112, -345, -304, -50,  -265, -235,  -235, -38,   -345, -176,
+      -119, -52,  -156, -49,  -98,  -184, -269, -46,   -139, -85,   -206, -15,
+      -64,  -75,  -25,  28,   -69,  -306, -216, -145,  -321, -127,  -396, -359,
+      -364, -161, -402, -370, -313, -329, -419, -337,  -409, -317,  -386, -319,
+      -505, -396, -337, -244, -368, -365, -380, -286,  -289, -230,  -330, -297,
+      -331, -175, -233, -184, -243, -195, -204, -144,  -176, -187,  -238, -158,
+      -239, -170, -186, -124, -214, -122, -133, -166,  -196, -122,  -151, -204,
+      -200, -147, -209, -159, -148, -140, -151, -108,  -106, -148,  -137, -107,
+      -109, -157, -138, -152, -159, -152, -143, -132,  -169, -84,   -121, -99,
+      -110, -53,  -57,  -49,  -65,  -56,  -59,  -72,   -39,  -80,   -76,  -50,
+      -24,  -78,  -31,  -4,   10,   -54,  43,   12,    18,   43,    54,   13,
+      -6,   86,   38,   22,   50,   62,   50,   38,    99,   31,    62,   60,
+      58,   29,   11,   38},
+     /* IRC_Composite_C_R0195_T345_P015.wav */
+     {-11,   11,   -12,  12,   -13,  13,   -14,  15,    -16,   16,    -17,
+      18,    -19,  20,   -22,  23,   -24,  26,   -27,   29,    -32,   34,
+      -38,   43,   -52,  76,   -328, -363, -214, -611,  -509,  -335,  -643,
+      -447,  -869, 31,   57,   1040, 355,  -374, 1188,  -2297, -4332, 5821,
+      -1811, 4636, 8422, 1800, 2573, 2446, 333,  -2543, 348,   441,   1038,
+      -1221, -120, 503,  1705, 557,  994,  621,  1108,  529,   1387,  553,
+      713,   268,  987,  1168, 1373, 742,  1063, 633,   1079,  800,   733,
+      478,   430,  -8,   448,  243,  389,  -34,  240,   -156,  40,    -68,
+      208,   -262, -163, -284, -85,  -119, 4,    -292,  -169,  -173,  -9,
+      -177,  -185, -284, -261, -239, -151, -333, -444,  -278,  -212,  -269,
+      -320,  -207, -353, -167, -122, 5,    -101, -146,  -236,  26,    158,
+      -193,  -6,   31,   -110, -13,  194,  -336, -291,  166,   -187,  -339,
+      71,    83,   -256, -36,  64,   -291, -253, -258,  -423,  -483,  -404,
+      -542,  -450, -398, -404, -517, -415, -395, -300,  -370,  -455,  -406,
+      -370,  -334, -365, -318, -452, -367, -267, -228,  -288,  -345,  -283,
+      -288,  -195, -237, -211, -198, -242, -243, -148,  -120,  -215,  -182,
+      -175,  -171, -178, -118, -143, -239, -153, -169,  -160,  -247,  -180,
+      -208,  -203, -145, -188, -199, -188, -106, -162,  -137,  -157,  -131,
+      -109,  -183, -154, -130, -109, -197, -150, -64,   -168,  -155,  -207,
+      -78,   -176, 15,   -176, -113, -192, -34,  -58,   -13,   -144,  -90,
+      -50,   -81,  -89,  24,   -35,  -100, -56,  -23,   -9,    29,    37,
+      -35,   -51,  58,   55,   55,   -8,   24,   51,    71,    28,    39,
+      46,    49,   51,   27,   -11,  48,   41,   22,    -3,    23,    -14,
+      20,    -19,  -13}};
+
+const int16_t irc_composite_c_r0195_p030[][256] =
+    {/* IRC_Composite_C_R0195_T000_P030.wav */
+     {-54,   55,   -55,   56,   -57,   57,    -58,  58,    -58,   59,   -58,
+      58,    -58,  56,    -55,  52,    -48,   42,   -34,   21,    1,    -38,
+      103,   -854, -293,  -223, -413,  -662,  -699, -525,  -649,  -153, -821,
+      -339,  218,  2022,  2054, -2330, -2792, 1747, -9777, 10764, 1443, 5425,
+      12121, 258,  -1458, 190,  2417,  73,    -781, 460,   -747,  -560, 1297,
+      150,   875,  435,   1456, 292,   1300,  -209, 948,   616,   962,  -100,
+      649,   571,  794,   577,  979,   597,   904,  611,   978,   804,  976,
+      113,   890,  241,   591,  176,   615,   -381, 92,    -14,   208,  -196,
+      144,   -25,  -87,   -290, -51,   -201,  -11,  -376,  13,    -401, 6,
+      -326,  16,   -348,  -73,  -269,  -196,  -276, -77,   -266,  -298, -332,
+      -42,   -326, -208,  -519, -147,  -423,  -93,  -432,  -257,  -451, -109,
+      -270,  -192, -375,  -151, -315,  -188,  -206, -37,   -459,  -168, -241,
+      -10,   -278, -73,   -257, 87,    -155,  -18,  -234,  -126,  -305, -82,
+      -294,  -405, -294,  -172, -409,  -47,   -89,  -22,   -89,   128,  -65,
+      30,    48,   -38,   -108, -174,  -371,  -254, -219,  -255,  -366, -375,
+      -485,  -192, -333,  -204, -374,  -171,  -415, -254,  -437,  -213, -353,
+      -257,  -320, -265,  -229, -230,  -199,  -344, -342,  -208,  -291, -328,
+      -415,  -224, -366,  -172, -311,  -156,  -289, -178,  -259,  -84,  -212,
+      -136,  -195, -136,  -204, -23,   -192,  -135, -219,  -38,   -190, -41,
+      -200,  -34,  -195,  -41,  -117,  -47,   -160, -96,   -111,  -97,  -131,
+      -73,   -186, -78,   -137, -70,   -202,  -93,  -119,  -55,   -149, -67,
+      -139,  -48,  -119,  -34,  -122,  -39,   -119, -31,   -62,   -1,   -61,
+      -28,   -48,  31,    6,    17,    -8,    75,   29,    40,    45,   65,
+      21,    45,   14},
+     /* IRC_Composite_C_R0195_T015_P030.wav */
+     {45,    -49,   52,   -57,  61,    -66,   71,   -77,    83,   -90,
+      98,    -107,  116,  -126, 138,   -150,  164,  -177,   188,  -179,
+      -73,   -966,  -102, -989, 560,   -1173, -327, -1684,  597,  -1115,
+      259,   -1628, 2183, 1183, 3440,  -5302, 3131, -10502, -529, 12738,
+      -6827, 21598, 5258, -7,   -3187, 1963,  495,  327,    502,  -665,
+      -1769, 1964,  -371, 1009, 596,   896,   151,  1819,   -137, 401,
+      915,   974,   249,  114,  559,   962,   751,  584,    371,  576,
+      838,   1123,  906,  351,  -23,   332,   521,  427,    -138, -265,
+      -100,  55,    -54,  59,   -205,  -261,  -77,  2,      -107, -229,
+      -219,  -403,  -235, -134, -143,  -470,  -234, -177,   -61,  -236,
+      -83,   -258,  -197, -192, -101,  -300,  -331, -254,   -216, -242,
+      -271,  -289,  -310, -261, -262,  -333,  -321, -293,   -270, -333,
+      -255,  -243,  -275, -316, -256,  -252,  -176, -243,   -182, -212,
+      -172,  -289,  -118, -228, -303,  -307,  -77,  -225,   -298, -230,
+      -122,  -403,  -282, -152, -387,  -451,  8,    13,     -160, -49,
+      124,   -112,  -104, 87,   -45,   -358,  -217, -135,   -206, -167,
+      -172,  -229,  -317, -197, -162,  -307,  -272, -363,   -290, -325,
+      -168,  -360,  -289, -359, -212,  -286,  -239, -278,   -241, -270,
+      -234,  -333,  -266, -308, -234,  -344,  -218, -357,   -274, -331,
+      -208,  -244,  -205, -284, -219,  -226,  -149, -177,   -181, -201,
+      -159,  -198,  -109, -172, -113,  -218,  -95,  -145,   -67,  -151,
+      -73,   -148,  -84,  -129, -52,   -118,  -51,  -142,   -54,  -113,
+      -29,   -150,  -66,  -118, -44,   -150,  -8,   -105,   -59,  -133,
+      -10,   -103,  -51,  -102, -26,   -114,  2,    -62,    -14,  -78,
+      24,    -62,   35,   -36,  28,    -33,   74,   -22,    57,   13,
+      101,   -23,   94,   36,   50,    32},
+     /* IRC_Composite_C_R0195_T030_P030.wav */
+     {22,   -22,   22,    -22,   22,     -22,   21,    -21,  20,   -19,  18,
+      -16,  13,    -9,    3,     6,      -23,   62,    -927, -68,  -994, 540,
+      -922, -246,  -1369, 328,   -1205,  -194,  -1442, 1439, 570,  2758, -885,
+      3393, -4925, -8851, 7444,  -11725, 26992, 7407,  2376, 4575, -342, -2312,
+      -724, 2866,  -938,  -1366, 86,     -585,  1150,  1637, 627,  500,  364,
+      713,  726,   1042,  -231,  616,    243,   1337,  315,  722,  212,  833,
+      509,  1128,  741,   763,   -23,    706,   -315,  392,  -241, 100,  -771,
+      -188, -212,  298,   -422,  -413,   -205,  37,    -281, -130, -523, -88,
+      -503, -112,  -573,  -211,  -454,   -80,   -504,  -132, -294, -80,  -239,
+      -22,  -204,  -163,  -320,  -82,    -296,  -297,  -476, -149, -313, -195,
+      -414, -189,  -328,  -173,  -333,   -197,  -351,  -236, -353, -212, -335,
+      -289, -358,  -212,  -232,  -235,   -269,  -127,  -247, -255, -385, -167,
+      -387, -267,  -392,  -201,  -373,   -245,  -334,  -180, -331, -245, -322,
+      -210, -238,  -125,  -198,  -59,    -138,  -44,   -137, -3,   -58,  9,
+      -199, -115,  -324,  -112,  -304,   -122,  -334,  -195, -342, -88,  -186,
+      -146, -296,  -157,  -238,  -181,   -256,  -164,  -321, -198, -320, -193,
+      -346, -189,  -318,  -169,  -297,   -240,  -369,  -178, -273, -261, -370,
+      -222, -299,  -226,  -327,  -186,   -311,  -192,  -283, -157, -277, -144,
+      -267, -140,  -249,  -109,  -220,   -95,   -203,  -109, -172, -51,  -152,
+      -88,  -149,  -49,   -133,  -83,    -123,  -30,   -107, -62,  -145, -55,
+      -94,  -23,   -81,   -49,   -86,    -1,    -42,   -24,  -119, -38,  -58,
+      -5,   -105,  -50,   -71,   1,      -68,   -36,   -53,  -6,   -19,  20,
+      -14,  18,    -29,   55,    -17,    56,    -9,    73,   3,    81,   35,
+      97,   32,    73},
+     /* IRC_Composite_C_R0195_T045_P030.wav */
+     {9,      -7,    5,     -3,    0,     3,     -7,   12,   -18,  25,    -34,
+      45,     -60,   79,    -107,  151,   -363,  -677, -240, -871, 518,   -941,
+      -269,   -1276, 265,   -1859, 1187,  -1350, 2149, 1,    5698, -4523, 4866,
+      -12427, -2128, 5167,  3666,  25146, 2435,  401,  417,  1825, -3164, 2605,
+      613,    -1851, -1719, 979,   -329,  2301,  -426, 1606, -126, 1675,  119,
+      1149,   -57,   595,   -80,   1644,  19,    793,  -129, 1326, 678,   1045,
+      -17,    856,   -22,   267,   -380,  190,   -844, -146, -664, 199,   -735,
+      -167,   -428,  32,    -402,  -217,  -411,  -308, -427, -284, -543,  -438,
+      -454,   -307,  -561,  -259,  -430,  -199,  -381, -181, -382, -148,  -228,
+      -87,    -405,  -169,  -303,  -91,   -361,  -175, -378, -291, -318,  -127,
+      -301,   -269,  -260,  -171,  -316,  -164,  -268, -241, -361, -178,  -341,
+      -229,   -322,  -201,  -335,  -221,  -314,  -343, -328, -221, -321,  -372,
+      -308,   -210,  -400,  -380,  -322,  -269,  -386, -304, -356, -223,  -292,
+      -166,   -317,  -192,  -91,   -100,  -158,  -51,  -73,  -116, -23,   -56,
+      -150,   -243,  -92,   -175,  -169,  -181,  -177, -294, -169, -187,  -210,
+      -286,   -130,  -227,  -213,  -253,  -159,  -238, -150, -211, -249,  -260,
+      -144,   -232,  -247,  -283,  -227,  -289,  -178, -293, -252, -299,  -148,
+      -285,   -211,  -271,  -197,  -276,  -200,  -222, -204, -252, -176,  -235,
+      -205,   -223,  -107,  -218,  -119,  -198,  -109, -182, -35,  -149,  -101,
+      -171,   -62,   -123,  -44,   -128,  -75,   -141, -37,  -172, -54,   -143,
+      -11,    -144,  -27,   -104,  26,    -103,  23,   -105, 37,   -73,   18,
+      -74,    15,    -63,   4,     -83,   16,    -68,  -5,   -50,  47,    -28,
+      24,     -35,   41,    -27,   78,    -21,   77,   -17,  105,  16,    96,
+      7,      122,   24},
+     /* IRC_Composite_C_R0195_T060_P030.wav */
+     {8,      -7,    7,     -6,   5,     -5,   4,     -3,   3,    -3,    3,
+      -3,     5,     -9,    20,   86,    -790, 103,   -872, 55,   -678,  -196,
+      -962,   -387,  -893,  162,  722,   1609, 608,   2929, 1713, -6837, -667,
+      -12691, 15713, 13906, 9242, 5982,  83,   -1809, 1076, 1596, 1326,  -1529,
+      -1392,  -366,  731,   78,   1368,  -246, 850,   592,  521,  699,   1115,
+      310,    930,   973,   592,  664,   159,  708,   710,  489,  -74,   485,
+      -697,   374,   -695,  249,  -1252, 204,  -915,  137,  -777, -91,   -241,
+      -403,   -350,  -466,  -269, -521,  -499, -540,  -380, -320, -709,  -233,
+      -517,   -228,  -634,  -244, -652,  -290, -447,  -147, -545, -241,  -352,
+      5,      -488,  -69,   -437, 53,    -465, -6,    -353, -78,  -525,  -24,
+      -335,   -157,  -418,  -27,  -367,  -103, -287,  -241, -397, -158,  -276,
+      -304,   -351,  -181,  -369, -225,  -401, -209,  -512, -184, -447,  -153,
+      -476,   -152,  -484,  -76,  -492,  -243, -498,  -127, -515, -186,  -392,
+      -135,   -408,  -53,   -239, -82,   -191, 13,    -138, -32,  -118,  -6,
+      -177,   -44,   -187,  -143, -246,  -50,  -262,  -108, -264, -71,   -256,
+      -48,    -325,  -90,   -349, -63,   -352, -110,  -280, -159, -311,  -153,
+      -350,   -187,  -295,  -152, -318,  -137, -192,  -178, -252, -142,  -284,
+      -157,   -277,  -190,  -286, -151,  -282, -119,  -257, -91,  -310,  -54,
+      -224,   -93,   -207,  -75,  -245,  -9,   -203,  -48,  -237, 12,    -182,
+      -40,    -201,  33,    -146, -26,   -200, -37,   -156, -43,  -159,  -28,
+      -130,   -32,   -114,  20,   -143,  6,    -113,  12,   -126, 58,    -103,
+      21,     -121,  59,    -91,  44,    -108, 59,    -43,  47,   -58,   60,
+      -38,    52,    -61,   83,   -36,   62,   3,     66,   27,   84,    84,
+      65,     66,    103},
+     /* IRC_Composite_C_R0195_T075_P030.wav */
+     {100,   -103,  106,   -109, 112,  -115, 118,  -120, 122,   -122,  119,
+      -110,  88,    -25,   53,   -874, 378,  -897, 369,  -1226, 537,   -1263,
+      224,   -1798, 1274,  98,   2195, -462, 4170, 361,  -3989, -3149, -11328,
+      17722, 9812,  11683, 3944, -511, -808, 1931, 2611, 1751,  -2704, -605,
+      -430,  728,   -349,  908,  612,  941,  -687, 931,  759,   -66,   894,
+      2123,  310,   1489,  449,  746,  475,  494,  -493, -725,  213,   -1074,
+      381,   -1100, -205,  -670, -115, -645, -170, -469, -431,  31,    -824,
+      -248,  -707,  -458,  -476, -267, -901, -228, -228, -601,  -420,  -335,
+      -514,  -545,  -367,  -632, -446, -566, -351, -391, -306,  -444,  -134,
+      -388,  -144,  -333,  -243, -194, -98,  -434, -105, -173,  -206,  -249,
+      -121,  -377,  -102,  -345, -159, -256, -146, -470, -106,  -327,  -152,
+      -330,  -205,  -368,  -192, -409, -197, -444, -310, -359,  -229,  -461,
+      -161,  -393,  -318,  -351, -227, -387, -261, -314, -291,  -270,  -249,
+      -242,  -213,  -299,  -192, -269, -108, -253, -73,  -278,  80,    -171,
+      -11,   -111,  7,     -198, -30,  -155, -148, -117, -164,  -168,  -181,
+      -155,  -172,  -221,  -156, -207, -176, -229, -130, -292,  -171,  -280,
+      -233,  -288,  -132,  -237, -242, -209, -166, -245, -147,  -263,  -160,
+      -240,  -126,  -261,  -109, -276, -150, -274, -92,  -247,  -135,  -225,
+      -43,   -207,  -57,   -174, -33,  -157, -10,  -172, -20,   -116,  -56,
+      -146,  22,    -132,  -61,  -119, -53,  -166, -22,  -107,  -81,   -99,
+      -40,   -65,   -89,   -70,  -32,  -77,  -31,  -75,  -25,   -75,   36,
+      -146,  3,     -52,   3,    -100, 13,   -55,  -5,   -17,   13,    -41,
+      5,     -12,   59,    -13,  32,   3,    61,   44,   107,   53,    129,
+      85,    118,   95},
+     /* IRC_Composite_C_R0195_T090_P030.wav */
+     {-85,    89,    -94,   98,    -102,  107,   -112,  116,   -119,  121,
+      -119,   106,   -58,   -455,  60,    99,    -1054, -36,   -88,   242,
+      -1635,  433,   -1255, 1196,  -1210, 2540,  -840,  5344,  -2798, 4670,
+      -10472, -874,  1281,  9198,  19443, 266,   2404,  -1705, 3816,  1662,
+      3038,   83,    -1499, -2490, 1368,  -37,   500,   -334,  1490,  -626,
+      909,    -15,   1075,  678,   1886,  658,   852,   386,   1418,  146,
+      400,    -1073, -338,  -633,  -165,  -1024, -383,  -1251, -264,  -407,
+      -25,    -1062, -140,  -559,  70,    -716,  -459,  -642,  -96,   -725,
+      -472,   -412,  -384,  -647,  -281,  -605,  -395,  -572,  -399,  -706,
+      -351,   -583,  -297,  -495,  -434,  -423,  -201,  -406,  -407,  -323,
+      -268,   -341,  -186,  -360,  -201,  -241,  -141,  -313,  -94,   -257,
+      -169,   -208,  -194,  -224,  -147,  -275,  -276,  -255,  -215,  -224,
+      -195,   -256,  -265,  -271,  -252,  -352,  -290,  -430,  -350,  -309,
+      -268,   -387,  -267,  -289,  -271,  -262,  -303,  -332,  -237,  -287,
+      -328,   -292,  -220,  -307,  -211,  -297,  -184,  -216,  -183,  -248,
+      -116,   -180,  -163,  -187,  -112,  -160,  -80,   -117,  -46,   -81,
+      -25,    -91,   -53,   -114,  -120,  -176,  -181,  -173,  -184,  -205,
+      -224,   -154,  -233,  -229,  -178,  -204,  -243,  -197,  -188,  -194,
+      -206,   -154,  -204,  -205,  -213,  -116,  -191,  -127,  -210,  -145,
+      -216,   -173,  -183,  -177,  -263,  -209,  -141,  -116,  -148,  -107,
+      -124,   -68,   -87,   -45,   -121,  -68,   -134,  -77,   -87,   -29,
+      -115,   -67,   -117,  -46,   -114,  -27,   -115,  -45,   -100,  -16,
+      -82,    -32,   -85,   -17,   -74,   -28,   -41,   -27,   -85,   -23,
+      -68,    -40,   -47,   -15,   -96,   -13,   -38,   11,    -39,   -38,
+      -19,    1,     -19,   -3,    -8,    -15,   -22,   28,    17,    70,
+      76,     128,   80,    124,   112,   151},
+     /* IRC_Composite_C_R0195_T105_P030.wav */
+     {104,   -111,  117,  -125, 133,  -143, 153,  -166, 180,   -197,  218,
+      -245,  288,   -401, 132,  -429, -287, -538, 503,  -701,  -185,  -1083,
+      867,   -1151, 1236, -192, 2589, 704,  2171, -548, -4634, -1245, -3979,
+      18483, 6281,  7009, -196, 501,  1762, 4063, 3838, -1,    -2732, -1085,
+      -409,  884,   444,  -393, -9,   733,  362,  138,  1179,  1594,  1300,
+      947,   317,   734,  700,  55,   -900, -943, -310, -712,  -521,  -933,
+      -377,  -1554, -143, -410, 48,   -924, -152, -440, -212,  -554,  -286,
+      -601,  -527,  -371, -454, -653, -533, -426, -450, -579,  -486,  -483,
+      -369,  -609,  -354, -543, -201, -457, -293, -586, -146,  -430,  -339,
+      -522,  -196,  -439, -324, -340, -141, -423, -157, -344,  -39,   -390,
+      -88,   -363,  -66,  -244, -152, -284, -176, -153, -268,  -264,  -245,
+      -148,  -256,  -290, -221, -325, -190, -340, -211, -490,  -165,  -333,
+      -238,  -471,  -203, -235, -228, -397, -218, -264, -147,  -365,  -267,
+      -367,  -142,  -319, -211, -356, -158, -290, -141, -243,  -127,  -217,
+      -86,   -135,  -64,  -128, -75,  -153, -67,  -155, -25,   -190,  -45,
+      -193,  -18,   -259, 1,    -235, -77,  -284, -88,  -227,  -151,  -250,
+      -186,  -245,  -111, -256, -126, -274, -46,  -272, -88,   -254,  -32,
+      -214,  -71,   -201, -144, -187, -98,  -154, -182, -196,  -78,   -154,
+      -85,   -159,  -36,  -221, -26,  -121, -4,   -188, -47,   -127,  -22,
+      -132,  -24,   -158, -49,  -155, -2,   -158, -21,  -135,  -30,   -121,
+      10,    -87,   -16,  -118, 10,   -48,  -22,  -86,  -12,   -80,   14,
+      -80,   13,    -86,  50,   -103, 29,   -96,  64,   -85,   45,    -104,
+      50,    -45,   64,   -67,  -11,  -39,  117,  -14,  158,   27,    234,
+      13,    225,   68},
+     /* IRC_Composite_C_R0195_T120_P030.wav */
+     {28,    -28,  29,    -29,  29,   -29,   29,   -27,   25,   -21,   15,
+      -5,    -12,  43,    -140, 88,   -559,  -189, 139,   -567, 198,   -314,
+      -739,  -383, 1538,  -450, 794,  642,   4608, -2302, 1277, -7331, 2563,
+      5148,  5379, 12280, 2713, -323, -1693, 7682, 3305,  1850, -1530, -2098,
+      -1411, 1431, -392,  275,  -715, 741,   -246, 1255,  618,  1870,  987,
+      1146,  368,  730,   -169, 18,   -891,  -805, -1522, -439, -421,  -616,
+      -1006, -319, -441,  -263, -511, -164,  -630, -92,   -636, -184,  -746,
+      -202,  -725, -323,  -658, -347, -529,  -471, -673,  -383, -415,  -310,
+      -627,  -358, -521,  -225, -567, -258,  -424, -264,  -432, -273,  -312,
+      -333,  -413, -308,  -330, -340, -367,  -280, -273,  -329, -276,  -223,
+      -220,  -254, -266,  -200, -177, -199,  -276, -232,  -217, -244,  -204,
+      -238,  -182, -299,  -214, -297, -195,  -311, -235,  -299, -177,  -299,
+      -241,  -285, -222,  -309, -277, -284,  -244, -293,  -235, -236,  -241,
+      -258,  -206, -250,  -223, -292, -255,  -245, -224,  -254, -179,  -205,
+      -152,  -237, -116,  -146, -78,  -191,  -75,  -140,  -27,  -176,  -63,
+      -134,  -82,  -199,  -79,  -164, -84,   -202, -123,  -164, -105,  -201,
+      -110,  -207, -105,  -211, -126, -202,  -134, -203,  -129, -204,  -131,
+      -154,  -100, -194,  -94,  -163, -74,   -166, -82,   -216, -107,  -149,
+      -27,   -161, -56,   -129, -40,  -130,  -55,  -114,  -53,  -102,  -63,
+      -77,   -31,  -89,   -106, -127, -83,   -123, -74,   -94,  -24,   -173,
+      -31,   -94,  11,    -133, -8,   -83,   -23,  -88,   12,   -65,   -46,
+      -111,  25,   -94,   13,   -117, 28,    -88,  37,    -79,  24,    -35,
+      51,    -27,  22,    7,    50,   -3,    53,   7,     103,  86,    163,
+      91,    162,  75},
+     /* IRC_Composite_C_R0195_T135_P030.wav */
+     {6,    -6,   7,    -7,   8,    -8,   9,    -9,    10,    -10,  11,   -12,
+      13,   -14,  16,   -18,  -114, -234, -98,  -110,  -115,  -173, -170, -232,
+      -274, 363,  317,  1353, 133,  2229, 1323, 703,   -7423, 4287, 1521, 5913,
+      8831, 4880, 804,  -872, 5713, 3941, 3130, -2817, -2290, -315, 304,  -50,
+      -26,  -209, 494,  295,  1100, 751,  1288, 533,   716,   964,  54,   -449,
+      -399, -528, -808, -934, -947, -951, -457, -538,  -157,  -625, -308, -385,
+      -95,  -571, -310, -374, -278, -494, -514, -429,  -461,  -556, -433, -529,
+      -432, -535, -327, -580, -236, -492, -287, -555,  -216,  -430, -280, -451,
+      -246, -351, -270, -341, -290, -325, -254, -404,  -265,  -383, -232, -420,
+      -232, -398, -198, -399, -199, -300, -210, -291,  -213,  -230, -263, -273,
+      -261, -215, -220, -262, -203, -257, -189, -275,  -199,  -298, -243, -292,
+      -198, -258, -246, -263, -220, -238, -243, -271,  -259,  -242, -227, -263,
+      -220, -247, -166, -281, -180, -274, -165, -251,  -184,  -245, -176, -216,
+      -217, -207, -158, -156, -155, -166, -106, -127,  -108,  -139, -81,  -152,
+      -98,  -167, -88,  -170, -109, -150, -125, -150,  -119,  -119, -124, -135,
+      -111, -142, -113, -144, -112, -168, -76,  -193,  -154,  -199, -103, -124,
+      -149, -137, -125, -90,  -105, -96,  -120, -90,   -84,   -94,  -99,  -96,
+      -69,  -90,  -69,  -82,  -94,  -80,  -67,  -60,   -94,   -70,  -110, -76,
+      -102, -51,  -103, -55,  -74,  -38,  -105, -57,   -91,   -76,  -96,  -51,
+      -36,  -59,  -52,  -57,  -56,  -61,  -46,  -70,   -31,   -32,  -25,  -47,
+      6,    -6,   17,   -33,  -1,   20,   49,   32,    43,    24,   63,   112,
+      138,  114,  43,   112},
+     /* IRC_Composite_C_R0195_T150_P030.wav */
+     {40,   -42,  44,   -47,  49,    -53,  56,   -60,  64,   -69,   75,    -82,
+      90,   -99,  110,  -125, 148,   -207, 85,   -316, 199,  -301,  148,   -341,
+      137,  -377, 122,  76,   967,   463,  1542, 593,  2935, -2335, -3746, 3892,
+      1126, 7018, 5853, 6116, -1287, 2364, 4148, 3574, 181,  -2393, -1125, -178,
+      512,  -398, -17,  -462, 310,   1083, 929,  1228, 538,  479,   431,   450,
+      -124, -446, -926, -681, -484,  -854, -858, -886, -425, -380,  -163,  -187,
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+      -398, -443, -447, -458, -316,  -355, -323, -348, -312, -366,  -331,  -282,
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+      -24,  -3,   21,   -10,  -1,    6,    97,   22,   44,   1,     101,   93,
+      134,  63,   82,   45},
+     /* IRC_Composite_C_R0195_T165_P030.wav */
+     {-5,    5,    -6,   6,    -6,   6,    -6,   7,    -7,   7,    -8,   8,
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+      -13,   -37,  25,   -7,   9,    -13,  -1,   53,   58,   59,   79,   109,
+      133,   124,  84,   60},
+     /* IRC_Composite_C_R0195_T180_P030.wav */
+     {11,   -12,  12,    -12,   13,   -13,  14,   -14,  15,   -15,  16,   -17,
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+      -53,  -1,   -6,    52,    20,   54,   20,   68,   72,   99,   107,  107,
+      128,  102,  141,   66},
+     /* IRC_Composite_C_R0195_T195_P030.wav */
+     {-2,   2,    -2,   2,    -2,   2,     -2,    2,    -2,   2,    -2,   2,
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+      -45,  -46,  -3,   16,   24,   24,    27,    58,   42,   84,   85,   129,
+      127,  141,  127,  125},
+     /* IRC_Composite_C_R0195_T210_P030.wav */
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+      -79,  -59,  -59,  -5,   -31,  13,   -2,   37,   21,   52,    67,   76,
+      116,  109,  131,  125},
+     /* IRC_Composite_C_R0195_T225_P030.wav */
+     {-1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,
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+      -89,  -78,  -59,  -67,  -27,  -39,  -8,   -19,  9,    9,    34,   44,
+      63,   65,   98,   113},
+     /* IRC_Composite_C_R0195_T240_P030.wav */
+     {0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
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+      35,   22,   73,   65},
+     /* IRC_Composite_C_R0195_T255_P030.wav */
+     {1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,
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+      -21,  13,   14,   31},
+     /* IRC_Composite_C_R0195_T270_P030.wav */
+     {0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
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+      4,    -20,  19,   6},
+     /* IRC_Composite_C_R0195_T285_P030.wav */
+     {0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
+      0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    -1,
+      1,    -1,   1,    -1,   1,    -1,   2,    -2,   3,    -3,   4,    -7,
+      13,   -190, -183, -235, -156, -218, -208, -304, -188, -211, -199, -139,
+      -161, 153,  336,  233,  -123, -164, -557, -685, 1147, 1908, 2035, 3160,
+      1492, 785,  1242, 1667, 1506, 818,  1143, 817,  1178, 780,  1057, 657,
+      807,  713,  683,  790,  563,  729,  595,  855,  599,  704,  615,  614,
+      540,  520,  362,  304,  245,  337,  219,  250,  133,  197,  59,   126,
+      -65,  -70,  -160, -89,  -159, -231, -254, -291, -203, -274, -185, -304,
+      -208, -285, -189, -264, -256, -331, -243, -311, -262, -308, -280, -304,
+      -246, -335, -250, -318, -269, -309, -282, -300, -260, -340, -268, -307,
+      -236, -284, -269, -325, -251, -300, -247, -310, -244, -277, -183, -239,
+      -198, -269, -229, -228, -237, -229, -245, -249, -264, -220, -228, -201,
+      -208, -179, -162, -96,  -70,  -76,  -134, -147, -121, -145, -142, -175,
+      -192, -136, -108, -69,  -153, -156, -210, -206, -236, -198, -320, -233,
+      -263, -173, -277, -125, -224, -144, -166, -131, -224, -164, -197, -216,
+      -260, -240, -244, -177, -196, -187, -157, -155, -171, -202, -195, -206,
+      -203, -187, -195, -236, -200, -188, -186, -176, -232, -177, -163, -133,
+      -189, -164, -179, -140, -153, -146, -152, -146, -135, -108, -114, -110,
+      -119, -88,  -135, -96,  -118, -89,  -127, -122, -110, -95,  -72,  -108,
+      -98,  -91,  -59,  -48,  -71,  -44,  -52,  -26,  -30,  -20,  -36,  -34,
+      -3,   -13,  9,    16},
+     /* IRC_Composite_C_R0195_T300_P030.wav */
+     {1,    -1,   2,     -2,   2,    -2,   2,    -2,   2,    -2,   2,    -2,
+      2,    -2,   2,     -3,   3,    -3,   3,    -3,   3,    -4,   4,    -4,
+      5,    -5,   5,     -6,   7,    -8,   9,    -10,  12,   -16,  27,   -117,
+      -227, -314, -248,  -219, -228, -232, -272, -389, -204, -111, 300,  56,
+      366,  86,   -1011, -298, 44,   -215, 2477, 2714, 2032, 2711, 645,  499,
+      536,  1654, 449,   374,  611,  764,  837,  821,  1350, 980,  1002, 808,
+      1149, 967,  761,   841,  707,  633,  745,  777,  758,  579,  642,  512,
+      570,  444,  429,   371,  457,  296,  276,  181,  191,  36,   17,   10,
+      -72,  -108, -107,  -232, -151, -205, -165, -300, -163, -265, -253, -269,
+      -203, -258, -301,  -275, -269, -292, -272, -260, -302, -297, -289, -229,
+      -300, -264, -305,  -243, -283, -264, -269, -279, -280, -287, -259, -269,
+      -276, -247, -263,  -262, -247, -231, -256, -218, -280, -241, -295, -239,
+      -252, -218, -256,  -230, -258, -241, -251, -148, -189, -159, -149, -45,
+      -91,  -37,  -119,  -83,  -194, -47,  -124, -58,  -134, -130, -231, -259,
+      -239, -276, -269,  -285, -258, -248, -198, -188, -240, -206, -250, -202,
+      -249, -205, -247,  -163, -226, -240, -217, -219, -189, -209, -212, -237,
+      -257, -142, -184,  -180, -176, -167, -213, -202, -187, -188, -191, -235,
+      -229, -241, -179,  -197, -185, -210, -199, -189, -160, -163, -174, -186,
+      -151, -189, -138,  -133, -147, -130, -137, -107, -118, -101, -102, -120,
+      -113, -110, -78,   -102, -64,  -110, -74,  -76,  -59,  -92,  -76,  -85,
+      -71,  -79,  -43,   -88,  -39,  -85,  -19,  -81,  13,   -60,  8,    -26,
+      31,   8,    16,    23},
+     /* IRC_Composite_C_R0195_T315_P030.wav */
+     {-3,    3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,    -3,   3,
+      -3,    3,    -3,   3,    -3,   3,    -3,   3,    -4,   4,    -4,   4,
+      -4,    5,    -5,   6,    -7,   10,   -15,  27,   -80,  -353, -110, -370,
+      -400,  -195, -345, -309, -394, -97,  -188, 193,  754,  56,   -290, -194,
+      -1922, 214,  1653, 1159, 4327, 2754, 1527, 1212, 743,  317,  539,  1293,
+      -56,   -21,  321,  1262, 295,  442,  708,  1213, 856,  813,  872,  1000,
+      917,   1073, 730,  897,  944,  834,  864,  775,  610,  537,  692,  383,
+      605,   370,  536,  245,  348,  155,  280,  101,  83,   31,   90,   -127,
+      -63,   -179, -119, -272, -116, -304, -232, -233, -212, -345, -176, -295,
+      -247,  -334, -204, -342, -233, -195, -284, -324, -229, -236, -255, -304,
+      -258,  -323, -258, -297, -224, -323, -252, -263, -224, -317, -252, -224,
+      -255,  -281, -221, -218, -205, -288, -209, -253, -193, -244, -249, -201,
+      -218,  -210, -200, -172, -180, -181, -84,  -203, -75,  -54,  19,   -108,
+      -2,    -46,  -171, -209, -203, -294, -294, -286, -264, -295, -154, -187,
+      -222,  -182, -252, -197, -249, -211, -305, -232, -289, -224, -284, -267,
+      -256,  -184, -221, -260, -295, -281, -237, -233, -228, -269, -251, -203,
+      -162,  -193, -192, -189, -166, -237, -188, -201, -208, -209, -181, -224,
+      -228,  -151, -192, -200, -209, -131, -196, -159, -162, -159, -166, -116,
+      -160,  -151, -131, -47,  -142, -119, -122, -73,  -93,  -108, -96,  -133,
+      -90,   -84,  -86,  -83,  -90,  -56,  -96,  -65,  -65,  -48,  -80,  -67,
+      -79,   -50,  -60,  -46,  -63,  -39,  -44,  -12,  -27,  9,    -35,  23,
+      -3,    24,   6,    41},
+     /* IRC_Composite_C_R0195_T330_P030.wav */
+     {-5,   5,    -5,   5,    -5,   5,    -5,   5,     -5,   6,    -6,    6,
+      -6,   6,    -6,   6,    -6,   6,    -7,   7,     -7,   7,    -8,    8,
+      -9,   9,    -11,  14,   -26,  -447, -195, -107,  -505, -457, -480,  -118,
+      -514, -260, -345, 82,   416,  511,  1319, -1763, -994, 103,  -1768, 3014,
+      4385, 2231, 5234, 1274, 290,  -49,  1679, 111,   -604, 148,  524,   573,
+      -122, 973,  360,  1000, 586,  1035, 552,  578,   788,  885,  854,   618,
+      749,  894,  854,  1021, 930,  968,  502,  1006,  765,  768,  438,   460,
+      339,  388,  236,  299,  159,  177,  -65,  162,   -6,   -74,  -134,  -51,
+      -230, -132, -193, -105, -327, -152, -257, -191,  -291, -227, -315,  -280,
+      -252, -192, -306, -253, -281, -182, -326, -252,  -283, -225, -353,  -255,
+      -282, -258, -322, -236, -271, -288, -246, -248,  -299, -265, -254,  -312,
+      -330, -129, -295, -141, -204, -129, -190, -22,   -154, -203, -125,  -172,
+      -156, -188, -111, -223, -112, -106, -241, -86,   -37,  -262, -238,  -193,
+      -103, -105, -7,   -250, -241, -39,  -63,  -184,  -238, -261, -364,  -243,
+      -263, -310, -361, -286, -407, -227, -301, -289,  -311, -319, -263,  -215,
+      -201, -304, -287, -339, -226, -246, -239, -335,  -276, -276, -181,  -234,
+      -181, -257, -180, -234, -151, -165, -203, -213,  -194, -170, -205,  -157,
+      -164, -191, -168, -154, -86,  -133, -128, -162,  -96,  -147, -81,   -165,
+      -68,  -173, -90,  -164, -64,  -127, -68,  -162,  -97,  -99,  -50,   -115,
+      -84,  -111, -102, -87,  -74,  -80,  -123, -76,   -57,  -95,  -59,   -42,
+      -45,  -74,  -38,  -27,  -9,   -30,  0,    -55,   36,   -23,  58,    -21,
+      61,   -7,   59,   29},
+     /* IRC_Composite_C_R0195_T345_P030.wav */
+     {0,    0,    0,    0,    0,    0,    0,     1,     -1,    1,    -1,
+      2,    -2,   2,    -3,   3,    -4,   5,     -5,    6,     -7,   8,
+      -9,   10,   -10,  -1,   -563, -328, -175,  -595,  -424,  -853, 130,
+      -981, 36,   -843, 608,  167,  1104, 1422,  -2010, -2282, 1073, -3818,
+      6278, 4848, 2400, 7303, 1229, -443, -1130, 2494,  95,    -616, -743,
+      335,  108,  968,  417,  599,  684,  1073,  630,   810,   221,  839,
+      469,  1136, 218,  739,  427,  945,  692,   942,   955,   757,  852,
+      1008, 762,  614,  598,  745,  200,  403,   51,    357,   29,   333,
+      -147, -65,  9,    35,   -114, -161, -231,  -52,   -197,  -139, -202,
+      -61,  -310, -211, -214, -165, -307, -207,  -293,  -261,  -259, -204,
+      -201, -310, -323, -244, -250, -260, -335,  -226,  -421,  -206, -333,
+      -191, -401, -188, -244, -264, -283, -225,  -208,  -338,  -233, -69,
+      -304, -181, -213, -18,  -265, 39,   -68,   -82,   -113,  -107, 16,
+      -329, -80,  -324, -205, -399, -184, -328,  -391,  -87,   -9,   36,
+      1,    59,   -85,  124,  -60,  -143, -307,  -231,  -406,  -296, -456,
+      -311, -297, -269, -433, -350, -189, -283,  -277,  -303,  -343, -325,
+      -287, -213, -328, -275, -279, -293, -296,  -268,  -217,  -284, -286,
+      -294, -277, -236, -247, -201, -305, -195,  -224,  -135,  -244, -185,
+      -128, -225, -126, -186, -74,  -235, -93,   -209,  -95,   -133, -90,
+      -131, -165, -80,  -89,  -69,  -148, -104,  -72,   -157,  -77,  -134,
+      -88,  -171, -67,  -113, -124, -110, -76,   -110,  -130,  -59,  -96,
+      -88,  -119, -61,  -109, -81,  -81,  -67,   -82,   -54,   -22,  -49,
+      -19,  -18,  2,    -15,  25,   13,   24,    8,     45,    57,   4,
+      20,   23,   60}};
+
+const int16_t irc_composite_c_r0195_p045[][256] =
+    {/* IRC_Composite_C_R0195_T000_P045.wav */
+     {-30,  30,   -31,  32,   -33,  34,    -35,   36,   -37,   38,    -39,
+      40,   -41,  42,   -43,  43,   -44,   44,    -42,  38,    -28,   -9,
+      -568, -169, -659, -127, -629, -131,  -879,  -529, -702,  -207,  -686,
+      -146, 757,  1500, 46,   1169, -4599, -1790, 163,  -4363, 15423, 4884,
+      3940, 951,  722,  -172, 1158, 1933,  -227,  -750, 303,   -613,  1137,
+      856,  493,  193,  1077, 347,  1002,  300,   420,  882,   521,   444,
+      512,  395,  442,  484,  453,  404,   610,   620,  839,   826,   758,
+      649,  736,  767,  608,  484,  493,   147,   169,  427,   10,    204,
+      -311, 19,   -245, 98,   -241, 0,     -167,  16,   -127,  -26,   -105,
+      -158, -191, -79,  -246, -206, -268,  -177,  -349, -216,  -328,  -216,
+      -366, -178, -332, -214, -308, -168,  -352,  -204, -312,  -132,  -301,
+      -215, -309, -152, -304, -202, -336,  -200,  -308, -229,  -340,  -205,
+      -331, -196, -326, -240, -274, -198,  -310,  -200, -321,  -159,  -282,
+      -190, -264, -176, -207, -74,  -246,  -157,  -167, -77,   -245,  18,
+      -155, -133, -274, -40,  -289, -273,  -267,  -139, -520,  -214,  -265,
+      -252, -317, -46,  -133, -17,  38,    121,   -153, 151,   27,    -13,
+      -114, 13,   -297, -74,  -188, -147,  -332,  -197, -268,  -297,  -199,
+      -221, -310, -309, -263, -206, -258,  -237,  -414, -155,  -291,  -200,
+      -367, -189, -287, -139, -241, -261,  -247,  -142, -190,  -233,  -239,
+      -158, -249, -110, -242, -173, -207,  -82,   -217, -136,  -180,  -136,
+      -164, -112, -177, -97,  -134, -40,   -164,  -32,  -115,  -16,   -91,
+      -12,  -112, 4,    -81,  -19,  -100,  -20,   -70,  -21,   -87,   -6,
+      -59,  22,   -68,  -7,   -66,  36,    -20,   13,   -47,   -2,    -27,
+      28,   -4,   13},
+     /* IRC_Composite_C_R0195_T015_P045.wav */
+     {16,   -14,   12,   -9,    7,     -4,    1,     3,     -7,   12,   -17,
+      22,   -29,   36,   -45,   56,    -69,   85,    -107,  143,  -247, -403,
+      -616, -230,  -517, -270,  -663,  -670,  -583,  -556,  -570, -300, 850,
+      1495, 298,   1490, -1579, -5393, 1137,  -9580, 17573, 8715, 3526, 4288,
+      -490, -1083, 280,  3534,  -555,  -1021, -341,  -114,  804,  1521, 139,
+      355,  844,   673,  325,   735,   -35,   1245,  91,    909,  -18,  912,
+      280,  465,   82,   792,   554,   703,   257,   849,   594,  952,  381,
+      794,  169,   438,  96,    219,   18,    183,   -220,  -11,  -303, 284,
+      -345, -44,   -429, -44,   -355,  -58,   -224,  -155,  -342, -30,  -264,
+      -140, -360,  -95,  -355,  -133,  -396,  -161,  -354,  -174, -312, -65,
+      -314, -194,  -357, -142,  -289,  -170,  -390,  -213,  -337, -152, -369,
+      -215, -364,  -191, -332,  -194,  -330,  -206,  -298,  -234, -318, -198,
+      -344, -225,  -368, -198,  -333,  -240,  -308,  -215,  -230, -223, -291,
+      -224, -258,  -133, -220,  -212,  -260,  -84,   -207,  -164, -294, -157,
+      -248, -263,  -285, -134,  -218,  -267,  -295,  -223,  -237, -192, -163,
+      -155, -68,   86,   -74,   -67,   -51,   -46,   -49,   -45,  -144, -181,
+      -126, -51,   -106, -136,  -213,  -189,  -163,  -189,  -238, -230, -238,
+      -276, -288,  -300, -274,  -264,  -268,  -228,  -288,  -202, -201, -180,
+      -232, -217,  -238, -174,  -222,  -220,  -242,  -187,  -203, -190, -211,
+      -143, -223,  -148, -187,  -110,  -202,  -179,  -175,  -124, -140, -207,
+      -165, -148,  -117, -138,  -129,  -91,   -111,  -52,   -96,  -31,  -72,
+      -57,  -38,   -41,  -39,   -63,   -12,   -65,   -2,    -67,  12,   -64,
+      37,   -48,   34,   -49,   20,    -4,    26,    -16,   4,    -9,   34,
+      -12,  34,    -23},
+     /* IRC_Composite_C_R0195_T030_P045.wav */
+     {53,    -54,  56,    -58,   59,    -62,   64,    -67,   70,   -73,  77,
+      -81,   87,   -93,   100,   -110,  122,   -140,  176,   -592, -266, -620,
+      -93,   -807, 111,   -1033, -244,  -1451, 661,   -1378, 1038, 309,  3279,
+      -1940, 4376, -7187, -1452, -4658, 38,    22380, 4913,  3629, 1106, -370,
+      -1337, 2958, 848,   -390,  -2691, 1295,  -1155, 2599,  -36,  956,  635,
+      787,   53,   821,   200,   581,   67,    983,   274,   565,  247,  646,
+      211,   751,  653,   273,   546,   759,   707,   776,   230,  704,  98,
+      258,   -440, 41,    -44,   -99,   -570,  -276,  125,   -89,  -316, -287,
+      -67,   -256, -409,  -175,  -243,  -300,  -369,  -371,  -102, -291, -350,
+      -382,  -109, -199,  -343,  -360,  -210,  -168,  -316,  -283, -194, -249,
+      -97,   -252, -166,  -374,  -161,  -245,  -317,  -297,  -364, -202, -337,
+      -267,  -324, -305,  -165,  -347,  -151,  -302,  -171,  -371, -213, -207,
+      -334,  -383, -313,  -191,  -297,  -398,  -235,  -273,  -140, -391, -214,
+      -292,  -119, -316,  -254,  -221,  -194,  -227,  -232,  -269, -229, -362,
+      -187,  -364, -206,  -335,  -144,  -247,  -146,  -237,  -158, -174, -97,
+      -117,  50,   -80,   66,    -91,   178,   -80,   -68,   -202, -96,  -198,
+      -132,  -223, -177,  -186,  -153,  -217,  -140,  -249,  -182, -298, -91,
+      -285,  -230, -334,  -153,  -250,  -218,  -295,  -154,  -260, -140, -294,
+      -141,  -233, -87,   -273,  -133,  -222,  -68,   -214,  -169, -225, -107,
+      -209,  -114, -264,  -92,   -242,  -49,   -295,  -99,   -226, -66,  -252,
+      -85,   -193, -65,   -189,  -57,   -171,  -14,   -172,  -4,   -179, 85,
+      -157,  3,    -134,  92,    -113,  -4,    -109,  69,    -51,  13,   -79,
+      68,    -27,  61,    -58,   66,    -43,   79,    -23,   54,   -24,  30,
+      -27,   66,   -41},
+     /* IRC_Composite_C_R0195_T045_P045.wav */
+     {-71,  74,   -76,   79,   -82,  84,   -87,  90,   -93,  96,   -99,   101,
+      -102, 102,  -97,   85,   -52,  -97,  -607, -227, -812, 438,  -1024, -204,
+      -760, 95,   -1636, 140,  -766, 2279, -763, 3056, 150,  2120, -5760, -6480,
+      1711, -14,  24602, 5276, 1036, -813, 1031, 1166, 848,  674,  -478,  -1487,
+      -945, 687,  1855,  -292, 799,  1002, 786,  207,  559,  505,  363,   364,
+      343,  530,  502,   11,   500,  785,  837,  217,  463,  907,  599,   564,
+      357,  380,  -197,  84,   -367, 169,  -530, -810, -282, -180, -171,  -519,
+      -179, -442, -72,   -160, -239, -494, -294, -88,  -553, -268, -512,  -278,
+      -503, -198, -473,  -229, -317, -213, -370, -187, -331, -249, -344,  -154,
+      -381, -135, -317,  -147, -279, -176, -355, -133, -274, -305, -330,  -235,
+      -314, -312, -303,  -290, -257, -159, -269, -234, -243, -173, -350,  -204,
+      -383, -239, -309,  -233, -404, -183, -289, -321, -344, -212, -319,  -251,
+      -253, -168, -303,  -204, -277, -189, -284, -216, -333, -204, -348,  -208,
+      -298, -166, -339,  -189, -186, -273, -152, -87,  -184, -166, 10,    48,
+      -40,  2,    0,     -18,  -97,  -3,   -134, -124, -166, -216, -161,  -149,
+      -234, -162, -175,  -239, -198, -150, -294, -241, -224, -190, -303,  -189,
+      -150, -236, -232,  -162, -159, -199, -136, -231, -166, -169, -136,  -155,
+      -152, -125, -146,  -133, -100, -123, -156, -121, -135, -145, -164,  -174,
+      -157, -151, -189,  -118, -198, -112, -121, -108, -138, -101, -93,   -63,
+      -79,  -82,  -83,   -26,  -44,  -58,  -96,  20,   -91,  -15,  -43,   22,
+      -69,  29,   5,     6,    -19,  24,   3,    1,    25,   26,   26,    17,
+      31,   13,   22,    22},
+     /* IRC_Composite_C_R0195_T060_P045.wav */
+     {59,    -58,   56,    -55,   52,    -49,  45,   -40,   34,   -25,  14,
+      1,     -23,   55,    -108,  214,   -619, -300, -123,  -537, -172, -393,
+      -415,  -586,  -104,  -1306, 69,    -335, 1375, 535,   2761, -686, 3824,
+      -8072, -1237, -7735, 11382, 19138, 4865, 2658, -2047, 1916, 571,  2806,
+      -624,  -694,  -1625, 70,    232,   950,  -64,  1404,  -414, 1196, 34,
+      1403,  -261,  1089,  343,   834,   -188, 587,  -251,  1086, 518,  827,
+      135,   641,   494,   842,   -31,   161,  -360, -291,  -317, 171,  -783,
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+      -323,  -261,  -320,  -293,  -286,  -298, -318, -332,  -305, -269, -323,
+      -212,  -266,  -206,  -301,  -143,  -304, -236, -268,  -212, -344, -239,
+      -265,  -225,  -364,  -157,  -292,  -187, -206, -129,  -241, -186, -152,
+      -141,  -106,  -103,  -15,   -115,  56,   -68,  26,    -47,  -92,  -125,
+      -141,  -154,  -176,  -196,  -181,  -250, -197, -136,  -112, -240, -89,
+      -180,  -145,  -235,  -146,  -223,  -248, -178, -229,  -152, -219, -112,
+      -250,  -98,   -141,  -152,  -155,  -116, -137, -150,  -110, -137, -128,
+      -94,   -113,  -72,   -189,  -29,   -176, -93,  -125,  -129, -123, -190,
+      -65,   -169,  -92,   -169,  -68,   -159, -48,  -109,  -46,  -109, -25,
+      -81,   -31,   -72,   -27,   -71,   7,    -65,  -16,   -30,  -30,  -44,
+      30,    -39,   8,     -56,   53,    -36,  81,   -51,   97,   -64,  95,
+      -35,   60,    -30},
+     /* IRC_Composite_C_R0195_T075_P045.wav */
+     {7,     -7,    6,     -6,    5,    -5,   4,     -3,   2,    0,    -2,
+      5,     -9,    14,    -23,   45,   -729, -86,   -463, -18,  -765, 398,
+      -1336, 518,   -1262, -130,  -544, 1878, -781,  3861, -694, 2694, -2479,
+      -6040, -3168, 1816,  21907, 5475, 5383, -2274, 1082, 1645, 3196, -180,
+      -862,  -1506, -288,  -182,  870,  -91,  1493,  -388, 779,  238,  870,
+      64,    614,   1007,  850,   122,  95,   413,   1397, 214,  480,  297,
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+      -336,  -296,  -307,  -215,  -240, -202, -250,  -161, -199, -207, -315,
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+      -223,  -340,  -127,  -300,  -165, -343, -198,  -307, -246, -285, -207,
+      -293,  -254,  -293,  -183,  -246, -202, -252,  -100, -213, -78,  -138,
+      -60,   -197,  -164,  -160,  -109, -141, -50,   -142, 2,    -155, 50,
+      -176,  0,     -217,  -67,   -265, -39,  -241,  -180, -198, -196, -186,
+      -182,  -111,  -135,  -134,  -125, -176, -98,   -184, -100, -212, -114,
+      -228,  -113,  -95,   -140,  -143, -141, -78,   -133, -100, -67,  -178,
+      -75,   -161,  -2,    -319,  35,   -236, -7,    -269, 53,   -223, -47,
+      -143,  -20,   -108,  -80,   -80,  -54,  -96,   -67,  -96,  -24,  -164,
+      -15,   -108,  -24,   -125,  -8,   -47,  -26,   -57,  4,    -8,   -39,
+      -35,   32,    -69,   30,    -49,  58,   -82,   89,   -80,  107,  -80,
+      104,   -56,   112},
+     /* IRC_Composite_C_R0195_T090_P045.wav */
+     {82,    -84,   86,    -88,   90,    -91,   93,   -94,  95,   -95,   94,
+      -92,   87,    -78,   61,    -33,   -240,  -426, 56,   -683, 113,   -744,
+      373,   -1306, 284,   -1194, 1644,  -667,  2652, -291, 4245, -2423, -2014,
+      -4070, -4168, 13696, 10251, 10475, -3031, 1940, 668,  3850, 1809,  -625,
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+      -134,  -87,   -100,  -96,   -67,   -104,  -58,  -178, -114, -154,  -134,
+      -226,  -145,  -224,  -215,  -130,  -174,  -117, -174, -71,  -88,   -63,
+      -109,  -58,   -78,   -120,  -50,   -84,   -79,  -101, -63,  -97,   -110,
+      -65,   -129,  -123,  -160,  -83,   -162,  -109, -152, -92,  -143,  -76,
+      -108,  -73,   -110,  -42,   -74,   -64,   -99,  -35,  -69,  -11,   -79,
+      -57,   -78,   -51,   -41,   -69,   -60,   -68,  -23,  -64,  48,    -114,
+      27,    -96,   75,    -109,  79,    -84,   99,   -121, 109,  -64,   69,
+      -52,   115,   -98},
+     /* IRC_Composite_C_R0195_T105_P045.wav */
+     {28,    -30,   31,    -33,   35,    -38,  40,    -43,  47,   -51,  55,
+      -60,   67,    -75,   85,    -99,   111,  -72,   -376, -88,  -540, 307,
+      -524,  -278,  -931,  669,   115,   481,  924,   2601, 65,   2397, -5614,
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+      -73,   -136,  -26,   -65,   -11,   -114, 16,    -69,  41,   -165, 27,
+      -83,   -32,   -83,   -83,   -89,   -127, -138,  -85,  -115, -105, -151,
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+      -31,   -63,   -14,   -64,   -36,   8,    -66,   36,   -43,  73,   -62,
+      61,    -61,   80},
+     /* IRC_Composite_C_R0195_T120_P045.wav */
+     {-12,   12,    -13,   13,    -14,   14,   -15,   16,   -17,  18,   -20,
+      22,    -25,   29,    -34,   41,    -54,  124,   -315, 231,  -915, 409,
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+      41,    -66,   -14,   -65,   -25,   -99,  -66,   -90,  -83,  -119, -73,
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+      -76,   -126,  -21,   -94,   -45,   -92,  -51,   -33,  3,    -63,  -11,
+      -52,   -9,    -73,   9,     -41,   -20,  -40,   3,    -44,  -34,  0,
+      0,     -32,   16},
+     /* IRC_Composite_C_R0195_T135_P045.wav */
+     {15,    -16,   17,    -18,  19,   -20,  21,    -22,   23,   -24,  25,
+      -27,   28,    -29,   29,   -28,  24,   -11,   -117,  -145, -90,  -121,
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+      -170,  -167,  -137,  -130, -154, -126, -141,  -104,  -142, -91,  -116,
+      -80,   -71,   -71,   -88,  -42,  -93,  -59,   -74,   -48,  -70,  -66,
+      -85,   -79,   -76,   -90,  -60,  -94,  -51,   -65,   -11,  -46,  -25,
+      -16,   -13,   -19,   -24,  -51,  -74,  -67,   -91,   -21,  -96,  -77,
+      -66,   2,     -73,   -36,  -56,  -44,  -134,  -42,   -35,  -70,  -118,
+      -47,   -68,   -116,  -72,  -82,  -75,  -136,  -24,   -85,  -36,  -90,
+      1,     -62,   -4,    -20,  -33,  -14,  1,     -17,   -33,  -28,  -35,
+      -37,   -9,    -16},
+     /* IRC_Composite_C_R0195_T150_P045.wav */
+     {13,    -14,  16,   -17,  19,   -20,  22,   -25,  27,   -30,  34,   -38,
+      42,    -48,  55,   -63,  75,   -91,  119,  -208, 14,   -179, 43,   -143,
+      14,    -237, 127,  -445, 347,  318,  882,  1092, 671,  2337, -2,   -2406,
+      -2033, 3403, 2686, 9080, 6065, 1654, -385, 2378, 3537, 1656, -964, -2514,
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+      788,   738,  691,  -486, -503, -169, -52,  -560, -771, -513, -275, -418,
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+      6,     -88,  -13,  -82,  -34,  -41,  -4,   -34,  -1,   -14,  9,    -50,
+      -23,   -69,  -16,  -101, -10,  -110, -17,  -119, -37,  -74,  -19,  -102,
+      -24,   -41,  -56,  -111, -39,  -111, -124, -147, -53,  -65,  -109, -46,
+      -41,   -68,  -52,  -7,   -29,  -64,  -43,  -18,  -36,  -49,  -47,  1,
+      -26,   -14,  -36,  -17},
+     /* IRC_Composite_C_R0195_T165_P045.wav */
+     {-5,    6,     -6,   7,    -7,   8,    -9,   9,    -10,  11,   -13,  14,
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+      -63,   -45,   -61,  -77},
+     /* IRC_Composite_C_R0195_T180_P045.wav */
+     {1,    -1,   1,     -1,    2,     -2,   2,    -2,   2,    -2,   2,    -2,
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+      -44,  -71,  -84,   -70},
+     /* IRC_Composite_C_R0195_T195_P045.wav */
+     {-10,  11,   -11,  11,   -12,  12,    -12,   13,   -13,  14,   -14,  15,
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+      -55,  -52,  -44,  -57},
+     /* IRC_Composite_C_R0195_T210_P045.wav */
+     {-1,   1,    -1,   1,    -1,   1,    -1,   1,    -2,    2,     -2,   2,
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+      -48,  -24,  -31,  -11},
+     /* IRC_Composite_C_R0195_T225_P045.wav */
+     {-3,   3,    -3,   3,    -3,   3,    -3,   4,    -4,   4,    -4,   4,
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+      -83,  -58,  -72,  -30},
+     /* IRC_Composite_C_R0195_T240_P045.wav */
+     {0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
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+      -88,  -79,  -72,  -47},
+     /* IRC_Composite_C_R0195_T255_P045.wav */
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+      -130, -91,  -134, -118, -138, -78,  -131, -110, -118, -103, -103, -97,
+      -94,  -71,  -62,  -45},
+     /* IRC_Composite_C_R0195_T270_P045.wav */
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+      -70,  -77,  -68,  -59},
+     /* IRC_Composite_C_R0195_T285_P045.wav */
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+      44,   18,   -5,   28,   17,   14,   19,   9,    27,   26,   56,   44,
+      54,   64,   65,   88},
+     /* IRC_Composite_C_R0195_T300_P045.wav */
+     {0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
+      0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,    0,
+      0,    0,    0,    0,    0,    1,    -2,   6,    -170, -102, -259, -234,
+      -199, -268, -51,  -528, -137, -354, -180, 92,   -180, 340,  604,  11,
+      -371, -583, -984, 10,   2699, 3142, 3080, 1412, 327,  1987, 956,  830,
+      764,  347,  196,  382,  1091, 294,  859,  889,  945,  523,  1199, 597,
+      582,  854,  625,  694,  469,  918,  446,  675,  763,  646,  482,  552,
+      534,  540,  563,  394,  300,  434,  396,  321,  144,  359,  -94,  142,
+      -43,  165,  -138, 6,    -183, -179, -221, -161, -328, -285, -267, -402,
+      -333, -327, -294, -327, -363, -296, -325, -259, -291, -320, -304, -304,
+      -298, -291, -297, -314, -280, -329, -297, -314, -281, -234, -352, -241,
+      -283, -217, -315, -280, -263, -279, -310, -268, -299, -296, -261, -277,
+      -280, -277, -225, -216, -261, -203, -214, -192, -239, -227, -233, -213,
+      -266, -208, -195, -200, -197, -191, -188, -186, -205, -150, -205, -153,
+      -112, -151, -87,  -35,  -120, -124, -179, -137, -157, -174, -117, -144,
+      -171, -153, -152, -138, -131, -174, -135, -128, -150, -114, -214, -174,
+      -206, -228, -215, -210, -194, -196, -208, -219, -189, -187, -156, -164,
+      -222, -168, -262, -148, -197, -176, -237, -214, -212, -100, -129, -146,
+      -213, -165, -123, -124, -133, -162, -159, -151, -116, -118, -122, -155,
+      -138, -137, -123, -99,  -108, -104, -101, -84,  -78,  -75,  -64,  -83,
+      -84,  -92,  -41,  -94,  -60,  -107, -53,  -86,  -49,  -66,  -60,  -68,
+      -31,  -65,  -31,  -29},
+     /* IRC_Composite_C_R0195_T315_P045.wav */
+     {-1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,    1,
+      -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,   1,    -1,    1,
+      -1,   2,    -2,   4,    -7,   17,   -100, -204, -340, -371, -160,  -344,
+      -275, -415, -419, -249, 41,   -182, 77,   891,  131,  2,    -1052, -1535,
+      924,  162,  4003, 4715, 1544, 1179, 621,  1424, 674,  1203, -197,  55,
+      273,  545,  402,  710,  753,  673,  897,  715,  647,  832,  758,   686,
+      732,  814,  620,  688,  779,  690,  604,  649,  586,  620,  586,   617,
+      478,  512,  430,  467,  385,  389,  275,  162,  250,  131,  159,   97,
+      100,  -42,  -92,  -107, -156, -193, -270, -280, -274, -295, -312,  -336,
+      -301, -367, -310, -306, -332, -327, -275, -299, -308, -302, -264,  -320,
+      -240, -345, -250, -325, -245, -353, -231, -309, -234, -308, -240,  -335,
+      -236, -284, -266, -330, -242, -316, -259, -281, -275, -309, -222,  -254,
+      -279, -277, -236, -254, -242, -235, -247, -199, -188, -240, -203,  -141,
+      -139, -214, -131, -200, -170, -111, -70,  -193, -145, -148, -101,  -173,
+      -117, -218, -163, -214, -197, -208, -111, -206, -113, -191, -104,  -100,
+      -70,  -142, -89,  -169, -106, -184, -156, -187, -185, -255, -195,  -151,
+      -192, -223, -180, -149, -215, -245, -216, -248, -232, -230, -274,  -292,
+      -203, -223, -171, -218, -208, -215, -141, -139, -194, -204, -163,  -184,
+      -155, -159, -154, -164, -134, -151, -130, -134, -130, -152, -122,  -129,
+      -133, -120, -106, -118, -102, -72,  -84,  -69,  -84,  -65,  -68,   -55,
+      -55,  -56,  -72,  -45,  -57,  -43,  -50,  -64,  -34,  -44,  -40,   -36,
+      -22,  -14,  -32,  2},
+     /* IRC_Composite_C_R0195_T330_P045.wav */
+     {0,    0,    0,    0,    0,    0,    0,    0,     0,     0,    0,    0,
+      0,    0,    1,    -1,   1,    -1,   1,    -1,    1,     -2,   2,    -2,
+      3,    -4,   5,    -8,   -246, -348, -392, -294,  -392,  -112, -819, -204,
+      -558, 205,  -381, 275,  768,  476,  319,  -1186, -2693, 1397, 173,  2904,
+      7126, 2191, 1840, -141, 1440, 643,  1148, 280,   -721,  280,  227,  474,
+      413,  704,  690,  753,  889,  309,  920,  476,   802,   503,  682,  628,
+      623,  869,  602,  783,  626,  695,  707,  691,   850,   635,  757,  447,
+      558,  418,  394,  227,  133,  146,  217,  152,   239,   143,  20,   23,
+      13,   -113, -130, -110, -235, -231, -231, -223,  -290,  -311, -279, -339,
+      -297, -335, -258, -358, -326, -332, -275, -283,  -293,  -287, -240, -279,
+      -296, -244, -248, -261, -257, -280, -251, -271,  -271,  -291, -272, -276,
+      -292, -293, -287, -273, -297, -261, -259, -291,  -282,  -238, -254, -285,
+      -269, -242, -239, -234, -237, -211, -232, -160,  -167,  -153, -54,  -121,
+      -133, -68,  -127, -55,  -93,  -170, -253, -307,  -209,  -255, -315, -307,
+      -329, -320, -102, -172, -106, -41,  -45,  45,    -125,  12,   -25,  -98,
+      -177, -185, -220, -166, -179, -213, -164, -210,  -152,  -240, -217, -183,
+      -300, -182, -230, -224, -316, -238, -276, -217,  -287,  -209, -290, -207,
+      -191, -176, -237, -198, -251, -191, -184, -201,  -245,  -147, -177, -164,
+      -190, -112, -183, -108, -171, -100, -163, -85,   -153,  -101, -137, -74,
+      -146, -57,  -120, -58,  -110, -43,  -90,  -50,   -94,   -38,  -79,  -64,
+      -64,  -54,  -43,  -59,  -47,  -30,  -23,  -30,   -38,   -10,  -18,  12,
+      -13,  10,   -24,  5},
+     /* IRC_Composite_C_R0195_T345_P045.wav */
+     {16,   -16,  16,   -16,  16,   -16,   16,   -16,   16,   -15,   15,   -14,
+      14,   -13,  11,   -9,   7,    -4,    0,    5,     -13,  25,    -45,  84,
+      -233, -338, -363, -485, -200, -447,  -145, -898,  -330, -612,  95,   -690,
+      294,  625,  1287, -151, 69,   -4722, 1521, -1345, 1556, 10396, 3052, 2742,
+      68,   1282, 287,  1583, 813,  -614,  -309, 335,   28,   768,   552,  558,
+      584,  902,  480,  734,  438,  738,   500,  576,   491,  546,   493,  681,
+      435,  611,  561,  752,  683,  864,   812,  738,   819,  537,   703,  581,
+      367,  218,  0,    254,  20,   344,   -127, 37,    -166, 211,   -28,  -62,
+      -54,  -160, -27,  -117, -166, -287,  -249, -103,  -237, -262,  -426, -232,
+      -222, -305, -355, -320, -241, -298,  -264, -311,  -247, -264,  -248, -253,
+      -288, -211, -240, -212, -321, -248,  -261, -248,  -306, -335,  -270, -295,
+      -235, -360, -279, -274, -198, -287,  -320, -253,  -257, -194,  -270, -236,
+      -243, -201, -255, -169, -230, -203,  -38,  -204,  -189, 64,    -47,  -32,
+      -96,  -24,  -149, -249, -211, -363,  -445, -322,  -199, -451,  -374, -212,
+      -206, -56,  -101, 52,   -41,  90,    144,  4,     -84,  38,    -111, -234,
+      -118, -217, -227, -246, -157, -274,  -235, -253,  -192, -341,  -200, -287,
+      -253, -258, -299, -242, -317, -247,  -273, -208,  -264, -246,  -273, -176,
+      -209, -205, -204, -238, -186, -184,  -158, -218,  -216, -193,  -143, -174,
+      -141, -179, -144, -152, -78,  -161,  -124, -122,  -54,  -126,  -102, -95,
+      -62,  -64,  -102, -54,  -75,  -51,   -32,  -78,   -64,  -53,   -39,  -81,
+      -33,  -75,  -14,  -32,  -27,  -68,   2,    -35,   2,    -22,   5,    -24,
+      24,   -12,  -6,   0}};
+
+const int16_t irc_composite_c_r0195_p060[][256] =
+    {/* IRC_Composite_C_R0195_T000_P060.wav */
+     {6,    -6,   6,    -6,   6,    -6,    6,     -6,    6,    -6,    6,
+      -6,   6,    -6,   5,    -5,   5,     -6,    7,     -9,   18,    -62,
+      -477, -304, -336, -458, -335, -491,  -402,  -829,  -409, -707,  -293,
+      -66,  983,  571,  504,  539,  -3665, -1218, -4247, 2405, 12687, 3792,
+      3387, 222,  562,  745,  2239, 1323,  -741,  -286,  -83,  813,   783,
+      665,  -6,   1300, 432,  669,  373,   824,   513,   741,  332,   740,
+      326,  679,  126,  625,  66,   777,   213,   737,   590,  515,   621,
+      819,  662,  597,  612,  571,  221,   378,   180,   456,  -17,   193,
+      -19,  201,  -103, 188,  -114, -50,   -223,  41,    -272, -7,    -204,
+      -73,  -274, -27,  -179, -118, -266,  -52,   -236,  -150, -298,  -185,
+      -328, -211, -330, -359, -321, -288,  -288,  -390,  -233, -301,  -238,
+      -287, -186, -257, -179, -231, -185,  -249,  -207,  -244, -242,  -257,
+      -261, -239, -326, -213, -312, -252,  -311,  -214,  -308, -230,  -270,
+      -210, -269, -227, -231, -256, -281,  -231,  -249,  -231, -210,  -238,
+      -174, -214, -204, -218, -156, -225,  -179,  -158,  -114, -151,  -171,
+      -128, -150, -184, -87,  -97,  -145,  -171,  -201,  -199, -179,  -259,
+      -239, -309, -310, -314, -227, -196,  -170,  -192,  -26,  -115,  43,
+      -61,  74,   -180, -52,  -139, -83,   -192,  -18,   -187, -104,  -111,
+      -67,  -175, -118, -149, -164, -160,  -118,  -201,  -219, -161,  -187,
+      -241, -189, -199, -195, -205, -163,  -179,  -154,  -122, -150,  -171,
+      -167, -152, -190, -183, -192, -193,  -159,  -180,  -150, -160,  -87,
+      -151, -131, -124, -50,  -117, -92,   -81,   -58,   -70,  -16,   -38,
+      -34,  -12,  39,   -10,  47,   38,    87,    17,    83,   31,    82,
+      15,   52,   1},
+     /* IRC_Composite_C_R0195_T030_P060.wav */
+     {20,   -21,  22,   -23,   24,    -25,   26,   -28,   29,   -31,  33,
+      -35,  37,   -40,  44,    -49,   57,    -69,  99,    -268, -386, -296,
+      -341, -306, -439, -461,  -219,  -926,  -450, -515,  -185, -151, 2064,
+      -248, 2208, -557, -2551, -2664, -6482, 6173, 13499, 9000, -215, 137,
+      1397, 897,  1847, 1331,  -989,  -1152, -31,  458,   894,  968,  314,
+      922,  758,  306,  312,   862,   265,   458,  604,   466,  114,  555,
+      424,  349,  321,  475,   497,   583,   497,  442,   464,  481,  505,
+      592,  441,  239,  76,    20,    163,   45,   -273,  -262, -336, -166,
+      -64,  -152, -189, -166,  -140,  -223,  -230, -243,  -348, -277, -328,
+      -243, -252, -223, -247,  -223,  -225,  -225, -211,  -310, -229, -259,
+      -253, -320, -259, -356,  -330,  -309,  -289, -302,  -271, -256, -257,
+      -245, -269, -281, -283,  -267,  -256,  -268, -267,  -280, -263, -261,
+      -261, -284, -289, -275,  -272,  -247,  -274, -279,  -271, -276, -305,
+      -292, -268, -282, -266,  -248,  -248,  -229, -247,  -221, -202, -217,
+      -233, -213, -204, -239,  -209,  -216,  -194, -227,  -155, -201, -185,
+      -243, -159, -264, -210,  -189,  -201,  -282, -175,  -151, -178, -186,
+      -73,  -151, -201, -182,  -71,   -254,  -132, -94,   -52,  -235, -22,
+      -27,  -101, -159, -17,   -130,  -166,  -55,  -23,   -206, -132, -126,
+      -159, -211, -63,  -187,  -189,  -148,  -84,  -174,  -142, -124, -126,
+      -185, -149, -155, -156,  -187,  -149,  -205, -136,  -153, -105, -164,
+      -138, -137, -75,  -158,  -128,  -135,  -115, -148,  -94,  -125, -133,
+      -123, -51,  -120, -102,  -85,   -18,   -84,  -52,   -42,  8,    -51,
+      10,   0,    9,    -20,   40,    16,    14,   24,    69,   21,   25,
+      23,   33,   2},
+     /* IRC_Composite_C_R0195_T060_P060.wav */
+     {49,   -49,  50,    -50,   49,    -49,   48,    -47,  45,   -43,   39,
+      -34,  27,   -17,   1,     24,    -67,   158,   -573, 27,   -273,  -533,
+      -199, -358, -57,   -1237, 277,   -801,  -585,  81,   1678, -52,   2584,
+      -635, 2281, -6550, -1907, -4733, 12287, 16138, 2134, 560,  -1141, 3390,
+      1047, 2071, -850,  -288,  -1730, 541,   499,   654,  282,  1081,  13,
+      1053, -59,  982,   296,   639,   439,   387,   97,   735,  -102,  457,
+      298,  541,  229,   951,   144,   642,   -123,  725,  239,  371,   -407,
+      173,  -217, 5,     -418,  64,    -709,  -238,  -708, -195, -402,  -221,
+      -414, -199, -238,  -175,  -279,  -243,  -447,  -227, -531, -270,  -429,
+      -178, -455, -173,  -350,  -170,  -312,  -238,  -284, -268, -355,  -248,
+      -402, -300, -358,  -251,  -472,  -244,  -350,  -274, -353, -195,  -309,
+      -279, -276, -243,  -232,  -248,  -185,  -198,  -344, -177, -263,  -274,
+      -355, -198, -341,  -222,  -358,  -178,  -344,  -238, -289, -271,  -338,
+      -286, -218, -316,  -206,  -256,  -230,  -264,  -214, -239, -277,  -261,
+      -259, -225, -268,  -221,  -134,  -246,  -182,  -192, -154, -260,  -116,
+      -218, -234, -234,  -227,  -254,  -254,  -224,  -164, -220, -284,  -56,
+      -257, -45,  -289,  -17,   -299,  22,    -225,  103,  -264, -36,   -33,
+      -158, -37,  -147,  39,    -238,  -62,   -117,  -69,  -262, 42,    -153,
+      -162, -86,  -22,   -214,  -31,   -193,  -19,   -225, -87,  -128,  -143,
+      -84,  -102, -114,  -95,   -106,  -146,  -143,  -5,   -308, -86,   -140,
+      -102, -254, 41,    -68,   -208,  37,    -26,   -47,  -124, 31,    -99,
+      -131, -35,  -104,  -59,   -141,  32,    -115,  3,    -48,  1,     -54,
+      32,   -67,  -4,    -31,   -40,   -50,   45,    -30,  -17,  43,    3,
+      8,    10,   37},
+     /* IRC_Composite_C_R0195_T090_P060.wav */
+     {20,   -22,   23,   -25,   26,   -28,   30,   -33,  35,    -38,  41,
+      -45,  49,    -54,  60,    -66,  75,    -87,  -10,  -413,  -67,  -233,
+      -277, -289,  -247, -282,  -617, -150,  -183, 967,  613,   1964, 132,
+      3132, -4890, -30,  -7492, 6867, 13211, 9263, 1898, -2424, 3417, 1010,
+      3400, -250,  -877, -1856, 320,  -121,  560,  520,  754,   353,  491,
+      -65,  931,   362,  494,   135,  792,   604,  460,  -39,   664,  258,
+      476,  82,    671,  151,   141,  64,    562,  -256, -248,  -550, -170,
+      -357, -396,  -602, -158,  -390, -415,  -610, -298, -495,  -302, -712,
+      -205, -411,  -237, -411,  -197, -452,  -250, -501, -239,  -421, -284,
+      -481, -170,  -438, -181,  -420, -184,  -419, -262, -417,  -218, -376,
+      -336, -394,  -333, -356,  -335, -390,  -304, -355, -247,  -352, -237,
+      -395, -250,  -384, -156,  -227, -147,  -266, -148, -197,  -207, -288,
+      -280, -286,  -312, -317,  -266, -316,  -238, -306, -223,  -320, -187,
+      -342, -177,  -339, -198,  -318, -212,  -319, -219, -304,  -241, -273,
+      -207, -279,  -190, -248,  -107, -255,  -126, -251, -112,  -272, -95,
+      -249, -163,  -250, -133,  -247, -177,  -254, -198, -235,  -194, -229,
+      -212, -194,  -191, -142,  -191, -92,   -153, -45,  -145,  -65,  -138,
+      -64,  -93,   -82,  -65,   -113, -49,   -140, -78,  -118,  -32,  -137,
+      -37,  -107,  -30,  -59,   -142, -106,  -132, -11,  -240,  49,   -159,
+      20,   -179,  55,   -117,  -17,  -81,   -27,  -68,  -101,  -72,  -98,
+      -48,  -107,  -136, -19,   -93,  -74,   -119, 94,   -211,  28,   -129,
+      49,   -157,  78,   -96,   21,   -143,  105,  -58,  -15,   -28,  13,
+      -42,  14,    -40,  -16,   -41,  -38,   -39,  21,   -40,   -15,  -19,
+      -23,  -62,   3},
+     /* IRC_Composite_C_R0195_T120_P060.wav */
+     {-11,  11,   -11,   11,    -11,  10,   -10,   10,   -9,   9,     -8,
+      7,    -6,   4,     -1,    -4,   11,   -23,   54,   -588, 228,   -347,
+      10,   -393, 78,    -279,  -133, -619, 373,   107,  880,  1033,  1066,
+      2024, -241, -1833, -4036, 1812, 3742, 13338, 5090, 1325, -1146, 2733,
+      2814, 1516, -118,  -2246, -488, 96,   262,   192,  815,  13,    476,
+      529,  495,  181,   475,   265,  780,  210,   341,  81,   498,   784,
+      541,  47,   -30,   -8,    36,   -29,  -17,   -592, -609, -210,  -251,
+      -604, -616, -485,  -190,  -407, -234, -662,  -290, -550, -301,  -639,
+      -361, -458, -250,  -462,  -279, -431, -205,  -390, -287, -423,  -271,
+      -396, -265, -443,  -263,  -432, -176, -477,  -278, -447, -188,  -475,
+      -218, -453, -216,  -462,  -223, -420, -282,  -431, -248, -385,  -293,
+      -403, -276, -425,  -204,  -367, -74,  -400,  -116, -375, -65,   -372,
+      -120, -338, -171,  -350,  -192, -321, -226,  -351, -196, -350,  -234,
+      -344, -164, -341,  -158,  -306, -138, -338,  -156, -268, -116,  -265,
+      -176, -243, -204,  -212,  -198, -229, -226,  -192, -144, -137,  -153,
+      -181, -152, -194,  -135,  -178, -195, -187,  -227, -175, -208,  -170,
+      -244, -159, -254,  -137,  -245, -163, -226,  -152, -162, -107,  -156,
+      -107, -123, -75,   -95,   -2,   -107, -23,   -62,  -12,  -74,   -60,
+      -54,  -88,  -41,   -86,   -11,  -107, -13,   -74,  -29,  -98,   -23,
+      -52,  -89,  -64,   -61,   -71,  -89,  -73,   19,   -96,  13,    -86,
+      41,   -106, 92,    -5,    1,    -25,  -7,    -37,  -56,  34,    -91,
+      -24,  -86,  10,    -49,   -33,  -31,  -56,   10,   -83,  56,    -62,
+      45,   -72,  47,    -53,   7,    -38,  12,    -40,  -11,  -4,    -41,
+      -34,  -33,  -33},
+     /* IRC_Composite_C_R0195_T150_P060.wav */
+     {-16,   16,    -17,   17,   -18,  18,   -19,  20,   -20,  21,   -21,  22,
+      -22,   23,    -23,   23,   -23,  22,   -20,  15,   -1,   -129, 61,   -83,
+      -118,  -17,   -51,   -30,  -188, -95,  109,  510,  730,  1005, 940,  2137,
+      -1039, -410,  -2971, 1165, 5406, 9365, 3937, 2137, -649, 991,  4216, 933,
+      -931,  -1885, 270,   39,   471,  -2,   92,   175,  614,  97,   357,  338,
+      829,   636,   315,   -32,  510,  435,  434,  61,   257,  -175, -293, -260,
+      -101,  -474,  -492,  -360, -265, -315, -367, -291, -300, -567, -400, -366,
+      -215,  -495,  -451,  -522, -369, -523, -275, -402, -310, -403, -263, -346,
+      -267,  -319,  -256,  -370, -300, -367, -298, -401, -295, -443, -304, -417,
+      -322,  -444,  -298,  -381, -312, -439, -329, -390, -317, -384, -301, -362,
+      -302,  -383,  -295,  -368, -322, -392, -272, -324, -246, -326, -211, -303,
+      -225,  -323,  -240,  -326, -214, -295, -235, -325, -259, -307, -231, -265,
+      -204,  -266,  -221,  -283, -193, -226, -150, -215, -156, -235, -140, -226,
+      -173,  -226,  -138,  -181, -139, -211, -149, -228, -146, -230, -122, -202,
+      -104,  -157,  -89,   -169, -122, -180, -141, -172, -119, -242, -182, -247,
+      -146,  -221,  -160,  -209, -160, -222, -134, -194, -158, -172, -105, -115,
+      -90,   -104,  -68,   -78,  -70,  -76,  -8,   -32,  19,   -27,  16,   -5,
+      -14,   -32,   0,     -60,  -11,  -71,  10,   -79,  -60,  -81,  -34,  -99,
+      -28,   -10,   -16,   0,    23,   9,    -66,  -19,  -7,   -4,   -45,  41,
+      7,     -5,    -25,   -4,   7,    -31,  -3,   22,   -19,  -42,  6,    -62,
+      -7,    -36,   56,    -32,  47,   -21,  39,   -84,  -6,   -47,  -14,  -45,
+      33,    -19,   -3,    -57},
+     /* IRC_Composite_C_R0195_T180_P060.wav */
+     {10,   -10,  10,   -11,  11,    -10,   10,   -10,  10,   -10,  10,   -10,
+      9,    -9,   8,    -7,   6,     -5,    3,    -1,   -2,   6,    -12,  19,
+      -22,  153,  -20,  -54,  110,   44,    10,   -80,  172,  36,   605,  639,
+      998,  719,  1758, -623, -249,  -1810, 1268, 4590, 6942, 3540, 1131, 148,
+      816,  4013, 533,  -755, -1438, 271,   -13,  605,  142,  55,   202,  291,
+      -131, 465,  665,  844,  217,   120,   213,  663,  92,   -57,  -100, 67,
+      -191, -270, -528, -381, -572,  -177,  -319, -6,   -278, -198, -211, 58,
+      -313, -463, -592, -248, -465,  -368,  -496, -344, -514, -270, -361, -242,
+      -341, -263, -349, -193, -335,  -278,  -416, -272, -416, -335, -439, -313,
+      -451, -348, -453, -349, -443,  -392,  -472, -363, -430, -386, -408, -312,
+      -392, -320, -379, -310, -370,  -331,  -403, -335, -385, -348, -366, -269,
+      -311, -276, -302, -234, -290,  -316,  -341, -275, -241, -217, -208, -227,
+      -228, -238, -222, -184, -224,  -208,  -241, -186, -232, -205, -245, -203,
+      -214, -163, -151, -128, -135,  -145,  -180, -187, -204, -171, -188, -145,
+      -158, -112, -157, -75,  -116,  -69,   -149, -129, -160, -136, -115, -133,
+      -152, -184, -177, -167, -165,  -197,  -218, -169, -194, -138, -205, -133,
+      -162, -112, -132, -87,  -117,  -57,   -79,  -38,  -55,  -39,  -50,  -29,
+      -58,  -18,  2,    13,   -3,    31,    17,   9,    8,    29,   -11,  -10,
+      -3,   19,   -49,  -35,  -14,   -39,   -21,  -64,  -32,  -53,  -24,  -15,
+      -14,  -23,  27,   17,   51,    -9,    -15,  -18,  18,   -37,  5,    -74,
+      -13,  -18,  63,   0,    46,    7,     63,   -15,  19,   -10,  7,    -30,
+      -32,  -44,  -79,  -107},
+     /* IRC_Composite_C_R0195_T210_P060.wav */
+     {-5,   5,    -5,   5,    -5,   5,    -5,   5,    -5,    5,    -5,   5,
+      -5,   5,    -5,   5,    -5,   5,    -5,   5,    -5,    5,    -4,   4,
+      -4,   4,    -4,   -36,  -70,  2,    -43,  -45,  -73,   17,   -83,  -76,
+      -100, 274,  192,  709,  499,  1049, 474,  192,  -1587, -214, 1449, 4357,
+      5043, 2092, 784,  976,  2790, 1402, 1131, -633, 70,    310,  385,  217,
+      300,  87,   393,  309,  640,  644,  573,  357,  469,   403,  414,  393,
+      291,  95,   9,    177,  41,   -38,  -147, -145, -178,  -91,  51,   -23,
+      82,   6,    81,   -66,  -34,  -205, -207, -298, -223,  -348, -275, -305,
+      -257, -377, -308, -322, -230, -324, -269, -300, -305,  -334, -286, -346,
+      -334, -384, -351, -411, -405, -389, -413, -415, -387,  -382, -401, -384,
+      -393, -405, -388, -417, -364, -399, -357, -398, -374,  -392, -376, -343,
+      -359, -316, -314, -243, -324, -280, -326, -283, -293,  -262, -262, -275,
+      -249, -231, -206, -195, -215, -224, -241, -210, -220,  -242, -276, -255,
+      -264, -198, -192, -190, -189, -210, -164, -172, -169,  -172, -189, -153,
+      -193, -127, -166, -117, -158, -112, -128, -133, -107,  -136, -117, -161,
+      -108, -110, -142, -127, -184, -138, -172, -147, -192,  -188, -193, -158,
+      -153, -183, -168, -189, -142, -164, -152, -151, -135,  -105, -91,  -78,
+      -46,  -70,  -9,   -22,  -10,  6,    -3,   11,   9,     0,    -18,  -34,
+      -49,  -47,  -78,  -56,  -80,  -88,  -84,  -41,  -76,   -89,  -89,  -34,
+      -62,  21,   -11,  29,   -63,  38,   13,   32,   -26,   -6,   -18,  -15,
+      -14,  -11,  -43,  -37,  -18,  40,   -10,  3,    -52,   3,    -65,  -18,
+      -66,  -33,  -83,  -53},
+     /* IRC_Composite_C_R0195_T240_P060.wav */
+     {-5,   5,    -5,   5,    -5,   6,    -6,   6,    -6,   6,    -6,   6,
+      -6,   6,    -6,   6,    -6,   6,    -6,   6,    -6,   7,    -7,   7,
+      -6,   6,    -6,   4,    1,    -68,  -141, -43,  -120, -98,  -71,  -122,
+      -111, -111, -115, -151, -61,  253,  299,  522,  567,  502,  242,  -1198,
+      -647, 857,  3187, 4265, 2111, 1152, 1068, 2588, 1288, 1124, 171,  189,
+      245,  613,  495,  306,  631,  410,  664,  513,  726,  614,  455,  361,
+      493,  538,  415,  305,  397,  398,  328,  162,  355,  127,  167,  95,
+      91,   92,   127,  159,  137,  126,  136,  112,  42,   -54,  -59,  -214,
+      -99,  -334, -210, -340, -229, -357, -264, -345, -232, -312, -253, -294,
+      -304, -352, -288, -347, -349, -364, -352, -373, -394, -376, -368, -434,
+      -376, -402, -393, -434, -406, -412, -357, -430, -351, -354, -374, -337,
+      -289, -323, -276, -287, -233, -304, -270, -311, -283, -321, -267, -321,
+      -256, -305, -231, -272, -202, -256, -217, -272, -229, -238, -234, -247,
+      -261, -261, -253, -218, -198, -218, -190, -226, -148, -209, -150, -201,
+      -174, -181, -174, -172, -158, -155, -141, -131, -122, -138, -117, -118,
+      -119, -130, -144, -143, -130, -146, -135, -155, -149, -172, -147, -158,
+      -167, -175, -169, -144, -176, -150, -108, -129, -129, -136, -93,  -143,
+      -97,  -99,  -87,  -71,  -74,  -15,  -38,  -48,  -75,  -66,  -95,  -101,
+      -91,  -93,  -103, -154, -122, -128, -116, -84,  -70,  -79,  -65,  -51,
+      -28,  -23,  -20,  -21,  -53,  -15,  -9,   15,   -49,  5,    -40,  -18,
+      -42,  -33,  -30,  -86,  -81,  -68,  -62,  -85,  -44,  -26,  -89,  -49,
+      -55,  -24,  -61,  -16},
+     /* IRC_Composite_C_R0195_T270_P060.wav */
+     {5,    -5,    6,    -6,   6,    -6,   6,    -6,   7,    -7,   7,    -7,
+      8,    -8,    8,    -9,   9,    -10,  10,   -11,  12,   -12,  13,   -15,
+      16,   -19,   22,   -27,  38,   -79,  -217, -116, -99,  -179, -211, -36,
+      -231, -204,  -130, -219, -231, -133, 80,   277,  396,  430,  429,  -88,
+      -768, -1080, 327,  3345, 4354, 1751, 849,  1821, 1217, 1828, 885,  517,
+      269,  311,   1017, 365,  820,  357,  716,  549,  590,  621,  516,  736,
+      326,  780,   263,  770,  281,  705,  378,  636,  371,  475,  311,  429,
+      255,  409,   238,  273,  140,  314,  213,  208,  138,  105,  115,  -23,
+      74,   -205,  -25,  -248, -155, -287, -280, -252, -297, -307, -288, -293,
+      -267, -333,  -245, -352, -304, -348, -348, -351, -433, -337, -451, -334,
+      -472, -324,  -423, -375, -381, -372, -314, -375, -276, -267, -321, -301,
+      -279, -273,  -291, -287, -302, -304, -304, -290, -236, -296, -289, -263,
+      -294, -284,  -324, -265, -281, -292, -245, -245, -266, -260, -214, -224,
+      -254, -200,  -260, -209, -252, -220, -200, -254, -196, -233, -158, -228,
+      -136, -185,  -148, -205, -143, -175, -148, -149, -166, -120, -148, -104,
+      -128, -140,  -125, -152, -101, -130, -77,  -199, -88,  -178, -108, -197,
+      -126, -198,  -117, -194, -113, -172, -133, -128, -87,  -131, -57,  -112,
+      -60,  -136,  -71,  -150, -125, -162, -106, -205, -139, -165, -138, -149,
+      -118, -134,  -84,  -134, -83,  -120, -64,  -109, -79,  -116, -53,  -90,
+      -64,  -49,   -97,  -69,  -54,  -86,  -38,  -63,  -120, -64,  -43,  -89,
+      -78,  -45,   -32,  -110, -9,   -49,  -30,  -91,  -36,  -87,  -34,  -90,
+      -12,  -62,   -20,  -67},
+     /* IRC_Composite_C_R0195_T300_P060.wav */
+     {5,    -5,   5,    -5,   6,    -6,   6,    -6,   6,    -6,   6,     -6,
+      6,    -7,   7,    -7,   7,    -7,   7,    -8,   8,    -8,   8,     -9,
+      9,    -9,   10,   -11,  16,   -103, -316, -221, -172, -255, -229,  -364,
+      -48,  -533, -236, -67,  -300, 259,  642,  185,  100,  427,  -1196, -1745,
+      476,  2467, 5309, 2589, 680,  1682, 925,  1569, 1041, 949,  -237,  234,
+      798,  421,  371,  569,  750,  604,  969,  544,  826,  529,  786,   297,
+      628,  582,  490,  514,  782,  585,  536,  518,  533,  446,  484,   585,
+      264,  478,  477,  367,  328,  417,  65,   219,  65,   246,  -91,   156,
+      -64,  72,   -138, 50,   -297, -48,  -254, -210, -356, -206, -256,  -287,
+      -317, -214, -367, -254, -373, -339, -401, -317, -430, -376, -368,  -350,
+      -418, -316, -298, -353, -273, -305, -281, -287, -227, -371, -293,  -252,
+      -310, -323, -260, -272, -324, -243, -257, -310, -290, -276, -297,  -255,
+      -201, -324, -280, -270, -230, -282, -221, -248, -298, -222, -268,  -241,
+      -260, -255, -277, -235, -187, -236, -210, -172, -224, -173, -145,  -191,
+      -157, -155, -157, -126, -186, -84,  -150, -177, -153, -72,  -128,  -133,
+      -127, -100, -229, -197, -160, -241, -236, -171, -176, -120, -75,   -47,
+      -120, -105, -128, -76,  -160, -120, -127, -109, -139, -135, -144,  -192,
+      -179, -152, -203, -173, -143, -161, -61,  -179, -143, -104, -126,  -147,
+      -78,  -176, -138, -177, -87,  -160, -110, -137, -186, -155, -86,   -135,
+      -127, -105, -97,  -124, -34,  -111, -50,  -117, -72,  -123, -75,   -65,
+      -94,  -51,  -83,  -75,  -56,  -47,  -45,  -13,  -39,  -55,  20,    -56,
+      36,   -41,  12,   -24},
+     /* IRC_Composite_C_R0195_T330_P060.wav */
+     {3,    -3,   4,    -4,   4,    -4,   5,     -5,   5,     -6,   6,    -6,
+      7,    -7,   8,    -9,   10,   -11,  13,    -15,  18,    -22,  28,   -41,
+      71,   -233, -373, -96,  -345, -318, -371,  -265, -252,  -594, -439, -312,
+      -197, 197,  323,  301,  1318, -757, -1616, -868, -2081, 2422, 6337, 5378,
+      1528, 381,  1195, 1007, 1545, 810,  158,   -165, 224,   315,  652,  555,
+      370,  728,  945,  600,  540,  730,  606,   560,  567,   567,  493,  662,
+      473,  574,  547,  639,  468,  638,  527,   625,  475,   584,  476,  643,
+      403,  465,  284,  372,  243,  235,  43,    108,  49,    102,  20,   -28,
+      -86,  -48,  -49,  -120, -149, -133, -123,  -173, -198,  -229, -246, -291,
+      -242, -300, -328, -373, -296, -357, -299,  -377, -351,  -356, -276, -311,
+      -324, -289, -282, -268, -271, -255, -256,  -273, -249,  -285, -230, -270,
+      -219, -299, -282, -269, -275, -287, -307,  -267, -290,  -268, -297, -284,
+      -263, -248, -257, -244, -243, -257, -290,  -273, -259,  -265, -237, -261,
+      -238, -273, -209, -234, -171, -171, -157,  -118, -120,  -137, -80,  -115,
+      -96,  -166, -45,  -140, -79,  -126, -114,  -259, -243,  -265, -228, -274,
+      -267, -264, -220, -203, -117, -137, -118,  -112, -33,   -82,  -41,  -93,
+      -102, -121, -120, -145, -175, -177, -147,  -143, -65,   -127, -138, -136,
+      -94,  -127, -130, -217, -183, -159, -168,  -202, -184,  -203, -240, -202,
+      -146, -195, -160, -147, -169, -143, -72,   -128, -191,  -139, -128, -170,
+      -166, -129, -160, -152, -97,  -112, -123,  -68,  -72,   -87,  -76,  -27,
+      -63,  -48,  -50,  -16,  -43,  16,   -15,   -10,  -1,    57,   47,   30,
+      49,   50,   35,   14}};
+
+const int16_t irc_composite_c_r0195_p075[][256] =
+    {/* IRC_Composite_C_R0195_T000_P075.wav */
+     {12,   -12,  13,   -13,   13,    -13,  13,   -13,  14,   -14,  14,   -14,
+      14,   -14,  15,   -15,   14,    -14,  13,   -12,  7,    8,    -224, -367,
+      -189, -595, -344, -165,  -801,  -581, -302, -836, -618, 407,  530,  134,
+      1455, -966, -265, -4329, -4113, 4525, 6992, 8542, 304,  -199, 2762, 1323,
+      1544, 1359, 195,  -1272, 815,   548,  454,  489,  1188, 303,  1217, 433,
+      606,  486,  940,  200,   865,   394,  415,  452,  439,  167,  687,  262,
+      388,  668,  618,  250,   776,   505,  518,  525,  594,  287,  552,  461,
+      358,  276,  304,  160,   239,   30,   35,   -21,  -62,  -125, -17,  -31,
+      -371, 39,   -211, -122,  -197,  40,   -391, 72,   -166, -93,  -262, -84,
+      -337, -76,  -325, -174,  -301,  -196, -378, -120, -367, -324, -293, -268,
+      -402, -208, -271, -377,  -322,  -212, -314, -262, -200, -245, -294, -186,
+      -222, -275, -226, -280,  -246,  -300, -262, -182, -254, -272, -206, -217,
+      -307, -182, -214, -248,  -250,  -184, -258, -303, -196, -229, -312, -240,
+      -213, -256, -268, -167,  -202,  -256, -145, -174, -141, -306, -39,  -235,
+      -192, -132, -161, -142,  -205,  -78,  -192, -93,  -268, -65,  -239, -166,
+      -229, -88,  -215, -227,  -134,  -262, -99,  -230, -94,  -297, -147, -247,
+      -15,  -245, -197, -51,   -171,  -94,  -46,  -60,  -97,  -14,  -100, 14,
+      -99,  -80,  -92,  -24,   -178,  55,   -112, -23,  -104, 47,   -121, -73,
+      -80,  -27,  -196, -98,   -74,   -188, -162, -54,  -110, -182, -133, -30,
+      -121, -83,  -72,  -116,  -145,  -93,  -89,  -136, -159, -163, -75,  -145,
+      -145, -85,  -61,  -103,  -107,  5,    -74,  -43,  -3,   -22,  -72,  -26,
+      -24,  6,    -85,  5},
+     /* IRC_Composite_C_R0195_T060_P075.wav */
+     {-14,  15,   -15,  16,   -17,   18,    -19,   20,   -21,   22,   -23,
+      24,   -25,  27,   -28,  30,    -32,   35,    -42,  65,    -281, -389,
+      123,  -583, -268, -119, -384,  -436,  -660,  282,  -1424, 931,  -191,
+      1870, -113, 1526, 243,  -2974, -2922, -3038, 6766, 14120, 3000, -177,
+      427,  2482, 2472, 607,  1128,  -2092, 683,   -982, 2038,  -839, 1456,
+      -30,  1050, 188,  720,  669,   391,   864,   257,  483,   214,  493,
+      238,  321,  378,  114,  592,   45,    886,   -9,   646,   103,  701,
+      13,   345,  57,   116,  121,   -88,   171,   -386, 97,    -335, -262,
+      -369, -373, -308, -409, 33,    -533,  5,     -484, 24,    -435, -237,
+      -319, -208, -233, -416, -146,  -317,  -166,  -309, -232,  -304, -340,
+      -158, -439, -169, -482, -113,  -521,  -200,  -450, -258,  -409, -296,
+      -298, -425, -296, -402, -280,  -434,  -185,  -317, -227,  -279, -194,
+      -241, -311, -232, -320, -314,  -306,  -233,  -329, -276,  -236, -245,
+      -297, -256, -187, -267, -262,  -210,  -246,  -287, -248,  -205, -331,
+      -278, -229, -232, -260, -191,  -174,  -217,  -208, -141,  -202, -243,
+      -151, -110, -284, -132, -132,  -134,  -277,  -115, -207,  -229, -254,
+      -153, -213, -329, -111, -236,  -248,  -245,  -82,  -281,  -102, -144,
+      -93,  -133, -52,  -82,  -139,  -23,   -121,  -40,  -135,  7,    -127,
+      -34,  -127, -9,   -67,  -119,  -37,   -34,   -134, 4,     -70,  -77,
+      -81,  -62,  -64,  -127, -18,   -109,  11,    -156, 60,    -224, 27,
+      -77,  -130, 6,    -101, -38,   -151,  93,    -175, -123,  0,    -32,
+      -129, -111, 115,  -229, -48,   12,    -46,   -196, -37,   -14,  -163,
+      -83,  -1,   -137, -48,  -101,  91,    -141,  15,   -60,   28,   -35,
+      -33,  -10,  -28},
+     /* IRC_Composite_C_R0195_T120_P075.wav */
+     {-13,  13,   -14,  14,   -15,  15,    -16,  16,    -17,  17,    -18,
+      18,   -19,  19,   -20,  20,   -19,   19,   -16,   12,   -1,    -57,
+      -48,  -189, -132, -75,  -227, -56,   -343, 28,    -501, 199,   201,
+      917,  964,  1130, 545,  1226, -3782, -981, -1166, 8252, 10379, 3341,
+      -521, -669, 4572, 428,  2096, -1386, -405, -1098, 1383, -2,    236,
+      525,  510,  124,  361,  340,  604,   403,  459,   20,   703,   550,
+      361,  4,    507,  76,   379,  200,   87,   66,    -36,  226,   133,
+      30,   -418, -183, -144, -119, -262,  -333, -294,  -550, -263,  -353,
+      -297, -518, -525, -481, -405, -172,  -382, -282,  -383, -195,  -329,
+      -205, -358, -340, -304, -296, -417,  -305, -321,  -319, -400,  -254,
+      -454, -283, -418, -294, -482, -262,  -427, -329,  -482, -271,  -419,
+      -319, -403, -345, -351, -373, -409,  -381, -310,  -297, -225,  -271,
+      -290, -302, -223, -252, -250, -249,  -220, -293,  -267, -259,  -270,
+      -267, -281, -238, -321, -208, -300,  -203, -357,  -197, -295,  -169,
+      -277, -177, -273, -210, -234, -196,  -252, -188,  -188, -144,  -221,
+      -121, -160, -150, -180, -111, -124,  -171, -110,  -188, -133,  -213,
+      -124, -220, -168, -203, -176, -182,  -229, -156,  -255, -117,  -221,
+      -99,  -240, -61,  -158, -122, -129,  -39,  -92,   -133, 4,     -74,
+      -53,  -77,  35,   -129, 8,    -50,   5,    -61,   -24,  -33,   -46,
+      31,   -56,  58,   -35,  39,   -37,   57,   -74,   2,    -108,  -10,
+      -136, 18,   -141, 8,    -114, 4,     -132, -22,   -37,  -31,   -20,
+      -71,  79,   -149, 14,   -32,  50,    -85,  -47,   94,   -59,   21,
+      -39,  82,   -145, 47,   -47,  -14,   -82,  -16,   -3,   -58,   -48,
+      -44,  -22,  -40},
+     /* IRC_Composite_C_R0195_T180_P075.wav */
+     {-17,  17,   -17,  17,   -16,  16,    -16,   16,   -15,  15,   -15,  14,
+      -14,  14,   -13,  12,   -12,  11,    -10,   9,    -7,   5,    -2,   -2,
+      11,   164,  -118, 156,  -74,  171,   -166,  281,  -259, 331,  -163, 647,
+      483,  1342, 96,   2125, -652, 369,   -2506, 1012, 3788, 6928, 5436, -854,
+      927,  1011, 2973, 419,  642,  -1377, -224,  371,  309,  -128, 78,   268,
+      228,  233,  434,  427,  397,  -91,   692,   541,  379,  -60,  222,  -23,
+      131,  178,  60,   -172, -56,  -82,   -145,  -311, -297, -431, -248, -412,
+      -107, -371, -72,  -353, -277, -448,  -332,  -408, -362, -461, -379, -530,
+      -388, -407, -290, -364, -334, -409,  -288,  -419, -285, -384, -338, -447,
+      -343, -386, -344, -453, -378, -390,  -377,  -502, -386, -429, -361, -457,
+      -343, -463, -441, -462, -348, -398,  -369,  -416, -375, -435, -312, -336,
+      -244, -309, -307, -321, -263, -272,  -272,  -262, -236, -196, -217, -232,
+      -215, -261, -261, -304, -256, -296,  -220,  -241, -218, -248, -199, -190,
+      -206, -213, -223, -195, -209, -182,  -150,  -161, -155, -165, -145, -155,
+      -129, -127, -102, -102, -126, -107,  -145,  -133, -143, -141, -131, -140,
+      -117, -119, -86,  -121, -100, -131,  -130,  -130, -139, -142, -162, -155,
+      -148, -128, -171, -136, -140, -117,  -101,  -76,  -112, -44,  -78,  -20,
+      -37,  -38,  -18,  15,   -6,   2,     39,    44,   55,   -20,  46,   41,
+      30,   8,    37,   20,   -14,  0,     -10,   0,    -57,  -38,  -55,  -62,
+      -113, -27,  -88,  -33,  -69,  -21,   -80,   1,    -86,  62,   -11,  16,
+      23,   128,  6,    102,  -12,  152,   3,     46,   29,   36,   -97,  2,
+      -15,  22,   -109, -54},
+     /* IRC_Composite_C_R0195_T240_P075.wav */
+     {3,    -3,   3,    -3,   3,    -3,   3,    -2,   2,     -2,    2,    -2,
+      1,    -1,   1,    0,    0,    1,    -2,   3,    -4,    5,     -7,   10,
+      -13,  20,   -42,  -111, -129, -71,  -115, -137, -61,   -159,  -123, -192,
+      -19,  -219, 406,  390,  680,  549,  886,  -353, -736,  -1471, 543,  4694,
+      6064, 1476, 785,  1081, 2341, 1848, 567,  793,  -1054, 878,   -19,  694,
+      -27,  624,  453,  574,  713,  561,  524,  289,  447,   341,   517,  436,
+      227,  504,  311,  411,  259,  396,  204,  319,  114,   96,    38,   -78,
+      39,   -23,  9,    123,  -104, 210,  -61,  44,   -245,  -60,   -225, -163,
+      -284, -292, -366, -364, -318, -284, -326, -284, -257,  -269,  -279, -348,
+      -246, -376, -367, -329, -388, -355, -428, -316, -418,  -375,  -426, -318,
+      -396, -436, -393, -407, -383, -405, -383, -442, -422,  -393,  -344, -375,
+      -309, -340, -270, -311, -270, -269, -256, -268, -233,  -283,  -233, -230,
+      -236, -283, -256, -265, -301, -274, -317, -261, -298,  -265,  -239, -262,
+      -221, -248, -188, -287, -192, -249, -152, -246, -140,  -231,  -148, -218,
+      -130, -170, -165, -154, -171, -127, -181, -99,  -138,  -109,  -120, -99,
+      -88,  -129, -61,  -150, -63,  -199, -44,  -206, -106,  -201,  -113, -189,
+      -182, -146, -180, -174, -177, -146, -150, -167, -103,  -131,  -76,  -138,
+      -14,  -119, -32,  -66,  -17,  -34,  -48,  8,    -80,   -30,   -22,  -34,
+      -12,  -52,  1,    -29,  -7,   12,   -34,  -30,  -36,   -43,   -95,  -29,
+      -80,  -102, -17,  -119, -24,  -119, 10,   -78,  -16,   -105,  -38,  -66,
+      -45,  -15,  9,    -36,  -32,  33,   -12,  -9,   -66,   -12,   -59,  8,
+      -23,  -10,  -6,   -22},
+     /* IRC_Composite_C_R0195_T300_P075.wav */
+     {5,    -5,   5,    -6,   6,    -6,   6,    -7,   7,     -7,   8,    -8,
+      9,    -9,   10,   -10,  11,   -12,  13,   -14,  16,    -19,  23,   -30,
+      49,   -174, -216, -220, -266, -146, -276, -310, -167,  -398, -257, -446,
+      -119, 2,    150,  904,  282,  399,  -249, -994, -2605, 51,   5569, 5353,
+      3304, -294, 1237, 2394, 748,  1912, 84,   -494, 511,   465,  372,  641,
+      761,  120,  1277, 373,  764,  283,  1087, -27,  731,   558,  213,  625,
+      490,  578,  331,  755,  261,  576,  308,  533,  270,   531,  241,  337,
+      345,  292,  249,  254,  130,  252,  95,   124,  0,     -94,  22,   -124,
+      -166, -120, -222, -195, -188, -102, -288, -138, -247,  -177, -285, -135,
+      -368, -230, -298, -370, -301, -355, -365, -394, -255,  -460, -300, -338,
+      -376, -356, -289, -388, -412, -303, -449, -224, -403,  -214, -327, -319,
+      -233, -280, -268, -283, -281, -308, -227, -300, -251,  -210, -330, -195,
+      -309, -196, -294, -197, -342, -211, -310, -249, -268,  -272, -295, -243,
+      -275, -237, -249, -206, -274, -172, -248, -191, -211,  -154, -258, -145,
+      -194, -183, -134, -143, -154, -131, -111, -163, -86,   -129, -136, -111,
+      -159, -135, -153, -128, -194, -126, -204, -111, -239,  -106, -205, -215,
+      -163, -175, -185, -155, -144, -130, -163, -92,  -105,  -142, -4,   -111,
+      -109, -19,  -118, 13,   -189, 9,    -76,  -118, -8,    -6,   -126, -54,
+      46,   -138, -64,  43,   -92,  -25,  -132, 48,   -164,  -57,  -43,  -117,
+      -199, -11,  -164, -58,  -105, -75,  -100, -150, -59,   -62,  -137, -80,
+      -148, -65,  -126, -49,  -62,  -33,  -77,  -10,  -18,   -11,  -21,  -15,
+      -36,  -38,  -38,  9}};
+
+const int16_t irc_composite_c_r0195_p090[][256] =
+    {/* IRC_Composite_C_R0195_T000_P090.wav */
+     {-1,   1,    -1,    1,     -1,    1,    -1,   2,    -2,   2,    -3,   3,
+      -3,   4,    -5,    5,     -7,    8,    -10,  14,   -25,  -385, -133, -356,
+      -344, -390, -312,  -438,  -400,  -852, -107, -995, -245, -186, 1358, -771,
+      1473, 28,   -1710, -3762, -3025, 4119, 8200, 5927, -785, 1558, 2250, 1837,
+      2098, 631,  -318,  150,   -77,   1215, -23,  792,  865,  600,  981,  659,
+      366,  950,  511,   766,   502,   516,  589,  299,  331,  571,  105,  609,
+      462,  485,  538,   594,   470,   640,  393,  432,  325,  604,  259,  355,
+      336,  454,  33,    263,   182,   -24,  -124, 121,  -231, -50,  -93,  49,
+      -230, -101, 25,    -196,  -123,  -21,  -160, -161, -132, -136, -155, -268,
+      -121, -146, -311,  -112,  -328,  -201, -220, -280, -214, -304, -233, -227,
+      -250, -242, -334,  -256,  -283,  -393, -267, -285, -332, -331, -224, -295,
+      -279, -171, -151,  -361,  -199,  -239, -227, -317, -168, -211, -370, -182,
+      -150, -343, -204,  -192,  -224,  -319, -203, -137, -357, -146, -223, -237,
+      -276, -162, -201,  -262,  -186,  -209, -147, -267, -46,  -252, -162, -146,
+      -182, -199, -143,  -218,  -204,  -108, -250, -163, -90,  -241, -137, -125,
+      -122, -272, -111,  -188,  -211,  -160, -160, -164, -201, -231, -114, -175,
+      -180, -212, -172,  -181,  -129,  -130, -77,  -150, -114, -39,  12,   -122,
+      -9,   -11,  -46,   -21,   -4,    7,    -73,  -20,  -50,  -21,  -80,  -102,
+      2,    -103, -66,   -67,   -8,    -41,  -119, -26,  29,   -208, 3,    -108,
+      -34,  -228, 114,   -196,  -84,   -28,  1,    -109, -92,  77,   -169, -27,
+      48,   -12,  -79,   3,     -100,  -44,  -53,  13,   -103, -90,  -14,  -91,
+      -23,  -64,  -21,   -116}};
+
+struct Elevation {
+  /**
+   * An array of |count| impulse responses of 256 samples for the left ear.
+   * The impulse responses in each elevation are at equally spaced azimuths
+   * for a full 360 degree revolution, ordered clockwise from in front the
+   * listener.
+   */
+  const int16_t (*azimuths)[256];
+  int count;
+};
+
+/**
+ * irc_composite_c_r0195 is an array with each element containing data for one
+ * elevation.
+ */
+const Elevation irc_composite_c_r0195[] = {
+    {irc_composite_c_r0195_p315, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p315)},
+    {irc_composite_c_r0195_p330, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p330)},
+    {irc_composite_c_r0195_p345, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p345)},
+    {irc_composite_c_r0195_p000, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p000)},
+    {irc_composite_c_r0195_p015, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p015)},
+    {irc_composite_c_r0195_p030, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p030)},
+    {irc_composite_c_r0195_p045, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p045)},
+    {irc_composite_c_r0195_p060, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p060)},
+    {irc_composite_c_r0195_p075, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p075)},
+    {irc_composite_c_r0195_p090, MOZ_ARRAY_LENGTH(irc_composite_c_r0195_p090)}};
+
+const int irc_composite_c_r0195_first_elevation = -45;   /* degrees */
+const int irc_composite_c_r0195_elevation_interval = 15; /* degrees */
+const int irc_composite_c_r0195_sample_rate = 44100;     /* Hz */
diff --git a/dom/media/webaudio/blink/PeriodicWave.cpp b/dom/media/webaudio/blink/PeriodicWave.cpp
new file mode 100644
index 0000000000..6b1d173008
--- /dev/null
+++ b/dom/media/webaudio/blink/PeriodicWave.cpp
@@ -0,0 +1,355 @@
+/*
+ * Copyright (C) 2012 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "PeriodicWave.h"
+#include <algorithm>
+#include <cmath>
+#include <limits>
+#include "mozilla/FFTBlock.h"
+
+const unsigned MinPeriodicWaveSize = 4096;  // This must be a power of two.
+const unsigned MaxPeriodicWaveSize = 8192;  // This must be a power of two.
+const float CentsPerRange = 1200 / 3;       // 1/3 Octave.
+
+using namespace mozilla;
+using mozilla::dom::OscillatorType;
+
+namespace WebCore {
+
+already_AddRefed<PeriodicWave> PeriodicWave::create(float sampleRate,
+                                                    const float* real,
+                                                    const float* imag,
+                                                    size_t numberOfComponents,
+                                                    bool disableNormalization) {
+  bool isGood = real && imag && numberOfComponents > 0;
+  MOZ_ASSERT(isGood);
+  if (isGood) {
+    RefPtr<PeriodicWave> periodicWave =
+        new PeriodicWave(sampleRate, numberOfComponents, disableNormalization);
+
+    // Limit the number of components used to those for frequencies below the
+    // Nyquist of the fixed length inverse FFT.
+    size_t halfSize = periodicWave->m_periodicWaveSize / 2;
+    numberOfComponents = std::min(numberOfComponents, halfSize);
+    periodicWave->m_numberOfComponents = numberOfComponents;
+    periodicWave->m_realComponents =
+        MakeUnique<AudioFloatArray>(numberOfComponents);
+    periodicWave->m_imagComponents =
+        MakeUnique<AudioFloatArray>(numberOfComponents);
+    memcpy(periodicWave->m_realComponents->Elements(), real,
+           numberOfComponents * sizeof(float));
+    memcpy(periodicWave->m_imagComponents->Elements(), imag,
+           numberOfComponents * sizeof(float));
+
+    return periodicWave.forget();
+  }
+  return nullptr;
+}
+
+already_AddRefed<PeriodicWave> PeriodicWave::createSine(float sampleRate) {
+  RefPtr<PeriodicWave> periodicWave =
+      new PeriodicWave(sampleRate, MinPeriodicWaveSize, false);
+  periodicWave->generateBasicWaveform(OscillatorType::Sine);
+  return periodicWave.forget();
+}
+
+already_AddRefed<PeriodicWave> PeriodicWave::createSquare(float sampleRate) {
+  RefPtr<PeriodicWave> periodicWave =
+      new PeriodicWave(sampleRate, MinPeriodicWaveSize, false);
+  periodicWave->generateBasicWaveform(OscillatorType::Square);
+  return periodicWave.forget();
+}
+
+already_AddRefed<PeriodicWave> PeriodicWave::createSawtooth(float sampleRate) {
+  RefPtr<PeriodicWave> periodicWave =
+      new PeriodicWave(sampleRate, MinPeriodicWaveSize, false);
+  periodicWave->generateBasicWaveform(OscillatorType::Sawtooth);
+  return periodicWave.forget();
+}
+
+already_AddRefed<PeriodicWave> PeriodicWave::createTriangle(float sampleRate) {
+  RefPtr<PeriodicWave> periodicWave =
+      new PeriodicWave(sampleRate, MinPeriodicWaveSize, false);
+  periodicWave->generateBasicWaveform(OscillatorType::Triangle);
+  return periodicWave.forget();
+}
+
+PeriodicWave::PeriodicWave(float sampleRate, size_t numberOfComponents,
+                           bool disableNormalization)
+    : m_sampleRate(sampleRate),
+      m_centsPerRange(CentsPerRange),
+      m_maxPartialsInBandLimitedTable(0),
+      m_normalizationScale(1.0f),
+      m_disableNormalization(disableNormalization) {
+  float nyquist = 0.5 * m_sampleRate;
+
+  if (numberOfComponents <= MinPeriodicWaveSize) {
+    m_periodicWaveSize = MinPeriodicWaveSize;
+  } else {
+    unsigned npow2 = fdlibm_exp2f(floorf(
+        fdlibm_logf(numberOfComponents - 1.0) / fdlibm_logf(2.0f) + 1.0f));
+    m_periodicWaveSize = std::min(MaxPeriodicWaveSize, npow2);
+  }
+
+  m_numberOfRanges =
+      (unsigned)(3.0f * fdlibm_logf(m_periodicWaveSize) / fdlibm_logf(2.0f));
+  m_bandLimitedTables.SetLength(m_numberOfRanges);
+  m_lowestFundamentalFrequency = nyquist / maxNumberOfPartials();
+  m_rateScale = m_periodicWaveSize / m_sampleRate;
+}
+
+size_t PeriodicWave::sizeOfIncludingThis(
+    mozilla::MallocSizeOf aMallocSizeOf) const {
+  size_t amount = aMallocSizeOf(this);
+
+  amount += m_bandLimitedTables.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < m_bandLimitedTables.Length(); i++) {
+    if (m_bandLimitedTables[i]) {
+      amount +=
+          m_bandLimitedTables[i]->ShallowSizeOfIncludingThis(aMallocSizeOf);
+    }
+  }
+
+  return amount;
+}
+
+void PeriodicWave::waveDataForFundamentalFrequency(
+    float fundamentalFrequency, float*& lowerWaveData, float*& higherWaveData,
+    float& tableInterpolationFactor) {
+  // Negative frequencies are allowed, in which case we alias
+  // to the positive frequency.
+  fundamentalFrequency = fabsf(fundamentalFrequency);
+
+  // We only need to rebuild to the tables if the new fundamental
+  // frequency is low enough to allow for more partials below the
+  // Nyquist frequency.
+  unsigned numberOfPartials = numberOfPartialsForRange(0);
+  float nyquist = 0.5 * m_sampleRate;
+  if (fundamentalFrequency != 0.0) {
+    numberOfPartials =
+        std::min(numberOfPartials, (unsigned)(nyquist / fundamentalFrequency));
+  }
+  if (numberOfPartials > m_maxPartialsInBandLimitedTable) {
+    for (unsigned rangeIndex = 0; rangeIndex < m_numberOfRanges; ++rangeIndex) {
+      m_bandLimitedTables[rangeIndex] = 0;
+    }
+
+    // We need to create the first table to determine the normalization
+    // constant.
+    createBandLimitedTables(fundamentalFrequency, 0);
+    m_maxPartialsInBandLimitedTable = numberOfPartials;
+  }
+
+  // Calculate the pitch range.
+  float ratio = fundamentalFrequency > 0
+                    ? fundamentalFrequency / m_lowestFundamentalFrequency
+                    : 0.5;
+  float centsAboveLowestFrequency =
+      fdlibm_logf(ratio) / fdlibm_logf(2.0f) * 1200;
+
+  // Add one to round-up to the next range just in time to truncate
+  // partials before aliasing occurs.
+  float pitchRange = 1 + centsAboveLowestFrequency / m_centsPerRange;
+
+  pitchRange = std::max(pitchRange, 0.0f);
+  pitchRange = std::min(pitchRange, static_cast<float>(m_numberOfRanges - 1));
+
+  // The words "lower" and "higher" refer to the table data having
+  // the lower and higher numbers of partials. It's a little confusing
+  // since the range index gets larger the more partials we cull out.
+  // So the lower table data will have a larger range index.
+  unsigned rangeIndex1 = static_cast<unsigned>(pitchRange);
+  unsigned rangeIndex2 =
+      rangeIndex1 < m_numberOfRanges - 1 ? rangeIndex1 + 1 : rangeIndex1;
+
+  if (!m_bandLimitedTables[rangeIndex1].get())
+    createBandLimitedTables(fundamentalFrequency, rangeIndex1);
+
+  if (!m_bandLimitedTables[rangeIndex2].get())
+    createBandLimitedTables(fundamentalFrequency, rangeIndex2);
+
+  lowerWaveData = m_bandLimitedTables[rangeIndex2]->Elements();
+  higherWaveData = m_bandLimitedTables[rangeIndex1]->Elements();
+
+  // Ranges from 0 -> 1 to interpolate between lower -> higher.
+  tableInterpolationFactor = rangeIndex2 - pitchRange;
+}
+
+unsigned PeriodicWave::maxNumberOfPartials() const {
+  return m_periodicWaveSize / 2;
+}
+
+unsigned PeriodicWave::numberOfPartialsForRange(unsigned rangeIndex) const {
+  // Number of cents below nyquist where we cull partials.
+  float centsToCull = rangeIndex * m_centsPerRange;
+
+  // A value from 0 -> 1 representing what fraction of the partials to keep.
+  float cullingScale = fdlibm_exp2f(-centsToCull / 1200);
+
+  // The very top range will have all the partials culled.
+  unsigned numberOfPartials = cullingScale * maxNumberOfPartials();
+
+  return numberOfPartials;
+}
+
+// Convert into time-domain wave buffers.
+// One table is created for each range for non-aliasing playback
+// at different playback rates. Thus, higher ranges have more
+// high-frequency partials culled out.
+void PeriodicWave::createBandLimitedTables(float fundamentalFrequency,
+                                           unsigned rangeIndex) {
+  unsigned fftSize = m_periodicWaveSize;
+  unsigned i;
+
+  const float* realData = m_realComponents->Elements();
+  const float* imagData = m_imagComponents->Elements();
+
+  // This FFTBlock is used to cull partials (represented by frequency bins).
+  FFTBlock frame(fftSize);
+
+  // Find the starting bin where we should start culling the aliasing
+  // partials for this pitch range.  We need to clear out the highest
+  // frequencies to band-limit the waveform.
+  unsigned numberOfPartials = numberOfPartialsForRange(rangeIndex);
+  // Also limit to the number of components that are provided.
+  numberOfPartials = std::min(numberOfPartials, m_numberOfComponents - 1);
+
+  // Limit number of partials to those below Nyquist frequency
+  float nyquist = 0.5 * m_sampleRate;
+  if (fundamentalFrequency != 0.0) {
+    numberOfPartials =
+        std::min(numberOfPartials, (unsigned)(nyquist / fundamentalFrequency));
+  }
+
+  // Copy from loaded frequency data and generate complex conjugate
+  // because of the way the inverse FFT is defined.
+  // The coefficients of higher partials remain zero, as initialized in
+  // the FFTBlock constructor.
+  for (i = 0; i < numberOfPartials + 1; ++i) {
+    frame.RealData(i) = realData[i];
+    frame.ImagData(i) = -imagData[i];
+  }
+
+  // Clear any DC-offset.
+  frame.RealData(0) = 0;
+  // Clear value which has no effect.
+  frame.ImagData(0) = 0;
+
+  // Create the band-limited table.
+  m_bandLimitedTables[rangeIndex] =
+      MakeUnique<AlignedAudioFloatArray>(m_periodicWaveSize);
+
+  // Apply an inverse FFT to generate the time-domain table data.
+  float* data = m_bandLimitedTables[rangeIndex]->Elements();
+  frame.GetInverseWithoutScaling(data);
+
+  // For the first range (which has the highest power), calculate
+  // its peak value then compute normalization scale.
+  if (m_disableNormalization) {
+    // See Bug 1424906, results need to be scaled by 0.5 even
+    // when normalization is disabled.
+    m_normalizationScale = 0.5;
+  } else if (!rangeIndex) {
+    float maxValue;
+    maxValue = AudioBufferPeakValue(data, m_periodicWaveSize);
+
+    if (maxValue) m_normalizationScale = 1.0f / maxValue;
+  }
+
+  // Apply normalization scale.
+  AudioBufferInPlaceScale(data, m_normalizationScale, m_periodicWaveSize);
+}
+
+void PeriodicWave::generateBasicWaveform(OscillatorType shape) {
+  const float piFloat = float(M_PI);
+  unsigned fftSize = periodicWaveSize();
+  unsigned halfSize = fftSize / 2;
+
+  m_numberOfComponents = halfSize;
+  m_realComponents = MakeUnique<AudioFloatArray>(halfSize);
+  m_imagComponents = MakeUnique<AudioFloatArray>(halfSize);
+  float* realP = m_realComponents->Elements();
+  float* imagP = m_imagComponents->Elements();
+
+  // Clear DC and imag value which is ignored.
+  realP[0] = 0;
+  imagP[0] = 0;
+
+  for (unsigned n = 1; n < halfSize; ++n) {
+    float omega = 2 * piFloat * n;
+    float invOmega = 1 / omega;
+
+    // Fourier coefficients according to standard definition.
+    float a;  // Coefficient for cos().
+    float b;  // Coefficient for sin().
+
+    // Calculate Fourier coefficients depending on the shape.
+    // Note that the overall scaling (magnitude) of the waveforms
+    // is normalized in createBandLimitedTables().
+    switch (shape) {
+      case OscillatorType::Sine:
+        // Standard sine wave function.
+        a = 0;
+        b = (n == 1) ? 1 : 0;
+        break;
+      case OscillatorType::Square:
+        // Square-shaped waveform with the first half its maximum value
+        // and the second half its minimum value.
+        a = 0;
+        b = invOmega * ((n & 1) ? 2 : 0);
+        break;
+      case OscillatorType::Sawtooth:
+        // Sawtooth-shaped waveform with the first half ramping from
+        // zero to maximum and the second half from minimum to zero.
+        a = 0;
+        b = -invOmega * fdlibm_cos(0.5 * omega);
+        break;
+      case OscillatorType::Triangle:
+        // Triangle-shaped waveform going from its maximum value to
+        // its minimum value then back to the maximum value.
+        a = 0;
+        if (n & 1) {
+          b = 2 * (2 / (n * piFloat) * 2 / (n * piFloat)) *
+              ((((n - 1) >> 1) & 1) ? -1 : 1);
+        } else {
+          b = 0;
+        }
+        break;
+      default:
+        MOZ_ASSERT_UNREACHABLE("invalid oscillator type");
+        a = 0;
+        b = 0;
+        break;
+    }
+
+    realP[n] = a;
+    imagP[n] = b;
+  }
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/PeriodicWave.h b/dom/media/webaudio/blink/PeriodicWave.h
new file mode 100644
index 0000000000..fd4e50287c
--- /dev/null
+++ b/dom/media/webaudio/blink/PeriodicWave.h
@@ -0,0 +1,123 @@
+/*
+ * Copyright (C) 2012 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef PeriodicWave_h
+#define PeriodicWave_h
+
+#include "mozilla/dom/OscillatorNodeBinding.h"
+#include "mozilla/UniquePtr.h"
+#include <nsTArray.h>
+#include "AlignedTArray.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace WebCore {
+
+typedef AlignedTArray<float> AlignedAudioFloatArray;
+typedef nsTArray<float> AudioFloatArray;
+
+using mozilla::UniquePtr;
+
+class PeriodicWave {
+ public:
+  NS_INLINE_DECL_THREADSAFE_REFCOUNTING(WebCore::PeriodicWave);
+
+  static already_AddRefed<PeriodicWave> createSine(float sampleRate);
+  static already_AddRefed<PeriodicWave> createSquare(float sampleRate);
+  static already_AddRefed<PeriodicWave> createSawtooth(float sampleRate);
+  static already_AddRefed<PeriodicWave> createTriangle(float sampleRate);
+
+  // Creates an arbitrary periodic wave given the frequency components
+  // (Fourier coefficients).
+  static already_AddRefed<PeriodicWave> create(float sampleRate,
+                                               const float* real,
+                                               const float* imag,
+                                               size_t numberOfComponents,
+                                               bool disableNormalization);
+
+  // Returns pointers to the lower and higher wave data for the pitch range
+  // containing the given fundamental frequency. These two tables are in
+  // adjacent "pitch" ranges where the higher table will have the maximum
+  // number of partials which won't alias when played back at this
+  // fundamental frequency. The lower wave is the next range containing fewer
+  // partials than the higher wave. Interpolation between these two tables
+  // can be made according to tableInterpolationFactor. Where values
+  // from 0 -> 1 interpolate between lower -> higher.
+  void waveDataForFundamentalFrequency(float, float*& lowerWaveData,
+                                       float*& higherWaveData,
+                                       float& tableInterpolationFactor);
+
+  // Returns the scalar multiplier to the oscillator frequency to calculate
+  // wave buffer phase increment.
+  float rateScale() const { return m_rateScale; }
+
+  unsigned periodicWaveSize() const { return m_periodicWaveSize; }
+  float sampleRate() const { return m_sampleRate; }
+
+  size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+ private:
+  explicit PeriodicWave(float sampleRate, size_t numberOfComponents,
+                        bool disableNormalization);
+  ~PeriodicWave() = default;
+
+  void generateBasicWaveform(mozilla::dom::OscillatorType);
+
+  float m_sampleRate;
+  unsigned m_periodicWaveSize;
+  unsigned m_numberOfRanges;
+  float m_centsPerRange;
+  unsigned m_numberOfComponents;
+  UniquePtr<AudioFloatArray> m_realComponents;
+  UniquePtr<AudioFloatArray> m_imagComponents;
+
+  // The lowest frequency (in Hertz) where playback will include all of the
+  // partials.  Playing back lower than this frequency will gradually lose
+  // more high-frequency information.
+  // This frequency is quite low (~10Hz @ // 44.1KHz)
+  float m_lowestFundamentalFrequency;
+
+  float m_rateScale;
+
+  unsigned numberOfRanges() const { return m_numberOfRanges; }
+
+  // Maximum possible number of partials (before culling).
+  unsigned maxNumberOfPartials() const;
+
+  unsigned numberOfPartialsForRange(unsigned rangeIndex) const;
+
+  // Creates table for specified index based on fundamental frequency.
+  void createBandLimitedTables(float fundamentalFrequency, unsigned rangeIndex);
+  unsigned m_maxPartialsInBandLimitedTable;
+  float m_normalizationScale;
+  bool m_disableNormalization;
+  nsTArray<UniquePtr<AlignedAudioFloatArray> > m_bandLimitedTables;
+};
+
+}  // namespace WebCore
+
+#endif  // PeriodicWave_h
diff --git a/dom/media/webaudio/blink/README b/dom/media/webaudio/blink/README
new file mode 100644
index 0000000000..96d209dfc8
--- /dev/null
+++ b/dom/media/webaudio/blink/README
@@ -0,0 +1,24 @@
+This directory contains the code originally borrowed from the Blink Web Audio
+implementation.  We are forking the code here because in many cases the burden
+of adopting Blink specific utilities is too large compared to the prospect of
+importing upstream fixes by just copying newer versions of the code in the
+future.
+
+The process of borrowing code from Blink is as follows:
+
+* Try to borrow utility classes only, and avoid borrowing code which depends
+  too much on the Blink specific utilities.
+* First, import the pristine files from the Blink repository before adding
+  them to the build system, noting the SVN revision of Blink from which the
+  original files were copied in the commit message.
+* In a separate commit, add the imported source files to the build system,
+  and apply the necessary changes to make it build successfully.
+* Use the code in a separate commit.
+* Never add headers as exported headers.  All headers should be included
+  using the following convention: #include "blink/Header.h".
+* Leave the imported code in the WebCore namespace, and import the needed
+  names into the Mozilla code via `using'.
+* Cherry-pick upsteam fixes manually when needed.  In case you fix a problem
+  that is not Mozilla specific locally, try to upstream your changes into
+  Blink.
+* Ping ehsan for any questions.
diff --git a/dom/media/webaudio/blink/Reverb.cpp b/dom/media/webaudio/blink/Reverb.cpp
new file mode 100644
index 0000000000..dfce966ca9
--- /dev/null
+++ b/dom/media/webaudio/blink/Reverb.cpp
@@ -0,0 +1,277 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "Reverb.h"
+#include "ReverbConvolverStage.h"
+
+#include <math.h>
+#include "ReverbConvolver.h"
+#include "mozilla/FloatingPoint.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+// Empirical gain calibration tested across many impulse responses to ensure
+// perceived volume is same as dry (unprocessed) signal
+const float GainCalibration = 0.00125f;
+const float GainCalibrationSampleRate = 44100;
+
+// A minimum power value to when normalizing a silent (or very quiet) impulse
+// response
+const float MinPower = 0.000125f;
+
+static float calculateNormalizationScale(const nsTArray<const float*>& response,
+                                         size_t aLength, float sampleRate) {
+  // Normalize by RMS power
+  size_t numberOfChannels = response.Length();
+
+  float power = 0;
+
+  for (size_t i = 0; i < numberOfChannels; ++i) {
+    float channelPower = AudioBufferSumOfSquares(response[i], aLength);
+    power += channelPower;
+  }
+
+  power = sqrt(power / (numberOfChannels * aLength));
+
+  // Protect against accidental overload
+  if (!std::isfinite(power) || std::isnan(power) || power < MinPower)
+    power = MinPower;
+
+  float scale = 1 / power;
+
+  scale *= GainCalibration;  // calibrate to make perceived volume same as
+                             // unprocessed
+
+  // Scale depends on sample-rate.
+  if (sampleRate) scale *= GainCalibrationSampleRate / sampleRate;
+
+  // True-stereo compensation
+  if (numberOfChannels == 4) scale *= 0.5f;
+
+  return scale;
+}
+
+Reverb::Reverb(const AudioChunk& impulseResponse, size_t maxFFTSize,
+               bool useBackgroundThreads, bool normalize, float sampleRate,
+               bool* aAllocationFailure) {
+  MOZ_ASSERT(aAllocationFailure);
+  size_t impulseResponseBufferLength = impulseResponse.mDuration;
+  float scale = impulseResponse.mVolume;
+
+  CopyableAutoTArray<const float*, 4> irChannels;
+  irChannels.AppendElements(impulseResponse.ChannelData<float>());
+  AutoTArray<float, 1024> tempBuf;
+
+  if (normalize) {
+    scale = calculateNormalizationScale(irChannels, impulseResponseBufferLength,
+                                        sampleRate);
+  }
+
+  if (scale != 1.0f) {
+    bool rv = tempBuf.SetLength(
+        irChannels.Length() * impulseResponseBufferLength, mozilla::fallible);
+    *aAllocationFailure = !rv;
+    if (*aAllocationFailure) {
+      return;
+    }
+
+    for (uint32_t i = 0; i < irChannels.Length(); ++i) {
+      float* buf = &tempBuf[i * impulseResponseBufferLength];
+      AudioBufferCopyWithScale(irChannels[i], scale, buf,
+                               impulseResponseBufferLength);
+      irChannels[i] = buf;
+    }
+  }
+
+  *aAllocationFailure = !initialize(irChannels, impulseResponseBufferLength,
+                                    maxFFTSize, useBackgroundThreads);
+}
+
+size_t Reverb::sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const {
+  size_t amount = aMallocSizeOf(this);
+  amount += m_convolvers.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < m_convolvers.Length(); i++) {
+    if (m_convolvers[i]) {
+      amount += m_convolvers[i]->sizeOfIncludingThis(aMallocSizeOf);
+    }
+  }
+
+  amount += m_tempBuffer.SizeOfExcludingThis(aMallocSizeOf, false);
+  return amount;
+}
+
+bool Reverb::initialize(const nsTArray<const float*>& impulseResponseBuffer,
+                        size_t impulseResponseBufferLength, size_t maxFFTSize,
+                        bool useBackgroundThreads) {
+  m_impulseResponseLength = impulseResponseBufferLength;
+
+  // The reverb can handle a mono impulse response and still do stereo
+  // processing
+  size_t numResponseChannels = impulseResponseBuffer.Length();
+  MOZ_ASSERT(numResponseChannels > 0);
+  // The number of convolvers required is at least the number of audio
+  // channels.  Even if there is initially only one audio channel, another
+  // may be added later, and so a second convolver is created now while the
+  // impulse response is available.
+  size_t numConvolvers = std::max<size_t>(numResponseChannels, 2);
+  m_convolvers.SetCapacity(numConvolvers);
+
+  int convolverRenderPhase = 0;
+  for (size_t i = 0; i < numConvolvers; ++i) {
+    size_t channelIndex = i < numResponseChannels ? i : 0;
+    const float* channel = impulseResponseBuffer[channelIndex];
+    size_t length = impulseResponseBufferLength;
+
+    bool allocationFailure;
+    UniquePtr<ReverbConvolver> convolver(
+        new ReverbConvolver(channel, length, maxFFTSize, convolverRenderPhase,
+                            useBackgroundThreads, &allocationFailure));
+    if (allocationFailure) {
+      return false;
+    }
+    m_convolvers.AppendElement(std::move(convolver));
+
+    convolverRenderPhase += WEBAUDIO_BLOCK_SIZE;
+  }
+
+  // For "True" stereo processing we allocate a temporary buffer to avoid
+  // repeatedly allocating it in the process() method. It can be bad to allocate
+  // memory in a real-time thread.
+  if (numResponseChannels == 4) {
+    m_tempBuffer.AllocateChannels(2);
+    WriteZeroesToAudioBlock(&m_tempBuffer, 0, WEBAUDIO_BLOCK_SIZE);
+  }
+  return true;
+}
+
+void Reverb::process(const AudioBlock* sourceBus, AudioBlock* destinationBus) {
+  // Do a fairly comprehensive sanity check.
+  // If these conditions are satisfied, all of the source and destination
+  // pointers will be valid for the various matrixing cases.
+  bool isSafeToProcess =
+      sourceBus && destinationBus && sourceBus->ChannelCount() > 0 &&
+      destinationBus->mChannelData.Length() > 0 &&
+      WEBAUDIO_BLOCK_SIZE <= MaxFrameSize &&
+      WEBAUDIO_BLOCK_SIZE <= size_t(sourceBus->GetDuration()) &&
+      WEBAUDIO_BLOCK_SIZE <= size_t(destinationBus->GetDuration());
+
+  MOZ_ASSERT(isSafeToProcess);
+  if (!isSafeToProcess) return;
+
+  // For now only handle mono or stereo output
+  MOZ_ASSERT(destinationBus->ChannelCount() <= 2);
+
+  float* destinationChannelL =
+      static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[0]));
+  const float* sourceBusL =
+      static_cast<const float*>(sourceBus->mChannelData[0]);
+
+  // Handle input -> output matrixing...
+  size_t numInputChannels = sourceBus->ChannelCount();
+  size_t numOutputChannels = destinationBus->ChannelCount();
+  size_t numReverbChannels = m_convolvers.Length();
+
+  if (numInputChannels == 2 && numReverbChannels == 2 &&
+      numOutputChannels == 2) {
+    // 2 -> 2 -> 2
+    const float* sourceBusR =
+        static_cast<const float*>(sourceBus->mChannelData[1]);
+    float* destinationChannelR =
+        static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+    m_convolvers[0]->process(sourceBusL, destinationChannelL);
+    m_convolvers[1]->process(sourceBusR, destinationChannelR);
+  } else if (numInputChannels == 1 && numOutputChannels == 2 &&
+             numReverbChannels == 2) {
+    // 1 -> 2 -> 2
+    for (int i = 0; i < 2; ++i) {
+      float* destinationChannel = static_cast<float*>(
+          const_cast<void*>(destinationBus->mChannelData[i]));
+      m_convolvers[i]->process(sourceBusL, destinationChannel);
+    }
+  } else if (numInputChannels == 1 && numOutputChannels == 1) {
+    // 1 -> 1 -> 1 (Only one of the convolvers is used.)
+    m_convolvers[0]->process(sourceBusL, destinationChannelL);
+  } else if (numInputChannels == 2 && numReverbChannels == 4 &&
+             numOutputChannels == 2) {
+    // 2 -> 4 -> 2 ("True" stereo)
+    const float* sourceBusR =
+        static_cast<const float*>(sourceBus->mChannelData[1]);
+    float* destinationChannelR =
+        static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+
+    float* tempChannelL =
+        static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
+    float* tempChannelR =
+        static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
+
+    // Process left virtual source
+    m_convolvers[0]->process(sourceBusL, destinationChannelL);
+    m_convolvers[1]->process(sourceBusL, destinationChannelR);
+
+    // Process right virtual source
+    m_convolvers[2]->process(sourceBusR, tempChannelL);
+    m_convolvers[3]->process(sourceBusR, tempChannelR);
+
+    AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL,
+                            sourceBus->GetDuration());
+    AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR,
+                            sourceBus->GetDuration());
+  } else if (numInputChannels == 1 && numReverbChannels == 4 &&
+             numOutputChannels == 2) {
+    // 1 -> 4 -> 2 (Processing mono with "True" stereo impulse response)
+    // This is an inefficient use of a four-channel impulse response, but we
+    // should handle the case.
+    float* destinationChannelR =
+        static_cast<float*>(const_cast<void*>(destinationBus->mChannelData[1]));
+
+    float* tempChannelL =
+        static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[0]));
+    float* tempChannelR =
+        static_cast<float*>(const_cast<void*>(m_tempBuffer.mChannelData[1]));
+
+    // Process left virtual source
+    m_convolvers[0]->process(sourceBusL, destinationChannelL);
+    m_convolvers[1]->process(sourceBusL, destinationChannelR);
+
+    // Process right virtual source
+    m_convolvers[2]->process(sourceBusL, tempChannelL);
+    m_convolvers[3]->process(sourceBusL, tempChannelR);
+
+    AudioBufferAddWithScale(tempChannelL, 1.0f, destinationChannelL,
+                            sourceBus->GetDuration());
+    AudioBufferAddWithScale(tempChannelR, 1.0f, destinationChannelR,
+                            sourceBus->GetDuration());
+  } else {
+    MOZ_ASSERT_UNREACHABLE("Unexpected Reverb configuration");
+    destinationBus->SetNull(destinationBus->GetDuration());
+  }
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/Reverb.h b/dom/media/webaudio/blink/Reverb.h
new file mode 100644
index 0000000000..16459532a9
--- /dev/null
+++ b/dom/media/webaudio/blink/Reverb.h
@@ -0,0 +1,80 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef Reverb_h
+#define Reverb_h
+
+#include "ReverbConvolver.h"
+#include "nsTArray.h"
+#include "AudioBlock.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/UniquePtr.h"
+
+namespace WebCore {
+
+using mozilla::UniquePtr;
+
+// Multi-channel convolution reverb with channel matrixing - one or more
+// ReverbConvolver objects are used internally.
+
+class Reverb {
+ public:
+  enum { MaxFrameSize = 256 };
+
+  // renderSliceSize is a rendering hint, so the FFTs can be optimized to not
+  // all occur at the same time (very bad when rendering on a real-time thread).
+  // aAllocation failure is to be checked by the caller. If false, internal
+  // memory could not be allocated, and this Reverb instance is not to be
+  // used.
+  Reverb(const mozilla::AudioChunk& impulseResponseBuffer, size_t maxFFTSize,
+         bool useBackgroundThreads, bool normalize, float sampleRate,
+         bool* aAllocationFailure);
+
+  void process(const mozilla::AudioBlock* sourceBus,
+               mozilla::AudioBlock* destinationBus);
+
+  size_t impulseResponseLength() const { return m_impulseResponseLength; }
+
+  size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+ private:
+  bool initialize(const nsTArray<const float*>& impulseResponseBuffer,
+                  size_t impulseResponseBufferLength, size_t maxFFTSize,
+                  bool useBackgroundThreads);
+
+  size_t m_impulseResponseLength;
+
+  nsTArray<UniquePtr<ReverbConvolver> > m_convolvers;
+
+  // For "True" stereo processing
+  mozilla::AudioBlock m_tempBuffer;
+};
+
+}  // namespace WebCore
+
+#endif  // Reverb_h
diff --git a/dom/media/webaudio/blink/ReverbAccumulationBuffer.cpp b/dom/media/webaudio/blink/ReverbAccumulationBuffer.cpp
new file mode 100644
index 0000000000..ec745c294f
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbAccumulationBuffer.cpp
@@ -0,0 +1,114 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "ReverbAccumulationBuffer.h"
+#include "AudioNodeEngine.h"
+#include "mozilla/PodOperations.h"
+#include <algorithm>
+
+using namespace mozilla;
+
+namespace WebCore {
+
+ReverbAccumulationBuffer::ReverbAccumulationBuffer()
+    : m_readIndex(0), m_readTimeFrame(0) {}
+
+bool ReverbAccumulationBuffer::allocate(size_t length) {
+  if (!m_buffer.SetLength(length, fallible)) {
+    return false;
+  }
+  PodZero(m_buffer.Elements(), length);
+  return true;
+}
+
+void ReverbAccumulationBuffer::readAndClear(float* destination,
+                                            size_t numberOfFrames) {
+  size_t bufferLength = m_buffer.Length();
+  bool isCopySafe =
+      m_readIndex <= bufferLength && numberOfFrames <= bufferLength;
+
+  MOZ_ASSERT(isCopySafe);
+  if (!isCopySafe) return;
+
+  size_t framesAvailable = bufferLength - m_readIndex;
+  size_t numberOfFrames1 = std::min(numberOfFrames, framesAvailable);
+  size_t numberOfFrames2 = numberOfFrames - numberOfFrames1;
+
+  float* source = m_buffer.Elements();
+  memcpy(destination, source + m_readIndex, sizeof(float) * numberOfFrames1);
+  memset(source + m_readIndex, 0, sizeof(float) * numberOfFrames1);
+
+  // Handle wrap-around if necessary
+  if (numberOfFrames2 > 0) {
+    memcpy(destination + numberOfFrames1, source,
+           sizeof(float) * numberOfFrames2);
+    memset(source, 0, sizeof(float) * numberOfFrames2);
+  }
+
+  m_readIndex = (m_readIndex + numberOfFrames) % bufferLength;
+  m_readTimeFrame += numberOfFrames;
+}
+
+void ReverbAccumulationBuffer::accumulate(const float* source,
+                                          size_t numberOfFrames,
+                                          size_t* readIndex,
+                                          size_t delayFrames) {
+  size_t bufferLength = m_buffer.Length();
+
+  size_t writeIndex = (*readIndex + delayFrames) % bufferLength;
+
+  // Update caller's readIndex
+  *readIndex = (*readIndex + numberOfFrames) % bufferLength;
+
+  size_t framesAvailable = bufferLength - writeIndex;
+  size_t numberOfFrames1 = std::min(numberOfFrames, framesAvailable);
+  size_t numberOfFrames2 = numberOfFrames - numberOfFrames1;
+
+  float* destination = m_buffer.Elements();
+
+  bool isSafe = writeIndex <= bufferLength &&
+                numberOfFrames1 + writeIndex <= bufferLength &&
+                numberOfFrames2 <= bufferLength;
+  MOZ_ASSERT(isSafe);
+  if (!isSafe) return;
+
+  AudioBufferAddWithScale(source, 1.0f, destination + writeIndex,
+                          numberOfFrames1);
+  if (numberOfFrames2 > 0) {
+    AudioBufferAddWithScale(source + numberOfFrames1, 1.0f, destination,
+                            numberOfFrames2);
+  }
+}
+
+void ReverbAccumulationBuffer::reset() {
+  PodZero(m_buffer.Elements(), m_buffer.Length());
+  m_readIndex = 0;
+  m_readTimeFrame = 0;
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/ReverbAccumulationBuffer.h b/dom/media/webaudio/blink/ReverbAccumulationBuffer.h
new file mode 100644
index 0000000000..8883a56a73
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbAccumulationBuffer.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef ReverbAccumulationBuffer_h
+#define ReverbAccumulationBuffer_h
+
+#include "AlignedTArray.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace WebCore {
+
+// ReverbAccumulationBuffer is a circular delay buffer with one client reading
+// from it and multiple clients writing/accumulating to it at different delay
+// offsets from the read position.  The read operation will zero the memory just
+// read from the buffer, so it will be ready for accumulation the next time
+// around.
+class ReverbAccumulationBuffer {
+ public:
+  ReverbAccumulationBuffer();
+  // Returns false on failure.
+  bool allocate(size_t length);
+
+  // This will read from, then clear-out numberOfFrames
+  void readAndClear(float* destination, size_t numberOfFrames);
+
+  // Each ReverbConvolverStage will accumulate its output at the appropriate
+  // delay from the read position. We need to pass in and update readIndex here,
+  // since each ReverbConvolverStage may be running in a different thread than
+  // the realtime thread calling ReadAndClear() and maintaining m_readIndex
+  void accumulate(const float* source, size_t numberOfFrames, size_t* readIndex,
+                  size_t delayFrames);
+
+  size_t readIndex() const { return m_readIndex; }
+
+  size_t readTimeFrame() const { return m_readTimeFrame; }
+
+  void reset();
+
+  size_t sizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const {
+    return m_buffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  AlignedTArray<float, 16> m_buffer;
+  size_t m_readIndex;
+  size_t m_readTimeFrame;  // for debugging (frame on continuous timeline)
+};
+
+}  // namespace WebCore
+
+#endif  // ReverbAccumulationBuffer_h
diff --git a/dom/media/webaudio/blink/ReverbConvolver.cpp b/dom/media/webaudio/blink/ReverbConvolver.cpp
new file mode 100644
index 0000000000..dcaf6e7467
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbConvolver.cpp
@@ -0,0 +1,272 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "ReverbConvolver.h"
+#include "ReverbConvolverStage.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+const int InputBufferSize = 8 * 16384;
+
+// We only process the leading portion of the impulse response in the real-time
+// thread.  We don't exceed this length. It turns out then, that the background
+// thread has about 278msec of scheduling slop. Empirically, this has been found
+// to be a good compromise between giving enough time for scheduling slop, while
+// still minimizing the amount of processing done in the primary (high-priority)
+// thread. This was found to be a good value on Mac OS X, and may work well on
+// other platforms as well, assuming the very rough scheduling latencies are
+// similar on these time-scales.  Of course, this code may need to be tuned for
+// individual platforms if this assumption is found to be incorrect.
+const size_t RealtimeFrameLimit = 8192 + 4096  // ~278msec @ 44.1KHz
+                                  - WEBAUDIO_BLOCK_SIZE;
+// First stage will have size MinFFTSize - successive stages will double in
+// size each time until we hit the maximum size.
+const size_t MinFFTSize = 256;
+// If we are using background threads then don't exceed this FFT size for the
+// stages which run in the real-time thread.  This avoids having only one or
+// two large stages (size 16384 or so) at the end which take a lot of time
+// every several processing slices.  This way we amortize the cost over more
+// processing slices.
+const size_t MaxRealtimeFFTSize = 4096;
+
+ReverbConvolver::ReverbConvolver(const float* impulseResponseData,
+                                 size_t impulseResponseLength,
+                                 size_t maxFFTSize, size_t convolverRenderPhase,
+                                 bool useBackgroundThreads,
+                                 bool* aAllocationFailure)
+    : m_impulseResponseLength(impulseResponseLength),
+      m_inputBuffer(InputBufferSize),
+      m_backgroundThread("ConvolverWorker"),
+      m_backgroundThreadMonitor("ConvolverMonitor"),
+      m_useBackgroundThreads(useBackgroundThreads),
+      m_wantsToExit(false),
+      m_moreInputBuffered(false) {
+  *aAllocationFailure = !m_accumulationBuffer.allocate(impulseResponseLength +
+                                                       WEBAUDIO_BLOCK_SIZE);
+  if (*aAllocationFailure) {
+    return;
+  }
+  // For the moment, a good way to know if we have real-time constraint is to
+  // check if we're using background threads. Otherwise, assume we're being run
+  // from a command-line tool.
+  bool hasRealtimeConstraint = useBackgroundThreads;
+
+  const float* response = impulseResponseData;
+  size_t totalResponseLength = impulseResponseLength;
+
+  // The total latency is zero because the first FFT stage is small enough
+  // to return output in the first block.
+  size_t reverbTotalLatency = 0;
+
+  size_t stageOffset = 0;
+  size_t stagePhase = 0;
+  size_t fftSize = MinFFTSize;
+  while (stageOffset < totalResponseLength) {
+    size_t stageSize = fftSize / 2;
+
+    // For the last stage, it's possible that stageOffset is such that we're
+    // straddling the end of the impulse response buffer (if we use stageSize),
+    // so reduce the last stage's length...
+    if (stageSize + stageOffset > totalResponseLength) {
+      stageSize = totalResponseLength - stageOffset;
+      // Use smallest FFT that is large enough to cover the last stage.
+      fftSize = MinFFTSize;
+      while (stageSize * 2 > fftSize) {
+        fftSize *= 2;
+      }
+    }
+
+    // This "staggers" the time when each FFT happens so they don't all happen
+    // at the same time
+    int renderPhase = convolverRenderPhase + stagePhase;
+
+    UniquePtr<ReverbConvolverStage> stage(new ReverbConvolverStage(
+        response, totalResponseLength, reverbTotalLatency, stageOffset,
+        stageSize, fftSize, renderPhase, &m_accumulationBuffer));
+
+    bool isBackgroundStage = false;
+
+    if (this->useBackgroundThreads() && stageOffset > RealtimeFrameLimit) {
+      m_backgroundStages.AppendElement(std::move(stage));
+      isBackgroundStage = true;
+    } else
+      m_stages.AppendElement(std::move(stage));
+
+    // Figure out next FFT size
+    fftSize *= 2;
+
+    stageOffset += stageSize;
+
+    if (hasRealtimeConstraint && !isBackgroundStage &&
+        fftSize > MaxRealtimeFFTSize) {
+      fftSize = MaxRealtimeFFTSize;
+      // Custom phase positions for all but the first of the realtime
+      // stages of largest size.  These spread out the work of the
+      // larger realtime stages.  None of the FFTs of size 1024, 2048 or
+      // 4096 are performed when processing the same block.  The first
+      // MaxRealtimeFFTSize = 4096 stage, at the end of the doubling,
+      // performs its FFT at block 7.  The FFTs of size 2048 are
+      // performed in blocks 3 + 8 * n and size 1024 at 1 + 4 * n.
+      const uint32_t phaseLookup[] = {14, 0, 10, 4};
+      stagePhase = WEBAUDIO_BLOCK_SIZE *
+                   phaseLookup[m_stages.Length() % ArrayLength(phaseLookup)];
+    } else if (fftSize > maxFFTSize) {
+      fftSize = maxFFTSize;
+      // A prime offset spreads out FFTs in a way that all
+      // available phase positions will be used if there are sufficient
+      // stages.
+      stagePhase += 5 * WEBAUDIO_BLOCK_SIZE;
+    } else if (stageSize > WEBAUDIO_BLOCK_SIZE) {
+      // As the stages are doubling in size, the next FFT will occur
+      // mid-way between FFTs for this stage.
+      stagePhase = stageSize - WEBAUDIO_BLOCK_SIZE;
+    }
+  }
+
+  // Start up background thread
+  // FIXME: would be better to up the thread priority here.  It doesn't need to
+  // be real-time, but higher than the default...
+  if (this->useBackgroundThreads() && m_backgroundStages.Length() > 0) {
+    if (!m_backgroundThread.Start()) {
+      NS_WARNING("Cannot start convolver thread.");
+      return;
+    }
+    m_backgroundThread.message_loop()->PostTask(NewNonOwningRunnableMethod(
+        "WebCore::ReverbConvolver::backgroundThreadEntry", this,
+        &ReverbConvolver::backgroundThreadEntry));
+  }
+}
+
+ReverbConvolver::~ReverbConvolver() {
+  // Wait for background thread to stop
+  if (useBackgroundThreads() && m_backgroundThread.IsRunning()) {
+    m_wantsToExit = true;
+
+    // Wake up thread so it can return
+    {
+      MonitorAutoLock locker(m_backgroundThreadMonitor);
+      m_moreInputBuffered = true;
+      m_backgroundThreadMonitor.Notify();
+    }
+
+    m_backgroundThread.Stop();
+  }
+}
+
+size_t ReverbConvolver::sizeOfIncludingThis(
+    mozilla::MallocSizeOf aMallocSizeOf) const {
+  size_t amount = aMallocSizeOf(this);
+  amount += m_stages.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < m_stages.Length(); i++) {
+    if (m_stages[i]) {
+      amount += m_stages[i]->sizeOfIncludingThis(aMallocSizeOf);
+    }
+  }
+
+  amount += m_backgroundStages.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  for (size_t i = 0; i < m_backgroundStages.Length(); i++) {
+    if (m_backgroundStages[i]) {
+      amount += m_backgroundStages[i]->sizeOfIncludingThis(aMallocSizeOf);
+    }
+  }
+
+  // NB: The buffer sizes are static, so even though they might be accessed
+  //     in another thread it's safe to measure them.
+  amount += m_accumulationBuffer.sizeOfExcludingThis(aMallocSizeOf);
+  amount += m_inputBuffer.sizeOfExcludingThis(aMallocSizeOf);
+
+  // Possible future measurements:
+  // - m_backgroundThread
+  // - m_backgroundThreadMonitor
+  return amount;
+}
+
+void ReverbConvolver::backgroundThreadEntry() {
+  while (!m_wantsToExit) {
+    // Wait for realtime thread to give us more input
+    m_moreInputBuffered = false;
+    {
+      MonitorAutoLock locker(m_backgroundThreadMonitor);
+      while (!m_moreInputBuffered && !m_wantsToExit)
+        m_backgroundThreadMonitor.Wait();
+    }
+
+    // Process all of the stages until their read indices reach the input
+    // buffer's write index
+    int writeIndex = m_inputBuffer.writeIndex();
+
+    // Even though it doesn't seem like every stage needs to maintain its own
+    // version of readIndex we do this in case we want to run in more than one
+    // background thread.
+    int readIndex;
+
+    while ((readIndex = m_backgroundStages[0]->inputReadIndex()) !=
+           writeIndex) {  // FIXME: do better to detect buffer overrun...
+      // Accumulate contributions from each stage
+      for (size_t i = 0; i < m_backgroundStages.Length(); ++i)
+        m_backgroundStages[i]->processInBackground(this);
+    }
+  }
+}
+
+void ReverbConvolver::process(const float* sourceChannelData,
+                              float* destinationChannelData) {
+  const float* source = sourceChannelData;
+  float* destination = destinationChannelData;
+  bool isDataSafe = source && destination;
+  MOZ_ASSERT(isDataSafe);
+  if (!isDataSafe) return;
+
+  // Feed input buffer (read by all threads)
+  m_inputBuffer.write(source, WEBAUDIO_BLOCK_SIZE);
+
+  // Accumulate contributions from each stage
+  for (size_t i = 0; i < m_stages.Length(); ++i) m_stages[i]->process(source);
+
+  // Finally read from accumulation buffer
+  m_accumulationBuffer.readAndClear(destination, WEBAUDIO_BLOCK_SIZE);
+
+  // Now that we've buffered more input, wake up our background thread.
+
+  // Not using a MonitorAutoLock looks strange, but we use a TryLock() instead
+  // because this is run on the real-time thread where it is a disaster for the
+  // lock to be contended (causes audio glitching).  It's OK if we fail to
+  // signal from time to time, since we'll get to it the next time we're called.
+  // We're called repeatedly and frequently (around every 3ms).  The background
+  // thread is processing well into the future and has a considerable amount of
+  // leeway here...
+  if (m_backgroundThreadMonitor.TryLock()) {
+    m_moreInputBuffered = true;
+    m_backgroundThreadMonitor.Notify();
+    m_backgroundThreadMonitor.Unlock();
+  }
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/ReverbConvolver.h b/dom/media/webaudio/blink/ReverbConvolver.h
new file mode 100644
index 0000000000..8000ba811e
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbConvolver.h
@@ -0,0 +1,94 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef ReverbConvolver_h
+#define ReverbConvolver_h
+
+#include "ReverbAccumulationBuffer.h"
+#include "ReverbInputBuffer.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/Monitor.h"
+#include "mozilla/UniquePtr.h"
+#ifdef LOG
+#  undef LOG
+#endif
+#include "base/thread.h"
+#include <atomic>
+
+namespace WebCore {
+
+using mozilla::UniquePtr;
+
+class ReverbConvolverStage;
+
+class ReverbConvolver {
+ public:
+  // maxFFTSize can be adjusted (from say 2048 to 32768) depending on how much
+  // precision is necessary. For certain tweaky de-convolving applications the
+  // phase errors add up quickly and lead to non-sensical results with larger
+  // FFT sizes and single-precision floats.  In these cases 2048 is a good size.
+  // If not doing multi-threaded convolution, then should not go > 8192.
+  ReverbConvolver(const float* impulseResponseData,
+                  size_t impulseResponseLength, size_t maxFFTSize,
+                  size_t convolverRenderPhase, bool useBackgroundThreads,
+                  bool* aAllocationFailure);
+  ~ReverbConvolver();
+
+  void process(const float* sourceChannelData, float* destinationChannelData);
+
+  size_t impulseResponseLength() const { return m_impulseResponseLength; }
+
+  ReverbInputBuffer* inputBuffer() { return &m_inputBuffer; }
+
+  bool useBackgroundThreads() const { return m_useBackgroundThreads; }
+  void backgroundThreadEntry();
+
+  size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+ private:
+  nsTArray<UniquePtr<ReverbConvolverStage> > m_stages;
+  nsTArray<UniquePtr<ReverbConvolverStage> > m_backgroundStages;
+  size_t m_impulseResponseLength;
+
+  ReverbAccumulationBuffer m_accumulationBuffer;
+
+  // One or more background threads read from this input buffer which is fed
+  // from the realtime thread.
+  ReverbInputBuffer m_inputBuffer;
+
+  // Background thread and synchronization
+  base::Thread m_backgroundThread;
+  mozilla::Monitor m_backgroundThreadMonitor MOZ_UNANNOTATED;
+  bool m_useBackgroundThreads;
+  std::atomic<bool> m_wantsToExit;
+  std::atomic<bool> m_moreInputBuffered;
+};
+
+}  // namespace WebCore
+
+#endif  // ReverbConvolver_h
diff --git a/dom/media/webaudio/blink/ReverbConvolverStage.cpp b/dom/media/webaudio/blink/ReverbConvolverStage.cpp
new file mode 100644
index 0000000000..56354f0271
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbConvolverStage.cpp
@@ -0,0 +1,101 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "ReverbConvolverStage.h"
+
+#include "ReverbAccumulationBuffer.h"
+#include "ReverbConvolver.h"
+#include "ReverbInputBuffer.h"
+#include "mozilla/PodOperations.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+ReverbConvolverStage::ReverbConvolverStage(
+    const float* impulseResponse, size_t, size_t reverbTotalLatency,
+    size_t stageOffset, size_t stageLength, size_t fftSize, size_t renderPhase,
+    ReverbAccumulationBuffer* accumulationBuffer)
+    : m_accumulationBuffer(accumulationBuffer),
+      m_accumulationReadIndex(0),
+      m_inputReadIndex(0) {
+  MOZ_ASSERT(impulseResponse);
+  MOZ_ASSERT(accumulationBuffer);
+
+  m_fftKernel = MakeUnique<FFTBlock>(fftSize);
+  m_fftKernel->PadAndMakeScaledDFT(impulseResponse + stageOffset, stageLength);
+  m_fftConvolver = MakeUnique<FFTConvolver>(fftSize, renderPhase);
+
+  // The convolution stage at offset stageOffset needs to have a corresponding
+  // delay to cancel out the offset.
+  size_t totalDelay = stageOffset + reverbTotalLatency;
+
+  // But, the FFT convolution itself incurs latency, so subtract this out...
+  size_t fftLatency = m_fftConvolver->latencyFrames();
+  MOZ_ASSERT(totalDelay >= fftLatency);
+  totalDelay -= fftLatency;
+
+  m_postDelayLength = totalDelay;
+}
+
+size_t ReverbConvolverStage::sizeOfIncludingThis(
+    mozilla::MallocSizeOf aMallocSizeOf) const {
+  size_t amount = aMallocSizeOf(this);
+
+  if (m_fftKernel) {
+    amount += m_fftKernel->SizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  if (m_fftConvolver) {
+    amount += m_fftConvolver->sizeOfIncludingThis(aMallocSizeOf);
+  }
+
+  return amount;
+}
+
+void ReverbConvolverStage::processInBackground(ReverbConvolver* convolver) {
+  ReverbInputBuffer* inputBuffer = convolver->inputBuffer();
+  float* source =
+      inputBuffer->directReadFrom(&m_inputReadIndex, WEBAUDIO_BLOCK_SIZE);
+  process(source);
+}
+
+void ReverbConvolverStage::process(const float* source) {
+  MOZ_ASSERT(source);
+  if (!source) return;
+
+  // Now, run the convolution (into the delay buffer).
+  // An expensive FFT will happen every fftSize / 2 frames.
+  const float* output = m_fftConvolver->process(m_fftKernel.get(), source);
+
+  // Now accumulate into reverb's accumulation buffer.
+  m_accumulationBuffer->accumulate(output, WEBAUDIO_BLOCK_SIZE,
+                                   &m_accumulationReadIndex, m_postDelayLength);
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/ReverbConvolverStage.h b/dom/media/webaudio/blink/ReverbConvolverStage.h
new file mode 100644
index 0000000000..f770e76d45
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbConvolverStage.h
@@ -0,0 +1,84 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef ReverbConvolverStage_h
+#define ReverbConvolverStage_h
+
+#include "FFTConvolver.h"
+
+#include "nsTArray.h"
+#include "mozilla/FFTBlock.h"
+#include "mozilla/MemoryReporting.h"
+#include "mozilla/UniquePtr.h"
+
+namespace WebCore {
+
+using mozilla::FFTBlock;
+
+class ReverbAccumulationBuffer;
+class ReverbConvolver;
+
+// A ReverbConvolverStage represents the convolution associated with a
+// sub-section of a large impulse response. It incorporates a delay line to
+// account for the offset of the sub-section within the larger impulse response.
+class ReverbConvolverStage {
+ public:
+  // renderPhase is useful to know so that we can manipulate the pre versus post
+  // delay so that stages will perform their heavy work (FFT processing) on
+  // different slices to balance the load in a real-time thread.
+  ReverbConvolverStage(const float* impulseResponse, size_t responseLength,
+                       size_t reverbTotalLatency, size_t stageOffset,
+                       size_t stageLength, size_t fftSize, size_t renderPhase,
+                       ReverbAccumulationBuffer*);
+
+  // |source| must point to an array of WEBAUDIO_BLOCK_SIZE elements.
+  void process(const float* source);
+
+  void processInBackground(ReverbConvolver* convolver);
+
+  // Useful for background processing
+  int inputReadIndex() const { return m_inputReadIndex; }
+
+  size_t sizeOfIncludingThis(mozilla::MallocSizeOf aMallocSizeOf) const;
+
+ private:
+  mozilla::UniquePtr<FFTBlock> m_fftKernel;
+  mozilla::UniquePtr<FFTConvolver> m_fftConvolver;
+
+  ReverbAccumulationBuffer* m_accumulationBuffer;
+  size_t m_accumulationReadIndex;
+  int m_inputReadIndex;
+
+  size_t m_postDelayLength;
+
+  nsTArray<float> m_temporaryBuffer;
+};
+
+}  // namespace WebCore
+
+#endif  // ReverbConvolverStage_h
diff --git a/dom/media/webaudio/blink/ReverbInputBuffer.cpp b/dom/media/webaudio/blink/ReverbInputBuffer.cpp
new file mode 100644
index 0000000000..d6a69263c9
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbInputBuffer.cpp
@@ -0,0 +1,85 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "ReverbInputBuffer.h"
+#include "mozilla/PodOperations.h"
+
+using namespace mozilla;
+
+namespace WebCore {
+
+ReverbInputBuffer::ReverbInputBuffer(size_t length) : m_writeIndex(0) {
+  m_buffer.SetLength(length);
+  PodZero(m_buffer.Elements(), length);
+}
+
+void ReverbInputBuffer::write(const float* sourceP, size_t numberOfFrames) {
+  // m_writeIndex is atomic and checked by other threads, so only touch
+  // it at the start and end.
+  size_t bufferLength = m_buffer.Length();
+  size_t index = m_writeIndex;
+  size_t newIndex = index + numberOfFrames;
+
+  MOZ_RELEASE_ASSERT(newIndex <= bufferLength);
+
+  memcpy(m_buffer.Elements() + index, sourceP, sizeof(float) * numberOfFrames);
+
+  if (newIndex >= bufferLength) {
+    newIndex = 0;
+  }
+  m_writeIndex = newIndex;
+}
+
+float* ReverbInputBuffer::directReadFrom(int* readIndex,
+                                         size_t numberOfFrames) {
+  size_t bufferLength = m_buffer.Length();
+  bool isPointerGood = readIndex && *readIndex >= 0 &&
+                       *readIndex + numberOfFrames <= bufferLength;
+  MOZ_ASSERT(isPointerGood);
+  if (!isPointerGood) {
+    // Should never happen in practice but return pointer to start of buffer
+    // (avoid crash)
+    if (readIndex) *readIndex = 0;
+    return m_buffer.Elements();
+  }
+
+  float* sourceP = m_buffer.Elements();
+  float* p = sourceP + *readIndex;
+
+  // Update readIndex
+  *readIndex = (*readIndex + numberOfFrames) % bufferLength;
+
+  return p;
+}
+
+void ReverbInputBuffer::reset() {
+  PodZero(m_buffer.Elements(), m_buffer.Length());
+  m_writeIndex = 0;
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/ReverbInputBuffer.h b/dom/media/webaudio/blink/ReverbInputBuffer.h
new file mode 100644
index 0000000000..93ef97a010
--- /dev/null
+++ b/dom/media/webaudio/blink/ReverbInputBuffer.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2010 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef ReverbInputBuffer_h
+#define ReverbInputBuffer_h
+
+#include "nsTArray.h"
+#include "mozilla/Atomics.h"
+#include "mozilla/MemoryReporting.h"
+
+namespace WebCore {
+
+// ReverbInputBuffer is used to buffer input samples for deferred processing by
+// the background threads.
+class ReverbInputBuffer {
+ public:
+  explicit ReverbInputBuffer(size_t length);
+
+  // The realtime audio thread keeps writing samples here.
+  // The assumption is that the buffer's length is evenly divisible by
+  // numberOfFrames (for nearly all cases this will be fine).
+  // FIXME: remove numberOfFrames restriction...
+  void write(const float* sourceP, size_t numberOfFrames);
+
+  // Background threads can call this to check if there's anything to read...
+  size_t writeIndex() const { return m_writeIndex; }
+
+  // The individual background threads read here (and hope that they can keep up
+  // with the buffer writing). readIndex is updated with the next readIndex to
+  // read from... The assumption is that the buffer's length is evenly divisible
+  // by numberOfFrames.
+  // FIXME: remove numberOfFrames restriction...
+  float* directReadFrom(int* readIndex, size_t numberOfFrames);
+
+  void reset();
+
+  size_t sizeOfExcludingThis(mozilla::MallocSizeOf aMallocSizeOf) const {
+    return m_buffer.ShallowSizeOfExcludingThis(aMallocSizeOf);
+  }
+
+ private:
+  nsTArray<float> m_buffer;
+  mozilla::Atomic<size_t, mozilla::ReleaseAcquire> m_writeIndex;
+};
+
+}  // namespace WebCore
+
+#endif  // ReverbInputBuffer_h
diff --git a/dom/media/webaudio/blink/ZeroPole.cpp b/dom/media/webaudio/blink/ZeroPole.cpp
new file mode 100644
index 0000000000..4d0888d758
--- /dev/null
+++ b/dom/media/webaudio/blink/ZeroPole.cpp
@@ -0,0 +1,83 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "ZeroPole.h"
+
+#include "DenormalDisabler.h"
+
+#include <cmath>
+#include <float.h>
+
+namespace WebCore {
+
+void ZeroPole::process(const float* source, float* destination,
+                       int framesToProcess) {
+  float zero = m_zero;
+  float pole = m_pole;
+
+  // Gain compensation to make 0dB @ 0Hz
+  const float k1 = 1 / (1 - zero);
+  const float k2 = 1 - pole;
+
+  // Member variables to locals.
+  float lastX = m_lastX;
+  float lastY = m_lastY;
+
+  for (int i = 0; i < framesToProcess; ++i) {
+    float input = source[i];
+
+    // Zero
+    float output1 = k1 * (input - zero * lastX);
+    lastX = input;
+
+    // Pole
+    float output2 = k2 * output1 + pole * lastY;
+    lastY = output2;
+
+    destination[i] = output2;
+  }
+
+// Locals to member variables. Flush denormals here so we don't
+// slow down the inner loop above.
+#ifndef HAVE_DENORMAL
+  if (lastX == 0.0f && lastY != 0.0f && fabsf(lastY) < FLT_MIN) {
+    // Flush future values to zero (until there is new input).
+    lastY = 0.0;
+
+    // Flush calculated values.
+    for (int i = framesToProcess; i-- && fabsf(destination[i]) < FLT_MIN;) {
+      destination[i] = 0.0f;
+    }
+  }
+#endif
+
+  m_lastX = lastX;
+  m_lastY = lastY;
+}
+
+}  // namespace WebCore
diff --git a/dom/media/webaudio/blink/ZeroPole.h b/dom/media/webaudio/blink/ZeroPole.h
new file mode 100644
index 0000000000..07f3f04f53
--- /dev/null
+++ b/dom/media/webaudio/blink/ZeroPole.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (C) 2011 Google Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * 1.  Redistributions of source code must retain the above copyright
+ *     notice, this list of conditions and the following disclaimer.
+ * 2.  Redistributions in binary form must reproduce the above copyright
+ *     notice, this list of conditions and the following disclaimer in the
+ *     documentation and/or other materials provided with the distribution.
+ * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
+ *     its contributors may be used to endorse or promote products derived
+ *     from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
+ * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
+ * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
+ * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
+ * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
+ * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
+ * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
+ * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef ZeroPole_h
+#define ZeroPole_h
+
+namespace WebCore {
+
+// ZeroPole is a simple filter with one zero and one pole.
+
+class ZeroPole {
+ public:
+  ZeroPole() : m_zero(0), m_pole(0), m_lastX(0), m_lastY(0) {}
+
+  void process(const float* source, float* destination, int framesToProcess);
+
+  // Reset filter state.
+  void reset() {
+    m_lastX = 0;
+    m_lastY = 0;
+  }
+
+  void setZero(float zero) { m_zero = zero; }
+  void setPole(float pole) { m_pole = pole; }
+
+  float zero() const { return m_zero; }
+  float pole() const { return m_pole; }
+
+ private:
+  float m_zero;
+  float m_pole;
+  float m_lastX;
+  float m_lastY;
+};
+
+}  // namespace WebCore
+
+#endif  // ZeroPole_h
diff --git a/dom/media/webaudio/blink/moz.build b/dom/media/webaudio/blink/moz.build
new file mode 100644
index 0000000000..5f00d01ed3
--- /dev/null
+++ b/dom/media/webaudio/blink/moz.build
@@ -0,0 +1,36 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# 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/.
+
+UNIFIED_SOURCES += [
+    "Biquad.cpp",
+    "DynamicsCompressor.cpp",
+    "DynamicsCompressorKernel.cpp",
+    "FFTConvolver.cpp",
+    "HRTFDatabase.cpp",
+    "HRTFDatabaseLoader.cpp",
+    "HRTFElevation.cpp",
+    "HRTFKernel.cpp",
+    "HRTFPanner.cpp",
+    "IIRFilter.cpp",
+    "PeriodicWave.cpp",
+    "Reverb.cpp",
+    "ReverbAccumulationBuffer.cpp",
+    "ReverbConvolver.cpp",
+    "ReverbConvolverStage.cpp",
+    "ReverbInputBuffer.cpp",
+    "ZeroPole.cpp",
+]
+
+# Are we targeting x86 or x64?  If so, build SSE2 files.
+if CONFIG["INTEL_ARCHITECTURE"]:
+    DEFINES["USE_SSE2"] = True
+
+include("/ipc/chromium/chromium-config.mozbuild")
+
+FINAL_LIBRARY = "xul"
+LOCAL_INCLUDES += [
+    "/dom/media/webaudio",
+]
diff --git a/dom/media/webaudio/moz.build b/dom/media/webaudio/moz.build
new file mode 100644
index 0000000000..2e82d3daa8
--- /dev/null
+++ b/dom/media/webaudio/moz.build
@@ -0,0 +1,153 @@
+# -*- Mode: python; indent-tabs-mode: nil; tab-width: 40 -*-
+# vim: set filetype=python:
+# 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/.
+
+with Files("*"):
+    BUG_COMPONENT = ("Core", "Web Audio")
+
+DIRS += ["blink"]
+
+MOCHITEST_MANIFESTS += [
+    "test/blink/mochitest.toml",
+    "test/mochitest.toml",
+    "test/mochitest_audio.toml",
+    "test/mochitest_bugs.toml",
+    "test/mochitest_media.toml",
+]
+
+TEST_HARNESS_FILES.testing.mochitest.tests.dom.media.webaudio.test.blink += [
+    "test/blink/audio-testing.js",
+    "test/blink/convolution-testing.js",
+    "test/blink/panner-model-testing.js",
+]
+
+EXPORTS += [
+    "AlignedTArray.h",
+    "AudioBlock.h",
+    "AudioEventTimeline.h",
+    "AudioNodeEngine.h",
+    "AudioNodeExternalInputTrack.h",
+    "AudioNodeTrack.h",
+    "AudioParamTimeline.h",
+    "MediaBufferDecoder.h",
+    "ThreeDPoint.h",
+    "WebAudioUtils.h",
+]
+
+EXPORTS.mozilla += [
+    "FFTBlock.h",
+    "MediaStreamAudioDestinationNode.h",
+]
+
+EXPORTS.mozilla.dom += [
+    "AnalyserNode.h",
+    "AudioBuffer.h",
+    "AudioBufferSourceNode.h",
+    "AudioContext.h",
+    "AudioDestinationNode.h",
+    "AudioListener.h",
+    "AudioNode.h",
+    "AudioParam.h",
+    "AudioParamDescriptorMap.h",
+    "AudioParamMap.h",
+    "AudioProcessingEvent.h",
+    "AudioScheduledSourceNode.h",
+    "AudioWorkletGlobalScope.h",
+    "AudioWorkletNode.h",
+    "AudioWorkletProcessor.h",
+    "BiquadFilterNode.h",
+    "ChannelMergerNode.h",
+    "ChannelSplitterNode.h",
+    "ConstantSourceNode.h",
+    "ConvolverNode.h",
+    "DelayNode.h",
+    "DynamicsCompressorNode.h",
+    "GainNode.h",
+    "IIRFilterNode.h",
+    "MediaElementAudioSourceNode.h",
+    "MediaStreamAudioDestinationNode.h",
+    "MediaStreamAudioSourceNode.h",
+    "MediaStreamTrackAudioSourceNode.h",
+    "OscillatorNode.h",
+    "PannerNode.h",
+    "PeriodicWave.h",
+    "ScriptProcessorNode.h",
+    "StereoPannerNode.h",
+    "WaveShaperNode.h",
+]
+
+UNIFIED_SOURCES += [
+    "AnalyserNode.cpp",
+    "AudioBlock.cpp",
+    "AudioBuffer.cpp",
+    "AudioBufferSourceNode.cpp",
+    "AudioContext.cpp",
+    "AudioDestinationNode.cpp",
+    "AudioEventTimeline.cpp",
+    "AudioListener.cpp",
+    "AudioNode.cpp",
+    "AudioNodeEngine.cpp",
+    "AudioNodeExternalInputTrack.cpp",
+    "AudioNodeTrack.cpp",
+    "AudioParam.cpp",
+    "AudioParamMap.cpp",
+    "AudioProcessingEvent.cpp",
+    "AudioScheduledSourceNode.cpp",
+    "AudioWorkletGlobalScope.cpp",
+    "AudioWorkletImpl.cpp",
+    "AudioWorkletNode.cpp",
+    "AudioWorkletProcessor.cpp",
+    "BiquadFilterNode.cpp",
+    "ChannelMergerNode.cpp",
+    "ChannelSplitterNode.cpp",
+    "ConstantSourceNode.cpp",
+    "ConvolverNode.cpp",
+    "DelayBuffer.cpp",
+    "DelayNode.cpp",
+    "DynamicsCompressorNode.cpp",
+    "FFTBlock.cpp",
+    "GainNode.cpp",
+    "IIRFilterNode.cpp",
+    "MediaBufferDecoder.cpp",
+    "MediaElementAudioSourceNode.cpp",
+    "MediaStreamAudioDestinationNode.cpp",
+    "MediaStreamAudioSourceNode.cpp",
+    "MediaStreamTrackAudioSourceNode.cpp",
+    "OscillatorNode.cpp",
+    "PannerNode.cpp",
+    "PeriodicWave.cpp",
+    "ScriptProcessorNode.cpp",
+    "StereoPannerNode.cpp",
+    "ThreeDPoint.cpp",
+    "WaveShaperNode.cpp",
+    "WebAudioUtils.cpp",
+]
+
+if CONFIG["TARGET_CPU"] == "aarch64" or CONFIG["BUILD_ARM_NEON"]:
+    DEFINES["USE_NEON"] = True
+    LOCAL_INCLUDES += ["/third_party/xsimd/include"]
+    SOURCES += ["AudioNodeEngineNEON.cpp"]
+    SOURCES["AudioNodeEngineNEON.cpp"].flags += CONFIG["NEON_FLAGS"]
+    if CONFIG["BUILD_ARM_NEON"]:
+        LOCAL_INCLUDES += ["/media/openmax_dl/dl/api/"]
+
+# Are we targeting x86 or x64?  If so, build SSEX files.
+if CONFIG["INTEL_ARCHITECTURE"]:
+    DEFINES["USE_SSE2"] = True
+    SOURCES += ["AudioNodeEngineSSE2.cpp"]
+    LOCAL_INCLUDES += ["/third_party/xsimd/include"]
+    SOURCES["AudioNodeEngineSSE2.cpp"].flags += CONFIG["SSE2_FLAGS"]
+    if CONFIG["SSE4_2_FLAGS"] and CONFIG["FMA_FLAGS"]:
+        DEFINES["USE_SSE4_2"] = True
+        DEFINES["USE_FMA3"] = True
+        SOURCES += ["AudioNodeEngineSSE4_2_FMA3.cpp"]
+        SOURCES["AudioNodeEngineSSE4_2_FMA3.cpp"].flags += (
+            CONFIG["SSE4_2_FLAGS"] + CONFIG["FMA_FLAGS"]
+        )
+
+include("/ipc/chromium/chromium-config.mozbuild")
+
+FINAL_LIBRARY = "xul"
+LOCAL_INCLUDES += [".."]
diff --git a/dom/media/webaudio/test/1856145.ogg b/dom/media/webaudio/test/1856145.ogg
new file mode 100644
index 0000000000..771fad7899
--- /dev/null
+++ b/dom/media/webaudio/test/1856145.ogg
diff --git a/dom/media/webaudio/test/8kHz-320kbps-6ch.aac b/dom/media/webaudio/test/8kHz-320kbps-6ch.aac
new file mode 100644
index 0000000000..8981d40dfd
--- /dev/null
+++ b/dom/media/webaudio/test/8kHz-320kbps-6ch.aac
diff --git a/dom/media/webaudio/test/audio-expected.wav b/dom/media/webaudio/test/audio-expected.wav
new file mode 100644
index 0000000000..384f76318a
--- /dev/null
+++ b/dom/media/webaudio/test/audio-expected.wav
diff --git a/dom/media/webaudio/test/audio-mono-expected-2.wav b/dom/media/webaudio/test/audio-mono-expected-2.wav
new file mode 100644
index 0000000000..68c90dfa1e
--- /dev/null
+++ b/dom/media/webaudio/test/audio-mono-expected-2.wav
diff --git a/dom/media/webaudio/test/audio-mono-expected.wav b/dom/media/webaudio/test/audio-mono-expected.wav
new file mode 100644
index 0000000000..bf00e5cdf2
--- /dev/null
+++ b/dom/media/webaudio/test/audio-mono-expected.wav
diff --git a/dom/media/webaudio/test/audio-quad.wav b/dom/media/webaudio/test/audio-quad.wav
new file mode 100644
index 0000000000..093f0197ae
--- /dev/null
+++ b/dom/media/webaudio/test/audio-quad.wav
diff --git a/dom/media/webaudio/test/audio.ogv b/dom/media/webaudio/test/audio.ogv
new file mode 100644
index 0000000000..68dee3cf2b
--- /dev/null
+++ b/dom/media/webaudio/test/audio.ogv
diff --git a/dom/media/webaudio/test/audioBufferSourceNodeDetached_worker.js b/dom/media/webaudio/test/audioBufferSourceNodeDetached_worker.js
new file mode 100644
index 0000000000..2a5a4bff89
--- /dev/null
+++ b/dom/media/webaudio/test/audioBufferSourceNodeDetached_worker.js
@@ -0,0 +1,3 @@
+onmessage = function (event) {
+  postMessage("Pong");
+};
diff --git a/dom/media/webaudio/test/audiovideo.mp4 b/dom/media/webaudio/test/audiovideo.mp4
new file mode 100644
index 0000000000..fe93122d29
--- /dev/null
+++ b/dom/media/webaudio/test/audiovideo.mp4
diff --git a/dom/media/webaudio/test/blink/README b/dom/media/webaudio/test/blink/README
new file mode 100644
index 0000000000..1d819221fd
--- /dev/null
+++ b/dom/media/webaudio/test/blink/README
@@ -0,0 +1,9 @@
+This directory contains tests originally borrowed from the Blink Web Audio test
+suite.
+
+The process of borrowing tests from Blink is as follows:
+
+* Import the pristine file from the Blink repo, noting the revision in the
+  commit message.
+* Modify the test files to turn the LayoutTest into a mochitest-plain and add
+* them to the test suite in a separate commit.
diff --git a/dom/media/webaudio/test/blink/audio-testing.js b/dom/media/webaudio/test/blink/audio-testing.js
new file mode 100644
index 0000000000..c66d32c7f2
--- /dev/null
+++ b/dom/media/webaudio/test/blink/audio-testing.js
@@ -0,0 +1,192 @@
+if (window.testRunner)
+    testRunner.overridePreference("WebKitWebAudioEnabled", "1");
+
+function writeString(s, a, offset) {
+    for (var i = 0; i < s.length; ++i) {
+        a[offset + i] = s.charCodeAt(i);
+    }
+}
+
+function writeInt16(n, a, offset) {
+    n = Math.floor(n);
+   
+    var b1 = n & 255;
+    var b2 = (n >> 8) & 255;
+
+    a[offset + 0] = b1;
+    a[offset + 1] = b2;
+}
+
+function writeInt32(n, a, offset) {
+    n = Math.floor(n);
+    var b1 = n & 255;
+    var b2 = (n >> 8) & 255;
+    var b3 = (n >> 16) & 255;
+    var b4 = (n >> 24) & 255;
+
+    a[offset + 0] = b1;
+    a[offset + 1] = b2;
+    a[offset + 2] = b3;
+    a[offset + 3] = b4;
+}
+
+function writeAudioBuffer(audioBuffer, a, offset) {
+    var n = audioBuffer.length;
+    var channels = audioBuffer.numberOfChannels;
+   
+    for (var i = 0; i < n; ++i) {
+        for (var k = 0; k < channels; ++k) {
+            var buffer = audioBuffer.getChannelData(k);
+            var sample = buffer[i] * 32768.0;
+
+            // Clip samples to the limitations of 16-bit.
+            // If we don't do this then we'll get nasty wrap-around distortion.
+            if (sample < -32768)
+                sample = -32768;
+            if (sample > 32767)
+                sample = 32767;
+
+            writeInt16(sample, a, offset);
+            offset += 2;
+        }
+    }
+}
+
+function createWaveFileData(audioBuffer) {
+    var frameLength = audioBuffer.length;
+    var numberOfChannels = audioBuffer.numberOfChannels;
+    var sampleRate = audioBuffer.sampleRate;
+    var bitsPerSample = 16;
+    var byteRate = sampleRate * numberOfChannels * bitsPerSample/8;
+    var blockAlign = numberOfChannels * bitsPerSample/8;
+    var wavDataByteLength = frameLength * numberOfChannels * 2; // 16-bit audio
+    var headerByteLength = 44;
+    var totalLength = headerByteLength + wavDataByteLength;
+
+    var waveFileData = new Uint8Array(totalLength);
+   
+    var subChunk1Size = 16; // for linear PCM
+    var subChunk2Size = wavDataByteLength;
+    var chunkSize = 4 + (8 + subChunk1Size) + (8 + subChunk2Size);
+
+    writeString("RIFF", waveFileData, 0);
+    writeInt32(chunkSize, waveFileData, 4);
+    writeString("WAVE", waveFileData, 8);
+    writeString("fmt ", waveFileData, 12);
+   
+    writeInt32(subChunk1Size, waveFileData, 16);      // SubChunk1Size (4)
+    writeInt16(1, waveFileData, 20);                  // AudioFormat (2)
+    writeInt16(numberOfChannels, waveFileData, 22);   // NumChannels (2)
+    writeInt32(sampleRate, waveFileData, 24);         // SampleRate (4)
+    writeInt32(byteRate, waveFileData, 28);           // ByteRate (4)
+    writeInt16(blockAlign, waveFileData, 32);         // BlockAlign (2)
+    writeInt32(bitsPerSample, waveFileData, 34);      // BitsPerSample (4)
+                                                     
+    writeString("data", waveFileData, 36);            
+    writeInt32(subChunk2Size, waveFileData, 40);      // SubChunk2Size (4)
+   
+    // Write actual audio data starting at offset 44.
+    writeAudioBuffer(audioBuffer, waveFileData, 44);
+   
+    return waveFileData;
+}
+
+function createAudioData(audioBuffer) {
+    return createWaveFileData(audioBuffer);
+}
+
+function finishAudioTest(event) {
+    var audioData = createAudioData(event.renderedBuffer);
+    testRunner.setAudioData(audioData);
+    testRunner.notifyDone();
+}
+
+// Create an impulse in a buffer of length sampleFrameLength
+function createImpulseBuffer(context, sampleFrameLength) {
+    var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+    var n = audioBuffer.length;
+    var dataL = audioBuffer.getChannelData(0);
+
+    for (var k = 0; k < n; ++k) {
+        dataL[k] = 0;
+    }
+    dataL[0] = 1;
+
+    return audioBuffer;
+}
+
+// Create a buffer of the given length with a linear ramp having values 0 <= x < 1.
+function createLinearRampBuffer(context, sampleFrameLength) {
+    var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+    var n = audioBuffer.length;
+    var dataL = audioBuffer.getChannelData(0);
+
+    for (var i = 0; i < n; ++i)
+        dataL[i] = i / n;
+
+    return audioBuffer;
+}
+
+// Create a buffer of the given length having a constant value.
+function createConstantBuffer(context, sampleFrameLength, constantValue) {
+    var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+    var n = audioBuffer.length;
+    var dataL = audioBuffer.getChannelData(0);
+
+    for (var i = 0; i < n; ++i)
+        dataL[i] = constantValue;
+
+    return audioBuffer;
+}
+
+// Create a stereo impulse in a buffer of length sampleFrameLength
+function createStereoImpulseBuffer(context, sampleFrameLength) {
+    var audioBuffer = context.createBuffer(2, sampleFrameLength, context.sampleRate);
+    var n = audioBuffer.length;
+    var dataL = audioBuffer.getChannelData(0);
+    var dataR = audioBuffer.getChannelData(1);
+
+    for (var k = 0; k < n; ++k) {
+        dataL[k] = 0;
+        dataR[k] = 0;
+    }
+    dataL[0] = 1;
+    dataR[0] = 1;
+
+    return audioBuffer;
+}
+
+// Convert time (in seconds) to sample frames.
+function timeToSampleFrame(time, sampleRate) {
+    return Math.floor(0.5 + time * sampleRate);
+}
+
+// Compute the number of sample frames consumed by start with
+// the specified |grainOffset|, |duration|, and |sampleRate|.
+function grainLengthInSampleFrames(grainOffset, duration, sampleRate) {
+    var startFrame = timeToSampleFrame(grainOffset, sampleRate);
+    var endFrame = timeToSampleFrame(grainOffset + duration, sampleRate);
+
+    return endFrame - startFrame;
+}
+
+// True if the number is not an infinity or NaN
+function isValidNumber(x) {
+    return !isNaN(x) && (x != Infinity) && (x != -Infinity);
+}
+
+function shouldThrowTypeError(func, text) {
+    var ok = false;
+    try {
+        func();
+    } catch (e) {
+        if (e instanceof TypeError) {
+            ok = true;
+        }
+    }
+    if (ok) {
+        testPassed(text + " threw TypeError.");
+    } else {
+        testFailed(text + " should throw TypeError.");
+    }
+}
diff --git a/dom/media/webaudio/test/blink/biquad-filters.js b/dom/media/webaudio/test/blink/biquad-filters.js
new file mode 100644
index 0000000000..06fff98b18
--- /dev/null
+++ b/dom/media/webaudio/test/blink/biquad-filters.js
@@ -0,0 +1,368 @@
+// Taken from WebKit/LayoutTests/webaudio/resources/biquad-filters.js
+
+// A biquad filter has a z-transform of
+// H(z) = (b0 + b1 / z + b2 / z^2) / (1 + a1 / z + a2 / z^2)
+//
+// The formulas for the various filters were taken from
+// http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt.
+
+
+// Lowpass filter.
+function createLowpassFilter(freq, q, gain) {
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+
+    if (freq == 1) {
+        // The formula below works, except for roundoff.  When freq = 1,
+        // the filter is just a wire, so hardwire the coefficients.
+        b0 = 1;
+        b1 = 0;
+        b2 = 0;
+        a0 = 1;
+        a1 = 0;
+        a2 = 0;
+    } else {
+        var w0 = Math.PI * freq;
+        var alpha = 0.5 * Math.sin(w0) / Math.pow(10, q / 20);
+        var cos_w0 = Math.cos(w0);
+
+        b0 = 0.5 * (1 - cos_w0);
+        b1 = 1 - cos_w0;
+        b2 = b0;
+        a0 = 1 + alpha;
+        a1 = -2.0 * cos_w0;
+        a2 = 1 - alpha;
+    }
+
+    return normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+}
+
+function createHighpassFilter(freq, q, gain) {
+    var b0;
+    var b1;
+    var b2;
+    var a1;
+    var a2;
+
+    if (freq == 1) {
+        // The filter is 0
+        b0 = 0;
+        b1 = 0;
+        b2 = 0;
+        a0 = 1;
+        a1 = 0;
+        a2 = 0;
+    } else if (freq == 0) {
+        // The filter is 1.  Computation of coefficients below is ok, but
+        // there's a pole at 1 and a zero at 1, so round-off could make
+        // the filter unstable.
+        b0 = 1;
+        b1 = 0;
+        b2 = 0;
+        a0 = 1;
+        a1 = 0;
+        a2 = 0;
+    } else {
+        var w0 = Math.PI * freq;
+        var alpha = 0.5 * Math.sin(w0) / Math.pow(10, q / 20);
+        var cos_w0 = Math.cos(w0);
+
+        b0 = 0.5 * (1 + cos_w0);
+        b1 = -1 - cos_w0;
+        b2 = b0;
+        a0 = 1 + alpha;
+        a1 = -2.0 * cos_w0;
+        a2 = 1 - alpha;
+    }
+
+    return normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+}
+
+function normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2) {
+    var scale = 1 / a0;
+
+    return {b0 : b0 * scale,
+            b1 : b1 * scale,
+            b2 : b2 * scale,
+            a1 : a1 * scale,
+            a2 : a2 * scale};
+}
+
+function createBandpassFilter(freq, q, gain) {
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+    var coef;
+
+    if (freq > 0 && freq < 1) {
+        var w0 = Math.PI * freq;
+        if (q > 0) {
+            var alpha = Math.sin(w0) / (2 * q);
+            var k = Math.cos(w0);
+
+            b0 = alpha;
+            b1 = 0;
+            b2 = -alpha;
+            a0 = 1 + alpha;
+            a1 = -2 * k;
+            a2 = 1 - alpha;
+
+            coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+        } else {
+            // q = 0, and frequency is not 0 or 1.  The above formula has a
+            // divide by zero problem.  The limit of the z-transform as q
+            // approaches 0 is 1, so set the filter that way.
+            coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+        }
+    } else {
+        // When freq = 0 or 1, the z-transform is identically 0,
+        // independent of q.
+        coef = {b0 : 0, b1 : 0, b2 : 0, a1 : 0, a2 : 0}
+    }
+
+    return coef;
+}
+
+function createLowShelfFilter(freq, q, gain) {
+    // q not used
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+    var coef;
+
+    var S = 1;
+    var A = Math.pow(10, gain / 40);
+
+    if (freq == 1) {
+        // The filter is just a constant gain
+        coef = {b0 : A * A, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    } else if (freq == 0) {
+        // The filter is 1
+        coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    } else {
+        var w0 = Math.PI * freq;
+        var alpha = 1 / 2 * Math.sin(w0) * Math.sqrt((A + 1 / A) * (1 / S - 1) + 2);
+        var k = Math.cos(w0);
+        var k2 = 2 * Math.sqrt(A) * alpha;
+        var Ap1 = A + 1;
+        var Am1 = A - 1;
+
+        b0 = A * (Ap1 - Am1 * k + k2);
+        b1 = 2 * A * (Am1 - Ap1 * k);
+        b2 = A * (Ap1 - Am1 * k - k2);
+        a0 = Ap1 + Am1 * k + k2;
+        a1 = -2 * (Am1 + Ap1 * k);
+        a2 = Ap1 + Am1 * k - k2;
+        coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+    }
+
+    return coef;
+}
+
+function createHighShelfFilter(freq, q, gain) {
+    // q not used
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+    var coef;
+
+    var A = Math.pow(10, gain / 40);
+
+    if (freq == 1) {
+        // When freq = 1, the z-transform is 1
+        coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    } else if (freq > 0) {
+        var w0 = Math.PI * freq;
+        var S = 1;
+        var alpha = 0.5 * Math.sin(w0) * Math.sqrt((A + 1 / A) * (1 / S - 1) + 2);
+        var k = Math.cos(w0);
+        var k2 = 2 * Math.sqrt(A) * alpha;
+        var Ap1 = A + 1;
+        var Am1 = A - 1;
+
+        b0 = A * (Ap1 + Am1 * k + k2);
+        b1 = -2 * A * (Am1 + Ap1 * k);
+        b2 = A * (Ap1 + Am1 * k - k2);
+        a0 = Ap1 - Am1 * k + k2;
+        a1 = 2 * (Am1 - Ap1*k);
+        a2 = Ap1 - Am1 * k-k2;
+
+        coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+    } else {
+        // When freq = 0, the filter is just a gain
+        coef = {b0 : A * A, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    }
+
+    return coef;
+}
+
+function createPeakingFilter(freq, q, gain) {
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+    var coef;
+
+    var A = Math.pow(10, gain / 40);
+
+    if (freq > 0 && freq < 1) {
+        if (q > 0) {
+            var w0 = Math.PI * freq;
+            var alpha = Math.sin(w0) / (2 * q);
+            var k = Math.cos(w0);
+
+            b0 = 1 + alpha * A;
+            b1 = -2 * k;
+            b2 = 1 - alpha * A;
+            a0 = 1 + alpha / A;
+            a1 = -2 * k;
+            a2 = 1 - alpha / A;
+
+            coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+        } else {
+            // q = 0, we have a divide by zero problem in the formulas
+            // above.  But if we look at the z-transform, we see that the
+            // limit as q approaches 0 is A^2.
+            coef = {b0 : A * A, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+        }
+    } else {
+        // freq = 0 or 1, the z-transform is 1
+        coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    }
+
+    return coef;
+}
+
+function createNotchFilter(freq, q, gain) {
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+    var coef;
+
+    if (freq > 0 && freq < 1) {
+        if (q > 0) {
+            var w0 = Math.PI * freq;
+            var alpha = Math.sin(w0) / (2 * q);
+            var k = Math.cos(w0);
+
+            b0 = 1;
+            b1 = -2 * k;
+            b2 = 1;
+            a0 = 1 + alpha;
+            a1 = -2 * k;
+            a2 = 1 - alpha;
+            coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+        } else {
+            // When q = 0, we get a divide by zero above.  The limit of the
+            // z-transform as q approaches 0 is 0, so set the coefficients
+            // appropriately.
+            coef = {b0 : 0, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+        }
+    } else {
+        // When freq = 0 or 1, the z-transform is 1
+        coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    }
+
+    return coef;
+}
+
+function createAllpassFilter(freq, q, gain) {
+    var b0;
+    var b1;
+    var b2;
+    var a0;
+    var a1;
+    var a2;
+    var coef;
+
+    if (freq > 0 && freq < 1) {
+        if (q > 0) {
+            var w0 = Math.PI * freq;
+            var alpha = Math.sin(w0) / (2 * q);
+            var k = Math.cos(w0);
+
+            b0 = 1 - alpha;
+            b1 = -2 * k;
+            b2 = 1 + alpha;
+            a0 = 1 + alpha;
+            a1 = -2 * k;
+            a2 = 1 - alpha;
+            coef = normalizeFilterCoefficients(b0, b1, b2, a0, a1, a2);
+        } else {
+            // q = 0
+            coef = {b0 : -1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+        }
+    } else {
+        coef = {b0 : 1, b1 : 0, b2 : 0, a1 : 0, a2 : 0};
+    }
+
+    return coef;
+}
+
+function filterData(filterCoef, signal, len) {
+    var y = new Array(len);
+    var b0 = filterCoef.b0;
+    var b1 = filterCoef.b1;
+    var b2 = filterCoef.b2;
+    var a1 = filterCoef.a1;
+    var a2 = filterCoef.a2;
+
+    // Prime the pump. (Assumes the signal has length >= 2!)
+    y[0] = b0 * signal[0];
+    y[1] = b0 * signal[1] + b1 * signal[0] - a1 * y[0];
+
+    // Filter all of the signal that we have.
+    for (var k = 2; k < Math.min(signal.length, len); ++k) {
+        y[k] = b0 * signal[k] + b1 * signal[k-1] + b2 * signal[k-2] - a1 * y[k-1] - a2 * y[k-2];
+    }
+
+    // If we need to filter more, but don't have any signal left,
+    // assume the signal is zero.
+    for (var k = signal.length; k < len; ++k) {
+        y[k] = - a1 * y[k-1] - a2 * y[k-2];
+    }
+
+    return y;
+}
+
+// Map the filter type name to a function that computes the filter coefficents for the given filter
+// type.
+var filterCreatorFunction = {"lowpass": createLowpassFilter,
+                             "highpass": createHighpassFilter,
+                             "bandpass": createBandpassFilter,
+                             "lowshelf": createLowShelfFilter,
+                             "highshelf": createHighShelfFilter,
+                             "peaking": createPeakingFilter,
+                             "notch": createNotchFilter,
+                             "allpass": createAllpassFilter};
+
+var filterTypeName = {"lowpass": "Lowpass filter",
+                      "highpass": "Highpass filter",
+                      "bandpass": "Bandpass filter",
+                      "lowshelf": "Lowshelf filter",
+                      "highshelf": "Highshelf filter",
+                      "peaking": "Peaking filter",
+                      "notch": "Notch filter",
+                      "allpass": "Allpass filter"};
+
+function createFilter(filterType, freq, q, gain) {
+    return filterCreatorFunction[filterType](freq, q, gain);
+}
diff --git a/dom/media/webaudio/test/blink/biquad-testing.js b/dom/media/webaudio/test/blink/biquad-testing.js
new file mode 100644
index 0000000000..795adf6012
--- /dev/null
+++ b/dom/media/webaudio/test/blink/biquad-testing.js
@@ -0,0 +1,153 @@
+// Globals, to make testing and debugging easier.
+var context;
+var filter;
+var signal;
+var renderedBuffer;
+var renderedData;
+
+var sampleRate = 44100.0;
+var pulseLengthFrames = .1 * sampleRate;
+
+// Maximum allowed error for the test to succeed.  Experimentally determined.
+var maxAllowedError = 5.9e-8;
+
+// This must be large enough so that the filtered result is
+// essentially zero.  See comments for createTestAndRun.
+var timeStep = .1;
+
+// Maximum number of filters we can process (mostly for setting the
+// render length correctly.)
+var maxFilters = 5;
+
+// How long to render.  Must be long enough for all of the filters we
+// want to test.
+var renderLengthSeconds = timeStep * (maxFilters + 1) ;
+
+var renderLengthSamples = Math.round(renderLengthSeconds * sampleRate);
+
+// Number of filters that will be processed.
+var nFilters;
+
+function createImpulseBuffer(context, length) {
+    var impulse = context.createBuffer(1, length, context.sampleRate);
+    var data = impulse.getChannelData(0);
+    for (var k = 1; k < data.length; ++k) {
+        data[k] = 0;
+    }
+    data[0] = 1;
+
+    return impulse;
+}
+
+
+function createTestAndRun(context, filterType, filterParameters) {
+    // To test the filters, we apply a signal (an impulse) to each of
+    // the specified filters, with each signal starting at a different
+    // time.  The output of the filters is summed together at the
+    // output.  Thus for filter k, the signal input to the filter
+    // starts at time k * timeStep.  For this to work well, timeStep
+    // must be large enough for the output of each filter to have
+    // decayed to zero with timeStep seconds.  That way the filter
+    // outputs don't interfere with each other.
+
+    nFilters = Math.min(filterParameters.length, maxFilters);
+
+    signal = new Array(nFilters);
+    filter = new Array(nFilters);
+
+    impulse = createImpulseBuffer(context, pulseLengthFrames);
+
+    // Create all of the signal sources and filters that we need.
+    for (var k = 0; k < nFilters; ++k) {
+        signal[k] = context.createBufferSource();
+        signal[k].buffer = impulse;
+
+        filter[k] = context.createBiquadFilter();
+        filter[k].type = filterType;
+        filter[k].frequency.value = context.sampleRate / 2 * filterParameters[k].cutoff;
+        filter[k].detune.value = (filterParameters[k].detune === undefined) ? 0 : filterParameters[k].detune;
+        filter[k].Q.value = filterParameters[k].q;
+        filter[k].gain.value = filterParameters[k].gain;
+
+        signal[k].connect(filter[k]);
+        filter[k].connect(context.destination);
+
+        signal[k].start(timeStep * k);
+    }
+
+    context.oncomplete = checkFilterResponse(filterType, filterParameters);
+    context.startRendering();
+}
+
+function addSignal(dest, src, destOffset) {
+    // Add src to dest at the given dest offset.
+    for (var k = destOffset, j = 0; k < dest.length, j < src.length; ++k, ++j) {
+        dest[k] += src[j];
+    }
+}
+
+function generateReference(filterType, filterParameters) {
+    var result = new Array(renderLengthSamples);
+    var data = new Array(renderLengthSamples);
+    // Initialize the result array and data.
+    for (var k = 0; k < result.length; ++k) {
+        result[k] = 0;
+        data[k] = 0;
+    }
+    // Make data an impulse.
+    data[0] = 1;
+
+    for (var k = 0; k < nFilters; ++k) {
+        // Filter an impulse
+        var detune = (filterParameters[k].detune === undefined) ? 0 : filterParameters[k].detune;
+        var frequency = filterParameters[k].cutoff * Math.pow(2, detune / 1200); // Apply detune, converting from Cents.
+
+        var filterCoef = createFilter(filterType,
+                                      frequency,
+                                      filterParameters[k].q,
+                                      filterParameters[k].gain);
+        var y = filterData(filterCoef, data, renderLengthSamples);
+
+        // Accumulate this filtered data into the final output at the desired offset.
+        addSignal(result, y, timeToSampleFrame(timeStep * k, sampleRate));
+    }
+
+    return result;
+}
+
+function checkFilterResponse(filterType, filterParameters) {
+    return function(event) {
+        renderedBuffer = event.renderedBuffer;
+        renderedData = renderedBuffer.getChannelData(0);
+
+        reference = generateReference(filterType, filterParameters);
+
+        var len = Math.min(renderedData.length, reference.length);
+
+        var success = true;
+
+        // Maximum error between rendered data and expected data
+        var maxError = 0;
+
+        // Sample offset where the maximum error occurred.
+        var maxPosition = 0;
+
+        // Number of infinities or NaNs that occurred in the rendered data.
+        var invalidNumberCount = 0;
+
+        ok(nFilters == filterParameters.length, "Test wanted " + filterParameters.length + " filters but only " + maxFilters + " allowed.");
+
+        compareChannels(renderedData, reference, len, 0, 0, true);
+
+        // Check for bad numbers in the rendered output too.
+        // There shouldn't be any.
+        for (var k = 0; k < len; ++k) {
+            if (!isValidNumber(renderedData[k])) {
+                ++invalidNumberCount;
+            }
+        }
+
+        ok(invalidNumberCount == 0, "Rendered output has " + invalidNumberCount + " infinities or NaNs.");
+        SimpleTest.finish();
+    }
+}
diff --git a/dom/media/webaudio/test/blink/convolution-testing.js b/dom/media/webaudio/test/blink/convolution-testing.js
new file mode 100644
index 0000000000..98ff0c7756
--- /dev/null
+++ b/dom/media/webaudio/test/blink/convolution-testing.js
@@ -0,0 +1,182 @@
+var sampleRate = 44100.0;
+
+var renderLengthSeconds = 8;
+var pulseLengthSeconds = 1;
+var pulseLengthFrames = pulseLengthSeconds * sampleRate;
+
+function createSquarePulseBuffer(context, sampleFrameLength) {
+    var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+
+    var n = audioBuffer.length;
+    var data = audioBuffer.getChannelData(0);
+
+    for (var i = 0; i < n; ++i)
+        data[i] = 1;
+
+    return audioBuffer;
+}
+
+// The triangle buffer holds the expected result of the convolution.
+// It linearly ramps up from 0 to its maximum value (at the center)
+// then linearly ramps down to 0.  The center value corresponds to the
+// point where the two square pulses overlap the most.
+function createTrianglePulseBuffer(context, sampleFrameLength) {
+    var audioBuffer = context.createBuffer(1, sampleFrameLength, context.sampleRate);
+
+    var n = audioBuffer.length;
+    var halfLength = n / 2;
+    var data = audioBuffer.getChannelData(0);
+    
+    for (var i = 0; i < halfLength; ++i)
+        data[i] = i + 1;
+
+    for (var i = halfLength; i < n; ++i)
+        data[i] = n - i - 1;
+
+    return audioBuffer;
+}
+
+function log10(x) {
+  return Math.log(x)/Math.LN10;
+}
+
+function linearToDecibel(x) {
+  return 20*log10(x);
+}
+
+// Verify that the rendered result is very close to the reference
+// triangular pulse.
+function checkTriangularPulse(rendered, reference) {
+    var match = true;
+    var maxDelta = 0;
+    var valueAtMaxDelta = 0;
+    var maxDeltaIndex = 0;
+
+    for (var i = 0; i < reference.length; ++i) {
+        var diff = rendered[i] - reference[i];
+        var x = Math.abs(diff);
+        if (x > maxDelta) {
+            maxDelta = x;
+            valueAtMaxDelta = reference[i];
+            maxDeltaIndex = i;
+        }
+    }
+
+    // allowedDeviationFraction was determined experimentally.  It
+    // is the threshold of the relative error at the maximum
+    // difference between the true triangular pulse and the
+    // rendered pulse.
+    var allowedDeviationDecibels = -129.4;
+    var maxDeviationDecibels = linearToDecibel(maxDelta / valueAtMaxDelta);
+
+    if (maxDeviationDecibels <= allowedDeviationDecibels) {
+        testPassed("Triangular portion of convolution is correct.");
+    } else {
+        testFailed("Triangular portion of convolution is not correct.  Max deviation = " + maxDeviationDecibels + " dB at " + maxDeltaIndex);
+        match = false;
+    }
+
+    return match;
+}        
+
+// Verify that the rendered data is close to zero for the first part
+// of the tail.
+function checkTail1(data, reference, breakpoint) {
+    var isZero = true;
+    var tail1Max = 0;
+
+    for (var i = reference.length; i < reference.length + breakpoint; ++i) {
+        var mag = Math.abs(data[i]);
+        if (mag > tail1Max) {
+            tail1Max = mag;
+        }
+    }
+
+    // Let's find the peak of the reference (even though we know a
+    // priori what it is).
+    var refMax = 0;
+    for (var i = 0; i < reference.length; ++i) {
+      refMax = Math.max(refMax, Math.abs(reference[i]));
+    }
+
+    // This threshold is experimentally determined by examining the
+    // value of tail1MaxDecibels.
+    var threshold1 = -129.7;
+
+    var tail1MaxDecibels = linearToDecibel(tail1Max/refMax);
+    if (tail1MaxDecibels <= threshold1) {
+        testPassed("First part of tail of convolution is sufficiently small.");
+    } else {
+        testFailed("First part of tail of convolution is not sufficiently small: " + tail1MaxDecibels + " dB");
+        isZero = false;
+    }
+
+    return isZero;
+}
+
+// Verify that the second part of the tail of the convolution is
+// exactly zero.
+function checkTail2(data, reference, breakpoint) {
+    var isZero = true;
+    var tail2Max = 0;
+    // For the second part of the tail, the maximum value should be
+    // exactly zero.
+    var threshold2 = 0;
+    for (var i = reference.length + breakpoint; i < data.length; ++i) {
+        if (Math.abs(data[i]) > 0) {
+            isZero = false; 
+            break;
+        }
+    }
+
+    if (isZero) {
+        testPassed("Rendered signal after tail of convolution is silent.");
+    } else {
+        testFailed("Rendered signal after tail of convolution should be silent.");
+    }
+
+    return isZero;
+}
+
+function checkConvolvedResult(trianglePulse) {
+    return function(event) {
+        var renderedBuffer = event.renderedBuffer;
+
+        var referenceData = trianglePulse.getChannelData(0);
+        var renderedData = renderedBuffer.getChannelData(0);
+    
+        var success = true;
+    
+        // Verify the triangular pulse is actually triangular.
+
+        success = success && checkTriangularPulse(renderedData, referenceData);
+        
+        // Make sure that portion after convolved portion is totally
+        // silent.  But round-off prevents this from being completely
+        // true.  At the end of the triangle, it should be close to
+        // zero.  If we go farther out, it should be even closer and
+        // eventually zero.
+
+        // For the tail of the convolution (where the result would be
+        // theoretically zero), we partition the tail into two
+        // parts.  The first is the at the beginning of the tail,
+        // where we tolerate a small but non-zero value.  The second part is
+        // farther along the tail where the result should be zero.
+        
+        // breakpoint is the point dividing the first two tail parts
+        // we're looking at.  Experimentally determined.
+        var breakpoint = 12800;
+
+        success = success && checkTail1(renderedData, referenceData, breakpoint);
+        
+        success = success && checkTail2(renderedData, referenceData, breakpoint);
+        
+        if (success) {
+            testPassed("Test signal was correctly convolved.");
+        } else {
+            testFailed("Test signal was not correctly convolved.");
+        }
+
+        finishJSTest();
+    }
+}
diff --git a/dom/media/webaudio/test/blink/mochitest.toml b/dom/media/webaudio/test/blink/mochitest.toml
new file mode 100644
index 0000000000..2b176d1225
--- /dev/null
+++ b/dom/media/webaudio/test/blink/mochitest.toml
@@ -0,0 +1,36 @@
+[DEFAULT]
+tags = "mtg webaudio"
+subsuite = "media"
+support-files = [
+  "biquad-filters.js",
+  "biquad-testing.js",
+  "../webaudio.js",
+]
+
+["test_biquadFilterNodeAllPass.html"]
+
+["test_biquadFilterNodeAutomation.html"]
+skip-if = ["true"] # Known problems with Biquad automation, e.g. Bug 1155709
+
+["test_biquadFilterNodeBandPass.html"]
+
+["test_biquadFilterNodeGetFrequencyResponse.html"]
+
+["test_biquadFilterNodeHighPass.html"]
+
+["test_biquadFilterNodeHighShelf.html"]
+
+["test_biquadFilterNodeLowPass.html"]
+
+["test_biquadFilterNodeLowShelf.html"]
+
+["test_biquadFilterNodeNotch.html"]
+
+["test_biquadFilterNodePeaking.html"]
+
+["test_biquadFilterNodeTail.html"]
+
+["test_iirFilterNode.html"]
+
+["test_iirFilterNodeGetFrequencyResponse.html"]
+
diff --git a/dom/media/webaudio/test/blink/panner-model-testing.js b/dom/media/webaudio/test/blink/panner-model-testing.js
new file mode 100644
index 0000000000..45460e2768
--- /dev/null
+++ b/dom/media/webaudio/test/blink/panner-model-testing.js
@@ -0,0 +1,210 @@
+var sampleRate = 48000.0;
+
+var numberOfChannels = 1;
+
+// Time step when each panner node starts.
+var timeStep = 0.001;
+
+// Length of the impulse signal.
+var pulseLengthFrames = Math.round(timeStep * sampleRate);
+
+// How many panner nodes to create for the test
+var nodesToCreate = 100;
+
+// Be sure we render long enough for all of our nodes.
+var renderLengthSeconds = timeStep * (nodesToCreate + 1);
+
+// These are global mostly for debugging.
+var context;
+var impulse;
+var bufferSource;
+var panner;
+var position;
+var time;
+      
+var renderedBuffer;
+var renderedLeft;
+var renderedRight;
+
+function createGraph(context, nodeCount) {
+    bufferSource = new Array(nodeCount);
+    panner = new Array(nodeCount);
+    position = new Array(nodeCount);
+    time = new Array(nodeCount);
+    // Angle between panner locations.  (nodeCount - 1 because we want
+    // to include both 0 and 180 deg.
+    var angleStep = Math.PI / (nodeCount - 1);
+
+    if (numberOfChannels == 2) {
+        impulse = createStereoImpulseBuffer(context, pulseLengthFrames);
+    }
+    else
+        impulse = createImpulseBuffer(context, pulseLengthFrames);
+
+    for (var k = 0; k < nodeCount; ++k) {
+        bufferSource[k] = context.createBufferSource();
+        bufferSource[k].buffer = impulse;
+
+        panner[k] = context.createPanner();
+        panner[k].panningModel = "equalpower";
+        panner[k].distanceModel = "linear";
+
+        var angle = angleStep * k;
+        position[k] = {angle : angle, x : Math.cos(angle), z : Math.sin(angle)};
+        panner[k].positionX.value = position[k].x;
+        panner[k].positionZ.value = position[k].z;
+
+        bufferSource[k].connect(panner[k]);
+        panner[k].connect(context.destination);
+
+        // Start the source
+        time[k] = k * timeStep;
+        bufferSource[k].start(time[k]);
+    }
+}
+
+function createTestAndRun(context, nodeCount, numberOfSourceChannels) {
+    numberOfChannels = numberOfSourceChannels;
+
+    createGraph(context, nodeCount);
+
+    context.oncomplete = checkResult;
+    context.startRendering();
+}
+
+// Map our position angle to the azimuth angle (in degrees).
+//
+// An angle of 0 corresponds to an azimuth of 90 deg; pi, to -90 deg.
+function angleToAzimuth(angle) {
+    return 90 - angle * 180 / Math.PI;
+}
+
+// The gain caused by the EQUALPOWER panning model
+function equalPowerGain(angle) {
+    var azimuth = angleToAzimuth(angle);
+
+    if (numberOfChannels == 1) {
+        var panPosition = (azimuth + 90) / 180;
+
+        var gainL = Math.cos(0.5 * Math.PI * panPosition);
+        var gainR = Math.sin(0.5 * Math.PI * panPosition);
+
+        return { left : gainL, right : gainR };
+    } else {
+        if (azimuth <= 0) {
+            var panPosition = (azimuth + 90) / 90;
+    
+            var gainL = 1 + Math.cos(0.5 * Math.PI * panPosition);
+            var gainR = Math.sin(0.5 * Math.PI * panPosition);
+    
+            return { left : gainL, right : gainR };
+        } else {
+            var panPosition = azimuth / 90;
+    
+            var gainL = Math.cos(0.5 * Math.PI * panPosition);
+            var gainR = 1 + Math.sin(0.5 * Math.PI * panPosition);
+    
+            return { left : gainL, right : gainR };
+        }
+    }
+}
+
+function checkResult(event) {
+    renderedBuffer = event.renderedBuffer;
+    renderedLeft = renderedBuffer.getChannelData(0);
+    renderedRight = renderedBuffer.getChannelData(1);
+
+    // The max error we allow between the rendered impulse and the
+    // expected value.  This value is experimentally determined.  Set
+    // to 0 to make the test fail to see what the actual error is.
+    var maxAllowedError = 1.3e-6;
+  
+    var success = true;
+
+    // Number of impulses found in the rendered result.
+    var impulseCount = 0;
+
+    // Max (relative) error and the index of the maxima for the left
+    // and right channels.
+    var maxErrorL = 0;
+    var maxErrorIndexL = 0;
+    var maxErrorR = 0;
+    var maxErrorIndexR = 0;
+
+    // Number of impulses that don't match our expected locations.
+    var timeCount = 0;
+
+    // Locations of where the impulses aren't at the expected locations.
+    var timeErrors = new Array();
+
+    for (var k = 0; k < renderedLeft.length; ++k) {
+        // We assume that the left and right channels start at the same instant.
+        if (renderedLeft[k] != 0 || renderedRight[k] != 0) {
+            // The expected gain for the left and right channels.
+            var pannerGain = equalPowerGain(position[impulseCount].angle);
+            var expectedL = pannerGain.left;
+            var expectedR = pannerGain.right;
+
+            // Absolute error in the gain.
+            var errorL = Math.abs(renderedLeft[k] - expectedL);
+            var errorR = Math.abs(renderedRight[k] - expectedR);
+
+            if (Math.abs(errorL) > maxErrorL) {
+                maxErrorL = Math.abs(errorL);
+                maxErrorIndexL = impulseCount;
+            }
+            if (Math.abs(errorR) > maxErrorR) {
+                maxErrorR = Math.abs(errorR);
+                maxErrorIndexR = impulseCount;
+            }
+
+            // Keep track of the impulses that didn't show up where we
+            // expected them to be.
+            var expectedOffset = timeToSampleFrame(time[impulseCount], sampleRate);
+            if (k != expectedOffset) {
+                timeErrors[timeCount] = { actual : k, expected : expectedOffset};
+                ++timeCount;
+            }
+            ++impulseCount;
+        }
+    }
+
+    if (impulseCount == nodesToCreate) {
+        testPassed("Number of impulses matches the number of panner nodes.");
+    } else {
+        testFailed("Number of impulses is incorrect.  (Found " + impulseCount + " but expected " + nodesToCreate + ")");
+        success = false;
+    }
+
+    if (timeErrors.length > 0) {
+        success = false;
+        testFailed(timeErrors.length + " timing errors found in " + nodesToCreate + " panner nodes.");
+        for (var k = 0; k < timeErrors.length; ++k) {
+            testFailed("Impulse at sample " + timeErrors[k].actual + " but expected " + timeErrors[k].expected);
+        }
+    } else {
+        testPassed("All impulses at expected offsets.");
+    }
+
+    if (maxErrorL <= maxAllowedError) {
+        testPassed("Left channel gain values are correct.");
+    } else {
+        testFailed("Left channel gain values are incorrect.  Max error = " + maxErrorL + " at time " + time[maxErrorIndexL] + " (threshold = " + maxAllowedError + ")");
+        success = false;
+    }
+
+    if (maxErrorR <= maxAllowedError) {
+        testPassed("Right channel gain values are correct.");
+    } else {
+        testFailed("Right channel gain values are incorrect.  Max error = " + maxErrorR + " at time " + time[maxErrorIndexR] + " (threshold = " + maxAllowedError + ")");
+        success = false;
+    }
+
+    if (success) {
+        testPassed("EqualPower panner test passed");
+    } else {
+        testFailed("EqualPower panner test failed");
+    }
+
+    finishJSTest();
+}
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeAllPass.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeAllPass.html
new file mode 100644
index 0000000000..024c2f50df
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeAllPass.html
@@ -0,0 +1,32 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode All Pass Filter</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    var filterParameters = [{cutoff : 0,    q : 10, gain : 1 },
+                            {cutoff : 1,    q : 10, gain : 1 },
+                            {cutoff : .5,   q :  0, gain : 1 },
+                            {cutoff : 0.25, q : 10, gain : 1 },
+                           ];
+    createTestAndRun(context, "allpass", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeAutomation.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeAutomation.html
new file mode 100644
index 0000000000..5a71ce46e5
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeAutomation.html
@@ -0,0 +1,351 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode All Pass Filter</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  // Don't need to run these tests at high sampling rate, so just use a low one to reduce memory
+  // usage and complexity.
+  var sampleRate = 16000;
+
+  // How long to render for each test.
+  var renderDuration = 1;
+
+  // The definition of the linear ramp automation function.
+  function linearRamp(t, v0, v1, t0, t1) {
+    return v0 + (v1 - v0) * (t - t0) / (t1 - t0);
+  }
+
+  // Generate the filter coefficients for the specified filter using the given parameters for
+  // the given duration.  |filterTypeFunction| is a function that returns the filter
+  // coefficients for one set of parameters.  |parameters| is a property bag that contains the
+  // start and end values (as an array) for each of the biquad attributes.  The properties are
+  // |freq|, |Q|, |gain|, and |detune|.  |duration| is the number of seconds for which the
+  // coefficients are generated.
+  //
+  // A property bag with properties |b0|, |b1|, |b2|, |a1|, |a2|.  Each propery is an array
+  // consisting of the coefficients for the time-varying biquad filter.
+  function generateFilterCoefficients(filterTypeFunction, parameters, duration) {
+     var endFrame = Math.ceil(duration * sampleRate);
+     var nCoef = endFrame;
+     var b0 = new Float64Array(nCoef);
+     var b1 = new Float64Array(nCoef);
+     var b2 = new Float64Array(nCoef);
+     var a1 = new Float64Array(nCoef);
+     var a2 = new Float64Array(nCoef);
+
+     var k = 0;
+     // If the property is not given, use the defaults.
+     var freqs = parameters.freq || [350, 350];
+     var qs = parameters.Q || [1, 1];
+     var gains = parameters.gain || [0, 0];
+     var detunes = parameters.detune || [0, 0];
+
+     for (var frame = 0; frame < endFrame; ++frame) {
+        // Apply linear ramp at frame |frame|.
+        var f = linearRamp(frame / sampleRate, freqs[0], freqs[1], 0, duration);
+        var q = linearRamp(frame / sampleRate, qs[0], qs[1], 0, duration);
+        var g = linearRamp(frame / sampleRate, gains[0], gains[1], 0, duration);
+        var d = linearRamp(frame / sampleRate, detunes[0], detunes[1], 0, duration);
+
+        // Compute actual frequency parameter
+        f = f * Math.pow(2, d / 1200);
+
+        // Compute filter coefficients
+        var coef = filterTypeFunction(f / (sampleRate / 2), q, g);
+        b0[k] = coef.b0;
+        b1[k] = coef.b1;
+        b2[k] = coef.b2;
+        a1[k] = coef.a1;
+        a2[k] = coef.a2;
+        ++k;
+     }
+
+     return {b0: b0, b1: b1, b2: b2, a1: a1, a2: a2};
+  }
+
+  // Apply the given time-varying biquad filter to the given signal, |signal|.  |coef| should be
+  // the time-varying coefficients of the filter, as returned by |generateFilterCoefficients|.
+  function timeVaryingFilter(signal, coef) {
+    var length = signal.length;
+    // Use double precision for the internal computations.
+    var y = new Float64Array(length);
+
+    // Prime the pump. (Assumes the signal has length >= 2!)
+    y[0] = coef.b0[0] * signal[0];
+    y[1] = coef.b0[1] * signal[1] + coef.b1[1] * signal[0] - coef.a1[1] * y[0];
+
+    for (var n = 2; n < length; ++n) {
+      y[n] = coef.b0[n] * signal[n] + coef.b1[n] * signal[n-1] + coef.b2[n] * signal[n-2];
+      y[n] -= coef.a1[n] * y[n-1] + coef.a2[n] * y[n-2];
+    }
+
+    // But convert the result to single precision for comparison.
+    return y.map(Math.fround);
+  }
+
+  // Configure the audio graph using |context|.  Returns the biquad filter node and the
+  // AudioBuffer used for the source.
+  function configureGraph(context, toneFrequency) {
+    // The source is just a simple sine wave.
+    var src = context.createBufferSource();
+    var b = context.createBuffer(1, renderDuration * sampleRate, sampleRate);
+    var data = b.getChannelData(0);
+    var omega = 2 * Math.PI * toneFrequency / sampleRate;
+    for (var k = 0; k < data.length; ++k) {
+      data[k] = Math.sin(omega * k);
+    }
+    src.buffer = b;
+    var f = context.createBiquadFilter();
+    src.connect(f);
+    f.connect(context.destination);
+
+    src.start();
+
+    return {filter: f, source: b};
+  }
+
+  function createFilterVerifier(filterCreator, threshold, parameters, input, message) {
+    return function (resultBuffer) {
+      var actual = resultBuffer.getChannelData(0);
+      var coefs = generateFilterCoefficients(filterCreator, parameters, renderDuration);
+
+      reference = timeVaryingFilter(input, coefs);
+
+      compareChannels(actual, reference);
+    };
+  }
+
+  var testPromises = [];
+
+  // Automate just the frequency parameter.  A bandpass filter is used where the center
+  // frequency is swept across the source (which is a simple tone).
+  testPromises.push(function () {
+    var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+
+    // Center frequency of bandpass filter and also the frequency of the test tone.
+    var centerFreq = 10*440;
+
+    // Sweep the frequency +/- 9*440 Hz from the center.  This should cause the output to low at
+    // the beginning and end of the test where the done is outside the pass band of the filter,
+    // but high in the center where the tone is near the center of the pass band.
+    var parameters = {
+      freq: [centerFreq - 9*440, centerFreq + 9*440]
+    }
+    var graph = configureGraph(context, centerFreq);
+    var f = graph.filter;
+    var b = graph.source;
+
+    f.type = "bandpass";
+    f.frequency.setValueAtTime(parameters.freq[0], 0);
+    f.frequency.linearRampToValueAtTime(parameters.freq[1], renderDuration);
+
+    return context.startRendering()
+      .then(createFilterVerifier(createBandpassFilter, 5e-5, parameters, b.getChannelData(0),
+        "Output of bandpass filter with frequency automation"));
+  }());
+
+  // Automate just the Q parameter.  A bandpass filter is used where the Q of the filter is
+  // swept.
+  testPromises.push(function() {
+    var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+
+    // The frequency of the test tone.
+    var centerFreq = 440;
+
+    // Sweep the Q paramter between 1 and 200.  This will cause the output of the filter to pass
+    // most of the tone at the beginning to passing less of the tone at the end.  This is
+    // because we set center frequency of the bandpass filter to be slightly off from the actual
+    // tone.
+    var parameters = {
+      Q: [1, 200],
+      // Center frequency of the bandpass filter is just 25 Hz above the tone frequency.
+      freq: [centerFreq + 25, centerFreq + 25]
+    };
+    var graph = configureGraph(context, centerFreq);
+    var f = graph.filter;
+    var b = graph.source;
+
+    f.type = "bandpass";
+    f.frequency.value = parameters.freq[0];
+    f.Q.setValueAtTime(parameters.Q[0], 0);
+    f.Q.linearRampToValueAtTime(parameters.Q[1], renderDuration);
+
+    return context.startRendering()
+      .then(createFilterVerifier(createBandpassFilter, 1.4e-6, parameters, b.getChannelData(0),
+        "Output of bandpass filter with Q automation"));
+  }());
+
+  // Automate just the gain of the lowshelf filter.  A test tone will be in the lowshelf part of
+  // the filter.  The output will vary as the gain of the lowshelf is changed.
+  testPromises.push(function() {
+    var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+
+    // Frequency of the test tone.
+    var centerFreq = 440;
+
+    // Set the cutoff frequency of the lowshelf to be significantly higher than the test tone.
+    // Sweep the gain from 20 dB to -20 dB.  (We go from 20 to -20 to easily verify that the
+    // filter didn't go unstable.)
+    var parameters = {
+      freq: [3500, 3500],
+      gain: [20, -20]
+    }
+    var graph = configureGraph(context, centerFreq);
+    var f = graph.filter;
+    var b = graph.source;
+
+    f.type = "lowshelf";
+    f.frequency.value = parameters.freq[0];
+    f.gain.setValueAtTime(parameters.gain[0], 0);
+    f.gain.linearRampToValueAtTime(parameters.gain[1], renderDuration);
+
+    context.startRendering()
+      .then(createFilterVerifier(createLowShelfFilter, 8e-6, parameters, b.getChannelData(0),
+        "Output of lowshelf filter with gain automation"));
+  }());
+
+  // Automate just the detune parameter.  Basically the same test as for the frequncy parameter
+  // but we just use the detune parameter to modulate the frequency parameter.
+  testPromises.push(function() {
+    var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+    var centerFreq = 10*440;
+    var parameters = {
+      freq: [centerFreq, centerFreq],
+      detune: [-10*1200, 10*1200]
+    };
+    var graph = configureGraph(context, centerFreq);
+    var f = graph.filter;
+    var b = graph.source;
+
+    f.type = "bandpass";
+    f.frequency.value = parameters.freq[0];
+    f.detune.setValueAtTime(parameters.detune[0], 0);
+    f.detune.linearRampToValueAtTime(parameters.detune[1], renderDuration);
+
+    context.startRendering()
+      .then(createFilterVerifier(createBandpassFilter, 5e-6, parameters, b.getChannelData(0),
+        "Output of bandpass filter with detune automation"));
+  }());
+
+  // Automate all of the filter parameters at once.  This is a basic check that everything is
+  // working.  A peaking filter is used because it uses all of the parameters.
+  testPromises.push(function() {
+    var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+    var graph = configureGraph(context, 10*440);
+    var f = graph.filter;
+    var b = graph.source;
+
+    // Sweep all of the filter parameters.  These are pretty much arbitrary.
+    var parameters = {
+      freq: [10000, 100],
+      Q: [f.Q.value, .0001],
+      gain: [f.gain.value, 20],
+      detune: [2400, -2400]
+    };
+
+    f.type = "peaking";
+    // Set starting points for all parameters of the filter.  Start at 10 kHz for the center
+    // frequency, and the defaults for Q and gain.
+    f.frequency.setValueAtTime(parameters.freq[0], 0);
+    f.Q.setValueAtTime(parameters.Q[0], 0);
+    f.gain.setValueAtTime(parameters.gain[0], 0);
+    f.detune.setValueAtTime(parameters.detune[0], 0);
+
+    // Linear ramp each parameter
+    f.frequency.linearRampToValueAtTime(parameters.freq[1], renderDuration);
+    f.Q.linearRampToValueAtTime(parameters.Q[1], renderDuration);
+    f.gain.linearRampToValueAtTime(parameters.gain[1], renderDuration);
+    f.detune.linearRampToValueAtTime(parameters.detune[1], renderDuration);
+
+    context.startRendering()
+      .then(createFilterVerifier(createPeakingFilter, 3.3e-4, parameters, b.getChannelData(0),
+        "Output of peaking filter with automation of all parameters"));
+  }());
+
+  // Test that modulation of the frequency parameter of the filter works.  A sinusoid of 440 Hz
+  // is the test signal that is applied to a bandpass biquad filter.  The frequency parameter of
+  // the filter is modulated by a sinusoid at 103 Hz, and the frequency modulation varies from
+  // 116 to 412 Hz.  (This test was taken from the description in
+  // https://github.com/WebAudio/web-audio-api/issues/509#issuecomment-94731355)
+  testPromises.push(function() {
+    var context = new OfflineAudioContext(1, renderDuration * sampleRate, sampleRate);
+
+    // Create a graph with the sinusoidal source at 440 Hz as the input to a biquad filter.
+    var graph = configureGraph(context, 440);
+    var f = graph.filter;
+    var b = graph.source;
+
+    f.type = "bandpass";
+    f.Q.value = 5;
+    f.frequency.value = 264;
+
+    // Create the modulation source, a sinusoid with frequency 103 Hz and amplitude 148.  (The
+    // amplitude of 148 is added to the filter's frequency value of 264 to produce a sinusoidal
+    // modulation of the frequency parameter from 116 to 412 Hz.)
+    var mod = context.createBufferSource();
+    var mbuffer = context.createBuffer(1, renderDuration * sampleRate, sampleRate);
+    var d = mbuffer.getChannelData(0);
+    var omega = 2 * Math.PI * 103 / sampleRate;
+    for (var k = 0; k < d.length; ++k) {
+      d[k] = 148 * Math.sin(omega * k);
+    }
+    mod.buffer = mbuffer;
+
+    mod.connect(f.frequency);
+
+    mod.start();
+    return context.startRendering()
+      .then(function (resultBuffer) {
+         var actual = resultBuffer.getChannelData(0);
+         // Compute the filter coefficients using the mod sine wave
+
+         var endFrame = Math.ceil(renderDuration * sampleRate);
+         var nCoef = endFrame;
+         var b0 = new Float64Array(nCoef);
+         var b1 = new Float64Array(nCoef);
+         var b2 = new Float64Array(nCoef);
+         var a1 = new Float64Array(nCoef);
+         var a2 = new Float64Array(nCoef);
+
+         // Generate the filter coefficients when the frequency varies from 116 to 248 Hz using
+         // the 103 Hz sinusoid.
+         for (var k = 0; k < nCoef; ++k) {
+           var freq = f.frequency.value + d[k];
+           var c = createBandpassFilter(freq / (sampleRate / 2), f.Q.value, f.gain.value);
+           b0[k] = c.b0;
+           b1[k] = c.b1;
+           b2[k] = c.b2;
+           a1[k] = c.a1;
+           a2[k] = c.a2;
+         }
+         reference = timeVaryingFilter(b.getChannelData(0),
+           {b0: b0, b1: b1, b2: b2, a1: a1, a2: a2});
+
+         compareChannels(actual, reference);
+       });
+  }());
+
+  // Wait for all tests
+  Promise.all(testPromises).then(function () {
+    SimpleTest.finish();
+  }, function () {
+    SimpleTest.finish();
+  });
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeBandPass.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeBandPass.html
new file mode 100644
index 0000000000..05c4c3a265
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeBandPass.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode Band Pass Filter</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    // The filters we want to test.
+    var filterParameters = [{cutoff : 0,    q : 0, gain : 1 },
+                            {cutoff : 1,    q : 0, gain : 1 },
+                            {cutoff : 0.5,  q : 0, gain : 1 },
+                            {cutoff : 0.25, q : 1, gain : 1 },
+                           ];
+
+    createTestAndRun(context, "bandpass", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeGetFrequencyResponse.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeGetFrequencyResponse.html
new file mode 100644
index 0000000000..c2b6612034
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeGetFrequencyResponse.html
@@ -0,0 +1,261 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode All Pass Filter</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+// Test the frequency response of a biquad filter.  We compute the frequency response for a simple
+// peaking biquad filter and compare it with the expected frequency response.  The actual filter
+// used doesn't matter since we're testing getFrequencyResponse and not the actual filter output.
+// The filters are extensively tested in other biquad tests.
+
+var context;
+
+// The biquad filter node.
+var filter;
+
+// The magnitude response of the biquad filter.
+var magResponse;
+
+// The phase response of the biquad filter.
+var phaseResponse;
+
+// Number of frequency samples to take.
+var numberOfFrequencies = 1000;
+
+// The filter parameters.
+var filterCutoff = 1000; // Hz.
+var filterQ = 1;
+var filterGain = 5; // Decibels.
+
+// The maximum allowed error in the magnitude response.
+var maxAllowedMagError = 5.7e-7;
+
+// The maximum allowed error in the phase response.
+var maxAllowedPhaseError = 4.7e-8;
+
+// The magnitudes and phases of the reference frequency response.
+var magResponse;
+var phaseResponse;
+
+// The magnitudes and phases of the reference frequency response.
+var expectedMagnitudes;
+var expectedPhases;
+
+// Convert frequency in Hz to a normalized frequency between 0 to 1 with 1 corresponding to the
+// Nyquist frequency.
+function normalizedFrequency(freqHz, sampleRate)
+{
+    var nyquist = sampleRate / 2;
+    return freqHz / nyquist;
+}
+
+// Get the filter response at a (normalized) frequency |f| for the filter with coefficients |coef|.
+function getResponseAt(coef, f)
+{
+    var b0 = coef.b0;
+    var b1 = coef.b1;
+    var b2 = coef.b2;
+    var a1 = coef.a1;
+    var a2 = coef.a2;
+
+    // H(z) = (b0 + b1 / z + b2 / z^2) / (1 + a1 / z + a2 / z^2)
+    //
+    // Compute H(exp(i * pi * f)).  No native complex numbers in javascript, so break H(exp(i * pi * // f))
+    // in to the real and imaginary parts of the numerator and denominator.  Let omega = pi * f.
+    // Then the numerator is
+    //
+    // b0 + b1 * cos(omega) + b2 * cos(2 * omega) - i * (b1 * sin(omega) + b2 * sin(2 * omega))
+    //
+    // and the denominator is
+    //
+    // 1 + a1 * cos(omega) + a2 * cos(2 * omega) - i * (a1 * sin(omega) + a2 * sin(2 * omega))
+    //
+    // Compute the magnitude and phase from the real and imaginary parts.
+
+    var omega = Math.PI * f;
+    var numeratorReal = b0 + b1 * Math.cos(omega) + b2 * Math.cos(2 * omega);
+    var numeratorImag = -(b1 * Math.sin(omega) + b2 * Math.sin(2 * omega));
+    var denominatorReal = 1 + a1 * Math.cos(omega) + a2 * Math.cos(2 * omega);
+    var denominatorImag = -(a1 * Math.sin(omega) + a2 * Math.sin(2 * omega));
+
+    var magnitude = Math.sqrt((numeratorReal * numeratorReal + numeratorImag * numeratorImag)
+                              / (denominatorReal * denominatorReal + denominatorImag * denominatorImag));
+    var phase = Math.atan2(numeratorImag, numeratorReal) - Math.atan2(denominatorImag, denominatorReal);
+
+    if (phase >= Math.PI) {
+        phase -= 2 * Math.PI;
+    } else if (phase <= -Math.PI) {
+        phase += 2 * Math.PI;
+    }
+
+    return {magnitude : magnitude, phase : phase};
+}
+
+// Compute the reference frequency response for the biquad filter |filter| at the frequency samples
+// given by |frequencies|.
+function frequencyResponseReference(filter, frequencies)
+{
+    var sampleRate = filter.context.sampleRate;
+    var normalizedFreq = normalizedFrequency(filter.frequency.value, sampleRate);
+    var filterCoefficients = createFilter(filter.type, normalizedFreq, filter.Q.value, filter.gain.value);
+
+    var magnitudes = [];
+    var phases = [];
+
+    for (var k = 0; k < frequencies.length; ++k) {
+        var response = getResponseAt(filterCoefficients, normalizedFrequency(frequencies[k], sampleRate));
+        magnitudes.push(response.magnitude);
+        phases.push(response.phase);
+    }
+
+    return {magnitudes : magnitudes, phases : phases};
+}
+
+// Compute a set of linearly spaced frequencies.
+function createFrequencies(nFrequencies, sampleRate)
+{
+    var frequencies = new Float32Array(nFrequencies);
+    var nyquist = sampleRate / 2;
+    var freqDelta = nyquist / nFrequencies;
+
+    for (var k = 0; k < nFrequencies; ++k) {
+        frequencies[k] = k * freqDelta;
+    }
+
+    return frequencies;
+}
+
+function linearToDecibels(x)
+{
+    if (x) {
+        return 20 * Math.log(x) / Math.LN10;
+    } else {
+        return -1000;
+    }
+}
+
+// Look through the array and find any NaN or infinity. Returns the index of the first occurence or
+// -1 if none.
+function findBadNumber(signal)
+{
+    for (var k = 0; k < signal.length; ++k) {
+        if (!isValidNumber(signal[k])) {
+           return k;
+        }
+    }
+    return -1;
+}
+
+// Compute absolute value of the difference between phase angles, taking into account the wrapping
+// of phases.
+function absolutePhaseDifference(x, y)
+{
+    var diff = Math.abs(x - y);
+
+    if (diff > Math.PI) {
+        diff = 2 * Math.PI - diff;
+    }
+    return diff;
+}
+
+// Compare the frequency response with our expected response.
+function compareResponses(filter, frequencies, magResponse, phaseResponse)
+{
+    var expectedResponse = frequencyResponseReference(filter, frequencies);
+
+    expectedMagnitudes = expectedResponse.magnitudes;
+    expectedPhases = expectedResponse.phases;
+
+    var n = magResponse.length;
+    var success = true;
+    var badResponse = false;
+
+    var maxMagError = -1;
+    var maxMagErrorIndex = -1;
+
+    var k;
+    var hasBadNumber;
+
+    hasBadNumber = findBadNumber(magResponse);
+    ok (hasBadNumber < 0, "Magnitude response has NaN or infinity at " + hasBadNumber);
+
+    hasBadNumber = findBadNumber(phaseResponse);
+    ok (hasBadNumber < 0, "Phase response has NaN or infinity at " + hasBadNumber);
+
+    // These aren't testing the implementation itself.  Instead, these are sanity checks on the
+    // reference.  Failure here does not imply an error in the implementation.
+    hasBadNumber = findBadNumber(expectedMagnitudes);
+    ok (hasBadNumber < 0, "Expected magnitude response has NaN or infinity at " + hasBadNumber);
+
+    hasBadNumber = findBadNumber(expectedPhases);
+    ok (hasBadNumber < 0, "Expected phase response has NaN or infinity at " + hasBadNumber);
+
+    for (k = 0; k < n; ++k) {
+        var error = Math.abs(linearToDecibels(magResponse[k]) - linearToDecibels(expectedMagnitudes[k]));
+        if (error > maxMagError) {
+            maxMagError = error;
+            maxMagErrorIndex = k;
+        }
+    }
+
+    var message = "Magnitude error (" + maxMagError + " dB)";
+    message += " exceeded threshold at " + frequencies[maxMagErrorIndex];
+    message += " Hz.  Actual: " + linearToDecibels(magResponse[maxMagErrorIndex]);
+    message += " dB, expected: " + linearToDecibels(expectedMagnitudes[maxMagErrorIndex]) + " dB.";
+    ok(maxMagError < maxAllowedMagError, message);
+
+    var maxPhaseError = -1;
+    var maxPhaseErrorIndex = -1;
+
+    for (k = 0; k < n; ++k) {
+        var error = absolutePhaseDifference(phaseResponse[k], expectedPhases[k]);
+        if (error > maxPhaseError) {
+            maxPhaseError = error;
+            maxPhaseErrorIndex = k;
+        }
+    }
+
+    message = "Phase error (radians) (" + maxPhaseError;
+    message += ") exceeded threshold at " + frequencies[maxPhaseErrorIndex];
+    message += " Hz.  Actual: " + phaseResponse[maxPhaseErrorIndex];
+    message += " expected: " + expectedPhases[maxPhaseErrorIndex];
+
+    ok(maxPhaseError < maxAllowedPhaseError, message);
+}
+
+context = new AudioContext();
+
+filter = context.createBiquadFilter();
+
+// Arbitrarily test a peaking filter, but any kind of filter can be tested.
+filter.type = "peaking";
+filter.frequency.value = filterCutoff;
+filter.Q.value = filterQ;
+filter.gain.value = filterGain;
+
+var frequencies = createFrequencies(numberOfFrequencies, context.sampleRate);
+magResponse = new Float32Array(numberOfFrequencies);
+phaseResponse = new Float32Array(numberOfFrequencies);
+
+filter.getFrequencyResponse(frequencies, magResponse, phaseResponse);
+compareResponses(filter, frequencies, magResponse, phaseResponse);
+
+SimpleTest.finish();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeHighPass.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeHighPass.html
new file mode 100644
index 0000000000..a615574c2d
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeHighPass.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode High Pass Filter</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    // The filters we want to test.
+    var filterParameters = [{cutoff : 0,    q : 1, gain : 1 },
+                            {cutoff : 1,    q : 1, gain : 1 },
+                            {cutoff : 0.25, q : 1, gain : 1 },
+                           ];
+
+    createTestAndRun(context, "highpass", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeHighShelf.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeHighShelf.html
new file mode 100644
index 0000000000..c8f6815930
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeHighShelf.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode High Shelf Filter</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    // The filters we want to test.
+    var filterParameters = [{cutoff : 0,    q : 10, gain : 10 },
+                            {cutoff : 1,    q : 10, gain : 10 },
+                            {cutoff : 0.25, q : 10, gain : 10 },
+                           ];
+
+    createTestAndRun(context, "highshelf", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeLowPass.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeLowPass.html
new file mode 100644
index 0000000000..dcea18551a
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeLowPass.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode Low Pass Filter</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    // The filters we want to test.
+    var filterParameters = [{cutoff : 0,    q : 1, gain : 1 },
+                            {cutoff : 1,    q : 1, gain : 1 },
+                            {cutoff : 0.25, q : 1, gain : 1 },
+                            {cutoff : 0.25, q : 1, gain : 1, detune : 100 },
+                            {cutoff : 0.01, q : 1, gain : 1, detune : -200 },
+                           ];
+    createTestAndRun(context, "lowpass", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeLowShelf.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeLowShelf.html
new file mode 100644
index 0000000000..c1349f8e7e
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeLowShelf.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode Low Shelf Filter</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+ 
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    // The filters we want to test.
+    var filterParameters = [{cutoff : 0,    q : 10, gain : 10 },
+                            {cutoff : 1,    q : 10, gain : 10 },
+                            {cutoff : 0.25, q : 10, gain : 10 },
+                           ];
+
+    createTestAndRun(context, "lowshelf", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeNotch.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeNotch.html
new file mode 100644
index 0000000000..0fcbc5546e
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeNotch.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode Notch Filter</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    var filterParameters = [{cutoff : 0,    q : 10, gain : 1 },
+                            {cutoff : 1,    q : 10, gain : 1 },
+                            {cutoff : .5,   q :  0, gain : 1 },
+                            {cutoff : 0.25, q : 10, gain : 1 },
+                           ];
+
+    createTestAndRun(context, "notch", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodePeaking.html b/dom/media/webaudio/test/blink/test_biquadFilterNodePeaking.html
new file mode 100644
index 0000000000..8e4727a37e
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodePeaking.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode Low Pass Filter</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    // The filters we want to test.
+    var filterParameters = [{cutoff : 0,    q : 10, gain : 10 },
+                            {cutoff : 1,    q : 10, gain : 10 },
+                            {cutoff : .5,   q :  0, gain : 10 },
+                            {cutoff : 0.25, q : 10, gain : 10 },
+                           ];
+
+    createTestAndRun(context, "peaking", filterParameters);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_biquadFilterNodeTail.html b/dom/media/webaudio/test/blink/test_biquadFilterNodeTail.html
new file mode 100644
index 0000000000..cc170df2a5
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_biquadFilterNodeTail.html
@@ -0,0 +1,76 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode All Pass Filter</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="audio-testing.js"></script>
+<script src="biquad-filters.js"></script>
+<script src="biquad-testing.js"></script>
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  // A high sample rate shows the issue more clearly.
+  var sampleRate = 192000;
+  // Some short duration because we don't need to run the test for very long.
+  var testDurationSec = 0.5;
+  var testDurationFrames = testDurationSec * sampleRate;
+
+  // Amplitude experimentally determined to give a biquad output close to 1. (No attempt was
+  // made to produce exactly 1; it's not needed.)
+  var sourceAmplitude = 100;
+
+  // The output of the biquad filter should not change by more than this much between output
+  // samples.  Threshold was determined experimentally.
+  var glitchThreshold = 0.01292;
+
+  // Test that a Biquad filter doesn't have it's output terminated because the input has gone
+  // away.  Generally, when a source node is finished, it disconnects itself from any downstream
+  // nodes.  This is the correct behavior.  Nodes that have no inputs (disconnected) are
+  // generally assumed to output zeroes.  This is also desired behavior.  However, biquad
+  // filters have memory so they should not suddenly output zeroes when the input is
+  // disconnected.  This test checks to see if the output doesn't suddenly change to zero.
+  var context = new OfflineAudioContext(1, testDurationFrames, sampleRate);
+
+  // Create an impulse source.
+  var buffer = context.createBuffer(1, 1, context.sampleRate);
+  buffer.getChannelData(0)[0] = sourceAmplitude;
+  var source = context.createBufferSource();
+  source.buffer = buffer;
+
+  // Create the biquad filter. It doesn't really matter what kind, so the default filter type
+  // and parameters is fine.  Connect the source to it.
+  var biquad = context.createBiquadFilter();
+  source.connect(biquad);
+  biquad.connect(context.destination);
+
+  source.start();
+
+  context.startRendering().then(function(result) {
+    // There should be no large discontinuities in the output
+    var buffer = result.getChannelData(0);
+    var maxGlitchIndex = 0;
+    var maxGlitchValue = 0.0;
+    for (var i = 1; i < buffer.length; i++) {
+      var diff = Math.abs(buffer[i-1] - buffer[i]);
+      if (diff >= glitchThreshold) {
+        if (diff > maxGlitchValue) {
+          maxGlitchIndex = i;
+          maxGlitchValue = diff;
+        }
+      }
+    }
+    ok(maxGlitchIndex == 0, 'glitches detected in biquad output: maximum glitch at ' + maxGlitchIndex + ' with diff of ' + maxGlitchValue);
+    SimpleTest.finish();
+  })
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_iirFilterNode.html b/dom/media/webaudio/test/blink/test_iirFilterNode.html
new file mode 100644
index 0000000000..0ef7a37e3b
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_iirFilterNode.html
@@ -0,0 +1,467 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test IIRFilterNode GetFrequencyResponse</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <script type="text/javascript" src="biquad-filters.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  var sampleRate = 48000;
+  var testDurationSec = 1;
+  var testFrames = testDurationSec * sampleRate;
+
+  var testPromises = []
+  testPromises.push(function () {
+    // Test that the feedback coefficients are normalized.  Do this be creating two
+    // IIRFilterNodes.  One has normalized coefficients, and one doesn't.  Compute the
+    // difference and make sure they're the same.
+    var context = new OfflineAudioContext(2, testFrames, sampleRate);
+
+    // Use a simple impulse as the source.
+    var buffer = context.createBuffer(1, 1, sampleRate);
+    buffer.getChannelData(0)[0] = 1;
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    // Gain node for computing the difference between the filters.
+    var gain = context.createGain();
+    gain.gain.value = -1;
+
+    // The IIR filters.  Use a common feedforward array.
+    var ff = [1];
+
+    var fb1 = [1, .9];
+
+    var fb2 = new Float64Array(2);
+    // Scale the feedback coefficients by an arbitrary factor.
+    var coefScaleFactor = 2;
+    for (var k = 0; k < fb2.length; ++k) {
+      fb2[k] = coefScaleFactor * fb1[k];
+    }
+
+    var iir1 = context.createIIRFilter(ff, fb1);
+    var iir2 = context.createIIRFilter(ff, fb2);
+
+    // Create the graph.  The output of iir1 (normalized coefficients) is channel 0, and the
+    // output of iir2 (unnormalized coefficients), with appropriate scaling, is channel 1.
+    var merger = context.createChannelMerger(2);
+    source.connect(iir1);
+    source.connect(iir2);
+    iir1.connect(merger, 0, 0);
+    iir2.connect(gain);
+
+    // The gain for the gain node should be set to compensate for the scaling of the
+    // coefficients.  Since iir2 has scaled the coefficients by coefScaleFactor, the output is
+    // reduced by the same factor, so adjust the gain to scale the output of iir2 back up.
+    gain.gain.value = coefScaleFactor;
+    gain.connect(merger, 0, 1);
+
+    merger.connect(context.destination);
+
+    source.start();
+
+    // Rock and roll!
+
+    return context.startRendering().then(function (result) {
+      // Find the max amplitude of the result, which should be near zero.
+      var iir1Data = result.getChannelData(0);
+      var iir2Data = result.getChannelData(1);
+
+      // Threshold isn't exactly zero because the arithmetic is done differently between the
+      // IIRFilterNode and the BiquadFilterNode.
+      compareChannels(iir1Data, iir2Data);
+    });
+  }());
+
+  testPromises.push(function () {
+    // Create a simple 1-zero filter and compare with the expected output.
+    var context = new OfflineAudioContext(1, testFrames, sampleRate);
+
+    // Use a simple impulse as the source
+    var buffer = context.createBuffer(1, 1, sampleRate);
+    buffer.getChannelData(0)[0] = 1;
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    // The filter is y(n) = 0.5*(x(n) + x(n-1)), a simple 2-point moving average.  This is
+    // rather arbitrary; keep it simple.
+
+    var iir = context.createIIRFilter([0.5, 0.5], [1]);
+
+    // Create the graph
+    source.connect(iir);
+    iir.connect(context.destination);
+
+    // Rock and roll!
+    source.start();
+
+    return context.startRendering().then(function (result) {
+      var actual = result.getChannelData(0);
+      var expected = new Float64Array(testFrames);
+      // The filter is a simple 2-point moving average of an impulse, so the first two values
+      // are non-zero and the rest are zero.
+      expected[0] = 0.5;
+      expected[1] = 0.5;
+      compareChannels(actual, expected);
+    });
+  }());
+
+  testPromises.push(function () {
+    // Create a simple 1-pole filter and compare with the expected output.
+
+    // The filter is y(n) + c*y(n-1)= x(n).  The analytical response is (-c)^n, so choose a
+    // suitable number of frames to run the test for where the output isn't flushed to zero.
+    var c = 0.9;
+    var eps = 1e-20;
+    var duration = Math.floor(Math.log(eps) / Math.log(Math.abs(c)));
+    var context = new OfflineAudioContext(1, duration, sampleRate);
+
+    // Use a simple impulse as the source
+    var buffer = context.createBuffer(1, 1, sampleRate);
+    buffer.getChannelData(0)[0] = 1;
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    var iir = context.createIIRFilter([1], [1, c]);
+
+    // Create the graph
+    source.connect(iir);
+    iir.connect(context.destination);
+
+    // Rock and roll!
+    source.start();
+
+    return context.startRendering().then(function (result) {
+      var actual = result.getChannelData(0);
+      var expected = new Float64Array(actual.length);
+
+      // The filter is a simple 1-pole filter: y(n) = -c*y(n-k)+x(n), with an impulse as the
+      // input.
+      expected[0] = 1;
+      for (k = 1; k < testFrames; ++k) {
+        expected[k] = -c * expected[k-1];
+      }
+
+      compareChannels(actual, expected);
+    });
+  }());
+
+  // This function creates an IIRFilterNode equivalent to the specified
+  // BiquadFilterNode and compares the outputs.  The
+  // outputs from the two filters should be virtually identical.
+  function testWithBiquadFilter(filterType) {
+    var context = new OfflineAudioContext(2, testFrames, sampleRate);
+
+    // Use a constant (step function) as the source
+    var buffer = context.createBuffer(1, testFrames, context.sampleRate);
+    for (var i = 0; i < testFrames; ++i) {
+      buffer.getChannelData(0)[i] = 1;
+    }
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    // Create the biquad.  Choose some rather arbitrary values for Q and gain for the biquad
+    // so that the shelf filters aren't identical.
+    var biquad = context.createBiquadFilter();
+    biquad.type = filterType;
+    biquad.Q.value = 10;
+    biquad.gain.value = 10;
+
+    // Create the equivalent IIR Filter node by computing the coefficients of the given biquad
+    // filter type.
+    var nyquist = sampleRate / 2;
+    var coef = createFilter(filterType,
+                            biquad.frequency.value / nyquist,
+                            biquad.Q.value,
+                            biquad.gain.value);
+
+    var iir = context.createIIRFilter([coef.b0, coef.b1, coef.b2], [1, coef.a1, coef.a2]);
+
+    var merger = context.createChannelMerger(2);
+    // Create the graph
+    source.connect(biquad);
+    source.connect(iir);
+
+    biquad.connect(merger, 0, 0);
+    iir.connect(merger, 0, 1);
+
+    merger.connect(context.destination);
+
+    // Rock and roll!
+    source.start();
+
+    return context.startRendering().then(function (result) {
+      // Find the max amplitude of the result, which should be near zero.
+      var expected = result.getChannelData(0);
+      var actual = result.getChannelData(1);
+      compareChannels(actual, expected);
+    });
+  }
+
+  biquadFilterTypes = ["lowpass", "highpass", "bandpass", "notch",
+                       "allpass", "lowshelf", "highshelf", "peaking"];
+
+  // Create a set of tasks based on biquadTestConfigs.
+  for (var i = 0; i < biquadFilterTypes.length; ++i) {
+    testPromises.push(testWithBiquadFilter(biquadFilterTypes[i]));
+  }
+
+  testPromises.push(function () {
+    // Multi-channel test.  Create a biquad filter and the equivalent IIR filter.  Filter the
+    // same multichannel signal and compare the results.
+    var nChannels = 3;
+    var context = new OfflineAudioContext(nChannels, testFrames, sampleRate);
+
+    // Create a set of oscillators as the multi-channel source.
+    var source = [];
+
+    for (k = 0; k < nChannels; ++k) {
+      source[k] = context.createOscillator();
+      source[k].type = "sawtooth";
+      // The frequency of the oscillator is pretty arbitrary, but each oscillator should have a
+      // different frequency.
+      source[k].frequency.value = 100 + k * 100;
+    }
+
+    var merger = context.createChannelMerger(3);
+
+    var biquad = context.createBiquadFilter();
+
+    // Create the equivalent IIR Filter node.
+    var nyquist = sampleRate / 2;
+    var coef = createFilter(biquad.type,
+      biquad.frequency.value / nyquist,
+      biquad.Q.value,
+      biquad.gain.value);
+    var fb = [1, coef.a1, coef.a2];
+    var ff = [coef.b0, coef.b1, coef.b2];
+
+    var iir = context.createIIRFilter(ff, fb);
+    // Gain node to compute the difference between the IIR and biquad filter.
+    var gain = context.createGain();
+    gain.gain.value = -1;
+
+    // Create the graph.
+    for (k = 0; k < nChannels; ++k)
+      source[k].connect(merger, 0, k);
+
+    merger.connect(biquad);
+    merger.connect(iir);
+    iir.connect(gain);
+    biquad.connect(context.destination);
+    gain.connect(context.destination);
+
+    for (k = 0; k < nChannels; ++k)
+      source[k].start();
+
+    return context.startRendering().then(function (result) {
+      var errorThresholds = [3.7671e-5, 3.0071e-5, 2.6241e-5];
+
+      // Check the difference signal on each channel
+      for (channel = 0; channel < result.numberOfChannels; ++channel) {
+        // Find the max amplitude of the result, which should be near zero.
+        var data = result.getChannelData(channel);
+        var maxError = data.reduce(function(reducedValue, currentValue) {
+          return Math.max(reducedValue, Math.abs(currentValue));
+        });
+
+        ok(maxError <= errorThresholds[channel], "Max difference between IIR and Biquad on channel " + channel);
+      }
+    });
+  }());
+
+  testPromises.push(function () {
+    // Apply an IIRFilter to the given input signal.
+    //
+    // IIR filter in the time domain is
+    //
+    //   y[n] = sum(ff[k]*x[n-k], k, 0, M) - sum(fb[k]*y[n-k], k, 1, N)
+    //
+    function iirFilter(input, feedforward, feedback) {
+      // For simplicity, create an x buffer that contains the input, and a y buffer that contains
+      // the output.  Both of these buffers have an initial work space to implement the initial
+      // memory of the filter.
+      var workSize = Math.max(feedforward.length, feedback.length);
+      var x = new Float32Array(input.length + workSize);
+
+      // Float64 because we want to match the implementation that uses doubles to minimize
+      // roundoff.
+      var y = new Float64Array(input.length + workSize);
+
+      // Copy the input over.
+      for (var k = 0; k < input.length; ++k)
+        x[k + feedforward.length] = input[k];
+
+      // Run the filter
+      for (var n = 0; n < input.length; ++n) {
+        var index = n + workSize;
+        var yn = 0;
+        for (var k = 0; k < feedforward.length; ++k)
+          yn += feedforward[k] * x[index - k];
+        for (var k = 0; k < feedback.length; ++k)
+          yn -= feedback[k] * y[index - k];
+
+        y[index] = yn;
+      }
+
+      return y.slice(workSize).map(Math.fround);
+    }
+
+    // Cascade the two given biquad filters to create one IIR filter.
+    function cascadeBiquads(f1Coef, f2Coef) {
+      // The biquad filters are:
+      //
+      // f1 = (b10 + b11/z + b12/z^2)/(1 + a11/z + a12/z^2);
+      // f2 = (b20 + b21/z + b22/z^2)/(1 + a21/z + a22/z^2);
+      //
+      // To cascade them, multiply the two transforms together to get a fourth order IIR filter.
+
+      var numProduct = [f1Coef.b0 * f2Coef.b0,
+        f1Coef.b0 * f2Coef.b1 + f1Coef.b1 * f2Coef.b0,
+        f1Coef.b0 * f2Coef.b2 + f1Coef.b1 * f2Coef.b1 + f1Coef.b2 * f2Coef.b0,
+        f1Coef.b1 * f2Coef.b2 + f1Coef.b2 * f2Coef.b1,
+        f1Coef.b2 * f2Coef.b2
+      ];
+
+      var denProduct = [1,
+        f2Coef.a1 + f1Coef.a1,
+        f2Coef.a2 + f1Coef.a1 * f2Coef.a1 + f1Coef.a2,
+        f1Coef.a1 * f2Coef.a2 + f1Coef.a2 * f2Coef.a1,
+        f1Coef.a2 * f2Coef.a2
+      ];
+
+      return {
+        ff: numProduct,
+        fb: denProduct
+      }
+    }
+
+    // Find the magnitude of the root of the quadratic that has the maximum magnitude.
+    //
+    // The quadratic is z^2 + a1 * z + a2 and we want the root z that has the largest magnitude.
+    function largestRootMagnitude(a1, a2) {
+      var discriminant = a1 * a1 - 4 * a2;
+      if (discriminant < 0) {
+        // Complex roots:  -a1/2 +/- i*sqrt(-d)/2.  Thus the magnitude of each root is the same
+        // and is sqrt(a1^2/4 + |d|/4)
+        var d = Math.sqrt(-discriminant);
+        return Math.hypot(a1 / 2, d / 2);
+      } else {
+        // Real roots
+        var d = Math.sqrt(discriminant);
+        return Math.max(Math.abs((-a1 + d) / 2), Math.abs((-a1 - d) / 2));
+      }
+    }
+
+    // Cascade 2 lowpass biquad filters and compare that with the equivalent 4th order IIR
+    // filter.
+
+    var nyquist = sampleRate / 2;
+    // Compute the coefficients of a lowpass filter.
+
+    // First some preliminary stuff.  Compute the coefficients of the biquad.  This is used to
+    // figure out how frames to use in the test.
+    var biquadType = "lowpass";
+    var biquadCutoff = 350;
+    var biquadQ = 5;
+    var biquadGain = 1;
+
+    var coef = createFilter(biquadType,
+      biquadCutoff / nyquist,
+      biquadQ,
+      biquadGain);
+
+    // Cascade the biquads together to create an equivalent IIR filter.
+    var cascade = cascadeBiquads(coef, coef);
+
+    // Since we're cascading two identical biquads, the root of denominator of the IIR filter is
+    // repeated, so the root of the denominator with the largest magnitude occurs twice.  The
+    // impulse response of the IIR filter will be roughly c*(r*r)^n at time n, where r is the
+    // root of largest magnitude.  This approximation gets better as n increases.  We can use
+    // this to get a rough idea of when the response has died down to a small value.
+
+    // This is the value we will use to determine how many frames to render.  Rendering too many
+    // is a waste of time and also makes it hard to compare the actual result to the expected
+    // because the magnitudes are so small that they could be mostly round-off noise.
+    //
+    // Find magnitude of the root with largest magnitude
+    var rootMagnitude = largestRootMagnitude(coef.a1, coef.a2);
+
+    // Find n such that |r|^(2*n) <= eps.  That is, n = log(eps)/(2*log(r)).  Somewhat
+    // arbitrarily choose eps = 1e-20;
+    var eps = 1e-20;
+    var framesForTest = Math.floor(Math.log(eps) / (2 * Math.log(rootMagnitude)));
+
+    // We're ready to create the graph for the test.  The offline context has two channels:
+    // channel 0 is the expected (cascaded biquad) result and channel 1 is the actual IIR filter
+    // result.
+    var context = new OfflineAudioContext(2, framesForTest, sampleRate);
+
+    // Use a simple impulse with a large (arbitrary) amplitude as the source
+    var amplitude = 1;
+    var buffer = context.createBuffer(1, testFrames, sampleRate);
+    buffer.getChannelData(0)[0] = amplitude;
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    // Create the two biquad filters.  Doesn't really matter what, but for simplicity we choose
+    // identical lowpass filters with the same parameters.
+    var biquad1 = context.createBiquadFilter();
+    biquad1.type = biquadType;
+    biquad1.frequency.value = biquadCutoff;
+    biquad1.Q.value = biquadQ;
+
+    var biquad2 = context.createBiquadFilter();
+    biquad2.type = biquadType;
+    biquad2.frequency.value = biquadCutoff;
+    biquad2.Q.value = biquadQ;
+
+    var iir = context.createIIRFilter(cascade.ff, cascade.fb);
+
+    // Create the merger to get the signals into multiple channels
+    var merger = context.createChannelMerger(2);
+
+    // Create the graph, filtering the source through two biquads.
+    source.connect(biquad1);
+    biquad1.connect(biquad2);
+    biquad2.connect(merger, 0, 0);
+
+    source.connect(iir);
+    iir.connect(merger, 0, 1);
+
+    merger.connect(context.destination);
+
+    // Now filter the source through the IIR filter.
+    var y = iirFilter(buffer.getChannelData(0), cascade.ff, cascade.fb);
+
+    // Rock and roll!
+    source.start();
+
+    return context.startRendering().then(function(result) {
+      var expected = result.getChannelData(0);
+      var actual = result.getChannelData(1);
+
+      compareChannels(actual, expected);
+
+    });
+  }());
+
+  // Wait for all tests
+  Promise.all(testPromises).then(function () {
+    SimpleTest.finish();
+  }, function () {
+    SimpleTest.finish();
+  });
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/blink/test_iirFilterNodeGetFrequencyResponse.html b/dom/media/webaudio/test/blink/test_iirFilterNodeGetFrequencyResponse.html
new file mode 100644
index 0000000000..09782f73be
--- /dev/null
+++ b/dom/media/webaudio/test/blink/test_iirFilterNodeGetFrequencyResponse.html
@@ -0,0 +1,97 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test IIRFilterNode GetFrequencyResponse</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <script type="text/javascript" src="biquad-filters.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  // Modified from WebKit/LayoutTests/webaudio/iirfilter-getFrequencyResponse.html
+  var sampleRate = 48000;
+  var testDurationSec = 0.01;
+
+  // Compute a set of linearly spaced frequencies.
+  function createFrequencies(nFrequencies, sampleRate)
+  {
+    var frequencies = new Float32Array(nFrequencies);
+    var nyquist = sampleRate / 2;
+    var freqDelta = nyquist / nFrequencies;
+
+    for (var k = 0; k < nFrequencies; ++k) {
+        frequencies[k] = k * freqDelta;
+    }
+
+    return frequencies;
+  }
+
+  // Number of frequency samples to take.
+  var numberOfFrequencies = 1000;
+
+  var context = new OfflineAudioContext(1, testDurationSec * sampleRate, sampleRate);
+
+  var frequencies = createFrequencies(numberOfFrequencies, context.sampleRate);
+
+  // 1-Pole IIR Filter
+  var iir = context.createIIRFilter([1], [1, -0.9]);
+
+  var iirMag = new Float32Array(numberOfFrequencies);
+  var iirPhase = new Float32Array(numberOfFrequencies);
+  var trueMag = new Float32Array(numberOfFrequencies);
+  var truePhase = new Float32Array(numberOfFrequencies);
+
+  // The IIR filter is
+  //   H(z) = 1/(1 - 0.9*z^(-1)).
+  //
+  // The frequency response is
+  //   H(exp(j*w)) = 1/(1 - 0.9*exp(-j*w)).
+  //
+  // Thus, the magnitude is
+  //   |H(exp(j*w))| = 1/sqrt(1.81-1.8*cos(w)).
+  //
+  // The phase is
+  //   arg(H(exp(j*w)) = atan(0.9*sin(w)/(.9*cos(w)-1))
+
+  var frequencyScale = Math.PI / (sampleRate / 2);
+
+  for (var k = 0; k < frequencies.length; ++k) {
+    var omega = frequencyScale * frequencies[k];
+    trueMag[k] = 1/Math.sqrt(1.81-1.8*Math.cos(omega));
+    truePhase[k] = Math.atan(0.9 * Math.sin(omega) / (0.9 * Math.cos(omega) - 1));
+  }
+
+  iir.getFrequencyResponse(frequencies, iirMag, iirPhase);
+  compareChannels(iirMag, trueMag);
+  compareChannels(iirPhase, truePhase);
+
+  // Compare IIR and Biquad Filter
+  // Create an IIR filter equivalent to the biquad filter. Compute the frequency response for both and verify that they are the same.
+  var biquad = context.createBiquadFilter();
+  var coef = createFilter(biquad.type,
+                          biquad.frequency.value / (context.sampleRate / 2),
+                          biquad.Q.value,
+                          biquad.gain.value);
+
+  var iir = context.createIIRFilter([coef.b0, coef.b1, coef.b2], [1, coef.a1, coef.a2]);
+
+  var biquadMag = new Float32Array(numberOfFrequencies);
+  var biquadPhase = new Float32Array(numberOfFrequencies);
+  var iirMag = new Float32Array(numberOfFrequencies);
+  var iirPhase = new Float32Array(numberOfFrequencies);
+
+  biquad.getFrequencyResponse(frequencies, biquadMag, biquadPhase);
+  iir.getFrequencyResponse(frequencies, iirMag, iirPhase);
+  compareChannels(iirMag, biquadMag);
+  compareChannels(iirPhase, biquadPhase);
+
+  SimpleTest.finish();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/corsServer.sjs b/dom/media/webaudio/test/corsServer.sjs
new file mode 100644
index 0000000000..de77ef71ed
--- /dev/null
+++ b/dom/media/webaudio/test/corsServer.sjs
@@ -0,0 +1,26 @@
+function handleRequest(request, response) {
+  var file = Cc["@mozilla.org/file/directory_service;1"]
+    .getService(Ci.nsIProperties)
+    .get("CurWorkD", Ci.nsIFile);
+  var fis = Cc["@mozilla.org/network/file-input-stream;1"].createInstance(
+    Ci.nsIFileInputStream
+  );
+  var bis = Cc["@mozilla.org/binaryinputstream;1"].createInstance(
+    Ci.nsIBinaryInputStream
+  );
+  var paths = "tests/dom/media/webaudio/test/small-shot.ogg";
+  var split = paths.split("/");
+  for (var i = 0; i < split.length; ++i) {
+    file.append(split[i]);
+  }
+  fis.init(file, -1, -1, false);
+  bis.setInputStream(fis);
+  var bytes = bis.readBytes(bis.available());
+  response.setHeader("Content-Type", "video/ogg", false);
+  response.setHeader("Content-Length", "" + bytes.length, false);
+  response.setHeader("Access-Control-Allow-Origin", "*", false);
+  response.write(bytes, bytes.length);
+  response.processAsync();
+  response.finish();
+  bis.close();
+}
diff --git a/dom/media/webaudio/test/file_nodeCreationDocumentGone.html b/dom/media/webaudio/test/file_nodeCreationDocumentGone.html
new file mode 100644
index 0000000000..aedf16702f
--- /dev/null
+++ b/dom/media/webaudio/test/file_nodeCreationDocumentGone.html
@@ -0,0 +1,4 @@
+<!DOCTYPE html>
+<html><script>
+var context = new AudioContext();
+setTimeout(function(){window.close();},1000);</script></html>
diff --git a/dom/media/webaudio/test/generate-test-files.py b/dom/media/webaudio/test/generate-test-files.py
new file mode 100755
index 0000000000..af4bc3bb49
--- /dev/null
+++ b/dom/media/webaudio/test/generate-test-files.py
@@ -0,0 +1,52 @@
+#!/usr/bin/env python3
+
+import os
+
+rates = [44100, 48000]
+channels = [1, 2]
+duration = "0.5"
+frequency = "1000"
+volume = "-3dB"
+name = "half-a-second"
+formats = {
+    "aac-in-adts": [{"codec": "aac", "extension": "aac"}],
+    "mp3": [{"codec": "libmp3lame", "extension": "mp3"}],
+    "mp4": [
+        {
+            "codec": "libopus",
+            "extension": "mp4",
+        },
+        {"codec": "aac", "extension": "mp4"},
+    ],
+    "ogg": [
+        {"codec": "libvorbis", "extension": "ogg"},
+        {"codec": "libopus", "extension": "opus"},
+    ],
+    "flac": [
+        {"codec": "flac", "extension": "flac"},
+    ],
+    "webm": [
+        {"codec": "libopus", "extension": "webm"},
+        {"codec": "libvorbis", "extension": "webm"},
+    ],
+}
+
+for rate in rates:
+    for channel_count in channels:
+        wav_filename = "{}-{}ch-{}.wav".format(name, channel_count, rate)
+        wav_command = "sox -V -r {} -n -b 16 -c {} {}  synth {} sin {} vol {}".format(
+            rate, channel_count, wav_filename, duration, frequency, volume
+        )
+        print(wav_command)
+        os.system(wav_command)
+        for container, codecs in formats.items():
+            for codec in codecs:
+                encoded_filename = "{}-{}ch-{}-{}.{}".format(
+                    name, channel_count, rate, codec["codec"], codec["extension"]
+                )
+                print(encoded_filename)
+                encoded_command = "ffmpeg -y -i {} -acodec {} {}".format(
+                    wav_filename, codec["codec"], encoded_filename
+                )
+                print(encoded_command)
+                os.system(encoded_command)
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100-aac-afconvert.mp4 b/dom/media/webaudio/test/half-a-second-1ch-44100-aac-afconvert.mp4
new file mode 100644
index 0000000000..7e3b008376
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100-aac-afconvert.mp4
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100-aac.aac b/dom/media/webaudio/test/half-a-second-1ch-44100-aac.aac
new file mode 100644
index 0000000000..28435589ae
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100-aac.aac
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100-aac.mp4 b/dom/media/webaudio/test/half-a-second-1ch-44100-aac.mp4
new file mode 100644
index 0000000000..c603635f6e
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100-aac.mp4
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100-alaw.wav b/dom/media/webaudio/test/half-a-second-1ch-44100-alaw.wav
new file mode 100644
index 0000000000..4e8b6e7d45
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100-alaw.wav
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100-flac.flac b/dom/media/webaudio/test/half-a-second-1ch-44100-flac.flac
new file mode 100644
index 0000000000..49f71674f5
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100-flac.flac
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100-libmp3lame.mp3 b/dom/media/webaudio/test/half-a-second-1ch-44100-libmp3lame.mp3
new file mode 100644
index 0000000000..fb1ec45af2
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100-libmp3lame.mp3
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100-libopus.mp4 b/dom/media/webaudio/test/half-a-second-1ch-44100-libopus.mp4
new file mode 100644
index 0000000000..3a7df17582
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100-libopus.mp4
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100-libopus.opus b/dom/media/webaudio/test/half-a-second-1ch-44100-libopus.opus
new file mode 100644
index 0000000000..304b9f9d1d
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100-libopus.opus
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100-libopus.webm b/dom/media/webaudio/test/half-a-second-1ch-44100-libopus.webm
new file mode 100644
index 0000000000..71be30de9c
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100-libopus.webm
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100-libvorbis.ogg b/dom/media/webaudio/test/half-a-second-1ch-44100-libvorbis.ogg
new file mode 100644
index 0000000000..ab5ec06e50
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100-libvorbis.ogg
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100-libvorbis.webm b/dom/media/webaudio/test/half-a-second-1ch-44100-libvorbis.webm
new file mode 100644
index 0000000000..b5142703ba
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100-libvorbis.webm
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100-mulaw.wav b/dom/media/webaudio/test/half-a-second-1ch-44100-mulaw.wav
new file mode 100644
index 0000000000..5b6c1edcb4
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100-mulaw.wav
diff --git a/dom/media/webaudio/test/half-a-second-1ch-44100.wav b/dom/media/webaudio/test/half-a-second-1ch-44100.wav
new file mode 100644
index 0000000000..0028a66007
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-44100.wav
diff --git a/dom/media/webaudio/test/half-a-second-1ch-48000-aac.aac b/dom/media/webaudio/test/half-a-second-1ch-48000-aac.aac
new file mode 100644
index 0000000000..e1c5ba4631
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-48000-aac.aac
diff --git a/dom/media/webaudio/test/half-a-second-1ch-48000-aac.mp4 b/dom/media/webaudio/test/half-a-second-1ch-48000-aac.mp4
new file mode 100644
index 0000000000..089d2a93e1
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-48000-aac.mp4
diff --git a/dom/media/webaudio/test/half-a-second-1ch-48000-flac.flac b/dom/media/webaudio/test/half-a-second-1ch-48000-flac.flac
new file mode 100644
index 0000000000..783bbf2c97
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-48000-flac.flac
diff --git a/dom/media/webaudio/test/half-a-second-1ch-48000-libmp3lame.mp3 b/dom/media/webaudio/test/half-a-second-1ch-48000-libmp3lame.mp3
new file mode 100644
index 0000000000..f9dfe29a89
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-48000-libmp3lame.mp3
diff --git a/dom/media/webaudio/test/half-a-second-1ch-48000-libopus.mp4 b/dom/media/webaudio/test/half-a-second-1ch-48000-libopus.mp4
new file mode 100644
index 0000000000..eb48fdac1b
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-48000-libopus.mp4
diff --git a/dom/media/webaudio/test/half-a-second-1ch-48000-libopus.opus b/dom/media/webaudio/test/half-a-second-1ch-48000-libopus.opus
new file mode 100644
index 0000000000..1b7cefcb3f
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-48000-libopus.opus
diff --git a/dom/media/webaudio/test/half-a-second-1ch-48000-libopus.webm b/dom/media/webaudio/test/half-a-second-1ch-48000-libopus.webm
new file mode 100644
index 0000000000..c06e5d7583
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-48000-libopus.webm
diff --git a/dom/media/webaudio/test/half-a-second-1ch-48000-libvorbis.ogg b/dom/media/webaudio/test/half-a-second-1ch-48000-libvorbis.ogg
new file mode 100644
index 0000000000..ad88da968c
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-48000-libvorbis.ogg
diff --git a/dom/media/webaudio/test/half-a-second-1ch-48000-libvorbis.webm b/dom/media/webaudio/test/half-a-second-1ch-48000-libvorbis.webm
new file mode 100644
index 0000000000..d63e2c31d3
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-48000-libvorbis.webm
diff --git a/dom/media/webaudio/test/half-a-second-1ch-48000.wav b/dom/media/webaudio/test/half-a-second-1ch-48000.wav
new file mode 100644
index 0000000000..d1fcb21134
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-1ch-48000.wav
diff --git a/dom/media/webaudio/test/half-a-second-2ch-44100-aac.aac b/dom/media/webaudio/test/half-a-second-2ch-44100-aac.aac
new file mode 100644
index 0000000000..d2255e982b
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-44100-aac.aac
diff --git a/dom/media/webaudio/test/half-a-second-2ch-44100-aac.mp4 b/dom/media/webaudio/test/half-a-second-2ch-44100-aac.mp4
new file mode 100644
index 0000000000..fbdd17e416
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-44100-aac.mp4
diff --git a/dom/media/webaudio/test/half-a-second-2ch-44100-flac.flac b/dom/media/webaudio/test/half-a-second-2ch-44100-flac.flac
new file mode 100644
index 0000000000..9cc57c24bd
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-44100-flac.flac
diff --git a/dom/media/webaudio/test/half-a-second-2ch-44100-libmp3lame.mp3 b/dom/media/webaudio/test/half-a-second-2ch-44100-libmp3lame.mp3
new file mode 100644
index 0000000000..399df50839
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-44100-libmp3lame.mp3
diff --git a/dom/media/webaudio/test/half-a-second-2ch-44100-libopus.mp4 b/dom/media/webaudio/test/half-a-second-2ch-44100-libopus.mp4
new file mode 100644
index 0000000000..242fb3e12e
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-44100-libopus.mp4
diff --git a/dom/media/webaudio/test/half-a-second-2ch-44100-libopus.opus b/dom/media/webaudio/test/half-a-second-2ch-44100-libopus.opus
new file mode 100644
index 0000000000..a9311b5f9c
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-44100-libopus.opus
diff --git a/dom/media/webaudio/test/half-a-second-2ch-44100-libopus.webm b/dom/media/webaudio/test/half-a-second-2ch-44100-libopus.webm
new file mode 100644
index 0000000000..ca8876d5e1
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-44100-libopus.webm
diff --git a/dom/media/webaudio/test/half-a-second-2ch-44100-libvorbis.ogg b/dom/media/webaudio/test/half-a-second-2ch-44100-libvorbis.ogg
new file mode 100644
index 0000000000..edf76edf89
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-44100-libvorbis.ogg
diff --git a/dom/media/webaudio/test/half-a-second-2ch-44100-libvorbis.webm b/dom/media/webaudio/test/half-a-second-2ch-44100-libvorbis.webm
new file mode 100644
index 0000000000..b01575c526
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-44100-libvorbis.webm
diff --git a/dom/media/webaudio/test/half-a-second-2ch-44100.wav b/dom/media/webaudio/test/half-a-second-2ch-44100.wav
new file mode 100644
index 0000000000..ae37e12813
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-44100.wav
diff --git a/dom/media/webaudio/test/half-a-second-2ch-48000-aac.aac b/dom/media/webaudio/test/half-a-second-2ch-48000-aac.aac
new file mode 100644
index 0000000000..d26803d76f
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-48000-aac.aac
diff --git a/dom/media/webaudio/test/half-a-second-2ch-48000-aac.mp4 b/dom/media/webaudio/test/half-a-second-2ch-48000-aac.mp4
new file mode 100644
index 0000000000..d7e3140580
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-48000-aac.mp4
diff --git a/dom/media/webaudio/test/half-a-second-2ch-48000-flac.flac b/dom/media/webaudio/test/half-a-second-2ch-48000-flac.flac
new file mode 100644
index 0000000000..624e5280ff
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-48000-flac.flac
diff --git a/dom/media/webaudio/test/half-a-second-2ch-48000-libmp3lame.mp3 b/dom/media/webaudio/test/half-a-second-2ch-48000-libmp3lame.mp3
new file mode 100644
index 0000000000..bd009ebfb4
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-48000-libmp3lame.mp3
diff --git a/dom/media/webaudio/test/half-a-second-2ch-48000-libopus.mp4 b/dom/media/webaudio/test/half-a-second-2ch-48000-libopus.mp4
new file mode 100644
index 0000000000..89e2b19256
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-48000-libopus.mp4
diff --git a/dom/media/webaudio/test/half-a-second-2ch-48000-libopus.opus b/dom/media/webaudio/test/half-a-second-2ch-48000-libopus.opus
new file mode 100644
index 0000000000..1e3c72b7b2
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-48000-libopus.opus
diff --git a/dom/media/webaudio/test/half-a-second-2ch-48000-libopus.webm b/dom/media/webaudio/test/half-a-second-2ch-48000-libopus.webm
new file mode 100644
index 0000000000..c7306df2ef
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-48000-libopus.webm
diff --git a/dom/media/webaudio/test/half-a-second-2ch-48000-libvorbis.ogg b/dom/media/webaudio/test/half-a-second-2ch-48000-libvorbis.ogg
new file mode 100644
index 0000000000..63e6d2bb87
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-48000-libvorbis.ogg
diff --git a/dom/media/webaudio/test/half-a-second-2ch-48000-libvorbis.webm b/dom/media/webaudio/test/half-a-second-2ch-48000-libvorbis.webm
new file mode 100644
index 0000000000..589370d6e2
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-48000-libvorbis.webm
diff --git a/dom/media/webaudio/test/half-a-second-2ch-48000.wav b/dom/media/webaudio/test/half-a-second-2ch-48000.wav
new file mode 100644
index 0000000000..e9fc21e30b
--- /dev/null
+++ b/dom/media/webaudio/test/half-a-second-2ch-48000.wav
diff --git a/dom/media/webaudio/test/invalid.txt b/dom/media/webaudio/test/invalid.txt
new file mode 100644
index 0000000000..c44840faf1
--- /dev/null
+++ b/dom/media/webaudio/test/invalid.txt
@@ -0,0 +1 @@
+Surely this is not an audio file!
diff --git a/dom/media/webaudio/test/invalidContent.flac b/dom/media/webaudio/test/invalidContent.flac
new file mode 100644
index 0000000000..b2f4e1ff7a
--- /dev/null
+++ b/dom/media/webaudio/test/invalidContent.flac
@@ -0,0 +1 @@
+fLaC
diff --git a/dom/media/webaudio/test/layouttest-glue.js b/dom/media/webaudio/test/layouttest-glue.js
new file mode 100644
index 0000000000..0ed0b9dc90
--- /dev/null
+++ b/dom/media/webaudio/test/layouttest-glue.js
@@ -0,0 +1,18 @@
+// Reimplementation of the LayoutTest API from Blink so we can easily port
+// WebAudio tests to Simpletest, without touching the internals of the test.
+
+function testFailed(msg) {
+  ok(false, msg);
+}
+
+function testPassed(msg) {
+  ok(true, msg);
+}
+
+function finishJSTest() {
+  SimpleTest.finish();
+}
+
+function description(str) {
+  info(str);
+}
diff --git a/dom/media/webaudio/test/mochitest.toml b/dom/media/webaudio/test/mochitest.toml
new file mode 100644
index 0000000000..74d17d4fc9
--- /dev/null
+++ b/dom/media/webaudio/test/mochitest.toml
@@ -0,0 +1,332 @@
+[DEFAULT]
+tags = "mtg webaudio"
+subsuite = "media"
+support-files = [
+  "8kHz-320kbps-6ch.aac",
+  "audio-expected.wav",
+  "audio-mono-expected-2.wav",
+  "audio-mono-expected.wav",
+  "audio-quad.wav",
+  "audio.ogv",
+  "audiovideo.mp4",
+  "audioBufferSourceNodeDetached_worker.js",
+  "corsServer.sjs",
+  "!/dom/events/test/event_leak_utils.js",
+  "file_nodeCreationDocumentGone.html",
+  "invalid.txt",
+  "invalidContent.flac",
+  "layouttest-glue.js",
+  "nil-packet.ogg",
+  "noaudio.webm",
+  "small-shot-expected.wav",
+  "small-shot-mono-expected.wav",
+  "small-shot.ogg",
+  "small-shot.mp3",
+  "sweep-300-330-1sec.opus",
+  "ting-44.1k-1ch.ogg",
+  "ting-44.1k-2ch.ogg",
+  "ting-48k-1ch.ogg",
+  "ting-48k-2ch.ogg",
+  "ting-44.1k-1ch.wav",
+  "ting-44.1k-2ch.wav",
+  "ting-48k-1ch.wav",
+  "ting-48k-2ch.wav",
+  "sine-440-10s.opus",
+  "webaudio.js",
+  # See ./generate-test-files.py
+  "half-a-second-1ch-44100-aac.aac",
+  "half-a-second-1ch-44100-aac.mp4",
+  "half-a-second-1ch-44100-alaw.wav",
+  "half-a-second-1ch-44100-flac.flac",
+  "half-a-second-1ch-44100-libmp3lame.mp3",
+  "half-a-second-1ch-44100-libopus.mp4",
+  "half-a-second-1ch-44100-libopus.opus",
+  "half-a-second-1ch-44100-libopus.webm",
+  "half-a-second-1ch-44100-libvorbis.ogg",
+  "half-a-second-1ch-44100-libvorbis.webm",
+  "half-a-second-1ch-44100-mulaw.wav",
+  "half-a-second-1ch-44100.wav",
+  "half-a-second-1ch-48000-aac.aac",
+  "half-a-second-1ch-48000-aac.mp4",
+  "half-a-second-1ch-48000-flac.flac",
+  "half-a-second-1ch-48000-libmp3lame.mp3",
+  "half-a-second-1ch-48000-libopus.mp4",
+  "half-a-second-1ch-48000-libopus.opus",
+  "half-a-second-1ch-48000-libopus.webm",
+  "half-a-second-1ch-48000-libvorbis.ogg",
+  "half-a-second-1ch-48000-libvorbis.webm",
+  "half-a-second-1ch-48000.wav",
+  "half-a-second-2ch-44100-aac.aac",
+  "half-a-second-2ch-44100-aac.mp4",
+  "half-a-second-2ch-44100-flac.flac",
+  "half-a-second-2ch-44100-libmp3lame.mp3",
+  "half-a-second-2ch-44100-libopus.mp4",
+  "half-a-second-2ch-44100-libopus.opus",
+  "half-a-second-2ch-44100-libopus.webm",
+  "half-a-second-2ch-44100-libvorbis.ogg",
+  "half-a-second-2ch-44100-libvorbis.webm",
+  "half-a-second-2ch-44100.wav",
+  "half-a-second-2ch-48000-aac.aac",
+  "half-a-second-2ch-48000-aac.mp4",
+  "half-a-second-2ch-48000-flac.flac",
+  "half-a-second-2ch-48000-libmp3lame.mp3",
+  "half-a-second-2ch-48000-libopus.mp4",
+  "half-a-second-2ch-48000-libopus.opus",
+  "half-a-second-2ch-48000-libopus.webm",
+  "half-a-second-2ch-48000-libvorbis.ogg",
+  "half-a-second-2ch-48000-libvorbis.webm",
+  "half-a-second-2ch-48000.wav",
+  "half-a-second-1ch-44100-aac-afconvert.mp4",
+  "waveformatextensible.wav",
+  "waveformatextensiblebadmask.wav",
+  "sixteen-frames.mp3", # only 16 frames of valid audio
+  "../../webrtc/tests/mochitests/mediaStreamPlayback.js",
+  "../../webrtc/tests/mochitests/head.js",
+]
+
+["test_OfflineAudioContext.html"]
+
+["test_ScriptProcessorCollected1.html"]
+
+["test_WebAudioMemoryReporting.html"]
+
+["test_analyserNode.html"]
+skip-if = ["!asan && os != android"] # These are tested in web-platform-tests, except on ASan and Android which don't run WPT.
+
+["test_analyserNodeMinimum.html"]
+skip-if = ["!asan && os != android"] # These are tested in web-platform-tests, except on ASan and Android which don't run WPT.
+
+["test_analyserNodeOutput.html"]
+skip-if = ["!asan && os != android"] # These are tested in web-platform-tests, except on ASan and Android which don't run WPT.
+
+["test_analyserNodePassThrough.html"]
+
+["test_analyserNodeWithGain.html"]
+skip-if = ["!asan && os != android"] # These are tested in web-platform-tests, except on ASan and Android which don't run WPT.
+
+["test_analyserScale.html"]
+skip-if = ["!asan && os != android"] # These are tested in web-platform-tests, except on ASan and Android which don't run WPT.
+
+["test_audioContextParams_recordNonDefaultSampleRate.html"]
+
+["test_audioContextParams_sampleRate.html"]
+
+["test_channelMergerNode.html"]
+
+["test_channelMergerNodeWithVolume.html"]
+
+["test_channelSplitterNode.html"]
+
+["test_channelSplitterNodeWithVolume.html"]
+
+["test_convolver-upmixing-1-channel-response.html"]
+# This is a copy of
+# testing/web-platform/tests/webaudio/the-audio-api/the-convolvernode-interface/convolver-upmixing-1-channel-response.html,
+# but WPT are not run with ASan or Android builds.
+skip-if = ["!asan && os != android"]
+
+["test_convolverNode.html"]
+
+["test_convolverNodeChannelCount.html"]
+
+["test_convolverNodeChannelInterpretationChanges.html"]
+
+["test_convolverNodeDelay.html"]
+
+["test_convolverNodeFiniteInfluence.html"]
+
+["test_convolverNodeNormalization.html"]
+
+["test_convolverNodeOOM.html"]
+skip-if = [
+  "asan",
+  "tsan", # 1672869
+]
+
+["test_convolverNodePassThrough.html"]
+
+["test_convolverNodeWithGain.html"]
+
+["test_convolverNode_mono_mono.html"]
+
+["test_currentTime.html"]
+
+["test_decodeAudioDataOnDetachedBuffer.html"]
+
+["test_decodeAudioDataPromise.html"]
+
+["test_decodeAudioError.html"]
+
+["test_decodeMultichannel.html"]
+
+["test_decodeOpusTail.html"]
+
+["test_decoderDelay.html"]
+
+["test_delayNode.html"]
+
+["test_delayNodeAtMax.html"]
+
+["test_delayNodeChannelChanges.html"]
+
+["test_delayNodeCycles.html"]
+
+["test_delayNodePassThrough.html"]
+
+["test_delayNodeSmallMaxDelay.html"]
+
+["test_delayNodeTailIncrease.html"]
+
+["test_delayNodeTailWithDisconnect.html"]
+
+["test_delayNodeTailWithGain.html"]
+
+["test_delayNodeTailWithReconnect.html"]
+
+["test_delayNodeWithGain.html"]
+
+["test_delaynode-channel-count-1.html"]
+# This is a copy of
+# testing/web-platform/tests/webaudio/the-audio-api/the-delaynode-interface/delaynode-channel-count-1.html
+# but WPT are not run with ASan or Android builds.
+skip-if = ["!asan && os != android"]
+
+["test_disconnectAll.html"]
+
+["test_disconnectAudioParam.html"]
+
+["test_disconnectAudioParamFromOutput.html"]
+
+["test_disconnectExceptions.html"]
+
+["test_disconnectFromAudioNode.html"]
+
+["test_disconnectFromAudioNodeAndOutput.html"]
+
+["test_disconnectFromAudioNodeAndOutputAndInput.html"]
+
+["test_disconnectFromAudioNodeMultipleConnection.html"]
+
+["test_disconnectFromOutput.html"]
+
+["test_dynamicsCompressorNode.html"]
+
+["test_dynamicsCompressorNodePassThrough.html"]
+
+["test_dynamicsCompressorNodeWithGain.html"]
+
+["test_event_listener_leaks.html"]
+
+["test_gainNode.html"]
+
+["test_gainNodeInLoop.html"]
+
+["test_gainNodePassThrough.html"]
+
+["test_iirFilterNodePassThrough.html"]
+
+["test_maxChannelCount.html"]
+
+["test_mixingRules.html"]
+
+["test_nodeCreationDocumentGone.html"]
+
+["test_nodeToParamConnection.html"]
+
+["test_notAllowedToStartAudioContextGC.html"]
+
+["test_offlineDestinationChannelCountLess.html"]
+
+["test_offlineDestinationChannelCountMore.html"]
+
+["test_oscillatorNode.html"]
+
+["test_oscillatorNode2.html"]
+
+["test_oscillatorNodeNegativeFrequency.html"]
+
+["test_oscillatorNodePassThrough.html"]
+
+["test_oscillatorNodeStart.html"]
+
+["test_oscillatorTypeChange.html"]
+
+["test_pannerNode.html"]
+
+["test_pannerNodeAbove.html"]
+
+["test_pannerNodeAtZeroDistance.html"]
+
+["test_pannerNodeChannelCount.html"]
+
+["test_pannerNodeHRTFSymmetry.html"]
+
+["test_pannerNodeTail.html"]
+
+["test_pannerNode_audioparam_distance.html"]
+
+["test_pannerNode_equalPower.html"]
+
+["test_pannerNode_maxDistance.html"]
+
+["test_periodicWave.html"]
+
+["test_periodicWaveBandLimiting.html"]
+
+["test_periodicWaveDisableNormalization.html"]
+
+["test_retrospective-exponentialRampToValueAtTime.html"]
+
+["test_retrospective-linearRampToValueAtTime.html"]
+
+["test_retrospective-setTargetAtTime.html"]
+
+["test_retrospective-setValueAtTime.html"]
+
+["test_retrospective-setValueCurveAtTime.html"]
+
+["test_scriptProcessorNode.html"]
+
+["test_scriptProcessorNodeChannelCount.html"]
+
+["test_scriptProcessorNodeNotConnected.html"]
+
+["test_scriptProcessorNodePassThrough.html"]
+
+["test_scriptProcessorNodeZeroInputOutput.html"]
+
+["test_scriptProcessorNode_playbackTime1.html"]
+
+["test_sequentialBufferSourceWithResampling.html"]
+
+["test_setValueCurveWithNonFiniteElements.html"]
+
+["test_singleSourceDest.html"]
+
+["test_slowStart.html"]
+
+["test_stereoPannerNode.html"]
+
+["test_stereoPannerNodePassThrough.html"]
+
+["test_stereoPanningWithGain.html"]
+
+["test_waveDecoder.html"]
+
+["test_waveShaper.html"]
+
+["test_waveShaperGain.html"]
+
+["test_waveShaperInvalidLengthCurve.html"]
+
+["test_waveShaperNoCurve.html"]
+
+["test_waveShaperPassThrough.html"]
+
+["test_webAudio_muteTab.html"]
+scheme = "https"
+skip-if = [
+  "os == 'mac'",
+  "os == 'win'",
+  "os == 'android'", # Bug 1404995, no loopback devices on some platforms
+]
diff --git a/dom/media/webaudio/test/mochitest_audio.toml b/dom/media/webaudio/test/mochitest_audio.toml
new file mode 100644
index 0000000000..56e612b102
--- /dev/null
+++ b/dom/media/webaudio/test/mochitest_audio.toml
@@ -0,0 +1,100 @@
+[DEFAULT]
+tags = "mtg webaudio"
+subsuite = "media"
+support-files = [
+  "audio-expected.wav",
+  "audio-mono-expected-2.wav",
+  "audio-mono-expected.wav",
+  "audio-quad.wav",
+  "audio.ogv",
+  "audiovideo.mp4",
+  "audioBufferSourceNodeDetached_worker.js",
+  "corsServer.sjs",
+  "!/dom/events/test/event_leak_utils.js",
+  "file_nodeCreationDocumentGone.html",
+  "invalid.txt",
+  "invalidContent.flac",
+  "layouttest-glue.js",
+  "nil-packet.ogg",
+  "noaudio.webm",
+  "small-shot-expected.wav",
+  "small-shot-mono-expected.wav",
+  "small-shot.ogg",
+  "small-shot.mp3",
+  "sweep-300-330-1sec.opus",
+  "ting-44.1k-1ch.ogg",
+  "ting-44.1k-2ch.ogg",
+  "ting-48k-1ch.ogg",
+  "ting-48k-2ch.ogg",
+  "ting-44.1k-1ch.wav",
+  "ting-44.1k-2ch.wav",
+  "ting-48k-1ch.wav",
+  "ting-48k-2ch.wav",
+  "sine-440-10s.opus",
+  "1856145.ogg",
+  "webaudio.js",
+  "../../webrtc/tests/mochitests/mediaStreamPlayback.js",
+  "../../webrtc/tests/mochitests/head.js",
+]
+
+["test_AudioBuffer.html"]
+
+["test_AudioContext.html"]
+
+["test_AudioContext_disabled.html"]
+
+["test_AudioListener.html"]
+
+["test_AudioNodeDevtoolsAPI.html"]
+
+["test_AudioParamDevtoolsAPI.html"]
+
+["test_audioBufferSourceNode.html"]
+
+["test_audioBufferSourceNodeDetached.html"]
+
+["test_audioBufferSourceNodeEnded.html"]
+
+["test_audioBufferSourceNodeLazyLoopParam.html"]
+
+["test_audioBufferSourceNodeLoop.html"]
+
+["test_audioBufferSourceNodeLoopStartEnd.html"]
+
+["test_audioBufferSourceNodeLoopStartEndSame.html"]
+
+["test_audioBufferSourceNodeNoStart.html"]
+
+["test_audioBufferSourceNodeNullBuffer.html"]
+
+["test_audioBufferSourceNodeOffset.html"]
+
+["test_audioBufferSourceNodePassThrough.html"]
+
+["test_audioBufferSourceNodeRate.html"]
+
+["test_audioContextGC.html"]
+
+["test_audioContextSuspendResumeClose.html"]
+tags = "capturestream"
+
+["test_audioDestinationNode.html"]
+
+["test_audioParamChaining.html"]
+
+["test_audioParamExponentialRamp.html"]
+
+["test_audioParamGain.html"]
+
+["test_audioParamLinearRamp.html"]
+
+["test_audioParamSetCurveAtTime.html"]
+
+["test_audioParamSetTargetAtTime.html"]
+
+["test_audioParamSetTargetAtTimeZeroTimeConstant.html"]
+
+["test_audioParamSetValueAtTime.html"]
+
+["test_audioParamTimelineDestinationOffset.html"]
+
diff --git a/dom/media/webaudio/test/mochitest_bugs.toml b/dom/media/webaudio/test/mochitest_bugs.toml
new file mode 100644
index 0000000000..8e987d6065
--- /dev/null
+++ b/dom/media/webaudio/test/mochitest_bugs.toml
@@ -0,0 +1,89 @@
+[DEFAULT]
+tags = "mtg webaudio"
+subsuite = "media"
+support-files = [
+  "audio-expected.wav",
+  "audio-mono-expected-2.wav",
+  "audio-mono-expected.wav",
+  "audio-quad.wav",
+  "audio.ogv",
+  "audiovideo.mp4",
+  "audioBufferSourceNodeDetached_worker.js",
+  "corsServer.sjs",
+  "!/dom/events/test/event_leak_utils.js",
+  "file_nodeCreationDocumentGone.html",
+  "invalid.txt",
+  "invalidContent.flac",
+  "layouttest-glue.js",
+  "nil-packet.ogg",
+  "noaudio.webm",
+  "small-shot-expected.wav",
+  "small-shot-mono-expected.wav",
+  "small-shot.ogg",
+  "small-shot.mp3",
+  "sweep-300-330-1sec.opus",
+  "ting-44.1k-1ch.ogg",
+  "ting-44.1k-2ch.ogg",
+  "ting-48k-1ch.ogg",
+  "ting-48k-2ch.ogg",
+  "ting-44.1k-1ch.wav",
+  "ting-44.1k-2ch.wav",
+  "ting-48k-1ch.wav",
+  "ting-48k-2ch.wav",
+  "sine-440-10s.opus",
+  "webaudio.js",
+  "../../webrtc/tests/mochitests/mediaStreamPlayback.js",
+  "../../webrtc/tests/mochitests/head.js",
+]
+
+["test_bug808374.html"]
+
+["test_bug827541.html"]
+
+["test_bug839753.html"]
+
+["test_bug845960.html"]
+
+["test_bug856771.html"]
+
+["test_bug866570.html"]
+
+["test_bug866737.html"]
+
+["test_bug867089.html"]
+
+["test_bug867174.html"]
+
+["test_bug873335.html"]
+
+["test_bug875221.html"]
+
+["test_bug875402.html"]
+
+["test_bug894150.html"]
+
+["test_bug956489.html"]
+
+["test_bug964376.html"]
+
+["test_bug966247.html"]
+tags = "capturestream"
+
+["test_bug972678.html"]
+
+["test_bug1027864.html"]
+skip-if = ["true"] #Bug 1650930
+
+["test_bug1056032.html"]
+
+["test_bug1113634.html"]
+
+["test_bug1118372.html"]
+
+["test_bug1255618.html"]
+
+["test_bug1267579.html"]
+
+["test_bug1355798.html"]
+
+["test_bug1447273.html"]
diff --git a/dom/media/webaudio/test/mochitest_media.toml b/dom/media/webaudio/test/mochitest_media.toml
new file mode 100644
index 0000000000..c369afd942
--- /dev/null
+++ b/dom/media/webaudio/test/mochitest_media.toml
@@ -0,0 +1,76 @@
+[DEFAULT]
+tags = "mtg webaudio"
+subsuite = "media"
+support-files = [
+  "audio-expected.wav",
+  "audio-mono-expected-2.wav",
+  "audio-mono-expected.wav",
+  "audio-quad.wav",
+  "audio.ogv",
+  "audiovideo.mp4",
+  "audioBufferSourceNodeDetached_worker.js",
+  "corsServer.sjs",
+  "!/dom/events/test/event_leak_utils.js",
+  "file_nodeCreationDocumentGone.html",
+  "invalid.txt",
+  "invalidContent.flac",
+  "layouttest-glue.js",
+  "nil-packet.ogg",
+  "noaudio.webm",
+  "small-shot-expected.wav",
+  "small-shot-mono-expected.wav",
+  "small-shot.ogg",
+  "small-shot.mp3",
+  "sweep-300-330-1sec.opus",
+  "ting-44.1k-1ch.ogg",
+  "ting-44.1k-2ch.ogg",
+  "ting-48k-1ch.ogg",
+  "ting-48k-2ch.ogg",
+  "ting-44.1k-1ch.wav",
+  "ting-44.1k-2ch.wav",
+  "ting-48k-1ch.wav",
+  "ting-48k-2ch.wav",
+  "sine-440-10s.opus",
+  "webaudio.js",
+  "../../webrtc/tests/mochitests/mediaStreamPlayback.js",
+  "../../webrtc/tests/mochitests/head.js",
+]
+
+["test_mediaDecoding.html"]
+
+["test_mediaElementAudioSourceNode.html"]
+tags = "capturestream"
+
+["test_mediaElementAudioSourceNodeCrossOrigin.html"]
+tags = "capturestream"
+
+["test_mediaElementAudioSourceNodeFidelity.html"]
+tags = "capturestream"
+
+["test_mediaElementAudioSourceNodePassThrough.html"]
+tags = "capturestream"
+
+["test_mediaElementAudioSourceNodeVideo.html"]
+tags = "capturestream"
+
+["test_mediaStreamAudioDestinationNode.html"]
+
+["test_mediaStreamAudioSourceNode.html"]
+
+["test_mediaStreamAudioSourceNodeCrossOrigin.html"]
+tags = "capturestream"
+
+["test_mediaStreamAudioSourceNodeNoGC.html"]
+skip-if = ["os == 'mac' && debug"] # Bug 1756880 - lower frequency shutdown hangs
+scheme = "https"
+
+["test_mediaStreamAudioSourceNodePassThrough.html"]
+
+["test_mediaStreamAudioSourceNodeResampling.html"]
+tags = "capturestream"
+
+["test_mediaStreamTrackAudioSourceNode.html"]
+
+["test_mediaStreamTrackAudioSourceNodeCrossOrigin.html"]
+
+["test_mediaStreamTrackAudioSourceNodeVideo.html"]
diff --git a/dom/media/webaudio/test/nil-packet.ogg b/dom/media/webaudio/test/nil-packet.ogg
new file mode 100644
index 0000000000..7b00b5a63e
--- /dev/null
+++ b/dom/media/webaudio/test/nil-packet.ogg
diff --git a/dom/media/webaudio/test/noaudio.webm b/dom/media/webaudio/test/noaudio.webm
new file mode 100644
index 0000000000..9207017fb6
--- /dev/null
+++ b/dom/media/webaudio/test/noaudio.webm
diff --git a/dom/media/webaudio/test/sine-440-10s.opus b/dom/media/webaudio/test/sine-440-10s.opus
new file mode 100644
index 0000000000..eb91020168
--- /dev/null
+++ b/dom/media/webaudio/test/sine-440-10s.opus
diff --git a/dom/media/webaudio/test/sixteen-frames.mp3 b/dom/media/webaudio/test/sixteen-frames.mp3
new file mode 100644
index 0000000000..1d15dcad59
--- /dev/null
+++ b/dom/media/webaudio/test/sixteen-frames.mp3
diff --git a/dom/media/webaudio/test/small-shot-expected.wav b/dom/media/webaudio/test/small-shot-expected.wav
new file mode 100644
index 0000000000..2faaa8258b
--- /dev/null
+++ b/dom/media/webaudio/test/small-shot-expected.wav
diff --git a/dom/media/webaudio/test/small-shot-mono-expected.wav b/dom/media/webaudio/test/small-shot-mono-expected.wav
new file mode 100644
index 0000000000..d4e2647e42
--- /dev/null
+++ b/dom/media/webaudio/test/small-shot-mono-expected.wav
diff --git a/dom/media/webaudio/test/small-shot.mp3 b/dom/media/webaudio/test/small-shot.mp3
new file mode 100644
index 0000000000..f9397a5106
--- /dev/null
+++ b/dom/media/webaudio/test/small-shot.mp3
diff --git a/dom/media/webaudio/test/small-shot.ogg b/dom/media/webaudio/test/small-shot.ogg
new file mode 100644
index 0000000000..1a41623f81
--- /dev/null
+++ b/dom/media/webaudio/test/small-shot.ogg
diff --git a/dom/media/webaudio/test/sweep-300-330-1sec.opus b/dom/media/webaudio/test/sweep-300-330-1sec.opus
new file mode 100644
index 0000000000..619d1e0844
--- /dev/null
+++ b/dom/media/webaudio/test/sweep-300-330-1sec.opus
diff --git a/dom/media/webaudio/test/test_AudioBuffer.html b/dom/media/webaudio/test/test_AudioBuffer.html
new file mode 100644
index 0000000000..05957f679e
--- /dev/null
+++ b/dom/media/webaudio/test/test_AudioBuffer.html
@@ -0,0 +1,104 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(2, 2048, context.sampleRate);
+  SpecialPowers.gc(); // Make sure that our channels are accessible after GC
+  ok(buffer, "Buffer was allocated successfully");
+  is(buffer.sampleRate, context.sampleRate, "Correct sample rate");
+  is(buffer.length, 2048, "Correct length");
+  ok(Math.abs(buffer.duration - 2048 / context.sampleRate) < 0.0001, "Correct duration");
+  is(buffer.numberOfChannels, 2, "Correct number of channels");
+  for (var i = 0; i < buffer.numberOfChannels; ++i) {
+    var buf = buffer.getChannelData(i);
+    ok(buf, "Buffer index " + i + " exists");
+    ok(buf instanceof Float32Array, "Result is a typed array");
+    is(buf.length, buffer.length, "Correct length");
+    var foundNonZero = false;
+    for (var j = 0; j < buf.length; ++j) {
+      if (buf[j] != 0) {
+        foundNonZero = true;
+        break;
+      }
+      buf[j] = j;
+    }
+    ok(!foundNonZero, "Buffer " + i + " should be initialized to 0");
+  }
+
+  // Now test copying the channel data out of a normal buffer
+  var copy = new Float32Array(100);
+  buffer.copyFromChannel(copy, 0, 1024);
+  for (var i = 0; i < copy.length; ++i) {
+    is(copy[i], 1024 + i, "Correct sample");
+  }
+  // Test copying the channel data out of a playing buffer
+  var srcNode = context.createBufferSource();
+  srcNode.buffer = buffer;
+  srcNode.start(0);
+  copy = new Float32Array(100);
+  buffer.copyFromChannel(copy, 0, 1024);
+  for (var i = 0; i < copy.length; ++i) {
+    is(copy[i], 1024 + i, "Correct sample");
+  }
+
+  // Test copying to the channel data
+  var newData = new Float32Array(200);
+  buffer.copyToChannel(newData, 0, 100);
+  var changedData = buffer.getChannelData(0);
+  for (var i = 0; i < changedData.length; ++i) {
+    if (i < 100 || i >= 300) {
+      is(changedData[i], i, "Untouched sample");
+    } else {
+      is(changedData[i], 0, "Correct sample");
+    }
+  }
+
+  // Now, detach the array buffer
+  var worker = new Worker("audioBufferSourceNodeDetached_worker.js");
+  var data = buffer.getChannelData(0).buffer;
+  worker.postMessage(data, [data]);
+  SpecialPowers.gc();
+
+  expectNoException(function() {
+    buffer.copyFromChannel(copy, 0, 1024);
+  });
+
+  expectNoException(function() {
+    buffer.copyToChannel(newData, 0, 100);
+  });
+
+  expectException(function() {
+    context.createBuffer(2, 2048, 7999);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  expectException(function() {
+    context.createBuffer(2, 2048, 192001);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  context.createBuffer(2, 2048, 8000);  // no exception
+  context.createBuffer(2, 2048, 192000); // no exception
+  context.createBuffer(32, 2048, 48000); // no exception
+  // Null length
+  expectException(function() {
+    context.createBuffer(2, 0, 48000);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  // Null number of channels
+  expectException(function() {
+    context.createBuffer(0, 2048, 48000);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_AudioContext.html b/dom/media/webaudio/test/test_AudioContext.html
new file mode 100644
index 0000000000..50aeee489e
--- /dev/null
+++ b/dom/media/webaudio/test/test_AudioContext.html
@@ -0,0 +1,23 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ac = new AudioContext();
+  ok(ac, "Create a AudioContext object");
+  ok(ac instanceof EventTarget, "AudioContexts must be EventTargets");
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_AudioContext_disabled.html b/dom/media/webaudio/test/test_AudioContext_disabled.html
new file mode 100644
index 0000000000..9c3651a883
--- /dev/null
+++ b/dom/media/webaudio/test/test_AudioContext_disabled.html
@@ -0,0 +1,56 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can disable the AudioContext interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+const webaudio_interfaces = [
+  "AudioContext",
+  "OfflineAudioContext",
+  "AudioContext",
+  "OfflineAudioCompletionEvent",
+  "AudioNode",
+  "AudioDestinationNode",
+  "AudioParam",
+  "GainNode",
+  "DelayNode",
+  "AudioBuffer",
+  "AudioBufferSourceNode",
+  "MediaElementAudioSourceNode",
+  "ScriptProcessorNode",
+  "AudioProcessingEvent",
+  "PannerNode",
+  "AudioListener",
+  "StereoPannerNode",
+  "ConvolverNode",
+  "AnalyserNode",
+  "ChannelSplitterNode",
+  "ChannelMergerNode",
+  "DynamicsCompressorNode",
+  "BiquadFilterNode",
+  "IIRFilterNode",
+  "WaveShaperNode",
+  "OscillatorNode",
+  "PeriodicWave",
+  "MediaStreamAudioSourceNode",
+  "MediaStreamAudioDestinationNode"
+];
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  SpecialPowers.pushPrefEnv({"set": [["dom.webaudio.enabled", false]]}, function() {
+      webaudio_interfaces.forEach((e) => ok(!window[e], e + " must be disabled when the Web Audio API is disabled"));
+    SimpleTest.finish();
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_AudioListener.html b/dom/media/webaudio/test/test_AudioListener.html
new file mode 100644
index 0000000000..e3d605cbcc
--- /dev/null
+++ b/dom/media/webaudio/test/test_AudioListener.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioContext.listener and the AudioListener interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  ok("listener" in context, "AudioContext.listener should exist");
+  // The values set by the following cannot be read from script, but the
+  // implementation is simple enough, so we just make sure that nothing throws.
+  context.listener.setPosition(1.0, 1.0, 1.0);
+  context.listener.setOrientation(1.0, 1.0, 1.0, 1.0, 1.0, 1.0);
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_AudioNodeDevtoolsAPI.html b/dom/media/webaudio/test/test_AudioNodeDevtoolsAPI.html
new file mode 100644
index 0000000000..f032ed88f0
--- /dev/null
+++ b/dom/media/webaudio/test/test_AudioNodeDevtoolsAPI.html
@@ -0,0 +1,59 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the devtool AudioNode API</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+  SimpleTest.waitForExplicitFinish();
+
+  function id(node) {
+    return SpecialPowers.getPrivilegedProps(node, "id");
+  }
+
+  var ac = new AudioContext();
+  var ids;
+  var weak;
+  (function() {
+    var src1 = ac.createBufferSource();
+    var src2 = ac.createBufferSource();
+    ok(id(src2) > id(src1), "The ID should be monotonic");
+    ok(id(src1) > id(ac.destination), "The ID of the destination node should be the lowest");
+    ids = [id(src1), id(src2)];
+    weak = SpecialPowers.Cu.getWeakReference(src1);
+    is(SpecialPowers.unwrap(weak.get()), src1, "The node should support a weak reference");
+  })();
+  function observer(subject, topic, data) {
+    var id = parseInt(data);
+    var index = ids.indexOf(id);
+    if (index != -1) {
+      info("Dropping id " + id + " at index " + index);
+      ids.splice(index, 1);
+      if (!ids.length) {
+        SimpleTest.executeSoon(function() {
+          is(weak.get(), null, "The weak reference must be dropped now");
+          SpecialPowers.removeObserver(observer, "webaudio-node-demise");
+          SimpleTest.finish();
+        });
+      }
+    }
+  }
+  SpecialPowers.addObserver(observer, "webaudio-node-demise");
+
+  forceCC();
+  forceCC();
+
+  function forceCC() {
+    SpecialPowers.DOMWindowUtils.cycleCollect();
+    SpecialPowers.DOMWindowUtils.garbageCollect();
+    SpecialPowers.DOMWindowUtils.garbageCollect();
+  }
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_AudioParamDevtoolsAPI.html b/dom/media/webaudio/test/test_AudioParamDevtoolsAPI.html
new file mode 100644
index 0000000000..81fdd357c9
--- /dev/null
+++ b/dom/media/webaudio/test/test_AudioParamDevtoolsAPI.html
@@ -0,0 +1,49 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the devtool AudioParam API</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+  function checkIdAndName(node, name) {
+    is(SpecialPowers.getPrivilegedProps(node, "id"),
+       SpecialPowers.getPrivilegedProps(node[name], "parentNodeId"),
+       "The parent id should be correct");
+    is(SpecialPowers.getPrivilegedProps(node[name], "name"), name,
+       "The name of the AudioParam should be correct.");
+  }
+
+  var ac = new AudioContext(),
+      gain = ac.createGain(),
+      osc = ac.createOscillator(),
+      del = ac.createDelay(),
+      source = ac.createBufferSource(),
+      stereoPanner = ac.createStereoPanner(),
+      comp = ac.createDynamicsCompressor(),
+      biquad = ac.createBiquadFilter();
+
+  checkIdAndName(gain, "gain");
+  checkIdAndName(osc, "frequency");
+  checkIdAndName(osc, "detune");
+  checkIdAndName(del, "delayTime");
+  checkIdAndName(source, "playbackRate");
+  checkIdAndName(source, "detune");
+  checkIdAndName(stereoPanner, "pan");
+  checkIdAndName(comp, "threshold");
+  checkIdAndName(comp, "knee");
+  checkIdAndName(comp, "ratio");
+  checkIdAndName(comp, "attack");
+  checkIdAndName(comp, "release");
+  checkIdAndName(biquad, "frequency");
+  checkIdAndName(biquad, "detune");
+  checkIdAndName(biquad, "Q");
+  checkIdAndName(biquad, "gain");
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_OfflineAudioContext.html b/dom/media/webaudio/test/test_OfflineAudioContext.html
new file mode 100644
index 0000000000..d9403566ae
--- /dev/null
+++ b/dom/media/webaudio/test/test_OfflineAudioContext.html
@@ -0,0 +1,118 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test OfflineAudioContext</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var renderedBuffer = null;
+var finished = 0;
+
+function finish() {
+  finished++;
+  if (finished == 2) {
+    SimpleTest.finish();
+  }
+}
+
+function setOrCompareRenderedBuffer(aRenderedBuffer) {
+  if (renderedBuffer) {
+    is(renderedBuffer, aRenderedBuffer, "Rendered buffers from the event and the promise should be the same");
+    finish();
+  } else {
+    renderedBuffer = aRenderedBuffer;
+  }
+}
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  let ctxs = [
+    new OfflineAudioContext(2, 100, 22050),
+    new OfflineAudioContext({length: 100, sampleRate: 22050}),
+    new OfflineAudioContext({channels: 2, length: 100, sampleRate: 22050}),
+  ];
+
+  for (let ctx of ctxs) {
+    ok(ctx instanceof EventTarget, "OfflineAudioContexts must be EventTargets");
+    is(ctx.length, 100, "OfflineAudioContext.length is equal to the value passed to the ctor.");
+
+    var buf = ctx.createBuffer(2, 100, ctx.sampleRate);
+    for (var i = 0; i < 2; ++i) {
+      for (var j = 0; j < 100; ++j) {
+        buf.getChannelData(i)[j] = Math.sin(2 * Math.PI * 200 * j / ctx.sampleRate);
+      }
+    }
+  }
+
+  is(ctxs[1].destination.channelCount, 1, "OfflineAudioContext defaults to to correct channelCount.");
+
+  let ctx = ctxs[0];
+
+  expectException(function() {
+    new OfflineAudioContext(2, 100, 0);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  expectException(function() {
+    new OfflineAudioContext(2, 100, -1);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  expectException(function() {
+    new OfflineAudioContext(0, 100, 44100);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  new OfflineAudioContext(32, 100, 44100);
+  expectException(function() {
+    new OfflineAudioContext(33, 100, 44100);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  expectException(function() {
+    new OfflineAudioContext(2, 0, 44100);
+  }, DOMException.NOT_SUPPORTED_ERR);
+  expectTypeError(function() {
+    new OfflineAudioContext({});
+  });
+  expectTypeError(function() {
+    new OfflineAudioContext({sampleRate: 44100});
+  });
+  expectTypeError(function() {
+    new OfflineAudioContext({length: 44100*40});
+  });
+
+  var src = ctx.createBufferSource();
+  src.buffer = buf;
+  src.start(0);
+  src.connect(ctx.destination);
+
+  ctx.addEventListener("complete", function(e) {
+    ok(e instanceof OfflineAudioCompletionEvent, "Correct event received");
+    is(e.renderedBuffer.numberOfChannels, 2, "Correct expected number of buffers");
+    ok(renderedBuffer != null, "The event should be fired after the promise callback.");
+    expectNoException(function() {
+      ctx.startRendering().then(function() {
+        ok(false, "Promise should not resolve when startRendering is called a second time on an OfflineAudioContext")
+        finish();
+      }).catch(function(err) {
+        ok(true, "Promise should reject when startRendering is called a second time on an OfflineAudioContext")
+        finish();
+      });
+    });
+    compareBuffers(e.renderedBuffer, buf);
+    setOrCompareRenderedBuffer(e.renderedBuffer);
+
+  });
+
+  expectNoException(function() {
+    ctx.startRendering().then(function(b) {
+      is(renderedBuffer, null, "The promise callback should be called first.");
+      setOrCompareRenderedBuffer(b);
+    }).catch(function (error) {
+       ok(false, "The promise from OfflineAudioContext.startRendering should never be rejected");
+    });
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_ScriptProcessorCollected1.html b/dom/media/webaudio/test/test_ScriptProcessorCollected1.html
new file mode 100644
index 0000000000..8f05889d26
--- /dev/null
+++ b/dom/media/webaudio/test/test_ScriptProcessorCollected1.html
@@ -0,0 +1,77 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ScriptProcessorNode in cycle with no listener is collected</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+var observer = function(subject, topic, data) {
+  var id = parseInt(data);
+  var index = ids.indexOf(id);
+  if (index != -1) {
+    ok(true, "Collected AudioNode id " + id + " at index " + index);
+    ids.splice(index, 1);
+  }
+}
+
+SpecialPowers.addObserver(observer, "webaudio-node-demise");
+
+SimpleTest.registerCleanupFunction(function() {
+  if (observer) {
+    SpecialPowers.removeObserver(observer, "webaudio-node-demise");
+  }
+});
+
+var ac = new AudioContext();
+
+var testProcessor = ac.createScriptProcessor(256, 1, 0);
+var delay = ac.createDelay();
+testProcessor.connect(delay);
+delay.connect(testProcessor);
+
+var referenceProcessor = ac.createScriptProcessor(256, 1, 0);
+var gain = ac.createGain();
+gain.connect(referenceProcessor);
+
+var processCount = 0;
+testProcessor.onaudioprocess = function(event) {
+  ++processCount;
+  switch (processCount) {
+  case 1:
+    // Switch to listening to referenceProcessor;
+    referenceProcessor.onaudioprocess = event.target.onaudioprocess;
+    referenceProcessor = null;
+    event.target.onaudioprocess = null;
+    break;
+  case 2:
+    // There are no references to testProcessor and so GC can begin.
+    SpecialPowers.exactGC(function() {
+      SpecialPowers.removeObserver(observer, "webaudio-node-demise");
+      observer = null;
+      event.target.onaudioprocess = null;
+      ok(!ids.length, "All expected nodes should be collected");
+      SimpleTest.finish();
+    });
+    break;
+  }
+};
+
+function id(node) {
+  return SpecialPowers.getPrivilegedProps(node, "id");
+}
+
+// Nodes with these ids should be collected.
+var ids = [ id(testProcessor), id(delay), id(gain) ];
+testProcessor = null;
+delay = null;
+gain = null;
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_WebAudioMemoryReporting.html b/dom/media/webaudio/test/test_WebAudioMemoryReporting.html
new file mode 100644
index 0000000000..693e558304
--- /dev/null
+++ b/dom/media/webaudio/test/test_WebAudioMemoryReporting.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Web Audio memory reporting</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+var ac = new AudioContext();
+var sp = ac.createScriptProcessor(4096, 1, 1);
+sp.connect(ac.destination);
+
+// Not started so as to test
+// https://bugzilla.mozilla.org/show_bug.cgi?id=1225003#c2
+var oac = new OfflineAudioContext(1, 1, 48000);
+
+var nodeTypes = ["ScriptProcessorNode", "AudioDestinationNode"];
+var objectTypes = ["dom-nodes", "engine-objects", "track-objects"];
+
+var usages = { "explicit/webaudio/audiocontext": 0 };
+
+for (var i = 0; i < nodeTypes.length; ++i) {
+  for (var j = 0; j < objectTypes.length; ++j) {
+    usages["explicit/webaudio/audio-node/" +
+           nodeTypes[i] + "/" + objectTypes[j]] = 0;
+  }
+}
+
+var handleReport = function(aProcess, aPath, aKind, aUnits, aAmount, aDesc) {
+  if (aPath in usages) {
+    usages[aPath] += aAmount;
+  }
+}
+
+var finished = function () {
+  ok(true, "Yay didn't crash!");
+  for (var resource in usages) {
+    ok(usages[resource] > 0, "Non-zero usage for " + resource);
+  };
+  SimpleTest.finish();
+}
+
+SpecialPowers.Cc["@mozilla.org/memory-reporter-manager;1"].
+  getService(SpecialPowers.Ci.nsIMemoryReporterManager).
+  getReports(handleReport, null, finished, null, /* anonymized = */ false);
+
+// To test bug 1225003, run a failing decodeAudioData() job over a time when
+// the tasks from getReports() are expected to run.
+ac.decodeAudioData(new ArrayBuffer(4), function(){}, function(){});
+</script>
+</html>
diff --git a/dom/media/webaudio/test/test_analyserNode.html b/dom/media/webaudio/test/test_analyserNode.html
new file mode 100644
index 0000000000..0793eeb2cb
--- /dev/null
+++ b/dom/media/webaudio/test/test_analyserNode.html
@@ -0,0 +1,178 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AnalyserNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function testNode() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var destination = context.destination;
+
+  var source = context.createBufferSource();
+
+  var analyser = context.createAnalyser();
+
+  source.buffer = buffer;
+
+  source.connect(analyser);
+  analyser.connect(destination);
+
+  is(analyser.channelCount, 2, "analyser node has 2 input channels by default");
+  is(analyser.channelCountMode, "max", "Correct channelCountMode for the analyser node");
+  is(analyser.channelInterpretation, "speakers", "Correct channelCountInterpretation for the analyser node");
+
+  is(analyser.fftSize, 2048, "Correct default value for fftSize");
+  is(analyser.frequencyBinCount, 1024, "Correct default value for frequencyBinCount");
+  expectException(function() {
+    analyser.fftSize = 0;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 1;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 8;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 100; // non-power of two
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 2049;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 4097;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 8193;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 16385;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 32769;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.fftSize = 65536;
+  }, DOMException.INDEX_SIZE_ERR);
+  analyser.fftSize = 1024;
+  is(analyser.frequencyBinCount, 512, "Correct new value for frequencyBinCount");
+
+  is(analyser.minDecibels, -100, "Correct default value for minDecibels");
+  is(analyser.maxDecibels, -30, "Correct default value for maxDecibels");
+  expectException(function() {
+    analyser.minDecibels = -30;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.minDecibels = -29;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.maxDecibels = -100;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.maxDecibels = -101;
+  }, DOMException.INDEX_SIZE_ERR);
+
+  ok(Math.abs(analyser.smoothingTimeConstant - 0.8) < 0.001, "Correct default value for smoothingTimeConstant");
+  expectException(function() {
+    analyser.smoothingTimeConstant = -0.1;
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser.smoothingTimeConstant = 1.1;
+  }, DOMException.INDEX_SIZE_ERR);
+  analyser.smoothingTimeConstant = 0;
+  analyser.smoothingTimeConstant = 1;
+}
+
+function testConstructor() {
+  var context = new AudioContext();
+
+  var analyser = new AnalyserNode(context);
+  is(analyser.channelCount, 2, "analyser node has 2 input channels by default");
+  is(analyser.channelCountMode, "max", "Correct channelCountMode for the analyser node");
+  is(analyser.channelInterpretation, "speakers", "Correct channelCountInterpretation for the analyser node");
+
+  is(analyser.fftSize, 2048, "Correct default value for fftSize");
+  is(analyser.frequencyBinCount, 1024, "Correct default value for frequencyBinCount");
+  is(analyser.minDecibels, -100, "Correct default value for minDecibels");
+  is(analyser.maxDecibels, -30, "Correct default value for maxDecibels");
+  ok(Math.abs(analyser.smoothingTimeConstant - 0.8) < 0.001, "Correct default value for smoothingTimeConstant");
+
+  expectException(function() {
+    analyser = new AnalyserNode(context, { fftSize: 0 });
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { fftSize: 1 });
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { fftSize: 8 });
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { fftSize: 100 }); // non-power of two
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { fftSize: 2049 });
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { fftSize: 4097 });
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { fftSize: 8193 });
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { fftSize: 16385 });
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { fftSize: 32769 });
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { fftSize: 65536 });
+  }, DOMException.INDEX_SIZE_ERR);
+  analyser = new AnalyserNode(context, { fftSize: 1024 });
+  is(analyser.frequencyBinCount, 512, "Correct new value for frequencyBinCount");
+
+  expectException(function() {
+    analyser = new AnalyserNode(context, { minDecibels: -30 });
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { minDecibels: -29 });
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { maxDecibels: -100 });
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { maxDecibels: -101 });
+  }, DOMException.INDEX_SIZE_ERR);
+
+  expectException(function() {
+    analyser = new AnalyserNode(context, { smoothingTimeConstant: -0.1 });
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    analyser = new AnalyserNode(context, { smoothingTimeConstant: -1.1 });
+  }, DOMException.INDEX_SIZE_ERR);
+  analyser = new AnalyserNode(context, { smoothingTimeConstant: 0 });
+  analyser = new AnalyserNode(context, { smoothingTimeConstant: 1 });
+}
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+  testNode();
+  testConstructor();
+
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_analyserNodeMinimum.html b/dom/media/webaudio/test/test_analyserNodeMinimum.html
new file mode 100644
index 0000000000..950fd1812b
--- /dev/null
+++ b/dom/media/webaudio/test/test_analyserNodeMinimum.html
@@ -0,0 +1,51 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AnalyserNode when the input is silent</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+  var ac = new AudioContext();
+  var analyser = ac.createAnalyser();
+  var constant = ac.createConstantSource();
+  var sp = ac.createScriptProcessor(2048, 1, 0);
+
+  constant.offset.value = 0.0;
+
+  constant.connect(analyser)
+          .connect(ac.destination);
+
+  constant.connect(sp);
+
+  var buf = new Float32Array(analyser.frequencyBinCount);
+  var iteration_count = 10;
+  sp.onaudioprocess = function() {
+    analyser.getFloatFrequencyData(buf);
+    var correct = true;
+    for (var i = 0; i < buf.length; i++) {
+     correct &= buf[i] == -Infinity;
+    }
+    ok(correct, "silent input process -Infinity in decibel bins");
+    if(!(iteration_count--)) {
+      sp.onaudioprocess = null;
+      constant.stop();
+      ac.close();
+      SimpleTest.finish();
+    }
+  }
+
+  constant.start();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_analyserNodeOutput.html b/dom/media/webaudio/test/test_analyserNodeOutput.html
new file mode 100644
index 0000000000..27b354e92d
--- /dev/null
+++ b/dom/media/webaudio/test/test_analyserNodeOutput.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AnalyserNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+
+    var analyser = context.createAnalyser();
+
+    source.buffer = this.buffer;
+
+    source.connect(analyser);
+
+    source.start(0);
+    return analyser;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_analyserNodePassThrough.html b/dom/media/webaudio/test/test_analyserNodePassThrough.html
new file mode 100644
index 0000000000..50ff94a8c7
--- /dev/null
+++ b/dom/media/webaudio/test/test_analyserNodePassThrough.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AnalyserNode with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+
+    var analyser = context.createAnalyser();
+
+    source.buffer = this.buffer;
+
+    source.connect(analyser);
+
+    var analyserWrapped = SpecialPowers.wrap(analyser);
+    ok("passThrough" in analyserWrapped, "AnalyserNode should support the passThrough API");
+    analyserWrapped.passThrough = true;
+
+    source.start(0);
+    return analyser;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_analyserNodeWithGain.html b/dom/media/webaudio/test/test_analyserNodeWithGain.html
new file mode 100644
index 0000000000..fa0a2caa75
--- /dev/null
+++ b/dom/media/webaudio/test/test_analyserNodeWithGain.html
@@ -0,0 +1,47 @@
+<!DOCTYPE html>
+<title>Test effect of AnalyserNode on GainNode output</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+promise_test(function() {
+  // fftSize <= bufferSize so that the time domain data is full of input after
+  // processing the buffer.
+  const fftSize = 32;
+  const bufferSize = 128;
+
+  var context = new OfflineAudioContext(1, bufferSize, 48000);
+
+  var analyser1 = context.createAnalyser();
+  analyser1.fftSize = fftSize;
+  analyser1.connect(context.destination);
+  var analyser2 = context.createAnalyser();
+  analyser2.fftSize = fftSize;
+
+  var gain = context.createGain();
+  gain.gain.value = 2.0;
+  gain.connect(analyser1);
+  gain.connect(analyser2);
+
+  // Create a DC input to make getFloatTimeDomainData() output consistent at
+  // any time.
+  var buffer = context.createBuffer(1, 1, context.sampleRate);
+  buffer.getChannelData(0)[0] = 1.0 / gain.gain.value;
+  var source = context.createBufferSource();
+  source.buffer = buffer;
+  source.loop = true;
+  source.connect(gain);
+  source.start();
+
+  return context.startRendering().
+    then(function(buffer) {
+      assert_equals(buffer.getChannelData(0)[0], 1.0,
+                    "analyser1 output");
+
+      var data = new Float32Array(1);
+      analyser1.getFloatTimeDomainData(data);
+      assert_equals(data[0], 1.0, "analyser1 time domain data");
+      analyser2.getFloatTimeDomainData(data);
+      assert_equals(data[0], 1.0, "analyser2 time domain data");
+    });
+});
+</script>
diff --git a/dom/media/webaudio/test/test_analyserScale.html b/dom/media/webaudio/test/test_analyserScale.html
new file mode 100644
index 0000000000..dc4260c47a
--- /dev/null
+++ b/dom/media/webaudio/test/test_analyserScale.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AnalyserNode when the input is scaled </title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+  var context = new AudioContext();
+
+  var gain = context.createGain();
+  var analyser = context.createAnalyser();
+  var osc = context.createOscillator();
+
+
+  osc.connect(gain);
+  gain.connect(analyser);
+
+  osc.start();
+
+  var array = new Uint8Array(analyser.frequencyBinCount);
+
+  function getAnalyserData() {
+    gain.gain.setValueAtTime(currentGain, context.currentTime);
+    analyser.getByteTimeDomainData(array);
+    var max = -1;
+    for (var i = 0; i < array.length; i++) {
+      if (array[i] > max) {
+        max = Math.abs(array[i] - 128);
+      }
+    }
+    if (max <= currentGain * 128) {
+      ok(true, "Analyser got scaled data for " + currentGain);
+      currentGain = tests.shift();
+      if (currentGain == undefined) {
+        SimpleTest.finish();
+        return;
+      }
+    }
+    requestAnimationFrame(getAnalyserData);
+  }
+
+  var tests = [1.0, 0.5, 0.0];
+  var currentGain = tests.shift();
+  requestAnimationFrame(getAnalyserData);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNode.html b/dom/media/webaudio/test/test_audioBufferSourceNode.html
new file mode 100644
index 0000000000..fc7c0b48d1
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNode.html
@@ -0,0 +1,44 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  createGraph(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+    source.start(0);
+    source.buffer = buffer;
+    return source;
+  },
+  createExpectedBuffers(context) {
+    var buffers = [];
+    var buffer = context.createBuffer(2, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+      buffer.getChannelData(1)[i] = buffer.getChannelData(0)[i];
+    }
+    buffers.push(buffer);
+    buffers.push(getEmptyBuffer(context, 2048));
+    return buffers;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeDetached.html b/dom/media/webaudio/test/test_audioBufferSourceNodeDetached.html
new file mode 100644
index 0000000000..e84c33e585
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeDetached.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode when an AudioBuffer's getChanneData buffer is detached</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function createGarbage() {
+  var s = [];
+  for (var i = 0; i < 10000000; ++i) {
+    s.push(i);
+  }
+  var sum = 0;
+  for (var i = 0; i < s.length; ++i) {
+    sum += s[i];
+  }
+  return sum;
+}
+
+var worker = new Worker("audioBufferSourceNodeDetached_worker.js");
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var buffer = context.createBuffer(1, 10000000, context.sampleRate);
+    var data = buffer.getChannelData(0);
+    for (var i = 0; i < data.length; ++i) {
+      data[i] = (i%100)/100 - 0.5;
+    }
+
+    // Detach the buffer now
+    var data = buffer.getChannelData(0).buffer;
+    worker.postMessage(data, [data]);
+    // Create garbage and GC to replace the buffer data with garbage
+    SpecialPowers.gc();
+    createGarbage();
+    SpecialPowers.gc();
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.start();
+    // This should play silence
+    return source;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeEnded.html b/dom/media/webaudio/test/test_audioBufferSourceNodeEnded.html
new file mode 100644
index 0000000000..a11bb880a2
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeEnded.html
@@ -0,0 +1,36 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ended event on AudioBufferSourceNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var source = context.createBufferSource();
+
+  source.onended = function(e) {
+    is(e.target, source, "Correct target for the ended event");
+    SimpleTest.finish();
+  };
+
+  source.start(0);
+  source.buffer = buffer;
+  source.connect(context.destination);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeLazyLoopParam.html b/dom/media/webaudio/test/test_audioBufferSourceNodeLazyLoopParam.html
new file mode 100644
index 0000000000..757d4487c4
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeLazyLoopParam.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph(context) {
+    // silence for half of the buffer, ones after that.
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 1024; i < 2048; i++) {
+      buffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+
+    // we start at the 1024 frames, we should only have ones.
+    source.loop = true;
+    source.loopStart = 1024 / context.sampleRate;
+    source.loopEnd = 2048 / context.sampleRate;
+    source.buffer = buffer;
+    source.start(0, 1024 / context.sampleRate, 2048 / context.sampleRate);
+    return source;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 4096, context.sampleRate);
+    for (var i = 0; i < 2048; i++) {
+      expectedBuffer.getChannelData(0)[i] = 1;
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeLoop.html b/dom/media/webaudio/test/test_audioBufferSourceNodeLoop.html
new file mode 100644
index 0000000000..10d5d99108
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeLoop.html
@@ -0,0 +1,45 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode looping</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048 * 4,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    source.start(0);
+    source.loop = true;
+    return source;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048 * 4, context.sampleRate);
+    for (var i = 0; i < 4; ++i) {
+      for (var j = 0; j < 2048; ++j) {
+        expectedBuffer.getChannelData(0)[i * 2048 + j] = Math.sin(440 * 2 * Math.PI * j / context.sampleRate);
+      }
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeLoopStartEnd.html b/dom/media/webaudio/test/test_audioBufferSourceNodeLoopStartEnd.html
new file mode 100644
index 0000000000..1ef08e0b83
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeLoopStartEnd.html
@@ -0,0 +1,48 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode looping</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048 * 4,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = new AudioBufferSourceNode(context, {buffer, loop: true, loopStart: buffer.duration * 0.25, loopEnd: buffer.duration * 0.75 });
+    source.start(0);
+    return source;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048 * 4, context.sampleRate);
+    for (var i = 0; i < 1536; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    for (var i = 0; i < 6; ++i) {
+      for (var j = 512; j < 1536; ++j) {
+        expectedBuffer.getChannelData(0)[1536 + i * 1024 + j - 512] = Math.sin(440 * 2 * Math.PI * j / context.sampleRate);
+      }
+    }
+    for (var j = 7680; j < 2048 * 4; ++j) {
+      expectedBuffer.getChannelData(0)[j] = Math.sin(440 * 2 * Math.PI * (j - 7168) / context.sampleRate);
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeLoopStartEndSame.html b/dom/media/webaudio/test/test_audioBufferSourceNodeLoopStartEndSame.html
new file mode 100644
index 0000000000..cfe054f838
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeLoopStartEndSame.html
@@ -0,0 +1,44 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode looping</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    source.loop = true;
+    source.loopStart = source.loopEnd = 1 / context.sampleRate;
+    source.start(0);
+    return source;
+  },
+  createExpectedBuffers(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    return buffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeNoStart.html b/dom/media/webaudio/test/test_audioBufferSourceNodeNoStart.html
new file mode 100644
index 0000000000..e9a0472e2a
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeNoStart.html
@@ -0,0 +1,33 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode when start() is not called</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    var data = buffer.getChannelData(0);
+    for (var i = 0; i < data.length; ++i) {
+      data[i] = (i%100)/100 - 0.5;
+    }
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    return source;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeNullBuffer.html b/dom/media/webaudio/test/test_audioBufferSourceNodeNullBuffer.html
new file mode 100644
index 0000000000..b0b405b366
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeNullBuffer.html
@@ -0,0 +1,31 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+
+    source.start(0);
+    source.buffer = null;
+    is(source.buffer, null, "Try playing back a null buffer");
+    return source;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeOffset.html b/dom/media/webaudio/test/test_audioBufferSourceNodeOffset.html
new file mode 100644
index 0000000000..0411b74ce5
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeOffset.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the offset property on AudioBufferSourceNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var fuzz = 0.3;
+
+if (navigator.platform.startsWith("Mac")) {
+  // bug 895720
+  fuzz = 0.6;
+}
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var samplesFromSource = 0;
+  var context = new AudioContext();
+  var sp = context.createScriptProcessor(256);
+
+  sp.onaudioprocess = function(e) {
+    samplesFromSource += e.inputBuffer.length;
+  }
+
+  var buffer = context.createBuffer(1, context.sampleRate, context.sampleRate);
+  for (var i = 0; i < context.sampleRate; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var source = context.createBufferSource();
+
+  source.onended = function(e) {
+    // The timing at which the audioprocess and ended listeners are called can
+    // change, hence the fuzzy equal here.
+    var errorRatio = samplesFromSource / (0.5 * context.sampleRate);
+    ok(errorRatio > (1.0 - fuzz) && errorRatio < (1.0 + fuzz),
+       "Correct number of samples received (expected: " +
+       (0.5 * context.sampleRate) + ", actual: " + samplesFromSource + ").");
+    SimpleTest.finish();
+  };
+
+  source.buffer = buffer;
+  source.connect(sp);
+  source.start(0, 0.5);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodePassThrough.html b/dom/media/webaudio/test/test_audioBufferSourceNodePassThrough.html
new file mode 100644
index 0000000000..6cb0cccf99
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodePassThrough.html
@@ -0,0 +1,45 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+
+    source.buffer = buffer;
+
+    var srcWrapped = SpecialPowers.wrap(source);
+    ok("passThrough" in srcWrapped, "AudioBufferSourceNode should support the passThrough API");
+    srcWrapped.passThrough = true;
+
+    source.start(0);
+    return source;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+
+    return [expectedBuffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioBufferSourceNodeRate.html b/dom/media/webaudio/test/test_audioBufferSourceNodeRate.html
new file mode 100644
index 0000000000..60a065c2a9
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioBufferSourceNodeRate.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+var rate = 44100;
+var off = new OfflineAudioContext(1, rate, rate);
+var off2 = new OfflineAudioContext(1, rate, rate);
+
+var source = off.createBufferSource();
+var source2 = off2.createBufferSource();
+
+// a buffer of a 440Hz at half the length. If we detune by -1200 or set the
+// playbackRate to 0.5, we should get 44100 samples back with a sine at 220Hz.
+var buf = off.createBuffer(1, rate / 2, rate);
+var bufarray = buf.getChannelData(0);
+for (var i = 0; i < bufarray.length; i++) {
+  bufarray[i] = Math.sin(i * 440 * 2 * Math.PI / rate);
+}
+
+source.buffer = buf;
+source.playbackRate.value = 0.25; // slowdown to 25% speed
+source.connect(off.destination);
+source.start(0);
+
+source2.buffer = buf;
+source2.detune.value = -2400; // two octaves -> slowdown to 25% speed
+source2.connect(off2.destination);
+source2.start(0);
+
+off.startRendering().then((renderedPlaybackRate) => {
+  // we don't care about comparing the value here, we just want to know whether
+  // the second part is noisy.
+  var rmsValue = rms(renderedPlaybackRate, 0, 22050);
+  ok(rmsValue != 0, "Resampling happened (rms of the second part " + rmsValue + ")");
+
+  off2.startRendering().then((renderedDetune) => {
+    var rmsValue = rms(renderedDetune, 0, 22050);
+    ok(rmsValue != 0, "Resampling happened (rms of the second part " + rmsValue + ")");
+    // The two buffers should be the same: detune of -2400 is a slowdown to 25% speed
+    compareBuffers(renderedPlaybackRate, renderedDetune);
+    SimpleTest.finish();
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
\ No newline at end of file
diff --git a/dom/media/webaudio/test/test_audioContextGC.html b/dom/media/webaudio/test/test_audioContextGC.html
new file mode 100644
index 0000000000..be9990cfad
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioContextGC.html
@@ -0,0 +1,162 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test inactive AudioContext is garbage collected</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+let ids;
+
+const observer = (subject, topic, data) => {
+  const id = parseInt(data);
+  if (ids) {
+    ok(ids.delete(id), "Collected AudioNode id " + id);
+  }
+}
+SpecialPowers.addObserver(observer, "webaudio-node-demise");
+
+SimpleTest.registerCleanupFunction(function() {
+  if (observer) {
+    SpecialPowers.removeObserver(observer, "webaudio-node-demise");
+  }
+});
+
+function id(node) {
+  return SpecialPowers.getPrivilegedProps(node, "id");
+}
+
+let tests = [{
+  name: "Bare running AudioContext", setup: () => {
+    const ac = new AudioContext();
+    ids.add(id(ac.destination));
+    // Await state change notification before collection.
+    return new Promise((resolve) => {
+      ac.onstatechange = () => {
+        is(ac.state, "running", "ac.state");
+        resolve();
+      };
+    });
+  }
+}, {
+  name: "Stopped source", setup: () => {
+    const ac = new AudioContext();
+    ids.add(id(ac.destination));
+    const source = new ConstantSourceNode(ac);
+    ids.add(id(source));
+    source.start();
+    source.stop();
+    // Await ended notification before collection.
+    return new Promise((resolve) => {
+      source.onended = () => {
+        is(ac.state, "running", "ac.state");
+        resolve();
+      };
+    });
+  }
+}, {
+  name: "OfflineAudioContext not started", setup: () => {
+    const ac = new OfflineAudioContext({
+      numberOfChannels: 1, length: 1, sampleRate: 48000
+    });
+    ids.add(id(ac.destination));
+    const source = new ConstantSourceNode(ac);
+    ids.add(id(source));
+    source.start();
+  }
+}, {
+  name: "Completed OfflineAudioContext", setup: async () => {
+    const ac = new OfflineAudioContext({
+      numberOfChannels: 1, length: 1, sampleRate: 48000
+    });
+    ids.add(id(ac.destination));
+    const sourceBeforeStart = new ConstantSourceNode(ac);
+    ids.add(id(sourceBeforeStart));
+    sourceBeforeStart.start();
+    ac.startRendering();
+    await new Promise((resolve) => {
+      ac.oncomplete = () => {
+        resolve();
+      };
+    });
+    const sourceAfterComplete = new ConstantSourceNode(ac);
+    ids.add(id(sourceAfterComplete));
+    sourceAfterComplete.start();
+  }
+}, {
+  name: "suspended AudioContext", setup: async () => {
+    const ac = new AudioContext();
+    ids.add(id(ac.destination));
+    const sourceBeforeSuspend = new ConstantSourceNode(ac);
+    ids.add(id(sourceBeforeSuspend));
+    sourceBeforeSuspend.start();
+    ac.suspend();
+    const sourceAfterSuspend = new ConstantSourceNode(ac);
+    ids.add(id(sourceAfterSuspend));
+    sourceAfterSuspend.start();
+    await new Promise((resolve) => {
+      ac.onstatechange = () => {
+        if (ac.state == "suspended") {
+          resolve();
+        }
+      };
+    });
+    const sourceAfterSuspended = new ConstantSourceNode(ac);
+    ids.add(id(sourceAfterSuspended));
+    sourceAfterSuspended.start();
+  }
+}, {
+  name: "closed AudioContext", setup: async () => {
+    const ac = new AudioContext();
+    ids.add(id(ac.destination));
+    const sourceBeforeClose = new ConstantSourceNode(ac);
+    ids.add(id(sourceBeforeClose));
+    sourceBeforeClose.start();
+    ac.close();
+    const sourceAfterClose = new ConstantSourceNode(ac);
+    ids.add(id(sourceAfterClose));
+    sourceAfterClose.start();
+    await new Promise((resolve) => {
+      ac.onstatechange = () => {
+        if (ac.state == "closed") {
+          resolve();
+        }
+      };
+    });
+    const sourceAfterClosed = new ConstantSourceNode(ac);
+    ids.add(id(sourceAfterClosed));
+    sourceAfterClosed.start();
+  }
+}];
+
+const start_next_test = async () => {
+  const test = tests.shift();
+  if (!test) {
+    SimpleTest.finish();
+    return;
+  }
+  // Collect all audio nodes from previous tests.
+  if (!ids) {
+    await new Promise(resolve => {
+      SpecialPowers.exactGC(resolve);
+    });
+  }
+  ids = new Set();
+  await test.setup();
+  SpecialPowers.exactGC(() => {
+    is(ids.size, 0,
+       `All expected nodes for "${test.name}" should be collected`);
+    start_next_test();
+  });
+}
+
+start_next_test();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioContextParams_recordNonDefaultSampleRate.html b/dom/media/webaudio/test/test_audioContextParams_recordNonDefaultSampleRate.html
new file mode 100644
index 0000000000..9f2e6c9116
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioContextParams_recordNonDefaultSampleRate.html
@@ -0,0 +1,48 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  <script type="text/javascript" src="manifest.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<script class="testbody" type="text/javascript">
+function startTest() {
+  let ctx = new AudioContext({sampleRate: 32000});
+  let oscillator = ctx.createOscillator();
+  let dest = ctx.createMediaStreamDestination();
+  oscillator.connect(dest);
+  oscillator.start();
+  let stream = dest.stream;
+
+  let recorder = new MediaRecorder(stream);
+  recorder.ondataavailable = (e) => {
+      ok(true, 'recorder ondataavailable event');
+      if (recorder.state == 'recording') {
+        ok(e.data.size > 0, 'check blob has data');
+        recorder.stop();
+      }
+  }
+
+  recorder.onstop = () => {
+      ok(true, 'recorder stop event');
+      SimpleTest.finish();
+  }
+
+  try {
+    recorder.start(1000);
+    ok(true, 'recorder started');
+    is('recording', recorder.state, 'check record state recording');
+  } catch (e) {
+    ok(false, 'Can t record audio context');
+  }
+}
+
+startTest();
+SimpleTest.waitForExplicitFinish();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioContextParams_sampleRate.html b/dom/media/webaudio/test/test_audioContextParams_sampleRate.html
new file mode 100644
index 0000000000..65cee61243
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioContextParams_sampleRate.html
@@ -0,0 +1,82 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<script>
+/* import-globals-from ../../webrtc/tests/mochitests/mediaStreamPlayback.js */
+createHTML({
+  title: "Parallel MTG by setting AudioContextParam sample rate",
+  bug: "1387454",
+  visible: true
+});
+
+runTest(async () => {
+  // Test an AudioContext of specific sample rate.
+  // Verify that the oscillator produces a tone.
+  const rate1 = 500;
+  const ac1 = new AudioContext({sampleRate: 44100});
+  const dest_ac1 = ac1.createMediaStreamDestination();
+  const osc_ac1 = ac1.createOscillator();
+  osc_ac1.frequency.value = rate1;
+  osc_ac1.connect(dest_ac1);
+  osc_ac1.start(0);
+
+  const analyser = new AudioStreamAnalyser(ac1, dest_ac1.stream);
+  analyser.enableDebugCanvas();
+  await analyser.waitForAnalysisSuccess( array => {
+    const freg_50Hz   = array[analyser.binIndexForFrequency(50)];
+    const freq_rate1  = array[analyser.binIndexForFrequency(rate1)];
+    const freq_4000Hz = array[analyser.binIndexForFrequency(4000)];
+
+    info("Analysing audio frequency - low:target1:high = "
+            + freg_50Hz + ':' + freq_rate1 + ':' + freq_4000Hz);
+    return freg_50Hz < 50 && freq_rate1 > 200 && freq_4000Hz < 50;
+  })
+  osc_ac1.stop();
+
+  // Same test using a  new AudioContext of different sample rate.
+  const rate2 = 1500;
+  const ac2 = new AudioContext({sampleRate: 48000});
+  const dest_ac2 = ac2.createMediaStreamDestination();
+  const osc_ac2 = ac2.createOscillator();
+  osc_ac2.frequency.value = rate2;
+  osc_ac2.connect(dest_ac2);
+  osc_ac2.start(0);
+
+  const analyser2 = new AudioStreamAnalyser(ac2, dest_ac2.stream);
+  analyser2.enableDebugCanvas();
+  await analyser2.waitForAnalysisSuccess( array => {
+    const freg_50Hz   = array[analyser2.binIndexForFrequency(50)];
+    const freq_rate2  = array[analyser2.binIndexForFrequency(rate2)];
+    const freq_4000Hz = array[analyser2.binIndexForFrequency(4000)];
+
+    info("Analysing audio frequency - low:target2:high = "
+            + freg_50Hz + ':' + freq_rate2 + ':' + freq_4000Hz);
+    return freg_50Hz < 50 && freq_rate2 > 200 && freq_4000Hz < 50;
+  })
+  osc_ac2.stop();
+
+  // Two AudioContexts with different sample rate cannot communicate.
+  mustThrowWith("Connect nodes with different sample rate", "NotSupportedError",
+                () => ac2.createMediaStreamSource(dest_ac1.stream));
+
+  // Two AudioContext with the same sample rate can communicate.
+  const ac3 = new AudioContext({sampleRate: 48000});
+  const dest_ac3 = ac3.createMediaStreamDestination();
+  ac2.createMediaStreamSource(dest_ac3.stream);
+  ok(true, "Connect nodes with the same sample rate is ok");
+
+  mustThrowWith("Invalid zero samplerate", "NotSupportedError",
+    () => new AudioContext({sampleRate: 0}));
+
+  mustThrowWith("Invalid negative samplerate", "NotSupportedError",
+    () => new AudioContext({sampleRate: -1}));
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioContextSuspendResumeClose.html b/dom/media/webaudio/test/test_audioContextSuspendResumeClose.html
new file mode 100644
index 0000000000..36cf8f720c
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioContextSuspendResumeClose.html
@@ -0,0 +1,419 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test suspend, resume and close method of the AudioContext</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function tryToCreateNodeOnClosedContext(ctx) {
+  is(ctx.state, "closed", "The context is in closed state");
+
+  [ { name: "createBufferSource" },
+    { name: "createMediaStreamDestination",
+      onOfflineAudioContext: false},
+    { name: "createScriptProcessor" },
+    { name: "createStereoPanner" },
+    { name: "createAnalyser" },
+    { name: "createGain" },
+    { name: "createDelay" },
+    { name: "createBiquadFilter" },
+    { name: "createWaveShaper" },
+    { name: "createPanner" },
+    { name: "createConvolver" },
+    { name: "createChannelSplitter" },
+    { name: "createChannelMerger" },
+    { name: "createDynamicsCompressor" },
+    { name: "createOscillator" },
+    { name: "createMediaElementSource",
+      args: [new Audio()],
+      onOfflineAudioContext: false },
+    { name: "createMediaStreamSource",
+      args: [(new AudioContext()).createMediaStreamDestination().stream],
+      onOfflineAudioContext: false } ].forEach(function(e) {
+
+      if (e.onOfflineAudioContext == false &&
+          ctx instanceof OfflineAudioContext) {
+        return;
+      }
+
+      expectNoException(function() {
+        ctx[e.name].apply(ctx, e.args);
+      }, DOMException.INVALID_STATE_ERR);
+    });
+}
+
+function loadFile(url, callback) {
+  var xhr = new XMLHttpRequest();
+  xhr.open("GET", url, true);
+  xhr.responseType = "arraybuffer";
+  xhr.onload = function() {
+    callback(xhr.response);
+  };
+  xhr.send();
+}
+
+// createBuffer, createPeriodicWave and decodeAudioData should work on a context
+// that has `state` == "closed"
+function tryLegalOpeerationsOnClosedContext(ctx) {
+  is(ctx.state, "closed", "The context is in closed state");
+
+  [ { name: "createBuffer",
+      args: [1, 44100, 44100] },
+    { name: "createPeriodicWave",
+      args: [new Float32Array(10), new Float32Array(10)] }
+  ].forEach(function(e) {
+    expectNoException(function() {
+      ctx[e.name].apply(ctx, e.args);
+    });
+  });
+  loadFile("ting-44.1k-1ch.ogg", function(buf) {
+    ctx.decodeAudioData(buf).then(function(decodedBuf) {
+      ok(true, "decodeAudioData on a closed context should work, it did.")
+      finish();
+    }).catch(function(e){
+      ok(false, "decodeAudioData on a closed context should work, it did not");
+      finish();
+    });
+  });
+}
+
+// Test that MediaStreams that are the output of a suspended AudioContext are
+// producing silence
+// ac1 produce a sine fed to a MediaStreamAudioDestinationNode
+// ac2 is connected to ac1 with a MediaStreamAudioSourceNode, and check that
+// there is silence when ac1 is suspended
+function testMultiContextOutput() {
+  var ac1 = new AudioContext(),
+      ac2 = new AudioContext();
+
+  ac1.onstatechange = function() {
+    ac1.onstatechange = null;
+
+    var osc1 = ac1.createOscillator(),
+        mediaStreamDestination1 = ac1.createMediaStreamDestination();
+
+    var mediaStreamAudioSourceNode2 =
+      ac2.createMediaStreamSource(mediaStreamDestination1.stream),
+      sp2 = ac2.createScriptProcessor(),
+      silentBuffersInARow = 0;
+
+
+    sp2.onaudioprocess = function(e) {
+      ac1.suspend().then(function() {
+        is(ac1.state, "suspended", "ac1 is suspended");
+        sp2.onaudioprocess = checkSilence;
+      });
+      sp2.onaudioprocess = null;
+    }
+
+    function checkSilence(e) {
+      var input = e.inputBuffer.getChannelData(0);
+      var silent = true;
+      for (var i = 0; i < input.length; i++) {
+        if (input[i] != 0.0) {
+          silent = false;
+        }
+      }
+
+      if (silent) {
+        silentBuffersInARow++;
+        if (silentBuffersInARow == 10) {
+          ok(true,
+              "MediaStreams produce silence when their input is blocked.");
+          sp2.onaudioprocess = null;
+          ac1.close();
+          ac2.close();
+          finish();
+        }
+      } else {
+        is(silentBuffersInARow, 0,
+            "No non silent buffer inbetween silent buffers.");
+      }
+    }
+
+    osc1.connect(mediaStreamDestination1);
+
+    mediaStreamAudioSourceNode2.connect(sp2);
+    osc1.start();
+  }
+}
+
+
+// Test that there is no buffering between contexts when connecting a running
+// AudioContext to a suspended AudioContext. Our ScriptProcessorNode does some
+// buffering internally, so we ensure this by using a very very low frequency
+// on a sine, and oberve that the phase has changed by a big enough margin.
+function testMultiContextInput() {
+  var ac1 = new AudioContext(),
+      ac2 = new AudioContext();
+
+  ac1.onstatechange = function() {
+    ac1.onstatechange = null;
+
+    var osc1 = ac1.createOscillator(),
+        mediaStreamDestination1 = ac1.createMediaStreamDestination(),
+        sp1 = ac1.createScriptProcessor();
+
+    var mediaStreamAudioSourceNode2 =
+      ac2.createMediaStreamSource(mediaStreamDestination1.stream),
+      sp2 = ac2.createScriptProcessor(),
+      eventReceived = 0;
+
+
+    osc1.frequency.value = 0.0001;
+
+    function checkDiscontinuity(e) {
+      var inputBuffer = e.inputBuffer.getChannelData(0);
+      if (eventReceived++ == 3) {
+        var delta = Math.abs(inputBuffer[1] - sp2.value),
+            theoreticalIncrement = 2048 * 3 * Math.PI * 2 * osc1.frequency.value / ac1.sampleRate;
+        ok(delta >= theoreticalIncrement,
+            "Buffering did not occur when the context was suspended (delta:" + delta + " increment: " + theoreticalIncrement+")");
+        ac1.close();
+        ac2.close();
+        sp1.onaudioprocess = null;
+        sp2.onaudioprocess = null;
+        finish();
+      }
+    }
+
+    sp2.onaudioprocess = function(e) {
+      var inputBuffer = e.inputBuffer.getChannelData(0);
+      sp2.value = inputBuffer[inputBuffer.length - 1];
+      ac2.suspend().then(function() {
+          ac2.resume().then(function() {
+            sp2.onaudioprocess = checkDiscontinuity;
+            });
+          });
+    }
+
+    osc1.connect(mediaStreamDestination1);
+    osc1.connect(sp1);
+
+    mediaStreamAudioSourceNode2.connect(sp2);
+    osc1.start();
+  }
+}
+
+// Test that ScriptProcessorNode's onaudioprocess don't get called while the
+// context is suspended/closed. It is possible that we get the handler called
+// exactly once after suspend, because the event has already been sent to the
+// event loop.
+function testScriptProcessNodeSuspended() {
+  var ac = new AudioContext();
+  var sp = ac.createScriptProcessor();
+  var remainingIterations = 30;
+  var afterResume = false;
+  ac.onstatechange = function() {
+    ac.onstatechange = null;
+    sp.onaudioprocess = function() {
+      ok(ac.state == "running", "If onaudioprocess is called, the context" +
+          " must be running (was " + ac.state + ", remainingIterations:" + remainingIterations +")");
+      remainingIterations--;
+      if (!afterResume) {
+        if (remainingIterations == 0) {
+          ac.suspend().then(function() {
+            ac.resume().then(function() {
+              remainingIterations = 30;
+              afterResume = true;
+            });
+          });
+        }
+      } else {
+        sp.onaudioprocess = null;
+        finish();
+      }
+    }
+  }
+  sp.connect(ac.destination);
+}
+
+// Take an AudioContext, make sure it switches to running when the audio starts
+// flowing, and then, call suspend, resume and close on it, tracking its state.
+function testAudioContext() {
+  var ac = new AudioContext();
+  is(ac.state, "suspended", "AudioContext should start in suspended state.");
+  var stateTracker = {
+    previous: ac.state,
+     // no promise for the initial suspended -> running
+    initial: {  handler: false },
+    suspend: { promise: false, handler: false },
+    resume: { promise: false, handler: false },
+    close: { promise: false, handler: false }
+  };
+
+  function initialSuspendToRunning() {
+    ok(stateTracker.previous == "suspended" &&
+       ac.state == "running",
+       "AudioContext should switch to \"running\" when the audio hardware is" +
+       " ready.");
+
+    stateTracker.previous = ac.state;
+    ac.onstatechange = afterSuspend;
+    stateTracker.initial.handler = true;
+
+    ac.suspend().then(function() {
+      ok(!stateTracker.suspend.promise && !stateTracker.suspend.handler,
+        "Promise should be resolved before the callback, and only once.")
+      stateTracker.suspend.promise = true;
+    });
+  }
+
+  function afterSuspend() {
+    ok(stateTracker.previous == "running" &&
+       ac.state == "suspended",
+       "AudioContext should switch to \"suspend\" when the audio stream is" +
+       "suspended.");
+    ok(stateTracker.suspend.promise && !stateTracker.suspend.handler,
+        "Handler should be called after the callback, and only once");
+
+    stateTracker.suspend.handler = true;
+    stateTracker.previous = ac.state;
+    ac.onstatechange = afterResume;
+
+    ac.resume().then(function() {
+      ok(!stateTracker.resume.promise && !stateTracker.resume.handler,
+        "Promise should be called before the callback, and only once");
+      stateTracker.resume.promise = true;
+    });
+  }
+
+  function afterResume() {
+    ok(stateTracker.previous == "suspended" &&
+       ac.state == "running",
+   "AudioContext should switch to \"running\" when the audio stream resumes.");
+
+    ok(stateTracker.resume.promise && !stateTracker.resume.handler,
+       "Handler should be called after the callback, and only once");
+
+    stateTracker.resume.handler = true;
+    stateTracker.previous = ac.state;
+    ac.onstatechange = afterClose;
+
+    ac.close().then(function() {
+      ok(!stateTracker.close.promise && !stateTracker.close.handler,
+        "Promise should be called before the callback, and only once");
+      stateTracker.close.promise = true;
+      tryToCreateNodeOnClosedContext(ac);
+      tryLegalOpeerationsOnClosedContext(ac);
+    });
+  }
+
+  function afterClose() {
+    ok(stateTracker.previous == "running" &&
+       ac.state == "closed",
+       "AudioContext should switch to \"closed\" when the audio stream is" +
+       " closed.");
+    ok(stateTracker.close.promise && !stateTracker.close.handler,
+       "Handler should be called after the callback, and only once");
+  }
+
+  ac.onstatechange = initialSuspendToRunning;
+}
+
+function testOfflineAudioContext() {
+  var o = new OfflineAudioContext(1, 44100, 44100);
+  is(o.state, "suspended", "OfflineAudioContext should start in suspended state.");
+
+  expectRejectedPromise(o, "resume", "NotSupportedError");
+
+  var previousState = o.state,
+      finishedRendering = false;
+  function beforeStartRendering() {
+    ok(previousState == "suspended" && o.state == "running", "onstatechanged" +
+        "handler is called on state changed, and the new state is running");
+    previousState = o.state;
+    o.onstatechange = onRenderingFinished;
+  }
+
+  function onRenderingFinished() {
+    ok(previousState == "running" && o.state == "closed",
+        "onstatechanged handler is called when rendering finishes, " +
+        "and the new state is closed");
+    ok(finishedRendering, "The Promise that is resolved when the rendering is" +
+                    "done should be resolved earlier than the state change.");
+    previousState = o.state;
+    o.onstatechange = afterRenderingFinished;
+
+    tryToCreateNodeOnClosedContext(o);
+    tryLegalOpeerationsOnClosedContext(o);
+  }
+
+  function afterRenderingFinished() {
+    ok(false, "There should be no transition out of the closed state.");
+  }
+
+  o.onstatechange = beforeStartRendering;
+
+  o.startRendering().then(function(buffer) {
+    finishedRendering = true;
+  });
+}
+
+function testSuspendResumeEventLoop() {
+  var ac = new AudioContext();
+  var source = ac.createBufferSource();
+  source.buffer = ac.createBuffer(1, 44100, 44100);
+  source.onended = function() {
+    ok(true, "The AudioContext did resume.");
+    finish();
+  }
+  ac.onstatechange = function() {
+    ac.onstatechange = null;
+
+    ok(ac.state == "running", "initial state is running");
+    ac.suspend();
+    source.start();
+    ac.resume();
+  }
+}
+
+function testResumeInStateChangeForResumeCallback() {
+  // Regression test for bug 1468085.
+  var ac = new AudioContext;
+  ac.onstatechange = function() {
+    ac.resume().then(() => {
+        ok(true, "resume promise resolved as expected.");
+        finish();
+    });
+  }
+}
+
+var remaining = 0;
+function finish() {
+  remaining--;
+  if (remaining == 0) {
+    SimpleTest.finish();
+  }
+}
+
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var tests = [
+    testOfflineAudioContext,
+    testScriptProcessNodeSuspended,
+    testMultiContextOutput,
+    testMultiContextInput,
+    testSuspendResumeEventLoop,
+    testResumeInStateChangeForResumeCallback
+  ];
+
+  // See Bug 1305136, many intermittent failures on Linux
+  if (!navigator.platform.startsWith("Linux")) {
+    tests.push(testAudioContext);
+  }
+
+  remaining = tests.length;
+  tests.forEach(function(f) { f() });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioDestinationNode.html b/dom/media/webaudio/test/test_audioDestinationNode.html
new file mode 100644
index 0000000000..e7a8b091ba
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioDestinationNode.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioDestinationNode as EventTarget</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+var ac = new AudioContext()
+ac.destination.addEventListener("foo", function() {
+  ok(true, "Event received!");
+  SimpleTest.finish();
+});
+ac.destination.dispatchEvent(new CustomEvent("foo"));
+
+</script>
+</pre>
+</body>
+</html>
+
diff --git a/dom/media/webaudio/test/test_audioParamChaining.html b/dom/media/webaudio/test/test_audioParamChaining.html
new file mode 100644
index 0000000000..85b8099e2e
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamChaining.html
@@ -0,0 +1,77 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish()
+
+function frameToTime(frame, rate)
+{
+  return frame / rate;
+}
+
+const RATE = 44100;
+
+var oc = new OfflineAudioContext(1, 44100, RATE);
+// This allows us to have a source that is simply a DC offset.
+var source = oc.createBufferSource();
+var buf = oc.createBuffer(1, 1, RATE);
+buf.getChannelData(0)[0] = 1;
+source.loop = true;
+source.buffer = buf;
+
+source.start(0);
+
+var gain = oc.createGain();
+
+source.connect(gain).connect(oc.destination);
+
+var gain2 = oc.createGain();
+var rv2 = gain2.gain.linearRampToValueAtTime(0.1, 0.5);
+ok(rv2 instanceof AudioParam, "linearRampToValueAtTime returns an AudioParam.");
+ok(rv2 == gain2.gain, "linearRampToValueAtTime returns the right AudioParam.");
+
+rv2 = gain2.gain.exponentialRampToValueAtTime(0.01, 1.0);
+ok(rv2 instanceof AudioParam,
+    "exponentialRampToValueAtTime returns an AudioParam.");
+ok(rv2 == gain2.gain,
+    "exponentialRampToValueAtTime returns the right AudioParam.");
+
+rv2 = gain2.gain.setTargetAtTime(1.0, 2.0, 0.1);
+ok(rv2 instanceof AudioParam, "setTargetAtTime returns an AudioParam.");
+ok(rv2 == gain2.gain, "setTargetAtTime returns the right AudioParam.");
+
+var array = new Float32Array(10);
+rv2 = gain2.gain.setValueCurveAtTime(array, 10, 11);
+ok(rv2 instanceof AudioParam, "setValueCurveAtTime returns an AudioParam.");
+ok(rv2 == gain2.gain, "setValueCurveAtTime returns the right AudioParam.");
+
+// We chain three automation methods, making a gain step.
+var rv = gain.gain.setValueAtTime(0, frameToTime(0, RATE))
+                  .setValueAtTime(0.5, frameToTime(22000, RATE))
+                  .setValueAtTime(1, frameToTime(44000, RATE));
+
+ok(rv instanceof AudioParam, "setValueAtTime returns an AudioParam.");
+ok(rv == gain.gain, "setValueAtTime returns the right AudioParam.");
+
+oc.startRendering().then(function(rendered) {
+    console.log(rendered.getChannelData(0));
+    is(rendered.getChannelData(0)[0], 0,
+      "The value of the first step is correct.");
+    is(rendered.getChannelData(0)[22050], 0.5,
+      "The value of the second step is correct");
+    is(rendered.getChannelData(0)[44099], 1,
+      "The value of the third step is correct.");
+    SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamExponentialRamp.html b/dom/media/webaudio/test/test_audioParamExponentialRamp.html
new file mode 100644
index 0000000000..2416b5de14
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamExponentialRamp.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.exponentialRampToValue</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var V0 = 0.1;
+var V1 = 0.9;
+var T0 = 0;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.setValueAtTime(V0, 0);
+    gain.gain.exponentialRampToValueAtTime(V1, 2048/context.sampleRate);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers(context) {
+    var T1 = 2048 / context.sampleRate;
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      var t = i / context.sampleRate;
+      expectedBuffer.getChannelData(0)[i] = V0 * Math.pow(V1 / V0, (t - T0) / (T1 - T0));
+    }
+    return expectedBuffer;
+  },
+};
+
+
+SimpleTest.waitForExplicitFinish();
+// Comparing different AudioContexts may result in different timing reated information being reported
+// when we jitter time, as they are on different Relative Timelines.
+SpecialPowers.pushPrefEnv({"set": [["privacy.resistFingerprinting.reduceTimerPrecision.jitter", false]]}, runTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamGain.html b/dom/media/webaudio/test/test_audioParamGain.html
new file mode 100644
index 0000000000..3977b94703
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamGain.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam with pre-gain </title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+var ctx = new AudioContext();
+var source = ctx.createOscillator();
+var lfo = ctx.createOscillator();
+var lfoIntensity = ctx.createGain();
+var effect = ctx.createGain();
+var sp = ctx.createScriptProcessor(2048, 1);
+
+source.frequency.value = 440;
+lfo.frequency.value = 2;
+// Very low gain, so the LFO should have very little influence
+// on the source, its RMS value should be close to the nominal value
+// for a sine wave.
+lfoIntensity.gain.value = 0.0001;
+
+lfo.connect(lfoIntensity);
+lfoIntensity.connect(effect.gain);
+source.connect(effect);
+effect.connect(sp);
+
+sp.onaudioprocess = function(e) {
+  var buffer = e.inputBuffer.getChannelData(0);
+  var rms = 0;
+  for (var i = 0; i < buffer.length; i++) {
+    rms += buffer[i] * buffer[i];
+  }
+
+  rms /= buffer.length;
+  rms = Math.sqrt(rms);
+
+  // 1 / Math.sqrt(2) is the theoretical RMS value for a sine wave.
+  ok(fuzzyCompare(rms, 1 / Math.sqrt(2)),
+      "Gain correctly applied to the AudioParam.");
+
+  ctx = null;
+  sp.onaudioprocess = null;
+  lfo.stop(0);
+  source.stop(0);
+
+  SimpleTest.finish();
+}
+
+lfo.start(0);
+source.start(0);
+
+</script>
+</pre>
+</body>
diff --git a/dom/media/webaudio/test/test_audioParamLinearRamp.html b/dom/media/webaudio/test/test_audioParamLinearRamp.html
new file mode 100644
index 0000000000..5ec26467e8
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamLinearRamp.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.linearRampToValue</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var V0 = 0.1;
+var V1 = 0.9;
+var T0 = 0;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.setValueAtTime(V0, 0);
+    gain.gain.linearRampToValueAtTime(V1, 2048/context.sampleRate);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers(context) {
+    var T1 = 2048 / context.sampleRate;
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      var t = i / context.sampleRate;
+      expectedBuffer.getChannelData(0)[i] = V0 + (V1 - V0) * ((t - T0) / (T1 - T0));
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamSetCurveAtTime.html b/dom/media/webaudio/test/test_audioParamSetCurveAtTime.html
new file mode 100644
index 0000000000..e21b58bb19
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamSetCurveAtTime.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.linearRampToValue</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var T0 = 0;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createConstantSource();
+
+    var gain = context.createGain();
+    gain.gain.setValueCurveAtTime(this.curve, T0, this.duration);
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers(context) {
+    this.duration = 1024 / context.sampleRate;
+    this.curve = new Float32Array([1.0, 0.5, 0.75, 0.25]);
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    var data = expectedBuffer.getChannelData(0);
+    var step = 1024 / 3;
+    for (var i = 0; i < 2048; ++i) {
+      if (i < step) {
+        data[i] = 1.0 - 0.5*i/step;
+      } else if (i < 2*step) {
+        data[i] = 0.5 + 0.25*(i - step)/step;
+      } else if (i < 3*step) {
+        data[i] = 0.75 - 0.5*(i - 2*step)/step;
+      } else {
+        data[i] = 0.25;
+      }
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamSetTargetAtTime.html b/dom/media/webaudio/test/test_audioParamSetTargetAtTime.html
new file mode 100644
index 0000000000..442e209be6
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamSetTargetAtTime.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.setTargetAtTime</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var V0 = 0.9;
+var V1 = 0.1;
+var T0 = 0;
+var TimeConstant = 10;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.value = V0;
+    gain.gain.setTargetAtTime(V1, T0, TimeConstant);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      var t = i / context.sampleRate;
+      expectedBuffer.getChannelData(0)[i] = V1 + (V0 - V1) * Math.exp(-(t - T0) / TimeConstant);
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamSetTargetAtTimeZeroTimeConstant.html b/dom/media/webaudio/test/test_audioParamSetTargetAtTimeZeroTimeConstant.html
new file mode 100644
index 0000000000..2d18447a0d
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamSetTargetAtTimeZeroTimeConstant.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.setTargetAtTime with zero time constant</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var V0 = 0.9;
+var V1 = 0.1;
+var T0 = 0;
+var TimeConstant = 0;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.value = V0;
+    gain.gain.setTargetAtTime(V1, T0, TimeConstant);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    expectedBuffer.getChannelData(0)[0] = V0;
+    for (var i = 1; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = V1;
+    }
+    return expectedBuffer;
+  },
+};
+
+SimpleTest.waitForExplicitFinish();
+// Comparing different AudioContexts may result in different timing reated information being reported
+// when we jitter time, as they are on different Relative Timelines.
+SpecialPowers.pushPrefEnv({"set": [["privacy.resistFingerprinting.reduceTimerPrecision.jitter", false]]}, runTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamSetValueAtTime.html b/dom/media/webaudio/test/test_audioParamSetValueAtTime.html
new file mode 100644
index 0000000000..18c02837e6
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamSetValueAtTime.html
@@ -0,0 +1,52 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.linearRampToValue</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var V0 = 0.1;
+var V1 = 0.9;
+var T0 = 0;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.value = 0;
+    gain.gain.setValueAtTime(V0, 1024/context.sampleRate);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 1024; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = 0.1;
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_audioParamTimelineDestinationOffset.html b/dom/media/webaudio/test/test_audioParamTimelineDestinationOffset.html
new file mode 100644
index 0000000000..76db12f88a
--- /dev/null
+++ b/dom/media/webaudio/test/test_audioParamTimelineDestinationOffset.html
@@ -0,0 +1,45 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam timeline events scheduled after the destination stream has started playback</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.requestFlakyTimeout("This test needs to wait until the AudioDestinationNode's stream's timer starts.");
+
+var gTest = {
+  length: 16384,
+  numberOfChannels: 1,
+  createGraphAsync(context, callback) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    setTimeout(function() {
+      var source = context.createBufferSource();
+      source.buffer = sourceBuffer;
+      source.start(context.currentTime);
+      source.stop(context.currentTime + sourceBuffer.duration);
+
+      var gain = context.createGain();
+      gain.gain.setValueAtTime(0, context.currentTime);
+      gain.gain.setTargetAtTime(0, context.currentTime + sourceBuffer.duration, 1);
+      source.connect(gain);
+
+      callback(gain);
+    }, 100);
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_badConnect.html b/dom/media/webaudio/test/test_badConnect.html
new file mode 100644
index 0000000000..51e83f6088
--- /dev/null
+++ b/dom/media/webaudio/test/test_badConnect.html
@@ -0,0 +1,52 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context1 = new OfflineAudioContext(1, 128, 44100);
+  var context2 = new OfflineAudioContext(1, 128, 44100);
+
+  var destination1 = context1.destination;
+  var destination2 = context2.destination;
+  var gain1 = new GainNode(context2);
+
+  isnot(destination1, destination2, "Destination nodes should not be the same");
+  isnot(destination1.context, destination2.context, "Destination nodes should not have the same context");
+
+  var source1 = context1.createBufferSource();
+
+  expectException(function() {
+    source1.connect(destination1, 1);
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    source1.connect(destination1, 0, 1);
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    source1.connect(destination2);
+  }, DOMException.INVALID_ACCESS_ERR);
+  expectException(function() {
+    source1.connect(gain1.gain);
+  }, DOMException.INVALID_ACCESS_ERR);
+
+  source1.connect(destination1);
+
+  expectException(function() {
+    source1.disconnect(1);
+  }, DOMException.INDEX_SIZE_ERR);
+
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_biquadFilterNode.html b/dom/media/webaudio/test/test_biquadFilterNode.html
new file mode 100644
index 0000000000..561198c491
--- /dev/null
+++ b/dom/media/webaudio/test/test_biquadFilterNode.html
@@ -0,0 +1,86 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+function near(a, b, msg) {
+  ok(Math.abs(a - b) < 1e-3, msg);
+}
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var destination = context.destination;
+
+  var source = context.createBufferSource();
+
+  var filter = context.createBiquadFilter();
+
+  source.buffer = buffer;
+
+  source.connect(filter);
+  filter.connect(destination);
+
+  // Verify default values
+  is(filter.type, "lowpass", "Correct default value for type");
+  near(filter.frequency.defaultValue, 350, "Correct default value for filter frequency");
+  near(filter.detune.defaultValue, 0, "Correct default value for filter detune");
+  near(filter.Q.defaultValue, 1, "Correct default value for filter Q");
+  near(filter.gain.defaultValue, 0, "Correct default value for filter gain");
+  is(filter.channelCount, 2, "Biquad filter node has 2 input channels by default");
+  is(filter.channelCountMode, "max", "Correct channelCountMode for the biquad filter node");
+  is(filter.channelInterpretation, "speakers", "Correct channelCountInterpretation for the biquad filter node");
+
+  // Make sure that we can set all of the valid type values
+  var types = [
+    "lowpass",
+    "highpass",
+    "bandpass",
+    "lowshelf",
+    "highshelf",
+    "peaking",
+    "notch",
+    "allpass",
+  ];
+  for (var i = 0; i < types.length; ++i) {
+    filter.type = types[i];
+  }
+
+  // Make sure getFrequencyResponse handles invalid frequencies properly
+  var frequencies = new Float32Array([-1.0, context.sampleRate*0.5 - 1.0, context.sampleRate]);
+  var magResults = new Float32Array(3);
+  var phaseResults = new Float32Array(3);
+  filter.getFrequencyResponse(frequencies, magResults, phaseResults);
+  ok(isNaN(magResults[0]), "Invalid input frequency should give NaN magnitude response");
+  ok(!isNaN(magResults[1]), "Valid input frequency should not give NaN magnitude response");
+  ok(isNaN(magResults[2]), "Invalid input frequency should give NaN magnitude response");
+  ok(isNaN(phaseResults[0]), "Invalid input frquency should give NaN phase response");
+  ok(!isNaN(phaseResults[1]), "Valid input frquency should not give NaN phase response");
+  ok(isNaN(phaseResults[2]), "Invalid input frquency should give NaN phase response");
+
+  source.start(0);
+  SimpleTest.executeSoon(function() {
+    source.stop(0);
+    source.disconnect();
+    filter.disconnect();
+
+    SimpleTest.finish();
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_biquadFilterNodePassThrough.html b/dom/media/webaudio/test/test_biquadFilterNodePassThrough.html
new file mode 100644
index 0000000000..4db01b0b14
--- /dev/null
+++ b/dom/media/webaudio/test/test_biquadFilterNodePassThrough.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+
+    var filter = context.createBiquadFilter();
+
+    source.buffer = this.buffer;
+
+    source.connect(filter);
+
+    var filterWrapped = SpecialPowers.wrap(filter);
+    ok("passThrough" in filterWrapped, "BiquadFilterNode should support the passThrough API");
+    filterWrapped.passThrough = true;
+
+    source.start(0);
+    return filter;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_biquadFilterNodeWithGain.html b/dom/media/webaudio/test/test_biquadFilterNodeWithGain.html
new file mode 100644
index 0000000000..d97a753de9
--- /dev/null
+++ b/dom/media/webaudio/test/test_biquadFilterNodeWithGain.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test BiquadFilterNode after a GainNode and tail - Bugs 924286 and 924288</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+const signalLength = 2048;
+
+var gTest = {
+  length: signalLength,
+  numberOfChannels: 1,
+  createGraph(context) {
+    // Two oscillators scheduled sequentially
+    var signalDuration = signalLength / context.sampleRate;
+    var osc1 = context.createOscillator();
+    osc1.type = "square";
+    osc1.start(0);
+    osc1.stop(signalDuration / 2);
+    var osc2 = context.createOscillator();
+    osc2.start(signalDuration / 2);
+    osc2.stop(signalDuration);
+
+    // Comparing a biquad on each source with one on both sources checks that
+    // the biquad on the first source doesn't shut down early.
+    var biquad1 = context.createBiquadFilter();
+    osc1.connect(biquad1);
+    var biquad2 = context.createBiquadFilter();
+    osc2.connect(biquad2);
+
+    var gain = context.createGain();
+    gain.gain.value = -1;
+    osc1.connect(gain);
+    osc2.connect(gain);
+
+    var biquadWithGain = context.createBiquadFilter();
+    gain.connect(biquadWithGain);
+
+    // The output of biquadWithGain should be the inverse of the sum of the
+    // outputs of biquad1 and biquad2, so blend them together and expect
+    // silence.
+    var blend = context.createGain();
+    biquad1.connect(blend);
+    biquad2.connect(blend);
+    biquadWithGain.connect(blend);
+
+    return blend;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug1027864.html b/dom/media/webaudio/test/test_bug1027864.html
new file mode 100644
index 0000000000..847485ff88
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1027864.html
@@ -0,0 +1,74 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test bug 1027864</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+function observer(subject, topic, data) {
+  var id = parseInt(data);
+  var index = ids.indexOf(id);
+  if (index != -1) {
+    ok(true, "Dropping id " + id + " at index " + index);
+    ids.splice(index, 1);
+    if (!ids.length) {
+      SimpleTest.executeSoon(function() {
+        SimpleTest.finish();
+      });
+    }
+  }
+}
+
+function id(node) {
+  return SpecialPowers.getPrivilegedProps(node, "id");
+}
+
+SpecialPowers.addAsyncObserver(observer, "webaudio-node-demise", false);
+
+SimpleTest.registerCleanupFunction(function() {
+  SpecialPowers.removeAsyncObserver(observer, "webaudio-node-demise");
+});
+
+var ac = new AudioContext();
+var ids;
+
+(function() {
+  var delay = ac.createDelay();
+  delay.delayTime.value = 0.03;
+
+  var gain = ac.createGain();
+  gain.gain.value = 0.6;
+
+  delay.connect(gain);
+  gain.connect(delay);
+
+  gain.connect(ac.destination);
+
+  var source = ac.createOscillator();
+
+  source.connect(gain);
+  source.start(ac.currentTime);
+  source.stop(ac.currentTime + 0.1);
+
+  ids = [ id(delay), id(gain), id(source) ];
+})();
+
+setInterval(function() {
+  forceCC();
+}, 200);
+
+function forceCC() {
+  SpecialPowers.DOMWindowUtils.cycleCollect();
+  SpecialPowers.DOMWindowUtils.garbageCollect();
+}
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug1056032.html b/dom/media/webaudio/test/test_bug1056032.html
new file mode 100644
index 0000000000..ba38267e19
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1056032.html
@@ -0,0 +1,35 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset=utf-8>
+<head>
+  <title>Test that we can decode an mp3 (bug 1056032)</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+var filename = "small-shot.mp3";
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  var xhr = new XMLHttpRequest();
+  xhr.open("GET", filename);
+  xhr.responseType = "arraybuffer";
+  xhr.onload = function() {
+    var context = new AudioContext();
+    context.decodeAudioData(xhr.response, function(b) {
+      ok(true, "We can decode an mp3 using decodeAudioData");
+      SimpleTest.finish();
+    }, function() {
+      ok(false, "We should be able to decode an mp3 using decodeAudioData but couldn't");
+      SimpleTest.finish();
+    });
+  };
+  xhr.send(null);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug1113634.html b/dom/media/webaudio/test/test_bug1113634.html
new file mode 100644
index 0000000000..acdcba7c25
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1113634.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioParam.setTargetAtTime where the target time is the same as the time of a previous event</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var V0 = 0.9;
+var V1 = 0.1;
+var T0 = 0;
+var TimeConstant = 0.1;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+    gain.gain.setValueAtTime(V0, T0);
+    gain.gain.setTargetAtTime(V1, T0, TimeConstant);
+
+    source.connect(gain);
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      var t = i / context.sampleRate;
+      expectedBuffer.getChannelData(0)[i] = V1 + (V0 - V1) * Math.exp(-(t - T0) / TimeConstant);
+    }
+    return expectedBuffer;
+  },
+};
+
+SimpleTest.waitForExplicitFinish();
+// Comparing different AudioContexts may result in different timing reated information being reported
+// when we jitter time, as they are on different Relative Timelines.
+SpecialPowers.pushPrefEnv({"set": [["privacy.resistFingerprinting.reduceTimerPrecision.jitter", false]]}, runTest);
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug1118372.html b/dom/media/webaudio/test/test_bug1118372.html
new file mode 100644
index 0000000000..f049b221e8
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1118372.html
@@ -0,0 +1,46 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test WaveShaperNode with no curve</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  var context = new OfflineAudioContext(1, 2048, 44100);
+
+  var osc=context.createOscillator();
+  var gain=context.createGain();
+  var shaper=context.createWaveShaper();
+  gain.gain.value=0.1;
+  shaper.curve=new Float32Array([-0.5,-0.5,1,1]);
+
+  osc.connect(gain);
+  gain.connect(shaper);
+  shaper.connect(context.destination);
+  osc.start(0);
+
+  context.startRendering().then(function(buffer) {
+    var samples = buffer.getChannelData(0);
+    // the signal should be scaled
+    var failures = 0;
+    for (var i = 0; i < 2048; ++i) {
+      if (samples[i] > 0.5) {
+        failures = failures + 1;
+      }
+    }
+    ok(failures == 0, "signal should have been rescaled by gain: found " + failures + " points too loud.");
+    SimpleTest.finish();
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug1255618.html b/dom/media/webaudio/test/test_bug1255618.html
new file mode 100644
index 0000000000..15e7351995
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1255618.html
@@ -0,0 +1,41 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test sync XHR does not crash unlinked AudioContext</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script>
+SimpleTest.waitForExplicitFinish();
+
+const filename = "test_bug1255618.html";
+
+function collect_and_fetch() {
+    SpecialPowers.forceGC();
+    SpecialPowers.forceCC();
+
+    var xhr = new XMLHttpRequest();
+    xhr.open("GET", filename, false);
+    var ended = false;
+    xhr.onloadend = function() { ended = true; }
+    // Sync XHR will suspend timeouts, which involves any AudioContexts still
+    // registered with the window.
+    // See https://bugzilla.mozilla.org/show_bug.cgi?id=1255618#c0
+    xhr.send(null);
+
+    ok(ended, "No crash during fetch");
+    SimpleTest.finish();
+}
+
+var ac = new AudioContext();
+
+ac.onstatechange = function () {
+    ac.onstatechange = null;
+    is(ac.state, "running", "statechange to running");
+    ac = null;
+    SimpleTest.executeSoon(collect_and_fetch);
+}
+
+</script>
+</body>
diff --git a/dom/media/webaudio/test/test_bug1267579.html b/dom/media/webaudio/test/test_bug1267579.html
new file mode 100644
index 0000000000..7003b345f5
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1267579.html
@@ -0,0 +1,46 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test that PeriodicWave handles fundamental fequency of zero</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// This is the smallest value that the test framework will accept
+const testLength = 256;
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  runTest();
+});
+
+var gTest = {
+  numberOfChannels: 1,
+  createGraph(context) {
+    var osc = context.createOscillator();
+    osc.setPeriodicWave(context.
+                        createPeriodicWave(new Float32Array([0.0, 1.0]),
+                                           new Float32Array(2)));
+    osc.frequency.value = 0.0;
+    osc.start();
+    return osc;
+  },
+  createExpectedBuffers(context) {
+    var buffer = context.createBuffer(1, testLength, context.sampleRate);
+
+    for (var i = 0; i < buffer.length; ++i) {
+      buffer.getChannelData(0)[i] = 1.0;
+    }
+    return buffer;
+  },
+};
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug1355798.html b/dom/media/webaudio/test/test_bug1355798.html
new file mode 100644
index 0000000000..9b46322bbc
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1355798.html
@@ -0,0 +1,30 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode produces output even when the even when the distance is
+    from the listener is zero, and the cone gain is present, regression test for
+  bug 1355798.</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+var off = new OfflineAudioContext(1, 128, 44100);
+var panner = off.createPanner();
+var osc = off.createOscillator();
+panner.setPosition(1, 1, 1);
+off.listener.setPosition(1, 1, 1);
+osc.connect(panner).connect(off.destination);
+panner.coneOuterAngle = 359;
+osc.start();
+off.startRendering().then(function(b) {
+    is(b.getChannelData(0).filter(x => isNaN(x)).length, 0);
+    SimpleTest.finish();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug1447273.html b/dom/media/webaudio/test/test_bug1447273.html
new file mode 100644
index 0000000000..f4b473d8ab
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug1447273.html
@@ -0,0 +1,175 @@
+<!DOCTYPE HTML>

+<html>

+<head>

+  <title>Test bug 1447273 - GainNode with a stereo input and changing volume</title>

+  <script src="/tests/SimpleTest/SimpleTest.js"></script>

+  <script type="text/javascript" src="webaudio.js"></script>

+  <script type="text/javascript" src="head.js"></script>

+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />

+</head>

+<body>

+<pre id="test">

+<script class="testbody" type="text/javascript">

+/**

+ * Sets up a stereo BufferSource and plumbs it through different gain node

+ * configurations. A control gain path with no changes to gain is used along

+ * with 2 other paths which should increase their gain. The result should be

+ * that audio travelling along the increased gain paths is louder than the

+ * control path.

+ */

+

+SimpleTest.waitForExplicitFinish();

+SimpleTest.requestFlakyTimeout(

+  "This test uses a live audio context and uses a setTimeout to schedule a " +

+  "change to the graph.");

+addLoadEvent(function() {

+  let context = new AudioContext();

+

+  let numChannels = 2;

+  let sampleRate = context.sampleRate;

+  // 60 seconds to mitigate timing issues on slow test machines

+  let recordingLength = 60;

+  let bufferLength = sampleRate * recordingLength;

+  let gainExplicitlyIncreased = false;

+  let sourceFinished = false;

+

+  // Create source buffer

+  let sourceBuffer = context.createBuffer(numChannels, bufferLength, sampleRate);

+  for (let i = 0; i < bufferLength; ++i) {

+    sourceBuffer.getChannelData(0)[i] = 1;

+    sourceBuffer.getChannelData(1)[i] = 1;

+  }

+  let source = context.createBufferSource();

+  source.buffer = sourceBuffer;

+

+  let gainNoChange = context.createGain();

+  let gainExplicitAssignment = context.createGain();

+  let gainSetValueAtTime = context.createGain();

+

+  // All gain nodes start of with the same gain

+  gainNoChange.gain.value = 0.25;

+  gainExplicitAssignment.gain.value = 0.25;

+  gainSetValueAtTime.gain.value = 0.25;

+

+  // Connect source to gain nodes:

+  // source--> gainNoChange

+  //       |-> gainExplicitAssignment

+  //       \-> gainSetValueAtTime

+  source.connect(gainNoChange);

+  source.connect(gainExplicitAssignment);

+  source.connect(gainSetValueAtTime);

+

+  // Create intermediate media streams (required to repro bug 1447273)

+  let destNoChange = context.createMediaStreamDestination();

+  let destExplicitAssignement = context.createMediaStreamDestination();

+  let destSetValueAtTime = context.createMediaStreamDestination();

+

+  let sourceNoChange = context.createMediaStreamSource(destNoChange.stream);

+  let sourceExplicitAssignement = context.createMediaStreamSource(destExplicitAssignement.stream);

+  let sourceSetValueAtTime = context.createMediaStreamSource(destSetValueAtTime.stream);

+

+  // Connect gain nodes to our intermediate streams:

+  // source--> gainNoChange           -> destNoChange            -> sourceNoChange

+  //       |-> gainExplicitAssignment -> destExplicitAssignement -> sourceExplicitAssignement

+  //       \-> gainSetValueAtTime     -> destSetValueAtTime      -> sourceSetValueAtTime

+  gainNoChange.connect(destNoChange);

+  gainExplicitAssignment.connect(destExplicitAssignement);

+  gainSetValueAtTime.connect(destSetValueAtTime);

+

+  // Create analysers for each path

+  let analyserNoChange = context.createAnalyser();

+  let analyserExplicitAssignment = context.createAnalyser();

+  let analyserSetValueAtTime = context.createAnalyser();

+

+  // Connect our intermediate media streams to analysers:

+  // source--> gainNoChange           -> destNoChange            -> sourceNoChange            -> analyserNoChange

+  //       |-> gainExplicitAssignment -> destExplicitAssignement -> sourceExplicitAssignement -> analyserExplicitAssignment

+  //       \-> gainSetValueAtTime     -> destSetValueAtTime      -> sourceSetValueAtTime      -> analyserSetValueAtTime

+  sourceNoChange.connect(analyserNoChange);

+  sourceExplicitAssignement.connect(analyserExplicitAssignment);

+  sourceSetValueAtTime.connect(analyserSetValueAtTime);

+

+  // Two seconds in, increase gain for setValueAt path

+  gainSetValueAtTime.gain.setValueAtTime(0.5, 2);

+

+  // Maximum values seen at each analyser node, will be updated by

+  // checkAnalysersForMaxValues() during test.

+  let maxNoGainChange = 0;

+  let maxExplicitAssignment = 0;

+  let maxSetValueAtTime = 0;

+

+  // Poll analysers and check for max values

+  function checkAnalysersForMaxValues() {

+    let findMaxValue =

+      (array) => array.reduce((a, b) => Math.max(Math.abs(a), Math.abs(b)));

+

+    let dataArray = new Float32Array(analyserNoChange.fftSize);

+    analyserNoChange.getFloatTimeDomainData(dataArray);

+    maxNoGainChange = Math.max(maxNoGainChange, findMaxValue(dataArray));

+

+    analyserExplicitAssignment.getFloatTimeDomainData(dataArray);

+    maxExplicitAssignment = Math.max(maxExplicitAssignment, findMaxValue(dataArray));

+

+    analyserSetValueAtTime.getFloatTimeDomainData(dataArray);

+    maxSetValueAtTime = Math.max(maxSetValueAtTime, findMaxValue(dataArray));

+

+    // End test if we've met our conditions

+    // Add a small amount to initial gain to make sure we're not getting

+    // passes due to rounding errors.

+    let epsilon = 0.01;

+    if (maxExplicitAssignment > (maxNoGainChange + epsilon) &&

+        maxSetValueAtTime > (maxNoGainChange + epsilon)) {

+      source.stop();

+    }

+  }

+

+  source.onended = () => {

+    sourceFinished = true;

+    info(`maxNoGainChange: ${maxNoGainChange}`);

+    info(`maxExplicitAssignment: ${maxExplicitAssignment}`);

+    info(`maxSetValueAtTime: ${maxSetValueAtTime}`);

+    ok(gainExplicitlyIncreased,

+       "Gain should be explicitly assinged during test!");

+    // Add a small amount to initial gain to make sure we're not getting

+    // passes due to rounding errors.

+    let epsilon = 0.01;

+    ok(maxExplicitAssignment > (maxNoGainChange + epsilon),

+       "Volume should increase due to explicit assignment to gain.value");

+    ok(maxSetValueAtTime > (maxNoGainChange + epsilon),

+       "Volume should increase due to setValueAtTime on gain.value");

+    SimpleTest.finish();

+  };

+

+  // Start the graph

+  source.start(0);

+

+  // We'll use this callback to check our analysers for gain

+  function animationFrameCb() {

+    if (sourceFinished) {

+      return;

+    }

+    requestAnimationFrame(animationFrameCb);

+    checkAnalysersForMaxValues();

+  }

+

+  // Using timers is gross, but as of writing there doesn't appear to be a

+  // nicer way to perform gain.value = 0.5 on our node. When/if we support

+  // suspend(time) on offline contexts we could potentially use that instead.

+

+  // Roughly 2 seconds through our source buffer (setTimeout flakiness) increase

+  // our gain on gainExplicitAssignment path.

+  window.setTimeout(() => {

+    gainExplicitAssignment.gain.value = 0.5;

+    // Make debugging flaky timeouts in test easier

+    info("Gain explicitly set!")

+    gainExplicitlyIncreased = true;

+    // Start checking analysers, we do this only after changing volume to avoid

+    // possible starvation of this timeout from requestAnimationFrame calls.

+    animationFrameCb();

+  }, 2000);

+});

+

+</script>

+</pre>

+</body>

+</html>

diff --git a/dom/media/webaudio/test/test_bug808374.html b/dom/media/webaudio/test/test_bug808374.html
new file mode 100644
index 0000000000..b255c0b9c3
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug808374.html
@@ -0,0 +1,22 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 808374</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+try {
+  var ctx = new AudioContext();
+  ctx.createBuffer(0, 1, ctx.sampleRate);
+} catch (e) {
+  ok(true, "The test should not crash during CC");
+}
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug827541.html b/dom/media/webaudio/test/test_bug827541.html
new file mode 100644
index 0000000000..f205c7edf9
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug827541.html
@@ -0,0 +1,24 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Tell the cycle collector about the audio contexts owned by nsGlobalWindow</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+  var iframe = document.createElementNS("http://www.w3.org/1999/xhtml", "iframe");
+  document.body.appendChild(iframe);
+  var frameWin = iframe.contentWindow;
+  new frameWin.AudioContext();
+  document.body.removeChild(iframe);
+  expectException(() => new frameWin.AudioContext(),
+                  DOMException.INVALID_STATE_ERR);
+
+  // This test should not leak.
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug839753.html b/dom/media/webaudio/test/test_bug839753.html
new file mode 100644
index 0000000000..f3e6598116
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug839753.html
@@ -0,0 +1,18 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 839753</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+(new AudioContext()).destination.expando = null;
+ok(true, "The test should not trigger wrapper cache assertions");
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug845960.html b/dom/media/webaudio/test/test_bug845960.html
new file mode 100644
index 0000000000..17bf9a5700
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug845960.html
@@ -0,0 +1,18 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 845960</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+(new AudioContext()).decodeAudioData(new ArrayBuffer(0), function() {});
+ok(true, "Should not crash when the optional failure callback is not specified");
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug856771.html b/dom/media/webaudio/test/test_bug856771.html
new file mode 100644
index 0000000000..24a527ccd5
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug856771.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test for bug 856771</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+
+  var source = context.createBufferSource();
+  source.connect(context.destination);
+  ok(true, "Nothing should leak");
+
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug866570.html b/dom/media/webaudio/test/test_bug866570.html
new file mode 100644
index 0000000000..90bd8f6985
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug866570.html
@@ -0,0 +1,18 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 859600</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+(new AudioContext()).foo = null;
+ok(true, "The test should not fatally assert");
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug866737.html b/dom/media/webaudio/test/test_bug866737.html
new file mode 100644
index 0000000000..e8db6b76e8
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug866737.html
@@ -0,0 +1,36 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test for bug 866737</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var context = new AudioContext();
+
+(function() {
+  var d = context.createDelay();
+  var panner = context.createPanner();
+  d.connect(panner);
+  var gain = context.createGain();
+  panner.connect(gain);
+  gain.connect(context.destination);
+  gain.disconnect(0);
+})();
+
+SpecialPowers.forceGC();
+SpecialPowers.forceCC();
+
+var gain = context.createGain();
+gain.connect(context.destination);
+gain.disconnect(0);
+
+ok(true, "No crashes should happen!");
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug867089.html b/dom/media/webaudio/test/test_bug867089.html
new file mode 100644
index 0000000000..e5a5179530
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug867089.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 867089</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ctx = new AudioContext();
+
+  // Test invalid playbackRate values for AudioBufferSourceNode.
+  var source = ctx.createBufferSource();
+  var buffer = ctx.createBuffer(2, 2048, 8000);
+  source.buffer = buffer;
+  source.playbackRate.value = 0.0;
+  source.connect(ctx.destination);
+  source.start(0);
+
+  var source2 = ctx.createBufferSource();
+  source2.buffer = buffer;
+  source2.playbackRate.value = -1.0;
+  source2.connect(ctx.destination);
+  source2.start(0);
+
+  var source3 = ctx.createBufferSource();
+  source3.buffer = buffer;
+  source3.playbackRate.value = 3000000.0;
+  source3.connect(ctx.destination);
+  source3.start(0);
+  ok(true, "We did not crash.");
+  SimpleTest.finish();
+});
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug867174.html b/dom/media/webaudio/test/test_bug867174.html
new file mode 100644
index 0000000000..e949bcec41
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug867174.html
@@ -0,0 +1,38 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 867174</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ctx = new AudioContext();
+
+  var source = ctx.createBufferSource();
+  var buffer = ctx.createBuffer(2, 2048, 8000);
+  source.playbackRate.setTargetAtTime(0, 2, 3);
+  var sp = ctx.createScriptProcessor();
+  source.connect(sp);
+  sp.connect(ctx.destination);
+  source.start(0);
+
+  sp.onaudioprocess = function(e) {
+    // Now set the buffer
+    source.buffer = buffer;
+
+    ok(true, "We did not crash.");
+    sp.onaudioprocess = null;
+    SimpleTest.finish();
+  };
+});
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug873335.html b/dom/media/webaudio/test/test_bug873335.html
new file mode 100644
index 0000000000..19fd6d4dae
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug873335.html
@@ -0,0 +1,22 @@
+<html>
+<head>
+<meta charset="UTF-8">
+<script src="/tests/SimpleTest/SimpleTest.js"></script>
+<script>
+
+function boom()
+{
+  (new AudioContext()).createScriptProcessor().hamster = {};
+  SpecialPowers.forceCC();
+  SpecialPowers.forceGC();
+  ok(true, "test finished");
+  SimpleTest.finish();
+}
+
+SimpleTest.waitForExplicitFinish();
+
+</script>
+</head>
+
+<body onload="boom();"></body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug875221.html b/dom/media/webaudio/test/test_bug875221.html
new file mode 100644
index 0000000000..134316f9ae
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug875221.html
@@ -0,0 +1,243 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 875221</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+/* eslint-disable no-implied-eval */
+
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("This test is generated by a fuzzer, so we leave these setTimeouts untouched.");
+
+let o0, o1, o2, o3, o4, o5, o6, o7, o8, o9, o10;
+let o11, o12, o13, o14, o15, o16, o17, o18, o19;
+
+try { o0 = document.createElement('audio'); } catch(e) {  }
+try { (document.body || document.documentElement).appendChild(o0); } catch(e) {  }
+try { o1 = new OfflineAudioContext(1, 10, (new AudioContext()).sampleRate); } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o1.listener.dopplerFactor = 1; } catch(e) {  }
+try { o2 = o1.createScriptProcessor(); } catch(e) {  }
+try { o3 = o1.createChannelMerger(4); } catch(e) {  }
+try { o1.listener.dopplerFactor = 3; } catch(e) {  }
+try { o1.listener.setPosition(0, 134217728, 64) } catch(e) {  }
+try { o1.listener.dopplerFactor = 15; } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o4 = new OfflineAudioContext(1, 10, (new AudioContext()).sampleRate); } catch(e) {  }
+try { o4.listener.speedOfSound = 2048; } catch(e) {  }
+try { o4.listener.setPosition(32768, 1, 1) } catch(e) {  }
+try { o5 = o1.createChannelSplitter(4); } catch(e) {  }
+try { o4.listener.setVelocity(4, 1, 0) } catch(e) {  }
+try { o4.startRendering(); } catch(e) {  }
+try { o4.startRendering(); } catch(e) {  }
+try { o4.listener.setPosition(64, 1, 0) } catch(e) {  }
+try { o1.listener.setOrientation(4194304, 15, 8388608, 15, 1, 1) } catch(e) {  }
+try { o1.listener.dopplerFactor = 256; } catch(e) {  }
+try { o6 = o4.createDelay(16); } catch(e) {  }
+try { o4.startRendering(); } catch(e) {  }
+try { o4.listener.setOrientation(0, 1, 0, 0, 31, 1073741824) } catch(e) {  }
+try { o4.listener.speedOfSound = 1; } catch(e) {  }
+try { o1.listener.speedOfSound = 0; } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o6.connect(o3, 1, 0) } catch(e) {  }
+try { o1.listener.setPosition(4294967296, 32, 1) } catch(e) {  }
+try { o1.listener.speedOfSound = 0; } catch(e) {  }
+try { o1.listener.speedOfSound = 0; } catch(e) {  }
+try { o1.listener.setVelocity(1, 256, 0) } catch(e) {  }
+try { o4.startRendering(); } catch(e) {  }
+try { o3.disconnect() } catch(e) {  }
+setTimeout("try { o4.startRendering(); } catch(e) {  }",50)
+try { o4.listener.setOrientation(0, 0, 2048, 128, 16384, 127) } catch(e) {  }
+try { o4.listener.setVelocity(0, 4, 1) } catch(e) {  }
+try { o7 = o4.createScriptProcessor(1024, 4, 1); } catch(e) {  }
+try { o8 = o4.createDynamicsCompressor(); } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+SpecialPowers.forceCC();
+SpecialPowers.forceGC();
+try { o4.listener.setOrientation(8192, 1, 1, 512, 0, 15) } catch(e) {  }
+setTimeout("try { o7.onaudioprocess = function() {}; } catch(e) {  }",50)
+try { o1.startRendering(); } catch(e) {  }
+try { o1.listener.speedOfSound = 1073741824; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o9 = o4.createScriptProcessor(1024, 1, 4); } catch(e) {  }
+try { o10 = o4.createAnalyser(); } catch(e) {  }
+try { o4.listener.speedOfSound = 0; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+try { o4.listener.setVelocity(524288, 1, 65536) } catch(e) {  }
+setTimeout("try { o2.connect(o9); } catch(e) {  } setTimeout(done, 0);",1000)
+try { o7.connect(o4); } catch(e) {  }
+try { o1.listener.setVelocity(1, 127, 31) } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+setTimeout("try { o5.disconnect() } catch(e) {  }",100)
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o4.startRendering(); } catch(e) {  }
+setTimeout("try { o1.listener.dopplerFactor = 1; } catch(e) {  }",100)
+try { o5.disconnect() } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o10.disconnect() } catch(e) {  }
+try { o1.startRendering(); } catch(e) {  }
+try { o11 = o1.createGain(); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o4); } catch(e) {  }
+try { o4.listener.setOrientation(31, 0, 15, 0, 33554432, 1) } catch(e) {  }
+try { o4.listener.dopplerFactor = 1; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+setTimeout("try { o9.connect(o4); } catch(e) {  }",50)
+try { o2.connect(o9); } catch(e) {  }
+setTimeout("try { o9.connect(o1); } catch(e) {  }",200)
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+try { o12 = o4.createDynamicsCompressor(); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+try { o9.onaudioprocess = function() {}; } catch(e) {  }
+try { o1.listener.speedOfSound = 262144; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+setTimeout("try { o7.connect(o4); } catch(e) {  }",50)
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o13 = o4.createGain(); } catch(e) {  }
+try { o4.listener.dopplerFactor = 31; } catch(e) {  }
+try { o11.gain.value = 268435456; } catch(e) {  }
+try { o1.listener.setOrientation(63, 3, 1, 63, 1, 2147483648) } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o4.listener.setVelocity(1, 0, 1) } catch(e) {  }
+try { o11.gain.value = 65536; } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+setTimeout("try { o7.connect(o4); } catch(e) {  }",200)
+try { o14 = o4.createDynamicsCompressor(); } catch(e) {  }
+setTimeout("try { o2.connect(o9); } catch(e) {  }",50)
+try { o7.connect(o1); } catch(e) {  }
+try { o15 = o1.createWaveShaper(); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+try { o16 = o1.createWaveShaper(); } catch(e) {  }
+try { o11.gain.value = 1; } catch(e) {  }
+try { o1.listener.speedOfSound = 16; } catch(e) {  }
+try { o4.listener.setVelocity(0, 127, 15) } catch(e) {  }
+try { o1.listener.setVelocity(0, 2048, 16777216) } catch(e) {  }
+try { o13.gain.value = 0; } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+try { o17 = document.createElement('audio'); } catch(e) {  }
+try { (document.body || document.documentElement).appendChild(o0); } catch(e) {  }
+try { o4.listener.setVelocity(3, 1, 256) } catch(e) {  }
+try { o11.gain.cancelScheduledValues(1) } catch(e) {  }
+try { o1.listener.dopplerFactor = 524288; } catch(e) {  }
+try { o9.onaudioprocess = function() {}; } catch(e) {  }
+setTimeout("try { o7.connect(o13, 0, 0) } catch(e) {  }",50)
+try { o1.listener.speedOfSound = 0; } catch(e) {  }
+try { o10.disconnect() } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o9.connect(o4); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o4); } catch(e) {  }
+try { o1.listener.speedOfSound = 1; } catch(e) {  }
+try { o15.disconnect() } catch(e) {  }
+try { o11.gain.exponentialRampToValueAtTime(0, 15) } catch(e) {  }
+try { o15.curve = new Float32Array(15); } catch(e) {  }
+try { o4.listener.setVelocity(1, 1, 1) } catch(e) {  }
+try { o14.connect(o6, 0, 0) } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+setTimeout("try { o7.connect(o1); } catch(e) {  }",100)
+try { o4.listener.setVelocity(1, 7, 1) } catch(e) {  }
+try { o18 = document.createElement('audio'); } catch(e) {  }
+try { (document.body || document.documentElement).appendChild(o18); } catch(e) {  }
+try { o19 = o4.createGain(); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+try { o4.listener.dopplerFactor = 0; } catch(e) {  }
+try { o1.listener.setPosition(256, 16, 1) } catch(e) {  }
+setTimeout("try { o2.connect(o9); } catch(e) {  }",50)
+try { o7.connect(o1); } catch(e) {  }
+try { o4.listener.speedOfSound = 31; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+setTimeout("try { o9.connect(o4); } catch(e) {  }",1000)
+try { o11.gain.value = 127; } catch(e) {  }
+try { o7.connect(o7, 0, 0) } catch(e) {  }
+try { o4.listener.speedOfSound = 63; } catch(e) {  }
+try { o11.gain.value = 33554432; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o4); } catch(e) {  }
+try { o4.listener.speedOfSound = 16; } catch(e) {  }
+try { o4.listener.setVelocity(1048576, 0, 127) } catch(e) {  }
+try { o1.listener.dopplerFactor = 0; } catch(e) {  }
+try { o6.connect(o2, 0, 1) } catch(e) {  }
+try { o5.disconnect() } catch(e) {  }
+try { o3.disconnect() } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+try { o16.disconnect() } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+try { o9.disconnect() } catch(e) {  }
+try { o4.listener.speedOfSound = 1; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o11.gain.setValueCurveAtTime(new Float32Array(3), 2048, 3) } catch(e) {  }
+try { o13.gain.value = 8; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o4); } catch(e) {  }
+try { o4.listener.setOrientation(1, 2048, 1, 1, 0, 31) } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o1); } catch(e) {  }
+try { o1.listener.speedOfSound = 256; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o4); } catch(e) {  }
+try { o4.listener.setVelocity(1, 67108864, 128) } catch(e) {  }
+setTimeout("try { o1.listener.setVelocity(0, 1, 1) } catch(e) {  }",100)
+try { o2.connect(o9); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+setTimeout("try { o20 = o1.createBiquadFilter(); } catch(e) {  }",200)
+try { o13.gain.value = 4096; } catch(e) {  }
+try { o1.listener.dopplerFactor = 0; } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+setTimeout("try { o2.connect(o9); } catch(e) {  }",200)
+try { o7.connect(o1); } catch(e) {  }
+try { o3.connect(o15, 1, 1) } catch(e) {  }
+try { o2.connect(o12, 0, 0) } catch(e) {  }
+try { o19.gain.exponentialRampToValueAtTime(1, 0) } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+
+function done() {
+  ok(true, "We did not crash.");
+  SimpleTest.finish();
+}
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug875402.html b/dom/media/webaudio/test/test_bug875402.html
new file mode 100644
index 0000000000..bb14c82037
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug875402.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Crashtest for bug 875402</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+/* eslint-disable no-implied-eval */
+
+SimpleTest.waitForExplicitFinish();
+
+SimpleTest.requestFlakyTimeout("This test is generated by a fuzzer, so we leave these setTimeouts untouched.");
+
+let o1, o2, o4, o5, o7, o9, o13;
+
+try { o1 = new OfflineAudioContext(1, 10, (new AudioContext()).sampleRate); } catch(e) {  }
+try { o2 = o1.createScriptProcessor(); } catch(e) {  }
+try { o4 = new OfflineAudioContext(1, 10, (new AudioContext()).sampleRate); } catch(e) {  }
+try { o5 = o1.createChannelSplitter(4); } catch(e) {  }
+try { o7 = o4.createScriptProcessor(1024, 4, 1); } catch(e) {  }
+SpecialPowers.forceCC();
+SpecialPowers.forceGC();
+try { o1.startRendering(); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o7.connect(o4); } catch(e) {  }
+try { o9 = o4.createScriptProcessor(1024, 1, 4); } catch(e) {  }
+try { o2.connect(o7); } catch(e) {  }
+try { o9.connect(o1); } catch(e) {  }
+setTimeout("try { o2.connect(o9); } catch(e) {  } done();",1000)
+try { o7.connect(o4); } catch(e) {  }
+setTimeout("try { o5.disconnect() } catch(e) {  }",100)
+try { o2.connect(o9); } catch(e) {  }
+try { o4.startRendering(); } catch(e) {  }
+try { o2.connect(o9); } catch(e) {  }
+setTimeout("try { o7.connect(o4); } catch(e) {  }",50)
+try { o13 = o4.createGain(); } catch(e) {  }
+setTimeout("try { o7.connect(o13, 0, 0) } catch(e) {  }",50)
+
+function done() {
+  ok(true, "We did not crash.");
+  SimpleTest.finish();
+}
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug894150.html b/dom/media/webaudio/test/test_bug894150.html
new file mode 100644
index 0000000000..4577232d71
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug894150.html
@@ -0,0 +1,21 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script>
+
+var ac = new AudioContext();
+ac.createPanner();
+var listener = ac.listener;
+SpecialPowers.forceGC();
+SpecialPowers.forceCC();
+listener.setOrientation(0, 0, -1, 0, 0, 0);
+
+ok(true, "No crashes should happen!");
+
+</script>
+</body>
diff --git a/dom/media/webaudio/test/test_bug956489.html b/dom/media/webaudio/test/test_bug956489.html
new file mode 100644
index 0000000000..f0ae559b05
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug956489.html
@@ -0,0 +1,56 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test when and currentTime are in the same coordinate system</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("This test needs to wait a while for the AudioContext's timer to start.");
+addLoadEvent(function() {
+  var freq = 330;
+
+  var context = new AudioContext();
+
+  var buffer = context.createBuffer(1, context.sampleRate / freq, context.sampleRate);
+  for (var i = 0; i < buffer.length; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(2 * Math.PI * i / buffer.length);
+  }
+
+  var source = context.createBufferSource();
+  source.loop = true;
+  source.buffer = buffer;
+
+  setTimeout(function () {
+      var finished = false;
+
+      source.start(context.currentTime);
+      var processor = context.createScriptProcessor(256, 1, 1);
+      processor.onaudioprocess = function (e) {
+          if (finished) return;
+          var c = e.inputBuffer.getChannelData(0);
+          var result = true;
+
+          for (var i = 0; i < buffer.length; ++i) {
+              if (Math.abs(c[i] - buffer.getChannelData(0)[i]) > 1e-9) {
+                  result = false;
+                  break;
+              }
+          }
+          finished = true;
+          ok(result, "when and currentTime are in same time coordinate system");
+          SimpleTest.finish();
+      }
+      processor.connect(context.destination);
+      source.connect(processor);
+  }, 500);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug964376.html b/dom/media/webaudio/test/test_bug964376.html
new file mode 100644
index 0000000000..bc9d167dcd
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug964376.html
@@ -0,0 +1,64 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test repeating audio is not distorted</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function gcd(a, b) {
+  if (b === 0) {
+    return a;
+  }
+  return gcd(b, a % b);
+}
+
+var SAMPLE_PLACEMENT = 128;
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+
+  createGraph(context) {
+    var freq = Math.round(context.sampleRate / SAMPLE_PLACEMENT);
+    var dur = context.sampleRate / gcd(freq, context.sampleRate);
+    var buffer = context.createBuffer(1, dur, context.sampleRate);
+
+    for (var i = 0; i < context.sampleRate; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(freq * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.loop = true;
+    source.playbackRate.setValueAtTime(0.5, SAMPLE_PLACEMENT / context.sampleRate);
+    source.start(0);
+
+    return source;
+  },
+
+  createExpectedBuffers(context) {
+    var freq = Math.round(context.sampleRate / SAMPLE_PLACEMENT);
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    var c = expectedBuffer.getChannelData(0);
+    for (var i = 0; i < c.length; ++i) {
+      if (i < SAMPLE_PLACEMENT) {
+        c[i] = Math.sin(freq * 2 * Math.PI * i / context.sampleRate);
+      } else {
+        c[i] = Math.sin(freq / 2 * 2 * Math.PI * (i + SAMPLE_PLACEMENT) / context.sampleRate);
+      }
+    }
+
+    return expectedBuffer;
+  },
+};
+
+runTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug966247.html b/dom/media/webaudio/test/test_bug966247.html
new file mode 100644
index 0000000000..69831c33b6
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug966247.html
@@ -0,0 +1,46 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether an audio file played with a volume set to 0 plays silence</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<audio preload=none src="ting-48k-1ch.ogg" controls> </audio>
+<script>
+  SimpleTest.waitForExplicitFinish();
+
+  var count = 20;
+
+  function isSilent(b) {
+    for (var i = 0; i < b.length; i++) {
+      if (b[i] != 0.0) {
+        return false;
+      }
+    }
+    return true;
+  }
+
+  var a = document.getElementsByTagName("audio")[0];
+  a.volume = 0.0;
+  var ac = new AudioContext();
+  var measn = ac.createMediaElementSource(a);
+  var sp = ac.createScriptProcessor();
+
+  sp.onaudioprocess = function(e) {
+    var inputBuffer = e.inputBuffer.getChannelData(0);
+    ok(isSilent(inputBuffer), "The volume is set to 0, so all the elements of the buffer are supposed to be equal to 0.0");
+  }
+  // Connect the MediaElementAudioSourceNode to the ScriptProcessorNode to check
+  // the audio volume.
+  measn.connect(sp);
+  a.play();
+
+  a.addEventListener("ended", function() {
+    sp.onaudioprocess = null;
+    SimpleTest.finish();
+  });
+
+</script>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_bug972678.html b/dom/media/webaudio/test/test_bug972678.html
new file mode 100644
index 0000000000..ac96b4ac3b
--- /dev/null
+++ b/dom/media/webaudio/test/test_bug972678.html
@@ -0,0 +1,62 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test buffers do not interfere when scheduled in sequence</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+var OFFSETS = [0.005, 0.01, 0.02, 0.03];
+var LENGTH = 128;
+
+var gTest = {
+  length: 128 * OFFSETS.length,
+  numberOfChannels: 1,
+
+  createGraph(context) {
+    var gain = context.createGain();
+
+    // create a repeating sample
+    var repeatingSample = context.createBuffer(1, 2, context.sampleRate);
+    var c = repeatingSample.getChannelData(0);
+    for (var i = 0; i < repeatingSample.length; ++i) {
+      c[i] = i % 2 == 0 ? 1 : -1;
+    }
+
+    OFFSETS.forEach(function (offset, offsetIdx) {
+      // Schedule a set of nodes to repeat the sample.
+      for (var i = 0; i < LENGTH; i += repeatingSample.length) {
+        var source = context.createBufferSource();
+        source.buffer = repeatingSample;
+        source.connect(gain);
+        source.start((offsetIdx * LENGTH + i + offset) / context.sampleRate);
+      }
+
+      let buffer = context.createBuffer(1, LENGTH, context.sampleRate);
+      c = buffer.getChannelData(0);
+      for (var i = 0; i < buffer.length; ++i) {
+        c[i] = i % 2 == 0 ? -1 : 1;
+      }
+
+      var source = context.createBufferSource();
+      source.buffer = buffer;
+      source.connect(gain);
+      source.start((offsetIdx * LENGTH + offset) / context.sampleRate);
+    });
+
+    return gain;
+  },
+
+  createExpectedBuffers(context) {
+    return context.createBuffer(1, gTest.length, context.sampleRate);
+  },
+};
+
+runTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_channelMergerNode.html b/dom/media/webaudio/test/test_channelMergerNode.html
new file mode 100644
index 0000000000..b62d34d6ba
--- /dev/null
+++ b/dom/media/webaudio/test/test_channelMergerNode.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ChannelMergerNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 6,
+  createGraph(context) {
+    var buffers = [];
+    for (var j = 0; j < 6; ++j) {
+      var buffer = context.createBuffer(2, 2048, context.sampleRate);
+      for (var i = 0; i < 2048; ++i) {
+        buffer.getChannelData(0)[i] = Math.sin(440 * 2 * (j + 1) * Math.PI * i / context.sampleRate);
+        // Second channel is silent
+      }
+      buffers.push(buffer);
+    }
+
+    var merger = new ChannelMergerNode(context);
+    is(merger.channelCount, 1, "merger node has 1 input channels");
+    is(merger.channelCountMode, "explicit", "Correct channelCountMode for the merger node");
+    is(merger.channelInterpretation, "speakers", "Correct channelCountInterpretation for the merger node");
+
+    for (var i = 0; i < 6; ++i) {
+      var source = context.createBufferSource();
+      source.buffer = buffers[i];
+      source.connect(merger, 0, i);
+      source.start(0);
+    }
+
+    return merger;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(6, 2048, context.sampleRate);
+    for (var i = 0; i < 6; ++i) {
+      for (var j = 0; j < 2048; ++j) {
+        expectedBuffer.getChannelData(i)[j] = 0.5 * Math.sin(440 * 2 * (i + 1) * Math.PI * j / context.sampleRate);
+      }
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_channelMergerNodeWithVolume.html b/dom/media/webaudio/test/test_channelMergerNodeWithVolume.html
new file mode 100644
index 0000000000..55b9ec0c0b
--- /dev/null
+++ b/dom/media/webaudio/test/test_channelMergerNodeWithVolume.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ChannelMergerNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 6,
+  createGraph(context) {
+    var buffers = [];
+    for (var j = 0; j < 6; ++j) {
+      var buffer = context.createBuffer(2, 2048, context.sampleRate);
+      for (var i = 0; i < 2048; ++i) {
+        buffer.getChannelData(0)[i] = Math.sin(440 * 2 * (j + 1) * Math.PI * i / context.sampleRate);
+        // Second channel is silent
+      }
+      buffers.push(buffer);
+    }
+
+    var merger = context.createChannelMerger();
+    is(merger.channelCount, 1, "merger node has 1 input channels");
+    is(merger.channelCountMode, "explicit", "Correct channelCountMode for the merger node");
+    is(merger.channelInterpretation, "speakers", "Correct channelCountInterpretation for the merger node");
+
+    for (var i = 0; i < 6; ++i) {
+      var source = context.createBufferSource();
+      source.buffer = buffers[i];
+      var gain = context.createGain();
+      gain.gain.value = 0.5;
+      source.connect(gain);
+      gain.connect(merger, 0, i);
+      source.start(0);
+    }
+
+    return merger;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(6, 2048, context.sampleRate);
+    for (var i = 0; i < 6; ++i) {
+      for (var j = 0; j < 2048; ++j) {
+        expectedBuffer.getChannelData(i)[j] = 0.5 * 0.5 * Math.sin(440 * 2 * (i + 1) * Math.PI * j / context.sampleRate);
+      }
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_channelSplitterNode.html b/dom/media/webaudio/test/test_channelSplitterNode.html
new file mode 100644
index 0000000000..d74845f821
--- /dev/null
+++ b/dom/media/webaudio/test/test_channelSplitterNode.html
@@ -0,0 +1,71 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ChannelSplitterNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// We do not use our generic graph test framework here because
+// the splitter node is special in that it creates multiple
+// output ports.
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(4, 2048, context.sampleRate);
+  for (var j = 0; j < 4; ++j) {
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(j)[i] = Math.sin(440 * 2 * (j + 1) * Math.PI * i / context.sampleRate);
+    }
+  }
+  var emptyBuffer = context.createBuffer(1, 2048, context.sampleRate);
+
+  var destination = context.destination;
+
+  var source = context.createBufferSource();
+
+  var splitter = new ChannelSplitterNode(context);
+  is(splitter.channelCount, 6, "splitter node has 2 input channels by default");
+  is(splitter.channelCountMode, "explicit", "Correct channelCountMode for the splitter node");
+  is(splitter.channelInterpretation, "discrete", "Correct channelCountInterpretation for the splitter node");
+
+  source.buffer = buffer;
+  source.connect(splitter);
+
+  var channelsSeen = 0;
+  function createHandler(i) {
+    return function(e) {
+      is(e.inputBuffer.numberOfChannels, 1, "Correct input channel count");
+      if (i < 4) {
+        compareChannels(e.inputBuffer.getChannelData(0), buffer.getChannelData(i));
+      } else {
+        compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0));
+      }
+      e.target.onaudioprocess = null;
+      ++channelsSeen;
+
+      if (channelsSeen == 6) {
+        SimpleTest.finish();
+      }
+    };
+  }
+
+  for (var i = 0; i < 6; ++i) {
+    var sp = context.createScriptProcessor(2048, 1);
+    splitter.connect(sp, i);
+    sp.onaudioprocess = createHandler(i);
+    sp.connect(destination);
+  }
+
+  source.start(0);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_channelSplitterNodeWithVolume.html b/dom/media/webaudio/test/test_channelSplitterNodeWithVolume.html
new file mode 100644
index 0000000000..c03f6deeaf
--- /dev/null
+++ b/dom/media/webaudio/test/test_channelSplitterNodeWithVolume.html
@@ -0,0 +1,76 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ChannelSplitterNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// We do not use our generic graph test framework here because
+// the splitter node is special in that it creates multiple
+// output ports.
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(4, 2048, context.sampleRate);
+  var expectedBuffer = context.createBuffer(4, 2048, context.sampleRate);
+  for (var j = 0; j < 4; ++j) {
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(j)[i] = Math.sin(440 * 2 * (j + 1) * Math.PI * i / context.sampleRate);
+      expectedBuffer.getChannelData(j)[i] = buffer.getChannelData(j)[i] / 2;
+    }
+  }
+  var emptyBuffer = context.createBuffer(1, 2048, context.sampleRate);
+
+  var destination = context.destination;
+
+  var source = context.createBufferSource();
+
+  var splitter = context.createChannelSplitter();
+  is(splitter.channelCount, 6, "splitter node has 2 input channels by default");
+  is(splitter.channelCountMode, "explicit", "Correct channelCountMode for the splitter node");
+  is(splitter.channelInterpretation, "discrete", "Correct channelCountInterpretation for the splitter node");
+
+  source.buffer = buffer;
+  var gain = context.createGain();
+  gain.gain.value = 0.5;
+  source.connect(gain);
+  gain.connect(splitter);
+
+  var channelsSeen = 0;
+  function createHandler(i) {
+    return function(e) {
+      is(e.inputBuffer.numberOfChannels, 1, "Correct input channel count");
+      if (i < 4) {
+        compareBuffers(e.inputBuffer.getChannelData(0), expectedBuffer.getChannelData(i));
+      } else {
+        compareBuffers(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0));
+      }
+      e.target.onaudioprocess = null;
+      ++channelsSeen;
+
+      if (channelsSeen == 6) {
+        SimpleTest.finish();
+      }
+    };
+  }
+
+  for (var i = 0; i < 6; ++i) {
+    var sp = context.createScriptProcessor(2048, 1);
+    splitter.connect(sp, i);
+    sp.onaudioprocess = createHandler(i);
+    sp.connect(destination);
+  }
+
+  source.start(0);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_convolver-upmixing-1-channel-response.html b/dom/media/webaudio/test/test_convolver-upmixing-1-channel-response.html
new file mode 100644
index 0000000000..50bd594821
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolver-upmixing-1-channel-response.html
@@ -0,0 +1,143 @@
+<!DOCTYPE html>
+<title>Test that up-mixing signals in ConvolverNode processing is linear</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+const EPSILON = 3.0 * Math.pow(2, -22);
+// sampleRate is a power of two so that delay times are exact in base-2
+// floating point arithmetic.
+const SAMPLE_RATE = 32768;
+// Length of stereo convolver input in frames (arbitrary):
+const STEREO_FRAMES = 256;
+// Length of mono signal in frames.  This is more than two blocks to ensure
+// that at least one block will be mono, even if interpolation in the
+// DelayNode means that stereo is output one block earlier and later than
+// if frames are delayed without interpolation.
+const MONO_FRAMES = 384;
+// Length of response buffer:
+const RESPONSE_FRAMES = 256;
+
+function test_linear_upmixing(channelInterpretation, initial_mono_frames)
+{
+  let stereo_input_end = initial_mono_frames + STEREO_FRAMES;
+  // Total length:
+  let length = stereo_input_end + RESPONSE_FRAMES + MONO_FRAMES + STEREO_FRAMES;
+  // The first two channels contain signal where some up-mixing occurs
+  // internally to a ConvolverNode when a stereo signal is added and removed.
+  // The last two channels are expected to contain the same signal, but mono
+  // and stereo signals are convolved independently before up-mixing the mono
+  // output to mix with the stereo output.
+  let context = new OfflineAudioContext({numberOfChannels: 4,
+                                         length,
+                                         sampleRate: SAMPLE_RATE});
+
+  let response = new AudioBuffer({numberOfChannels: 1,
+                                  length: RESPONSE_FRAMES,
+                                  sampleRate: context.sampleRate});
+
+  // Two stereo channel splitters will collect test and reference outputs.
+  let destinationMerger = new ChannelMergerNode(context, {numberOfInputs: 4});
+  destinationMerger.connect(context.destination);
+  let testSplitter =
+      new ChannelSplitterNode(context, {numberOfOutputs: 2});
+  let referenceSplitter =
+      new ChannelSplitterNode(context, {numberOfOutputs: 2});
+  testSplitter.connect(destinationMerger, 0, 0);
+  testSplitter.connect(destinationMerger, 1, 1);
+  referenceSplitter.connect(destinationMerger, 0, 2);
+  referenceSplitter.connect(destinationMerger, 1, 3);
+
+  // A GainNode mixes reference stereo and mono signals because up-mixing
+  // cannot be performed at a channel splitter.
+  let referenceGain = new GainNode(context);
+  referenceGain.connect(referenceSplitter);
+  referenceGain.channelInterpretation = channelInterpretation;
+
+  // The impulse response for convolution contains two impulses so as to test
+  // effects in at least two processing blocks.
+  response.getChannelData(0)[0] = 0.5;
+  response.getChannelData(0)[response.length - 1] = 0.5;
+
+  let testConvolver = new ConvolverNode(context, {disableNormalization: true,
+                                                  buffer: response});
+  testConvolver.channelInterpretation = channelInterpretation;
+  let referenceMonoConvolver = new ConvolverNode(context,
+                                                 {disableNormalization: true,
+                                                  buffer: response});
+  let referenceStereoConvolver = new ConvolverNode(context,
+                                                   {disableNormalization: true,
+                                                    buffer: response});
+  // No need to set referenceStereoConvolver.channelInterpretation because
+  // input is either silent or stereo.
+  testConvolver.connect(testSplitter);
+  // Mix reference convolver output.
+  referenceMonoConvolver.connect(referenceGain);
+  referenceStereoConvolver.connect(referenceGain);
+
+  // The DelayNode initially has a single channel of silence, which is used to
+  // switch the stereo signal in and out.  The output of the delay node is
+  // first mono silence (if there is a non-zero initial_mono_frames), then
+  // stereo, then mono silence, and finally stereo again.  maxDelayTime is
+  // used to generate the middle mono silence period from the initial silence
+  // in the DelayNode and then generate the final period of stereo from its
+  // initial input.
+  let maxDelayTime = (length - STEREO_FRAMES) / context.sampleRate;
+  let delay =
+      new DelayNode(context,
+                    {maxDelayTime,
+                     delayTime: initial_mono_frames / context.sampleRate});
+  // Schedule an increase in the delay to return to mono silence.
+  delay.delayTime.setValueAtTime(maxDelayTime,
+                                 stereo_input_end / context.sampleRate);
+  delay.connect(testConvolver);
+  delay.connect(referenceStereoConvolver);
+
+  let stereoMerger = new ChannelMergerNode(context, {numberOfInputs: 2});
+  stereoMerger.connect(delay);
+
+  // Three independent signals
+  let monoSignal = new OscillatorNode(context, {frequency: 440});
+  let leftSignal = new OscillatorNode(context, {frequency: 450});
+  let rightSignal = new OscillatorNode(context, {frequency: 460});
+  monoSignal.connect(testConvolver);
+  monoSignal.connect(referenceMonoConvolver);
+  leftSignal.connect(stereoMerger, 0, 0);
+  rightSignal.connect(stereoMerger, 0, 1);
+  monoSignal.start();
+  leftSignal.start();
+  rightSignal.start();
+
+  return context.startRendering().
+    then((buffer) => {
+      let maxDiff = -1.0;
+      let frameIndex = 0;
+      let channelIndex = 0;
+      for (let c = 0; c < 2; ++c) {
+        let testOutput = buffer.getChannelData(0 + c);
+        let referenceOutput = buffer.getChannelData(2 + c);
+        for (var i = 0; i < buffer.length; ++i) {
+          var diff = Math.abs(testOutput[i] - referenceOutput[i]);
+          if (diff > maxDiff) {
+            maxDiff = diff;
+            frameIndex = i;
+            channelIndex = c;
+          }
+        }
+      }
+      assert_approx_equals(buffer.getChannelData(0 + channelIndex)[frameIndex],
+                           buffer.getChannelData(2 + channelIndex)[frameIndex],
+                           EPSILON,
+                           `output at ${frameIndex} ` +
+                             `in channel ${channelIndex}` );
+    });
+}
+
+promise_test(() => test_linear_upmixing("speakers", MONO_FRAMES),
+             "speakers, initially mono");
+promise_test(() => test_linear_upmixing("discrete", MONO_FRAMES),
+             "discrete");
+// Gecko uses a separate path for "speakers" up-mixing when the convolver's
+// first input is stereo, so test that separately.
+promise_test(() => test_linear_upmixing("speakers", 0),
+             "speakers, initially stereo");
+</script>
diff --git a/dom/media/webaudio/test/test_convolverNode.html b/dom/media/webaudio/test/test_convolverNode.html
new file mode 100644
index 0000000000..c1677aafab
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNode.html
@@ -0,0 +1,31 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the ConvolverNode interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var conv = new ConvolverNode(context);
+
+  is(conv.channelCount, 2, "Convolver node has 2 input channels by default");
+  is(conv.channelCountMode, "clamped-max", "Correct channelCountMode for the Convolver node");
+  is(conv.channelInterpretation, "speakers", "Correct channelCountInterpretation for the Convolver node");
+
+  is(conv.buffer, null, "Default buffer value");
+  is(conv.normalize, true, "Default normalize value");
+
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_convolverNodeChannelCount.html b/dom/media/webaudio/test/test_convolverNodeChannelCount.html
new file mode 100644
index 0000000000..03824578ea
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodeChannelCount.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ConvolverNode channel count</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+const signalLength = 2048;
+const responseLength = 1000;
+const outputLength = 2048; // < signalLength + responseLength to test bug 910171
+
+var gTest = {
+  length: outputLength,
+  numberOfChannels: 2,
+  createGraph(context) {
+    var buffer = context.createBuffer(2, signalLength, context.sampleRate);
+    for (var i = 0; i < signalLength; ++i) {
+      var sample = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+      // When mixed into a single channel, this produces silence
+      buffer.getChannelData(0)[i] = sample;
+      buffer.getChannelData(1)[i] = -sample;
+    }
+
+    var response = context.createBuffer(2, responseLength, context.sampleRate);
+    for (var i = 0; i < responseLength; ++i) {
+      response.getChannelData(0)[i] = i / responseLength;
+      response.getChannelData(1)[i] = 1 - (i / responseLength);
+    }
+
+    var convolver = context.createConvolver();
+    convolver.buffer = response;
+    convolver.channelCount = 1;
+
+    expectException(function() { convolver.channelCount = 3; },
+                    DOMException.NOT_SUPPORTED_ERR);
+    convolver.channelCountMode = "explicit";
+    expectException(function() { convolver.channelCountMode = "max"; },
+                    DOMException.NOT_SUPPORTED_ERR);
+    convolver.channelInterpretation = "discrete";
+    convolver.channelInterpretation = "speakers";
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.connect(convolver);
+    source.start(0);
+
+    return convolver;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_convolverNodeChannelInterpretationChanges.html b/dom/media/webaudio/test/test_convolverNodeChannelInterpretationChanges.html
new file mode 100644
index 0000000000..bede517b2e
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodeChannelInterpretationChanges.html
@@ -0,0 +1,169 @@
+<!DOCTYPE html>
+<title>Test up-mixing in ConvolverNode after ChannelInterpretation change</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+// This test is not in wpt because it requires that multiple changes to the
+// nodes in an AudioContext during a single event will be processed by the
+// audio thread in a single transaction.  Gecko provides that, but this is not
+// currently required by the Web Audio API.
+
+const EPSILON = Math.pow(2, -23);
+// sampleRate is a power of two so that delay times are exact in base-2
+// floating point arithmetic.
+const SAMPLE_RATE = 32768;
+// Length of initial mono signal in frames, if the test has an initial mono
+// signal.  This is more than one block to ensure that at least one block
+// will be mono, even if interpolation in the DelayNode means that stereo is
+// output one block earlier than if frames are delayed without interpolation.
+const MONO_FRAMES = 256;
+// Length of response buffer.  This is greater than 1 to ensure that the
+// convolver has stereo output at least one block after stereo input is
+// disconnected.
+const RESPONSE_FRAMES = 2;
+
+function test_interpretation_change(t, initialInterpretation, initialMonoFrames)
+{
+  let context = new AudioContext({sampleRate: SAMPLE_RATE});
+
+  // Three independent signals.  These are constant so that results are
+  // independent of the timing of the `ended` event.
+  let monoOffset = 0.25
+  let monoSource = new ConstantSourceNode(context, {offset: monoOffset});
+  let leftOffset = 0.125;
+  let rightOffset = 0.5;
+  let leftSource = new ConstantSourceNode(context, {offset: leftOffset});
+  let rightSource = new ConstantSourceNode(context, {offset: rightOffset});
+  monoSource.start();
+  leftSource.start();
+  rightSource.start();
+
+  let stereoMerger = new ChannelMergerNode(context, {numberOfInputs: 2});
+  leftSource.connect(stereoMerger, 0, 0);
+  rightSource.connect(stereoMerger, 0, 1);
+
+  // The DelayNode initially has a single channel of silence, and so the
+  // output of the delay node is first mono silence (if there is a non-zero
+  // initialMonoFrames), then stereo.  In Gecko, this triggers a convolver
+  // configuration that is different for different channelInterpretations.
+  let delay =
+      new DelayNode(context,
+                    {maxDelayTime: MONO_FRAMES / context.sampleRate,
+                     delayTime: initialMonoFrames / context.sampleRate});
+  stereoMerger.connect(delay);
+
+  // Two convolvers with the same impulse response.  The test convolver will
+  // process a mix of stereo and mono signals.  The reference convolver will
+  // always process stereo, including the up-mixed mono signal.
+  let response = new AudioBuffer({numberOfChannels: 1,
+                                  length: RESPONSE_FRAMES,
+                                  sampleRate: context.sampleRate});
+  response.getChannelData(0)[response.length - 1] = 1;
+
+  let testConvolver = new ConvolverNode(context,
+                                        {disableNormalization: true,
+                                         buffer: response});
+  testConvolver.channelInterpretation = initialInterpretation;
+  let referenceConvolver = new ConvolverNode(context,
+                                             {disableNormalization: true,
+                                              buffer: response});
+  // No need to set referenceConvolver.channelInterpretation because
+  // input is always stereo, due to up-mixing at gain node.
+  let referenceMixer = new GainNode(context);
+  referenceMixer.channelCount = 2;
+  referenceMixer.channelCountMode = "explicit";
+  referenceMixer.channelInterpretation = initialInterpretation;
+  referenceMixer.connect(referenceConvolver);
+
+  delay.connect(testConvolver);
+  delay.connect(referenceMixer);
+
+  monoSource.connect(testConvolver);
+  monoSource.connect(referenceMixer);
+
+  // A timer sends 'ended' when the convolvers are known to be processing
+  // stereo.
+  let timer = new ConstantSourceNode(context);
+  timer.start();
+  timer.stop((initialMonoFrames + 1) / context.sampleRate);
+
+  timer.onended = t.step_func(() => {
+    let changedInterpretation =
+        initialInterpretation == "speakers" ? "discrete" : "speakers";
+
+    // Switch channelInterpretation in test and reference paths.
+    testConvolver.channelInterpretation = changedInterpretation;
+    referenceMixer.channelInterpretation = changedInterpretation;
+
+    // Disconnect the stereo input from both test and reference convolvers.
+    // The disconnected convolvers will continue to output stereo for at least
+    // one frame.  The test convolver will up-mix its mono input into its two
+    // buffers.
+    delay.disconnect();
+
+    // Capture the outputs in a script processor.
+    //
+    // The first two channels contain signal where some up-mixing occurs
+    // internally to the test convolver.
+    //
+    // The last two channels are expected to contain the same signal, but
+    // up-mixing was performed at a GainNode prior to convolution.
+    //
+    // Two stereo splitters will collect test and reference outputs.
+    let testSplitter =
+        new ChannelSplitterNode(context, {numberOfOutputs: 2});
+    let referenceSplitter =
+        new ChannelSplitterNode(context, {numberOfOutputs: 2});
+    testConvolver.connect(testSplitter);
+    referenceConvolver.connect(referenceSplitter);
+
+    let outputMerger = new ChannelMergerNode(context, {numberOfInputs: 4});
+    testSplitter.connect(outputMerger, 0, 0);
+    testSplitter.connect(outputMerger, 1, 1);
+    referenceSplitter.connect(outputMerger, 0, 2);
+    referenceSplitter.connect(outputMerger, 1, 3);
+
+    let processor = context.createScriptProcessor(256, 4, 0);
+    outputMerger.connect(processor);
+
+    processor.onaudioprocess = t.step_func_done((e) => {
+      e.target.onaudioprocess = null;
+      outputMerger.disconnect();
+
+      // The test convolver output is stereo for the first block.
+      let length = 128;
+
+      let buffer = e.inputBuffer;
+      let maxDiff = -1.0;
+      let frameIndex = 0;
+      let channelIndex = 0;
+      for (let c = 0; c < 2; ++c) {
+        let testOutput = buffer.getChannelData(0 + c);
+        let referenceOutput = buffer.getChannelData(2 + c);
+        for (var i = 0; i < length; ++i) {
+          var diff = Math.abs(testOutput[i] - referenceOutput[i]);
+          if (diff > maxDiff) {
+            maxDiff = diff;
+            frameIndex = i;
+            channelIndex = c;
+          }
+        }
+      }
+      assert_approx_equals(buffer.getChannelData(0 + channelIndex)[frameIndex],
+                           buffer.getChannelData(2 + channelIndex)[frameIndex],
+                           EPSILON,
+                           `output at ${frameIndex} ` +
+                             `in channel ${channelIndex}` );
+    });
+  });
+}
+
+async_test((t) => test_interpretation_change(t, "speakers", MONO_FRAMES),
+           "speakers to discrete, initially mono");
+async_test((t) => test_interpretation_change(t, "discrete", MONO_FRAMES),
+           "discrete to speakers");
+// Gecko uses a separate path for "speakers" initial up-mixing when the
+// convolver's first input is stereo, so test that separately.
+async_test((t) => test_interpretation_change(t, "speakers", 0),
+           "speakers to discrete, initially stereo");
+</script>
diff --git a/dom/media/webaudio/test/test_convolverNodeDelay.html b/dom/media/webaudio/test/test_convolverNodeDelay.html
new file mode 100644
index 0000000000..2e8caf8027
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodeDelay.html
@@ -0,0 +1,72 @@
+<!DOCTYPE html>
+<title>Test convolution to delay a triangle pulse</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+const sampleRate = 48000;
+const LENGTH = 12800;
+// tolerate 16-bit math.
+const EPSILON = 1.0 / Math.pow(2, 15);
+
+// Triangle pulse
+var sourceBuffer = new OfflineAudioContext(1, 1, sampleRate).
+  createBuffer(1, 2 * 128, sampleRate);
+var channelData = sourceBuffer.getChannelData(0);
+for (var i = 0; i < 128; ++i) {
+  channelData[i] = i/128;
+  channelData[128 + i] = 1.0 - i/128;
+}
+
+function test_delay_index(delayIndex) {
+
+  var context = new OfflineAudioContext(2, LENGTH, sampleRate);
+
+  var merger = context.createChannelMerger(2);
+  merger.connect(context.destination);
+
+  var impulse = context.createBuffer(1, delayIndex + 1, sampleRate);
+  impulse.getChannelData(0)[delayIndex] = 1.0;
+  var convolver = context.createConvolver();
+  convolver.normalize = false;
+  convolver.buffer = impulse;
+  convolver.connect(merger, 0, 0);
+
+  var delayTime = delayIndex/sampleRate;
+  var delay = context.createDelay(delayTime || 1/sampleRate);
+  delay.delayTime.value = delayTime;
+  delay.connect(merger, 0, 1);
+
+  var source = context.createBufferSource();
+  source.buffer = sourceBuffer;
+  source.connect(convolver);
+  source.connect(delay);
+  source.start(0);
+
+  return context.startRendering().
+    then((buffer) => {
+      var convolverOutput = buffer.getChannelData(0);
+      var delayOutput = buffer.getChannelData(1);
+      var maxDiff = 0.0;
+      var maxIndex = 0;
+      for (var i = 0; i < buffer.length; ++i) {
+        var diff = Math.abs(convolverOutput[i] - delayOutput[i]);
+        if (diff > maxDiff) {
+          maxDiff = diff;
+          maxIndex = i;
+        }
+      }
+      // The convolver should produce similar output to the delay.
+      assert_approx_equals(convolverOutput[maxIndex], delayOutput[maxIndex],
+                           EPSILON, "output at " + maxIndex);
+    });
+}
+
+// The 5/4 ratio provides sampling across a range of delays and offsets within
+// blocks.
+for (var delayIndex = 0;
+     delayIndex < LENGTH;
+     delayIndex = Math.floor((5 * (delayIndex + 1)) / 4)) {
+  promise_test(test_delay_index.bind(null, delayIndex),
+               "Delay " + delayIndex);
+}
+</script>
diff --git a/dom/media/webaudio/test/test_convolverNodeFiniteInfluence.html b/dom/media/webaudio/test/test_convolverNodeFiniteInfluence.html
new file mode 100644
index 0000000000..1cfb51ce8a
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodeFiniteInfluence.html
@@ -0,0 +1,44 @@
+<!DOCTYPE html>
+<title>Test convolution effect has finite duration</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+promise_test(function() {
+
+  const responseLength = 256;
+  // Accept an influence period of twice the responseLength to accept FFT
+  // implementations.
+  const tolerancePeriod = 2 * responseLength;
+  const totalSize = tolerancePeriod + responseLength;
+
+  var context = new OfflineAudioContext(1, totalSize, 48000);
+
+  var responseBuffer =
+    context.createBuffer(1, responseLength, context.sampleRate);
+  var responseChannelData = responseBuffer.getChannelData(0);
+  responseChannelData[0] = 1;
+  responseChannelData[responseLength - 1] = 1;
+  var convolver = context.createConvolver();
+  convolver.buffer = responseBuffer;
+  convolver.connect(context.destination);
+
+  var sourceBuffer = context.createBuffer(1, totalSize, context.sampleRate);
+  sourceBuffer.getChannelData(0)[0] = NaN;
+  var source = context.createBufferSource();
+  source.buffer = sourceBuffer;
+  source.connect(convolver);
+  source.start();
+
+  return context.startRendering().
+    then((buffer) => {
+      var convolverOutput = buffer.getChannelData(0);
+      // There should be no non-zeros after the tolerance period.
+      var testIndex = tolerancePeriod;
+      for (;
+           testIndex < buffer.length - 1 && convolverOutput[testIndex] == 0;
+           ++testIndex) {
+      }
+      assert_equals(convolverOutput[testIndex], 0, "output at " + testIndex);
+    });
+});
+</script>
diff --git a/dom/media/webaudio/test/test_convolverNodeNormalization.html b/dom/media/webaudio/test/test_convolverNodeNormalization.html
new file mode 100644
index 0000000000..24cb7d1670
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodeNormalization.html
@@ -0,0 +1,83 @@
+<!DOCTYPE html>
+<title>Test normalization of convolution buffers</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+// Constants from
+// https://www.w3.org/TR/2015/WD-webaudio-20151208/#widl-ConvolverNode-normalize
+const GainCalibration = 0.00125;
+const GainCalibrationSampleRate = 44100;
+
+const sampleRate = GainCalibrationSampleRate;
+const LENGTH = 12800;
+// tolerate 16-bit math.
+const EPSILON = 1.0 / Math.pow(2, 15);
+
+function test_normalization_via_response_concat(delayIndex)
+{
+  var context = new OfflineAudioContext(1, LENGTH, sampleRate);
+
+  var impulse = context.createBuffer(1, 1, sampleRate);
+  impulse.getChannelData(0)[0] = 1.0;
+  var source = context.createBufferSource();
+  source.buffer = impulse;
+  source.start(0);
+
+  // Construct a set of adjacent responses in such a way that, when each is
+  // convolved with the impulse, they can be merged to produce a constant.
+
+  // The 5/4 ratio provides a range of lengths with different offsets within
+  // blocks.
+  var i = 0;
+  for (var responseEnd = 1;
+       i < LENGTH;
+       responseEnd = Math.floor((5 * responseEnd) / 4) + 1) {
+    var responseBuffer = context.createBuffer(1, responseEnd, sampleRate);
+    var response = responseBuffer.getChannelData(0);
+    var responseStart = i;
+    // The values in the response should be normalized, and so the output
+    // should not be dependent on the value.  Pick a changing value to test
+    // this.
+    var value = responseStart + 1;
+    for (; i < responseEnd; ++i) {
+      response[i] = value;
+    }
+    var convolver = context.createConvolver();
+    convolver.normalize = true;
+    convolver.buffer = responseBuffer;
+    convolver.connect(context.destination);
+    // Undo the normalization calibration by scaling the impulse so as to
+    // expect unit pulse output from the convolver.
+    var gain = context.createGain();
+    gain.gain.value =
+      Math.sqrt((responseEnd - responseStart) / responseEnd) / GainCalibration;
+    gain.connect(convolver);
+    source.connect(gain);
+  }
+
+  return context.startRendering().
+    then((buffer) => {
+      var output = buffer.getChannelData(0);
+      var max = output[0];
+      var maxIndex = 0;
+      var min = max;
+      var minIndex = 0;
+      for (var i = 1; i < buffer.length; ++i) {
+        if (output[i] > max) {
+          max = output[i];
+          maxIndex = i;
+        } else if (output[i] < min) {
+          min = output[i];
+          minIndex = i;
+        }
+      }
+      assert_approx_equals(output[maxIndex], 1.0, EPSILON,
+                           "max output at " + maxIndex);
+      assert_approx_equals(output[minIndex], 1.0, EPSILON,
+                           "min output at " + minIndex);
+    });
+}
+
+promise_test(test_normalization_via_response_concat,
+             "via response concatenation");
+</script>
diff --git a/dom/media/webaudio/test/test_convolverNodeOOM.html b/dom/media/webaudio/test/test_convolverNodeOOM.html
new file mode 100644
index 0000000000..80260267a8
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodeOOM.html
@@ -0,0 +1,46 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ConvolverNode with very large buffer that triggers an OOM</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  skipOfflineContextTests: true,
+  createGraph(context) {
+    var source = context.createOscillator();
+    var convolver = context.createConvolver();
+    // Very big buffer that results in an OOM
+    try {
+      var buffer = context.createBuffer(2, 300000000, context.sampleRate)
+      buffer.getChannelData(0);
+    } catch(e) {
+      // OOM when attempting to create the buffer, this can happen on 32bits
+      // OSes. Simply return here.
+      return convolver;
+    }
+    source.connect(convolver);
+    try {
+      convolver.buffer = buffer;
+    } catch (e) {
+      // This can also OOM.
+      return convolver;
+    }
+    source.start();
+    return convolver;
+  }
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_convolverNodePassThrough.html b/dom/media/webaudio/test/test_convolverNodePassThrough.html
new file mode 100644
index 0000000000..54682aee0c
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodePassThrough.html
@@ -0,0 +1,48 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ConvolverNode with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+
+    var convolver = context.createConvolver();
+
+    source.buffer = this.buffer;
+    convolver.buffer = this.buffer;
+
+    source.connect(convolver);
+
+    var convolverWrapped = SpecialPowers.wrap(convolver);
+    ok("passThrough" in convolverWrapped, "ConvolverNode should support the passThrough API");
+    convolverWrapped.passThrough = true;
+
+    source.start(0);
+    return convolver;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_convolverNodeWithGain.html b/dom/media/webaudio/test/test_convolverNodeWithGain.html
new file mode 100644
index 0000000000..0762f16329
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNodeWithGain.html
@@ -0,0 +1,62 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ConvolverNode after a GainNode - Bug 891254 </title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+const signalLength = 2048;
+const responseLength = 100;
+const outputLength = 4096; // > signalLength + responseLength
+
+var gTest = {
+  length: outputLength,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var buffer = context.createBuffer(1, signalLength, context.sampleRate);
+    for (var i = 0; i < signalLength; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(2 * Math.PI * i / signalLength);
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.start(0);
+
+    var response = context.createBuffer(1, responseLength, context.sampleRate);
+    for (var i = 0; i < responseLength; ++i) {
+      response.getChannelData(0)[i] = i / responseLength;
+    }
+
+    var gain = context.createGain();
+    gain.gain.value = -1;
+    source.connect(gain);
+
+    var convolver1 = context.createConvolver();
+    convolver1.buffer = response;
+    gain.connect(convolver1);
+
+    var convolver2 = context.createConvolver();
+    convolver2.buffer = response;
+    source.connect(convolver2);
+
+    // The output of convolver1 should be the inverse of convolver2, so blend
+    // them together and expect silence.
+    var blend = context.createGain();
+    convolver1.connect(blend);
+    convolver2.connect(blend);
+
+    return blend;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_convolverNode_mono_mono.html b/dom/media/webaudio/test/test_convolverNode_mono_mono.html
new file mode 100644
index 0000000000..f3bac8d5c3
--- /dev/null
+++ b/dom/media/webaudio/test/test_convolverNode_mono_mono.html
@@ -0,0 +1,72 @@
+<!DOCTYPE html>
+
+<html>
+<head>
+<script src="/tests/SimpleTest/SimpleTest.js"></script>
+<script type="text/javascript" src="webaudio.js"></script>
+<script type="text/javascript" src="layouttest-glue.js"></script>
+<script type="text/javascript" src="blink/audio-testing.js"></script>
+<script type="text/javascript" src="blink/convolution-testing.js"></script>
+<link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+
+<body>
+
+<div id="description"></div>
+<div id="console"></div>
+
+<script>
+description("Tests ConvolverNode processing a mono channel with mono impulse response.");
+SimpleTest.waitForExplicitFinish();
+
+// To test the convolver, we convolve two square pulses together to
+// produce a triangular pulse.  To verify the result is correct we
+// check several parts of the result.  First, we make sure the initial
+// part of the result is zero (due to the latency in the convolver).
+// Next, the triangular pulse should match the theoretical result to
+// within some roundoff.  After the triangular pulse, the result
+// should be exactly zero, but round-off prevents that.  We make sure
+// the part after the pulse is sufficiently close to zero.  Finally,
+// the result should be exactly zero because the inputs are exactly
+// zero.
+function runTest() {
+    if (window.testRunner) {
+        window.testRunner.dumpAsText();
+        window.testRunner.waitUntilDone();
+    }
+
+    window.jsTestIsAsync = true;
+
+    // Create offline audio context.
+    var context = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+
+    var squarePulse = createSquarePulseBuffer(context, pulseLengthFrames);
+    var trianglePulse = createTrianglePulseBuffer(context, 2 * pulseLengthFrames);
+
+    var bufferSource = context.createBufferSource();
+    bufferSource.buffer = squarePulse;
+
+    var convolver = context.createConvolver();
+    convolver.normalize = false;
+    convolver.buffer = squarePulse;
+
+    bufferSource.connect(convolver);
+    convolver.connect(context.destination);
+
+    bufferSource.start(0);
+
+    context.oncomplete = checkConvolvedResult(trianglePulse);
+    context.startRendering();
+}
+
+function finishJSTest() {
+  SimpleTest.finish();
+}
+
+runTest();
+
+</script>
+
+<script src="../fast/js/resources/js-test-post.js"></script>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_currentTime.html b/dom/media/webaudio/test/test_currentTime.html
new file mode 100644
index 0000000000..66fdf42653
--- /dev/null
+++ b/dom/media/webaudio/test/test_currentTime.html
@@ -0,0 +1,27 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioContext.currentTime</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("This test needs to wait a while for the AudioContext's timer to start.");
+addLoadEvent(function() {
+  var ac = new AudioContext();
+  is(ac.currentTime, 0, "AudioContext.currentTime should be 0 initially");
+  ac.onstatechange = function () {
+    ok(ac.state == "running", "AudioContext.currentTime should eventually start");
+    ok(ac.currentTime > 0, "AudioContext.currentTime should have increased by now");
+    SimpleTest.finish();
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_decodeAudioDataOnDetachedBuffer.html b/dom/media/webaudio/test/test_decodeAudioDataOnDetachedBuffer.html
new file mode 100644
index 0000000000..e7c6d2db0c
--- /dev/null
+++ b/dom/media/webaudio/test/test_decodeAudioDataOnDetachedBuffer.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+  <meta charset=utf-8>
+<head>
+  <title>Bug 1308434 - Test DecodeAudioData on detached buffer</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script type="text/javascript">
+var testDecodeAudioDataOnDetachedBuffer = function(buffer) {
+  var context = new AudioContext();
+
+  // make the buffer detached
+  context.decodeAudioData(buffer);
+
+  // check that the buffer is detached
+  is(buffer.byteLength, 0, "Buffer should be detached");
+
+  // call decodeAudioData on detached buffer
+  context.decodeAudioData(buffer).then(function(b) {
+    ok(false, "We should not be able to decode the detached buffer but we do");
+    SimpleTest.finish();
+  }, function(r) {
+    SimpleTest.isa(r, TypeError);
+    is(r.message, "BaseAudioContext.decodeAudioData: Buffer argument can't be a detached buffer", "Incorrect message");
+    SimpleTest.finish();
+  });
+};
+
+var filename = "small-shot.mp3";
+
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  var xhr = new XMLHttpRequest();
+  xhr.open("GET", filename);
+  xhr.responseType = "arraybuffer";
+
+  xhr.onload = function() {
+    testDecodeAudioDataOnDetachedBuffer(xhr.response);
+  };
+
+  xhr.send();
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_decodeAudioDataPromise.html b/dom/media/webaudio/test/test_decodeAudioDataPromise.html
new file mode 100644
index 0000000000..139a686db1
--- /dev/null
+++ b/dom/media/webaudio/test/test_decodeAudioDataPromise.html
@@ -0,0 +1,62 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the decodeAudioData API with Promise</title>
+
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  <script src="webaudio.js"></script>
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+var finished = 0;
+
+function finish() {
+  if (++finished == 2) {
+    SimpleTest.finish();
+  }
+}
+
+var ac = new AudioContext();
+// Test that a the promise is rejected with an invalid source buffer.
+expectNoException(function() {
+  var p = ac.decodeAudioData(" ");
+  ok(p instanceof Promise, "AudioContext.decodeAudioData should return a Promise");
+  p.then(function(data) {
+    ok(false, "Promise should not resolve with an invalid source buffer.");
+    finish();
+  }).catch(function(e) {
+    ok(true, "Promise should be rejected with an invalid source buffer.");
+    ok(e.name == "TypeError", "The error should be TypeError");
+    finish();
+  })
+});
+
+// Test that a the promise is resolved with a valid source buffer.
+var xhr = new XMLHttpRequest();
+xhr.open("GET", "ting-44.1k-1ch.ogg", true);
+xhr.responseType = "arraybuffer";
+xhr.onload = function() {
+  var p = ac.decodeAudioData(xhr.response);
+  ok(p instanceof Promise, "AudioContext.decodeAudioData should return a Promise");
+  p.then(function(data) {
+    ok(data instanceof AudioBuffer, "Promise should resolve, passing an AudioBuffer");
+    ok(true, "Promise should resolve with a valid source buffer.");
+    finish();
+  }).catch(function() {
+    ok(false, "Promise should not be rejected with a valid source buffer.");
+    finish();
+  });
+};
+xhr.send();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_decodeAudioError.html b/dom/media/webaudio/test/test_decodeAudioError.html
new file mode 100644
index 0000000000..f18b971ac4
--- /dev/null
+++ b/dom/media/webaudio/test/test_decodeAudioError.html
@@ -0,0 +1,74 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the decodeAudioData Errors</title>
+
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  <script src="webaudio.js"></script>
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+var finished = 0;
+
+var ctx = new AudioContext();
+
+function errorExpectedWithFile(file, errorMsg) {
+  var xhr = new XMLHttpRequest();
+  function test(e) {
+    ok(e instanceof DOMException,
+      "The exception should be an instance of DOMException");
+    ok(e.name == "EncodingError",
+      "The exception name should be EncodingError");
+    ok(e.message == errorMsg,
+       "The exception message is not the one intended.\n" +
+       "\tExpected : " + errorMsg + "\n" +
+       "\tGot      : " + e.message );
+    finish();
+  }
+  xhr.open("GET", file, true);
+  xhr.responseType = "arraybuffer";
+  xhr.onload = function() {
+    ctx.decodeAudioData(xhr.response, buffer => {
+      ok(false, "You should not be able to decode that");
+      finish();
+    }, e => test(e))
+    .then(buffer => {
+      ok(false, "You should not be able to decode that");
+      finish();
+    })
+    .catch(e => test(e));
+  };
+  xhr.send();
+}
+
+function finish() {
+  if (++finished == 4) {
+    SimpleTest.finish();
+  }
+}
+
+// Unknown Content
+errorExpectedWithFile("404", "The buffer passed to decodeAudioData contains an unknown content type.");
+
+// Invalid Content
+errorExpectedWithFile("invalidContent.flac", "The buffer passed to decodeAudioData contains invalid content which cannot be decoded successfully.");
+
+// No Audio
+// # Bug 1656032
+// Think about increasing the finish counter to 6 when activating this line
+// errorExpectedWithFile("noaudio.webm", "The buffer passed to decodeAudioData does not contain any audio.");
+
+// Unknown Error
+// errorExpectedWithFile("There is no file we can't handle", "An unknown error occurred while processing decodeAudioData.");
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_decodeMultichannel.html b/dom/media/webaudio/test/test_decodeMultichannel.html
new file mode 100644
index 0000000000..a799c641ee
--- /dev/null
+++ b/dom/media/webaudio/test/test_decodeMultichannel.html
@@ -0,0 +1,75 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset=utf-8>
+<head>
+  <title>Test that we can decode multichannel file with webaudio and &lt;audio&gt;</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+var testcases = [
+  {
+    filename: "audio-quad.wav",
+    channels: 4
+  },
+  {
+    filename: "8kHz-320kbps-6ch.aac",
+    channels: 6
+  }
+];
+
+SimpleTest.waitForExplicitFinish();
+
+function decodeUsingAudioElement(filename, resolve) {
+  var a = new Audio();
+  a.addEventListener("error", function() {
+    ok(false, "Error loading metadata");
+    resolve();
+  });
+  a.addEventListener("loadedmetadata", function() {
+    ok(true, "Metadata Loaded");
+    resolve();
+  });
+
+  a.src = filename;
+  a.load();
+}
+
+function testOne({filename, channels}) {
+  return new Promise(resolve => {
+    var xhr = new XMLHttpRequest();
+    xhr.open("GET", filename);
+    xhr.responseType = "arraybuffer";
+    xhr.onload = function() {
+      var context = new AudioContext();
+      context.decodeAudioData(xhr.response, function(b) {
+        ok(true, "Decoding of a wave file with four channels succeded.");
+        is(b.numberOfChannels,
+           channels,
+           `The AudioBuffer decoded from ${filename} should have ${channels} channels.`);
+        decodeUsingAudioElement(filename, resolve);
+      }, function() {
+        ok(false, `Decoding ${filename} failed)`);
+        decodeUsingAudioElement(filename, resolve);
+      });
+    };
+    xhr.send(null);
+  });
+}
+
+async function runTest() {
+  for (var testcase of testcases) {
+    await testOne(testcase);
+  }
+
+  SimpleTest.finish();
+}
+
+addLoadEvent(runTest);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_decodeOpusTail.html b/dom/media/webaudio/test/test_decodeOpusTail.html
new file mode 100644
index 0000000000..451b2b6a23
--- /dev/null
+++ b/dom/media/webaudio/test/test_decodeOpusTail.html
@@ -0,0 +1,28 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Regression test to check that opus files don't have a tail at the end.</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+// This gets a 1 second Opus file and decodes it to a buffer. The opus file is
+// decoded at 48kHz, and the OfflineAudioContext is also at 48kHz, no resampling
+// is taking place.
+fetch('sweep-300-330-1sec.opus')
+.then(function(response) { return response.arrayBuffer(); })
+.then(function(buffer) {
+  var off = new OfflineAudioContext(1, 128, 48000);
+  off.decodeAudioData(buffer, function(decoded) {
+    var pcm = decoded.getChannelData(0);
+    is(pcm.length, 48000, "The length of the decoded file is correct.");
+    SimpleTest.finish();
+  });
+});
+
+</script>
diff --git a/dom/media/webaudio/test/test_decoderDelay.html b/dom/media/webaudio/test/test_decoderDelay.html
new file mode 100644
index 0000000000..4ed379dec5
--- /dev/null
+++ b/dom/media/webaudio/test/test_decoderDelay.html
@@ -0,0 +1,154 @@
+<!DOCTYPE html>
+<html>
+<head>
+  <meta charset="utf-8" />
+  <title>Test that decoder delay is handled</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+  <script class="testbody" type="text/javascript">
+    SimpleTest.waitForExplicitFinish();
+    const {AppConstants} = SpecialPowers.ChromeUtils.importESModule(
+      "resource://gre/modules/AppConstants.sys.mjs"
+    );
+
+    var tests_half_a_second = [
+      "half-a-second-1ch-44100-aac.mp4",
+      "half-a-second-1ch-44100-flac.flac",
+      "half-a-second-1ch-44100-libmp3lame.mp3",
+      "half-a-second-1ch-44100-libopus.opus",
+      "half-a-second-1ch-44100-libopus.webm",
+      "half-a-second-1ch-44100-libvorbis.ogg",
+      "half-a-second-1ch-44100.wav",
+      "half-a-second-1ch-48000-aac.mp4",
+      "half-a-second-1ch-48000-flac.flac",
+      "half-a-second-1ch-48000-libmp3lame.mp3",
+      "half-a-second-1ch-48000-libopus.opus",
+      "half-a-second-1ch-48000-libopus.webm",
+      "half-a-second-1ch-48000-libvorbis.ogg",
+      "half-a-second-1ch-48000.wav",
+      "half-a-second-2ch-44100-aac.mp4",
+      "half-a-second-2ch-44100-flac.flac",
+      "half-a-second-2ch-44100-libmp3lame.mp3",
+      "half-a-second-2ch-44100-libopus.opus",
+      "half-a-second-2ch-44100-libopus.webm",
+      "half-a-second-2ch-44100-libvorbis.ogg",
+      "half-a-second-2ch-44100.wav",
+      "half-a-second-2ch-48000-aac.mp4",
+      "half-a-second-2ch-48000-flac.flac",
+      "half-a-second-2ch-48000-libmp3lame.mp3",
+      "half-a-second-2ch-48000-libopus.opus",
+      "half-a-second-2ch-48000-libopus.webm",
+      "half-a-second-2ch-48000-libvorbis.ogg",
+      "half-a-second-2ch-48000.wav",
+    ];
+
+    // Those files are almost exactly half a second, but don't have enough pre-roll/padding
+    // information in the container, or the container isn't parsed properly, so
+    // aren't trimmed appropriately.
+    // vorbis webm, opus mp4, aac adts
+    var tests_adts = [
+      "half-a-second-1ch-44100-aac.aac",
+      "half-a-second-1ch-44100-libopus.mp4",
+      "half-a-second-1ch-44100-libvorbis.webm",
+      "half-a-second-1ch-48000-aac.aac",
+      "half-a-second-1ch-48000-libopus.mp4",
+      "half-a-second-1ch-48000-libvorbis.webm",
+      "half-a-second-2ch-44100-aac.aac",
+      "half-a-second-2ch-44100-libopus.mp4",
+      "half-a-second-2ch-44100-libvorbis.webm",
+      "half-a-second-2ch-48000-aac.aac",
+      "half-a-second-2ch-48000-libopus.mp4",
+      "half-a-second-2ch-48000-libvorbis.webm",
+    ];
+
+    // Other files that have interesting characteristics.
+    var tests_others = [
+      {
+        // Very short VBR file, 16 frames of audio at 44100. Padding spanning two
+        // packets.
+        "path": "sixteen-frames.mp3",
+        "frameCount": 16,
+        "samplerate": 44100,
+        "fuzz": {}
+      },
+      {
+        // This is incorrect (the duration should be 0.5s exactly)
+        // This is tracked in https://github.com/mozilla/mp4parse-rust/issues/404
+        "path":"half-a-second-1ch-44100-aac-afconvert.mp4",
+        "frameCount": 22464,
+        "samplerate": 44100,
+        "fuzz": {
+          "android": 2
+        }
+      },
+      {
+        // Bug 1856145 - Invalid OGG file with busted granulepos in the
+        // bytestrem, but we should be able to recover and play it properly.
+        // This triggers a bug in the decoder delay trimming logic.
+        "path": "1856145.ogg",
+        "samplerate": 8000,
+        "frameCount": 8192,
+        "fuzz" : {}
+      }
+    ];
+
+    var all_tests = [tests_half_a_second, tests_adts, tests_others].flat();
+
+    var count = 0;
+    function checkDone() {
+      if (++count == all_tests.length) {
+        SimpleTest.finish();
+      }
+    }
+
+    async function doit() {
+      var context = new OfflineAudioContext(1, 128, 48000);
+      tests_half_a_second.forEach(async testfile => {
+        var response = await fetch(testfile);
+        var buffer = await response.arrayBuffer();
+        var decoded = await context.decodeAudioData(buffer);
+        is(
+          decoded.duration,
+          0.5,
+          "The file " + testfile + " is half a second."
+        );
+        // Value found empirically after looking at the files. The initial
+        // amplitude should be 0 at phase 0 because those files are sine wave.
+        // The compression is sometimes lossy and the first sample is not always
+        // exactly 0.0.
+        ok(
+          Math.abs(decoded.getChannelData(0)[0]) <= 0.022,
+          `The start point for ${testfile} is correct ${ decoded.getChannelData(0)[0] }`
+        );
+        checkDone();
+      });
+      tests_adts.forEach(async testfile => {
+        var response = await fetch(testfile);
+        var buffer = await response.arrayBuffer();
+        var decoded = await context.decodeAudioData(buffer);
+        // Value found empirically after looking at the files. ADTS containers
+        // don't have encoder delay / padding info so we can't trim correctly.
+        ok(
+          Math.abs(decoded.duration - 0.5) < 0.02,
+          `The ADTS file ${testfile} is about half a second (${decoded.duration}, error: ${Math.abs(decoded.duration-0.5)}).`
+        );
+        checkDone();
+      });
+      tests_others.forEach(async test => {
+        // Get an context at a specific rate to avoid duration changes due to resampling.
+        var contextAtRate = new OfflineAudioContext(1, 128, test.samplerate);
+        var response = await fetch(test.path);
+        var buffer = await response.arrayBuffer();
+        var decoded = await contextAtRate.decodeAudioData(buffer);
+        const fuzz = test.fuzz[AppConstants.platform] ?? 0;
+        ok(Math.abs(decoded.length - test.frameCount) <= fuzz, `${test.path} is ${decoded.length} frames long`);
+        checkDone();
+      });
+    }
+
+    doit();
+  </script>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNode.html b/dom/media/webaudio/test/test_delayNode.html
new file mode 100644
index 0000000000..89172aa86f
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNode.html
@@ -0,0 +1,101 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DelayNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var delay = new DelayNode(context);
+    ok(delay.delayTime, "The audioparam member must exist");
+    is(delay.delayTime.value, 0, "Correct initial value");
+    is(delay.delayTime.defaultValue, 0, "Correct default value");
+    delay.delayTime.value = 0.5;
+    is(delay.delayTime.value, 0.5, "Correct initial value");
+    is(delay.delayTime.defaultValue, 0, "Correct default value");
+
+    delay = new DelayNode(context, { delayTime: 0.5 });
+    ok(delay.delayTime, "The audioparam member must exist");
+    is(delay.delayTime.value, 0.5, "Correct initial value");
+    is(delay.delayTime.defaultValue, 0, "Correct default value");
+
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+
+    delay = context.createDelay();
+
+    source.buffer = buffer;
+
+    source.connect(delay);
+
+    ok(delay.delayTime, "The audioparam member must exist");
+    is(delay.delayTime.value, 0, "Correct initial value");
+    is(delay.delayTime.defaultValue, 0, "Correct default value");
+    delay.delayTime.value = 0.5;
+    is(delay.delayTime.value, 0.5, "Correct initial value");
+    is(delay.delayTime.defaultValue, 0, "Correct default value");
+    is(delay.channelCount, 2, "delay node has 2 input channels by default");
+    is(delay.channelCountMode, "max", "Correct channelCountMode for the delay node");
+    is(delay.channelInterpretation, "speakers", "Correct channelCountInterpretation for the delay node");
+
+    expectException(function() {
+      context.createDelay(0);
+    }, DOMException.NOT_SUPPORTED_ERR);
+    expectException(function() {
+      context.createDelay(180);
+    }, DOMException.NOT_SUPPORTED_ERR);
+    expectTypeError(function() {
+      context.createDelay(NaN);
+    }, DOMException.NOT_SUPPORTED_ERR);
+    expectException(function() {
+      context.createDelay(-1);
+    }, DOMException.NOT_SUPPORTED_ERR);
+
+    expectException(function() {
+      new DelayNode(context, { maxDelayTime: 0 });
+    }, DOMException.NOT_SUPPORTED_ERR);
+    expectException(function() {
+      new DelayNode(context, { maxDelayTime: 180 });
+    }, DOMException.NOT_SUPPORTED_ERR);
+    expectTypeError(function() {
+      new DelayNode(context, { maxDelayTime: NaN });
+    }, DOMException.NOT_SUPPORTED_ERR);
+    expectException(function() {
+      new DelayNode(context, { maxDelayTime: -1 });
+    }, DOMException.NOT_SUPPORTED_ERR);
+
+    context.createDelay(1); // should not throw
+
+    // Delay the source stream by 2048 frames
+    delay.delayTime.value = 2048 / context.sampleRate;
+
+    source.start(0);
+    return delay;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048 * 2, context.sampleRate);
+    for (var i = 2048; i < 2048 * 2; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * (i - 2048) / context.sampleRate);
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeAtMax.html b/dom/media/webaudio/test/test_delayNodeAtMax.html
new file mode 100644
index 0000000000..3d0afba0ac
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeAtMax.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DelayNode with maxDelayTime delay - bug 890528</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+const signalLength = 2048;
+const delayLength = 1000; // Not on a block boundary
+const outputLength = 4096 // > signalLength + 2 * delayLength;
+
+function applySignal(buffer, offset) {
+  for (var i = 0; i < signalLength; ++i) {
+    buffer.getChannelData(0)[offset + i] = Math.cos(Math.PI * i / signalLength);
+  }
+}
+
+var gTest = {
+  numberOfChannels: 1,
+  createGraph(context) {
+    var buffer = context.createBuffer(1, signalLength, context.sampleRate);
+    applySignal(buffer, 0);
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    const delayTime = delayLength / context.sampleRate;
+    var delay = context.createDelay(delayTime);
+    delay.delayTime.value = delayTime;
+
+    source.connect(delay);
+
+    source.start(0);
+    return delay;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, outputLength, context.sampleRate);
+    applySignal(expectedBuffer, delayLength);
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeChannelChanges.html b/dom/media/webaudio/test/test_delayNodeChannelChanges.html
new file mode 100644
index 0000000000..c58d81f467
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeChannelChanges.html
@@ -0,0 +1,98 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>test DelayNode channel count changes</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestCompleteLog();
+
+const bufferSize = 4096;
+
+var ctx;
+var testDelay;
+var stereoDelay;
+var invertor;
+
+function compareOutputs(callback) {
+  var processor = ctx.createScriptProcessor(bufferSize, 2, 0);
+  testDelay.connect(processor);
+  invertor.connect(processor);
+  processor.onaudioprocess =
+    function(e) {
+      compareBuffers(e.inputBuffer,
+                     ctx.createBuffer(2, bufferSize, ctx.sampleRate));
+      e.target.onaudioprocess = null;
+      callback();
+    }
+}
+
+function startTest() {
+  // And a two-channel signal
+  var merger = ctx.createChannelMerger();
+  merger.connect(testDelay);
+  merger.connect(stereoDelay);
+  var oscL = ctx.createOscillator();
+  oscL.connect(merger, 0, 0);
+  oscL.start(0);
+  var oscR = ctx.createOscillator();
+  oscR.type = "sawtooth";
+  oscR.connect(merger, 0, 1);
+  oscR.start(0);
+
+  compareOutputs(
+    function () {
+      // Disconnect the two-channel signal and test again
+      merger.disconnect();
+      compareOutputs(SimpleTest.finish);
+    });
+}
+
+function prepareTest() {
+  ctx = new AudioContext();
+
+  // The output of a test delay node with mono and stereo input will be
+  // compared with that of separate mono and stereo delay nodes.
+  const delayTime = 0.3 * bufferSize / ctx.sampleRate;
+  testDelay = ctx.createDelay(delayTime);
+  testDelay.delayTime.value = delayTime;
+  let monoDelay = ctx.createDelay(delayTime);
+  monoDelay.delayTime.value = delayTime;
+  stereoDelay = ctx.createDelay(delayTime);
+  stereoDelay.delayTime.value = delayTime;
+
+  // Create a one-channel signal and connect to the delay nodes
+  var monoOsc = ctx.createOscillator();
+  monoOsc.frequency.value = 110;
+  monoOsc.connect(testDelay);
+  monoOsc.connect(monoDelay);
+  monoOsc.start(0);
+
+  // Invert the expected so that mixing with the test will find the difference.
+  invertor = ctx.createGain();
+  invertor.gain.value = -1.0;
+  monoDelay.connect(invertor);
+  stereoDelay.connect(invertor);
+
+  // Start the test after the delay nodes have begun processing.
+  var processor = ctx.createScriptProcessor(bufferSize, 1, 0);
+  processor.connect(ctx.destination);
+
+  processor.onaudioprocess =
+    function(e) {
+      e.target.onaudioprocess = null;
+      processor.disconnect();
+      startTest();
+    };
+}
+prepareTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeCycles.html b/dom/media/webaudio/test/test_delayNodeCycles.html
new file mode 100644
index 0000000000..fb4b6c5202
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeCycles.html
@@ -0,0 +1,156 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the support of cycles.</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+const sampleRate = 48000;
+const inputLength = 2048;
+
+addLoadEvent(function() {
+  function addSine(b) {
+    for (var i = 0; i < b.length; i++) {
+      b[i] += Math.sin(440 * 2 * Math.PI * i / sampleRate);
+    }
+  }
+
+  function getSineBuffer(ctx) {
+    var buffer = ctx.createBuffer(1, inputLength, ctx.sampleRate);
+    addSine(buffer.getChannelData(0));
+    return buffer;
+  }
+
+  function createAndPlayWithCycleAndDelayNode(ctx, delayFrames) {
+    var source = ctx.createBufferSource();
+    source.buffer = getSineBuffer(ctx);
+
+    var gain = ctx.createGain();
+    var delay = ctx.createDelay();
+    delay.delayTime.value = delayFrames/ctx.sampleRate;
+
+    source.connect(gain);
+    gain.connect(delay);
+    delay.connect(ctx.destination);
+    // cycle
+    delay.connect(gain);
+
+    source.start(0);
+  }
+
+  function createAndPlayWithCycleAndNoDelayNode(ctx) {
+    var source = ctx.createBufferSource();
+    source.loop = true;
+    source.buffer = getSineBuffer(ctx);
+
+    var gain = ctx.createGain();
+    var gain2 = ctx.createGain();
+
+    source.connect(gain);
+    gain.connect(gain2);
+    // cycle
+    gain2.connect(gain);
+    gain2.connect(ctx.destination);
+
+    source.start(0);
+  }
+
+  function createAndPlayWithCycleAndNoDelayNodeInCycle(ctx) {
+    var source = ctx.createBufferSource();
+    source.loop = true;
+    source.buffer = getSineBuffer(ctx);
+
+    var delay = ctx.createDelay();
+    var gain = ctx.createGain();
+    var gain2 = ctx.createGain();
+
+    // Their is a cycle, a delay, but the delay is not in the cycle.
+    source.connect(delay);
+    delay.connect(gain);
+    gain.connect(gain2);
+    // cycle
+    gain2.connect(gain);
+    gain2.connect(ctx.destination);
+
+    source.start(0);
+  }
+
+  var remainingTests = 0;
+  function finish() {
+    if (--remainingTests == 0) {
+      SimpleTest.finish();
+    }
+  }
+
+  function getOfflineContext(oncomplete) {
+    var ctx = new OfflineAudioContext(1, sampleRate, sampleRate);
+    ctx.oncomplete = oncomplete;
+    return ctx;
+  }
+
+  function checkSilentBuffer(e) {
+    var buffer = e.renderedBuffer.getChannelData(0);
+    for (var i = 0; i < buffer.length; i++) {
+      if (buffer[i] != 0.0) {
+        ok(false, "buffer should be silent.");
+        finish();
+        return;
+      }
+    }
+    ok(true, "buffer should be silent.");
+    finish();
+  }
+
+  function checkNoisyBuffer(e, aDelayFrames) {
+    let delayFrames = Math.max(128, aDelayFrames);
+
+    var expected = new Float32Array(e.renderedBuffer.length);
+    for (var i = delayFrames; i < expected.length; i += delayFrames) {
+      addSine(expected.subarray(i, i + inputLength));
+    }
+
+    compareChannels(e.renderedBuffer.getChannelData(0), expected);
+    finish();
+  }
+
+  function expectSilentOutput(f) {
+    remainingTests++;
+    var ctx = getOfflineContext(checkSilentBuffer);
+    f(ctx);
+    ctx.startRendering();
+  }
+
+  function expectNoisyOutput(delayFrames) {
+    remainingTests++;
+    var ctx = getOfflineContext();
+    ctx.oncomplete = function(e) { checkNoisyBuffer(e, delayFrames); };
+    createAndPlayWithCycleAndDelayNode(ctx, delayFrames);
+    ctx.startRendering();
+  }
+
+  // This is trying to make a graph with a cycle and no DelayNode in the graph.
+  // The cycle subgraph should be muted, in this graph the output should be silent.
+  expectSilentOutput(createAndPlayWithCycleAndNoDelayNode);
+  // This is trying to make a graph with a cycle and a DelayNode in the graph, but
+  // not part of the cycle.
+  // The cycle subgraph should be muted, in this graph the output should be silent.
+  expectSilentOutput(createAndPlayWithCycleAndNoDelayNodeInCycle);
+  // Those are making legal graphs, with at least one DelayNode in the cycle.
+  // There should be some non-silent output.
+  expectNoisyOutput(sampleRate/4);
+  // DelayNode.delayTime will be clamped to 128/ctx.sampleRate.
+  // There should be some non-silent output.
+  expectNoisyOutput(0);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodePassThrough.html b/dom/media/webaudio/test/test_delayNodePassThrough.html
new file mode 100644
index 0000000000..0c2d1db30a
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodePassThrough.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DelayNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+
+    var delay = context.createDelay();
+
+    source.buffer = this.buffer;
+
+    source.connect(delay);
+
+    delay.delayTime.value = 0.5;
+
+    // Delay the source stream by 2048 frames
+    delay.delayTime.value = 2048 / context.sampleRate;
+
+    var delayWrapped = SpecialPowers.wrap(delay);
+    ok("passThrough" in delayWrapped, "DelayNode should support the passThrough API");
+    delayWrapped.passThrough = true;
+
+    source.start(0);
+    return delay;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    var silence = context.createBuffer(1, 2048, context.sampleRate);
+
+    return [this.buffer, silence];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeSmallMaxDelay.html b/dom/media/webaudio/test/test_delayNodeSmallMaxDelay.html
new file mode 100644
index 0000000000..235a55a4c8
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeSmallMaxDelay.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DelayNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+
+    var delay = context.createDelay(0.02);
+
+    source.buffer = this.buffer;
+
+    source.connect(delay);
+
+    source.start(0);
+    return delay;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    this.buffer = expectedBuffer;
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeTailIncrease.html b/dom/media/webaudio/test/test_delayNodeTailIncrease.html
new file mode 100644
index 0000000000..a511a4a3ff
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeTailIncrease.html
@@ -0,0 +1,71 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test increasing delay of DelayNode after input finishes</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+const signalLength = 100;
+const bufferSize = 1024;
+// Delay should be long enough to allow CC to run
+const delayBufferCount = 50;
+const delayLength = delayBufferCount * bufferSize + 700;
+
+var count = 0;
+
+function applySignal(buffer, offset) {
+  for (var i = 0; i < signalLength; ++i) {
+    buffer.getChannelData(0)[offset + i] = Math.cos(Math.PI * i / signalLength);
+  }
+}
+
+function onAudioProcess(e) {
+  switch(count) {
+  case 5:
+    SpecialPowers.forceGC();
+    SpecialPowers.forceCC();
+    break;
+  case delayBufferCount:
+    var offset = delayLength - count * bufferSize;
+    var ctx = e.target.context;
+    var expected = ctx.createBuffer(1, bufferSize, ctx.sampleRate);
+    applySignal(expected, offset);
+    compareBuffers(e.inputBuffer, expected);
+    SimpleTest.finish();
+  }
+  count++;
+}
+
+function startTest() {
+  var ctx = new AudioContext();
+  var processor = ctx.createScriptProcessor(bufferSize, 1, 0);
+  processor.onaudioprocess = onAudioProcess;
+
+  // Switch on delay at a time in the future.
+  var delayDuration = delayLength / ctx.sampleRate;
+  var delayStartTime = (delayLength - bufferSize) / ctx.sampleRate;
+  var delay = ctx.createDelay(delayDuration);
+  delay.delayTime.setValueAtTime(delayDuration, delayStartTime);
+  delay.connect(processor);
+
+  // Short signal that finishes before switching to long delay
+  var buffer = ctx.createBuffer(1, signalLength, ctx.sampleRate);
+  applySignal(buffer, 0);
+  var source = ctx.createBufferSource();
+  source.buffer = buffer;
+  source.start();
+  source.connect(delay);
+};
+
+startTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeTailWithDisconnect.html b/dom/media/webaudio/test/test_delayNodeTailWithDisconnect.html
new file mode 100644
index 0000000000..c6723f643d
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeTailWithDisconnect.html
@@ -0,0 +1,95 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test tail time lifetime of DelayNode after input is disconnected</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// Web Audio doesn't provide a means to precisely time disconnect()s but we
+// can test that the output of delay nodes matches the output from their
+// sources before they are disconnected.
+
+SimpleTest.waitForExplicitFinish();
+
+const signalLength = 128;
+const bufferSize = 4096;
+const sourceCount = bufferSize / signalLength;
+// Delay should be long enough to allow CC to run
+var delayBufferCount = 20;
+const delayLength = delayBufferCount * bufferSize;
+
+var sourceOutput = new Float32Array(bufferSize);
+var delayOutputCount = 0;
+var sources = [];
+
+function onDelayOutput(e) {
+  if (delayOutputCount < delayBufferCount) {
+    delayOutputCount++;
+    return;
+  }
+
+  compareChannels(e.inputBuffer.getChannelData(0), sourceOutput);
+  e.target.onaudioprocess = null;
+  SimpleTest.finish();
+}
+
+function onSourceOutput(e) {
+  // Record the first buffer
+  e.inputBuffer.copyFromChannel(sourceOutput, 0);
+  e.target.onaudioprocess = null;
+}
+
+function disconnectSources() {
+  for (var i = 0; i < sourceCount; ++i) {
+    sources[i].disconnect();
+  }
+
+  SpecialPowers.forceGC();
+  SpecialPowers.forceCC();
+}
+
+function startTest() {
+  var ctx = new AudioContext();
+
+  var sourceProcessor = ctx.createScriptProcessor(bufferSize, 1, 0);
+  sourceProcessor.onaudioprocess = onSourceOutput;
+  // Keep audioprocess events going after source disconnect.
+  sourceProcessor.connect(ctx.destination);
+
+  var delayProcessor = ctx.createScriptProcessor(bufferSize, 1, 0);
+  delayProcessor.onaudioprocess = onDelayOutput;
+
+  var delayDuration = delayLength / ctx.sampleRate;
+  for (var i = 0; i < sourceCount; ++i) {
+    var delay = ctx.createDelay(delayDuration);
+    delay.delayTime.value = delayDuration;
+    delay.connect(delayProcessor);
+
+    var source = ctx.createOscillator();
+    source.frequency.value = 440 + 10 * i
+    source.start(i * signalLength / ctx.sampleRate);
+    source.stop((i + 1) * signalLength / ctx.sampleRate);
+    source.connect(delay);
+    source.connect(sourceProcessor);
+
+    sources[i] = source;
+  }
+
+  // Assuming the above Web Audio operations have already scheduled an event
+  // to run in stable state and start the graph thread, schedule a subsequent
+  // event to disconnect the sources, which will remove main thread connection
+  // references before it knows the graph thread has started using the source
+  // streams.
+  SimpleTest.executeSoon(disconnectSources);
+};
+
+startTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeTailWithGain.html b/dom/media/webaudio/test/test_delayNodeTailWithGain.html
new file mode 100644
index 0000000000..60cca276c0
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeTailWithGain.html
@@ -0,0 +1,72 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test tail time lifetime of DelayNode indirectly connected to source</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+const signalLength = 130;
+const bufferSize = 1024;
+// Delay should be long enough to allow CC to run
+const delayBufferCount = 50;
+const delayLength = delayBufferCount * bufferSize + 700;
+
+var count = 0;
+
+function applySignal(buffer, offset) {
+  for (var i = 0; i < signalLength; ++i) {
+    buffer.getChannelData(0)[offset + i] = Math.cos(Math.PI * i / signalLength);
+  }
+}
+
+function onAudioProcess(e) {
+  switch(count) {
+  case 5:
+    SpecialPowers.forceGC();
+    SpecialPowers.forceCC();
+    break;
+  case delayBufferCount:
+    var offset = delayLength - count * bufferSize;
+    var ctx = e.target.context;
+    var expected = ctx.createBuffer(1, bufferSize, ctx.sampleRate);
+    applySignal(expected, offset);
+    compareBuffers(e.inputBuffer, expected);
+    SimpleTest.finish();
+  }
+  count++;
+}
+
+function startTest() {
+  var ctx = new AudioContext();
+  var processor = ctx.createScriptProcessor(bufferSize, 1, 0);
+  processor.onaudioprocess = onAudioProcess;
+
+  var delayDuration = delayLength / ctx.sampleRate;
+  var delay = ctx.createDelay(delayDuration);
+  delay.delayTime.value = delayDuration;
+  delay.connect(processor);
+
+  var gain = ctx.createGain();
+  gain.connect(delay);
+
+  // Short signal that finishes before garbage collection
+  var buffer = ctx.createBuffer(1, signalLength, ctx.sampleRate);
+  applySignal(buffer, 0);
+  var source = ctx.createBufferSource();
+  source.buffer = buffer;
+  source.start();
+  source.connect(gain);
+};
+
+startTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeTailWithReconnect.html b/dom/media/webaudio/test/test_delayNodeTailWithReconnect.html
new file mode 100644
index 0000000000..da5f02b052
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeTailWithReconnect.html
@@ -0,0 +1,136 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test tail time lifetime of DelayNode after input finishes and new input added</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+// The buffer source will start on a block boundary, so keeping the signal
+// within one block ensures that it will not cross AudioProcessingEvent buffer
+// boundaries.
+const signalLength = 128;
+const bufferSize = 1024;
+// Delay should be long enough to allow CC to run
+var delayBufferCount = 50;
+var delayBufferOffset;
+const delayLength = delayBufferCount * bufferSize;
+
+var phase = "initial";
+var sourceCount = 0;
+var delayCount = 0;
+var oscillator;
+var delay;
+var source;
+
+function applySignal(buffer, offset) {
+  for (var i = 0; i < signalLength; ++i) {
+    buffer.getChannelData(0)[offset + i] = Math.cos(Math.PI * i / signalLength);
+  }
+}
+
+function bufferIsSilent(buffer, out) {
+  for (var i = 0; i < buffer.length; ++i) {
+    if (buffer.getChannelData(0)[i] != 0) {
+      if (out) {
+        out.soundOffset = i;
+      }
+      return false;
+    }
+  }
+  return true;
+}
+
+function onDelayOutput(e) {
+  switch(phase) {
+
+  case "initial":
+    // Wait for oscillator sound to exit delay
+    if (bufferIsSilent(e.inputBuffer))
+      break;
+
+    phase = "played oscillator";
+    break;
+
+  case "played oscillator":
+    // First tail time has expired.  Start second source and remove references
+    // to the delay and connected second source.
+    oscillator.disconnect();
+    source.connect(delay);
+    source.start();
+    source = null;
+    delay = null;
+    phase = "started second source";
+    break;
+
+  case "second tail time":
+    if (delayCount == delayBufferCount) {
+      var ctx = e.target.context;
+      var expected = ctx.createBuffer(1, bufferSize, ctx.sampleRate);
+      applySignal(expected, delayBufferOffset);
+      compareBuffers(e.inputBuffer, expected);
+      e.target.onaudioprocess = null;
+      SimpleTest.finish();
+    }
+  }
+
+  delayCount++;
+}
+
+function onSourceOutput(e) {
+  switch(phase) {
+  case "started second source":
+    var out = {};
+    if (!bufferIsSilent(e.inputBuffer, out)) {
+      delayBufferCount += sourceCount;
+      delayBufferOffset = out.soundOffset;
+      phase = "played second source";
+    }
+    break;
+  case "played second source":
+    SpecialPowers.forceGC();
+    SpecialPowers.forceCC();
+    phase = "second tail time";
+    e.target.onaudioprocess = null;
+  }
+
+  sourceCount++;
+}
+
+function startTest() {
+  var ctx = new AudioContext();
+  var delayDuration = delayLength / ctx.sampleRate;
+  delay = ctx.createDelay(delayDuration);
+  delay.delayTime.value = delayDuration;
+  var processor1 = ctx.createScriptProcessor(bufferSize, 1, 0);
+  delay.connect(processor1);
+  processor1.onaudioprocess = onDelayOutput;
+
+  // Signal to trigger initial tail time reference
+  oscillator = ctx.createOscillator();
+  oscillator.start(0);
+  oscillator.stop(100/ctx.sampleRate);
+  oscillator.connect(delay);
+
+  // Short signal, not started yet, with a ScriptProcessor to detect when it
+  // starts.  It should finish before garbage collection.
+  var buffer = ctx.createBuffer(1, signalLength, ctx.sampleRate);
+  applySignal(buffer, 0);
+  source = ctx.createBufferSource();
+  source.buffer = buffer;
+  var processor2 = ctx.createScriptProcessor(bufferSize, 1, 0);
+  source.connect(processor2);
+  processor2.onaudioprocess = onSourceOutput;
+};
+
+startTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delayNodeWithGain.html b/dom/media/webaudio/test/test_delayNodeWithGain.html
new file mode 100644
index 0000000000..af075c7439
--- /dev/null
+++ b/dom/media/webaudio/test/test_delayNodeWithGain.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DelayNode with a GainNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+
+    var delay = context.createDelay();
+
+    source.buffer = buffer;
+
+    var gain = context.createGain();
+    gain.gain.value = 0.5;
+
+    source.connect(gain);
+    gain.connect(delay);
+
+    // Delay the source stream by 2048 frames
+    delay.delayTime.value = 2048 / context.sampleRate;
+
+    source.start(0);
+    return delay;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048 * 2, context.sampleRate);
+    for (var i = 2048; i < 2048 * 2; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * (i - 2048) / context.sampleRate) / 2;
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_delaynode-channel-count-1.html b/dom/media/webaudio/test/test_delaynode-channel-count-1.html
new file mode 100644
index 0000000000..dd964ef9e3
--- /dev/null
+++ b/dom/media/webaudio/test/test_delaynode-channel-count-1.html
@@ -0,0 +1,104 @@
+<!DOCTYPE html>
+<title>Test that DelayNode output channelCount matches that of the delayed input</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+// See https://github.com/WebAudio/web-audio-api/issues/25
+
+// sampleRate is a power of two so that delay times are exact in base-2
+// floating point arithmetic.
+const SAMPLE_RATE = 32768;
+// Arbitrary delay time in frames (but this is assumed a multiple of block
+// size below):
+const DELAY_FRAMES = 3 * 128;
+// Implementations may apply interpolation to input samples, which can spread
+// the effect of input with larger channel counts over neighbouring blocks.
+// This test ignores enough neighbouring blocks to ignore the effects of
+// filter radius of up to this number of frames:
+const INTERPOLATION_GRACE = 128;
+// Number of frames of DelayNode output that are known to be stereo:
+const STEREO_FRAMES = 128;
+// The delay will be increased at this frame to switch DelayNode output back
+// to mono.
+const MONO_OUTPUT_START_FRAME =
+  DELAY_FRAMES + INTERPOLATION_GRACE + STEREO_FRAMES;
+// Number of frames of output that are known to be mono after the known stereo
+// and interpolation grace.
+const MONO_FRAMES = 128;
+// Total length allows for interpolation after effects of stereo input are
+// finished and one block to test return to mono output:
+const TOTAL_LENGTH =
+  MONO_OUTPUT_START_FRAME + INTERPOLATION_GRACE + MONO_FRAMES;
+// maxDelayTime, is a multiple of block size, because the Gecko implementation
+// once had a bug with delayTime = maxDelayTime in this situation:
+const MAX_DELAY_FRAMES = TOTAL_LENGTH + INTERPOLATION_GRACE;
+
+promise_test(() => {
+  let context = new OfflineAudioContext({numberOfChannels: 1,
+                                         length: TOTAL_LENGTH,
+                                         sampleRate: SAMPLE_RATE});
+
+  // Only channel 1 of the splitter is connected to the destination.
+  let splitter = new ChannelSplitterNode(context, {numberOfOutputs: 2});
+  splitter.connect(context.destination, 1);
+
+  // A gain node has channelCountMode "max" and channelInterpretation
+  // "speakers", and so will up-mix a mono input when there is stereo input.
+  let gain = new GainNode(context);
+  gain.connect(splitter);
+
+  // The delay node initially outputs a single channel of silence, when it
+  // does not have enough signal in its history to output what it has
+  // previously received.  After the delay period, it will then output the
+  // stereo signal it received.
+  let delay =
+      new DelayNode(context,
+                    {maxDelayTime: MAX_DELAY_FRAMES / context.sampleRate,
+                     delayTime: DELAY_FRAMES / context.sampleRate});
+  // Schedule an increase in the delay to return to mono silent output from
+  // the unfilled portion of the DelayNode's buffer.
+  delay.delayTime.setValueAtTime(MAX_DELAY_FRAMES / context.sampleRate,
+                                 MONO_OUTPUT_START_FRAME / context.sampleRate);
+  delay.connect(gain);
+
+  let stereoMerger = new ChannelMergerNode(context, {numberOfInputs: 2});
+  stereoMerger.connect(delay);
+
+  let leftOffset = 0.125;
+  let rightOffset = 0.5;
+  let leftSource = new ConstantSourceNode(context, {offset: leftOffset});
+  let rightSource = new ConstantSourceNode(context, {offset: rightOffset});
+  leftSource.start();
+  rightSource.start();
+  leftSource.connect(stereoMerger, 0, 0);
+  rightSource.connect(stereoMerger, 0, 1);
+  // Connect a mono source directly to the gain, so that even stereo silence
+  // will be detected in channel 1 of the gain output because it will cause
+  // the mono source to be up-mixed.
+  let monoOffset = 0.25
+  let monoSource = new ConstantSourceNode(context, {offset: monoOffset});
+  monoSource.start();
+  monoSource.connect(gain);
+
+  return context.startRendering().
+    then((buffer) => {
+      let output = buffer.getChannelData(0);
+
+      function assert_samples_equal(startIndex, length, expected, description)
+      {
+        for (let i = startIndex; i < startIndex + length; ++i) {
+          assert_equals(output[i], expected, description + ` at ${i}`);
+        }
+      }
+
+      assert_samples_equal(0, DELAY_FRAMES - INTERPOLATION_GRACE,
+                           0, "Initial mono");
+      assert_samples_equal(DELAY_FRAMES + INTERPOLATION_GRACE, STEREO_FRAMES,
+                           monoOffset + rightOffset, "Stereo");
+      assert_samples_equal(MONO_OUTPUT_START_FRAME + INTERPOLATION_GRACE,
+                           MONO_FRAMES,
+                           0, "Final mono");
+    });
+});
+
+</script>
diff --git a/dom/media/webaudio/test/test_disconnectAll.html b/dom/media/webaudio/test/test_disconnectAll.html
new file mode 100644
index 0000000000..f67b969949
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectAll.html
@@ -0,0 +1,51 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioNode</title>
+    <script src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 1,
+        createGraph(context) {
+    var sourceBuffer = context.createBuffer(1, 256, context.sampleRate);
+    var data = sourceBuffer.getChannelData(0);
+    for (var j = 0; j < data.length; j++) {
+        data[j] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain1 = context.createGain();
+    var gain2 = context.createGain();
+    var gain3 = context.createGain();
+    var merger = context.createChannelMerger(3);
+
+    source.connect(gain1);
+    source.connect(gain2);
+    source.connect(gain3);
+    gain1.connect(merger);
+    gain2.connect(merger);
+    gain3.connect(merger);
+    source.start();
+
+    source.disconnect();
+
+    return merger;
+  }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
\ No newline at end of file
diff --git a/dom/media/webaudio/test/test_disconnectAudioParam.html b/dom/media/webaudio/test/test_disconnectAudioParam.html
new file mode 100644
index 0000000000..87df902b8e
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectAudioParam.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioParam</title>
+    <script src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 1,
+        createGraph(context) {
+    var sourceBuffer = context.createBuffer(1, 256, context.sampleRate);
+    var data = sourceBuffer.getChannelData(0);
+    for (var j = 0; j < data.length; j++) {
+        data[j] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var half = context.createGain();
+    var gain1 = context.createGain();
+    var gain2 = context.createGain();
+
+    half.gain.value = 0.5;
+
+    source.connect(gain1);
+    gain1.connect(gain2);
+    source.connect(half);
+
+    half.connect(gain1.gain);
+    half.connect(gain2.gain);
+
+    half.disconnect(gain2.gain);
+
+    source.start();
+
+    return gain2;
+  },
+        createExpectedBuffers(context) {
+    let expectedBuffer = context.createBuffer(1, 256, context.sampleRate);
+    for (var i = 0; i < 256; ++i) {
+      expectedBuffer.getChannelData(0)[i] = 1.5;
+    }
+
+    return expectedBuffer;
+  }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectAudioParamFromOutput.html b/dom/media/webaudio/test/test_disconnectAudioParamFromOutput.html
new file mode 100644
index 0000000000..533091f920
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectAudioParamFromOutput.html
@@ -0,0 +1,67 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioParam</title>
+    <script src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 2,
+        createGraph(context) {
+    var sourceBuffer = context.createBuffer(2, 256, context.sampleRate);
+    for (var i = 1; i <= 2; i++) {
+      var data = sourceBuffer.getChannelData(i-1);
+      for (var j = 0; j < data.length; j++) {
+          data[j] = i;
+      }
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var half = context.createGain();
+    var gain1 = context.createGain();
+    var gain2 = context.createGain();
+    var splitter = context.createChannelSplitter(2);
+
+    half.gain.value = 0.5;
+
+    source.connect(gain1);
+    gain1.connect(gain2);
+    source.connect(half);
+    half.connect(splitter);
+    splitter.connect(gain1.gain, 0);
+    splitter.connect(gain2.gain, 1);
+
+    splitter.disconnect(gain2.gain, 1);
+
+    source.start();
+
+    return gain2;
+  },
+        createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(2, 256, context.sampleRate);
+    for (var i = 1; i <= 2; i++) {
+      var data = expectedBuffer.getChannelData(i-1);
+      for (var j = 0; j < data.length; j++) {
+        data[j] = (i == 1) ? 1.5 : 3.0;
+      }
+    }
+
+    return expectedBuffer;
+  }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectExceptions.html b/dom/media/webaudio/test/test_disconnectExceptions.html
new file mode 100644
index 0000000000..54fde4df8d
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectExceptions.html
@@ -0,0 +1,75 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioNode</title>
+    <script src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var ctx = new AudioContext();
+      var sourceBuffer = ctx.createBuffer(2, 256, ctx.sampleRate);
+      for (var i = 1; i <= 2; i++) {
+        var data = sourceBuffer.getChannelData(i-1);
+        for (var j = 0; j < data.length; j++) {
+            data[j] = i;
+        }
+      }
+
+      var source = ctx.createBufferSource();
+      source.buffer = sourceBuffer;
+
+      var gain1 = ctx.createGain();
+      var splitter = ctx.createChannelSplitter(2);
+      var merger = ctx.createChannelMerger(2);
+      var gain2 = ctx.createGain();
+      var gain3 = ctx.createGain();
+
+      gain1.connect(splitter);
+      splitter.connect(gain2, 0);
+      splitter.connect(gain3, 1);
+      splitter.connect(merger, 0, 0);
+      splitter.connect(merger, 1, 1);
+      gain2.connect(gain3);
+      gain3.connect(ctx.destination);
+      merger.connect(ctx.destination);
+
+      expectException(function() {
+          splitter.disconnect(2);
+      }, DOMException.INDEX_SIZE_ERR);
+
+      expectNoException(function() {
+        splitter.disconnect(1);
+        splitter.disconnect(1);
+      });
+
+      expectException(function() {
+        gain1.disconnect(gain2);
+      }, DOMException.INVALID_ACCESS_ERR);
+
+      expectException(function() {
+        gain1.disconnect(gain3);
+        ok(false, 'Should get InvalidAccessError exception');
+      }, DOMException.INVALID_ACCESS_ERR);
+
+      expectException(function() {
+        splitter.disconnect(gain2, 2);
+      }, DOMException.INDEX_SIZE_ERR);
+
+      expectException(function() {
+        splitter.disconnect(gain1, 0);
+      }, DOMException.INVALID_ACCESS_ERR);
+
+      expectException(function() {
+        splitter.disconnect(gain3, 0, 0);
+      }, DOMException.INVALID_ACCESS_ERR);
+
+      expectException(function() {
+        splitter.disconnect(merger, 3, 0);
+      }, DOMException.INDEX_SIZE_ERR);
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectFromAudioNode.html b/dom/media/webaudio/test/test_disconnectFromAudioNode.html
new file mode 100644
index 0000000000..5f80bbfbe8
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectFromAudioNode.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioNode</title>
+    <script src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 1,
+        createGraph(context) {
+    var sourceBuffer = context.createBuffer(1, 256, context.sampleRate);
+    var data = sourceBuffer.getChannelData(0);
+    for (var j = 0; j < data.length; j++) {
+        data[j] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain1 = context.createGain();
+    var gain2 = context.createGain();
+    var gain3 = context.createGain();
+
+    source.connect(gain1);
+    source.connect(gain2);
+
+    gain1.connect(gain3);
+    gain2.connect(gain3);
+
+    source.start();
+
+    source.disconnect(gain2);
+
+    return gain3;
+  },
+        createExpectedBuffers(context) {
+    let expectedBuffer = context.createBuffer(1, 256, context.sampleRate);
+    for (var i = 0; i < 256; ++i) {
+      expectedBuffer.getChannelData(0)[i] = 1.0;
+    }
+
+    return expectedBuffer;
+  }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectFromAudioNodeAndOutput.html b/dom/media/webaudio/test/test_disconnectFromAudioNodeAndOutput.html
new file mode 100644
index 0000000000..a0812bdb09
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectFromAudioNodeAndOutput.html
@@ -0,0 +1,59 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioNode</title>
+    <script src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 2,
+        createGraph(context) {
+    var sourceBuffer = context.createBuffer(2, 256, context.sampleRate);
+    for (var i = 1; i <= 2; i++) {
+      var data = sourceBuffer.getChannelData(i-1);
+      for (var j = 0; j < data.length; j++) {
+          data[j] = i;
+      }
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var splitter = context.createChannelSplitter(2);
+    var gain1 = context.createGain();
+    var gain2 = context.createGain();
+    var merger = context.createChannelMerger(2);
+
+    source.connect(splitter);
+    splitter.connect(gain1, 0);
+    splitter.connect(gain2, 0);
+    splitter.connect(gain2, 1);
+    gain1.connect(merger, 0, 1);
+    gain2.connect(merger, 0, 1);
+    source.start();
+
+    splitter.disconnect(gain2, 0);
+
+    return merger;
+  },
+        createExpectedBuffers(context) {
+    let expectedBuffer = context.createBuffer(2, 256, context.sampleRate);
+    for (var i = 0; i < 256; ++i) {
+      expectedBuffer.getChannelData(0)[i] = 0;
+      expectedBuffer.getChannelData(1)[i] = 3;
+    }
+
+    return expectedBuffer;
+  }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectFromAudioNodeAndOutputAndInput.html b/dom/media/webaudio/test/test_disconnectFromAudioNodeAndOutputAndInput.html
new file mode 100644
index 0000000000..478768c62d
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectFromAudioNodeAndOutputAndInput.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioNode</title>
+    <script src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 3,
+        createGraph(context) {
+    var sourceBuffer = context.createBuffer(3, 256, context.sampleRate);
+    for (var i = 1; i <= 3; i++) {
+      var data = sourceBuffer.getChannelData(i-1);
+      for (var j = 0; j < data.length; j++) {
+          data[j] = i;
+      }
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var splitter = context.createChannelSplitter(3);
+    var merger = context.createChannelMerger(3);
+
+    source.connect(splitter);
+    splitter.connect(merger, 0, 0);
+    splitter.connect(merger, 1, 1);
+    splitter.connect(merger, 2, 2);
+    source.start();
+
+    splitter.disconnect(merger, 2, 2);
+
+    return merger;
+  },
+        createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(3, 256, context.sampleRate);
+    for (var i = 1; i <= 3; i++) {
+      var data = expectedBuffer.getChannelData(i-1);
+      for (var j = 0; j < data.length; j++) {
+        data[j] = (i == 3) ? 0 : i;
+      }
+    }
+
+    return expectedBuffer;
+  }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
\ No newline at end of file
diff --git a/dom/media/webaudio/test/test_disconnectFromAudioNodeMultipleConnection.html b/dom/media/webaudio/test/test_disconnectFromAudioNodeMultipleConnection.html
new file mode 100644
index 0000000000..9bd8e4d6f6
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectFromAudioNodeMultipleConnection.html
@@ -0,0 +1,56 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>
+      Test whether we can disconnect all outbound connection of an AudioNode
+    </title>
+    <script src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 2,
+        createGraph(context) {
+    var sourceBuffer = context.createBuffer(1, 256, context.sampleRate);
+    var data = sourceBuffer.getChannelData(0);
+    for (var j = 0; j < data.length; j++) {
+        data[j] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var merger = context.createChannelMerger(2);
+    var gain = context.createGain();
+
+    source.connect(merger, 0, 0);
+    source.connect(gain);
+    source.connect(merger, 0, 1);
+
+    source.disconnect(merger);
+
+    source.start();
+
+    return merger;
+  },
+        createExpectedBuffers(context) {
+    let expectedBuffer = context.createBuffer(2, 256, context.sampleRate);
+    for (var channel = 0; channel < 2; channel++) {
+      for (var i = 0; i < 256; ++i) {
+        expectedBuffer.getChannelData(0)[i] = 0;
+      }
+    }
+
+    return expectedBuffer;
+  }
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_disconnectFromOutput.html b/dom/media/webaudio/test/test_disconnectFromOutput.html
new file mode 100644
index 0000000000..35080bf609
--- /dev/null
+++ b/dom/media/webaudio/test/test_disconnectFromOutput.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+  <head>
+    <title>Test whether we can disconnect an AudioNode</title>
+    <script src="/tests/SimpleTest/SimpleTest.js"></script>
+    <script type="text/javascript" src="webaudio.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+  <body>
+    <pre id="test">
+      <script class="testbody" type="text/javascript">
+      var gTest = {
+        length: 256,
+        numberOfChannels: 1,
+        createGraph(context) {
+    var sourceBuffer = context.createBuffer(3, 256, context.sampleRate);
+    for (var i = 1; i <= 3; i++) {
+      var data = sourceBuffer.getChannelData(i-1);
+      for (var j = 0; j < data.length; j++) {
+          data[j] = i;
+      }
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var splitter = context.createChannelSplitter(3);
+    var sum = context.createGain();
+
+    source.connect(splitter);
+    splitter.connect(sum, 0);
+    splitter.connect(sum, 1);
+    splitter.connect(sum, 2);
+    source.start();
+
+    splitter.disconnect(1);
+
+    return sum;
+  },
+        createExpectedBuffers(context) {
+    let expectedBuffer = context.createBuffer(1, 256, context.sampleRate);
+    for (var i = 0; i < 256; ++i) {
+      expectedBuffer.getChannelData(0)[i] = 4;
+    }
+
+    return expectedBuffer;
+  },
+      };
+
+      runTest();
+      </script>
+    </pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_dynamicsCompressorNode.html b/dom/media/webaudio/test/test_dynamicsCompressorNode.html
new file mode 100644
index 0000000000..05b6887a53
--- /dev/null
+++ b/dom/media/webaudio/test/test_dynamicsCompressorNode.html
@@ -0,0 +1,68 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DynamicsCompressorNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function near(a, b, msg) {
+  ok(Math.abs(a - b) < 1e-4, msg);
+}
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+
+  var osc = context.createOscillator();
+  var sp = context.createScriptProcessor();
+
+  var compressor = new DynamicsCompressorNode(context);
+
+  osc.connect(compressor);
+  osc.connect(sp);
+  compressor.connect(context.destination);
+
+  is(compressor.channelCount, 2, "compressor node has 2 input channels by default");
+  is(compressor.channelCountMode, "clamped-max", "Correct channelCountMode for the compressor node");
+  is(compressor.channelInterpretation, "speakers", "Correct channelCountInterpretation for the compressor node");
+
+  // Verify default values
+  ok(compressor.threshold instanceof AudioParam, "treshold is an AudioParam");
+  near(compressor.threshold.defaultValue, -24, "Correct default value for threshold");
+  ok(compressor.knee instanceof AudioParam, "knee is an AudioParam");
+  near(compressor.knee.defaultValue, 30, "Correct default value for knee");
+  ok(compressor.ratio instanceof AudioParam, "knee is an AudioParam");
+  near(compressor.ratio.defaultValue, 12, "Correct default value for ratio");
+  is(typeof compressor.reduction, "number", "reduction is a number");
+  near(compressor.reduction, 0, "Correct default value for reduction");
+  ok(compressor.attack instanceof AudioParam, "attack is an AudioParam");
+  near(compressor.attack.defaultValue, 0.003, "Correct default value for attack");
+  ok(compressor.release instanceof AudioParam, "release is an AudioParam");
+  near(compressor.release.defaultValue, 0.25, "Correct default value for release");
+
+  compressor.threshold.value = -80;
+
+  osc.start();
+  var iteration = 0;
+  sp.onaudioprocess = function(e) {
+    if (iteration > 10) {
+      ok(compressor.reduction < 0,
+         "Feeding a full-scale sine to a compressor should result in an db" +
+         "reduction.");
+      sp.onaudioprocess = null;
+      osc.stop(0);
+
+      SimpleTest.finish();
+    }
+    iteration++;
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_dynamicsCompressorNodePassThrough.html b/dom/media/webaudio/test/test_dynamicsCompressorNodePassThrough.html
new file mode 100644
index 0000000000..9e8d794547
--- /dev/null
+++ b/dom/media/webaudio/test/test_dynamicsCompressorNodePassThrough.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test DynamicsCompressorNode with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+
+    var compressor = context.createDynamicsCompressor();
+
+    source.buffer = this.buffer;
+
+    source.connect(compressor);
+
+    var compressorWrapped = SpecialPowers.wrap(compressor);
+    ok("passThrough" in compressorWrapped, "DynamicsCompressorNode should support the passThrough API");
+    compressorWrapped.passThrough = true;
+
+    source.start(0);
+    return compressor;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_dynamicsCompressorNodeWithGain.html b/dom/media/webaudio/test/test_dynamicsCompressorNodeWithGain.html
new file mode 100644
index 0000000000..7e6487e3f8
--- /dev/null
+++ b/dom/media/webaudio/test/test_dynamicsCompressorNodeWithGain.html
@@ -0,0 +1,51 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+<meta charset="utf-8">
+  <title>Test DynamicsCompressor with Gain</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+addLoadEvent(function() {
+  var samplerate = 44100;
+  var context = new OfflineAudioContext(1, samplerate/100, samplerate);
+
+  var osc = context.createOscillator();
+  osc.frequency.value = 2400;
+
+  var gain = context.createGain();
+  gain.gain.value = 1.5;
+
+  // These numbers are borrowed from the example code on MDN
+  // https://developer.mozilla.org/en-US/docs/Web/API/DynamicsCompressorNode
+  var compressor = context.createDynamicsCompressor();
+  compressor.threshold.value = -50;
+  compressor.knee.value = 40;
+  compressor.ratio.value = 12;
+  compressor.reduction.value = -20;
+  compressor.attack.value = 0;
+  compressor.release.value = 0.25;
+
+  osc.connect(gain);
+  gain.connect(compressor);
+  compressor.connect(context.destination);
+  osc.start();
+
+  context.startRendering().then(buffer => {
+    var peak = Math.max(...buffer.getChannelData(0));
+    console.log(peak);
+    // These values are experimentally determined. Without dynamics compression
+    // the peak should be just under 1.5. We also check for a minimum value
+    // to make sure we are not getting all zeros.
+    ok(peak >= 0.2 && peak < 1.0, "Peak value should be greater than 0.25 and less than 1.0");
+    SimpleTest.finish();
+  });
+});
+</script>
+<pre>
+</pre>
+</body>
diff --git a/dom/media/webaudio/test/test_event_listener_leaks.html b/dom/media/webaudio/test/test_event_listener_leaks.html
new file mode 100644
index 0000000000..a3bcc9259e
--- /dev/null
+++ b/dom/media/webaudio/test/test_event_listener_leaks.html
@@ -0,0 +1,47 @@
+<!--
+  Any copyright is dedicated to the Public Domain.
+  http://creativecommons.org/publicdomain/zero/1.0/
+-->
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Bug 1450358 - Test AudioContext event listener leak conditions</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="/tests/dom/events/test/event_leak_utils.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<script class="testbody" type="text/javascript">
+// Manipulate AudioContext objects in the frame's context.
+// Its important here that we create a listener callback from
+// the DOM objects back to the frame's global in order to
+// exercise the leak condition.
+async function useAudioContext(contentWindow) {
+  let ctx = new contentWindow.AudioContext();
+  ctx.onstatechange = e => {
+    contentWindow.stateChangeCount += 1;
+  };
+
+  let osc = ctx.createOscillator();
+  osc.type = "sine";
+  osc.frequency.value = 440;
+  osc.start();
+}
+
+async function runTest() {
+  try {
+    await checkForEventListenerLeaks("AudioContext", useAudioContext);
+  } catch (e) {
+    ok(false, e);
+  } finally {
+    SimpleTest.finish();
+  }
+}
+
+SimpleTest.waitForExplicitFinish();
+addEventListener("load", runTest, { once: true });
+</script>
+</pre>
+</body>
+</html>
+
diff --git a/dom/media/webaudio/test/test_gainNode.html b/dom/media/webaudio/test/test_gainNode.html
new file mode 100644
index 0000000000..77b0ae88b0
--- /dev/null
+++ b/dom/media/webaudio/test/test_gainNode.html
@@ -0,0 +1,72 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test GainNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+
+    var gain = new GainNode(context);
+    ok(gain.gain, "The audioparam member must exist");
+    is(gain.gain.value, 1.0, "Correct initial value");
+    is(gain.gain.defaultValue, 1.0, "Correct default value");
+    gain.gain.value = 0.5;
+    is(gain.gain.value, 0.5, "Correct initial value");
+    is(gain.gain.defaultValue, 1.0, "Correct default value");
+
+    gain = new GainNode(context, { gain: 0.5 });
+    ok(gain.gain, "The audioparam member must exist");
+    is(gain.gain.value, 0.5, "Correct initial value");
+    is(gain.gain.defaultValue, 1.0, "Correct default value");
+    gain.gain.value = 0.5;
+    is(gain.gain.value, 0.5, "Correct initial value");
+    is(gain.gain.defaultValue, 1.0, "Correct default value");
+
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+
+    gain = context.createGain();
+    source.connect(gain);
+
+    ok(gain.gain, "The audioparam member must exist");
+    is(gain.gain.value, 1.0, "Correct initial value");
+    is(gain.gain.defaultValue, 1.0, "Correct default value");
+    gain.gain.value = 0.5;
+    is(gain.gain.value, 0.5, "Correct initial value");
+    is(gain.gain.defaultValue, 1.0, "Correct default value");
+    is(gain.channelCount, 2, "gain node has 2 input channels by default");
+    is(gain.channelCountMode, "max", "Correct channelCountMode for the gain node");
+    is(gain.channelInterpretation, "speakers", "Correct channelCountInterpretation for the gain node");
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate) / 2;
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_gainNodeInLoop.html b/dom/media/webaudio/test/test_gainNodeInLoop.html
new file mode 100644
index 0000000000..90dedc0ef4
--- /dev/null
+++ b/dom/media/webaudio/test/test_gainNodeInLoop.html
@@ -0,0 +1,48 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test GainNode in presence of loops</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var sourceBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+    }
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+    source.loop = true;
+    source.start(0);
+    source.stop(sourceBuffer.duration * 2);
+
+    var gain = context.createGain();
+    // Adjust the gain in a way that we don't just end up modifying AudioChunk::mVolume
+    gain.gain.setValueAtTime(0.5, 0);
+    source.connect(gain);
+    return gain;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 4096, context.sampleRate);
+    for (var i = 0; i < 4096; ++i) {
+      expectedBuffer.getChannelData(0)[i] = 0.5;
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_gainNodePassThrough.html b/dom/media/webaudio/test/test_gainNodePassThrough.html
new file mode 100644
index 0000000000..5a973ccaa2
--- /dev/null
+++ b/dom/media/webaudio/test/test_gainNodePassThrough.html
@@ -0,0 +1,49 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test GainNode with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+
+    var gain = context.createGain();
+
+    source.buffer = this.buffer;
+
+    source.connect(gain);
+
+    gain.gain.value = 0.5;
+
+    var gainWrapped = SpecialPowers.wrap(gain);
+    ok("passThrough" in gainWrapped, "GainNode should support the passThrough API");
+    gainWrapped.passThrough = true;
+
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_iirFilterNodePassThrough.html b/dom/media/webaudio/test/test_iirFilterNodePassThrough.html
new file mode 100644
index 0000000000..ab54499e04
--- /dev/null
+++ b/dom/media/webaudio/test/test_iirFilterNodePassThrough.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test IIRFilterNode with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+
+    var filter = context.createIIRFilter([0.5, 0.5], [1.0]);
+
+    source.buffer = this.buffer;
+
+    source.connect(filter);
+
+    var filterWrapped = SpecialPowers.wrap(filter);
+    ok("passThrough" in filterWrapped, "BiquadFilterNode should support the passThrough API");
+    filterWrapped.passThrough = true;
+
+    source.start(0);
+    return filter;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_maxChannelCount.html b/dom/media/webaudio/test/test_maxChannelCount.html
new file mode 100644
index 0000000000..1a4e5c856e
--- /dev/null
+++ b/dom/media/webaudio/test/test_maxChannelCount.html
@@ -0,0 +1,38 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the AudioContext.destination interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// Work around bug 911777
+SpecialPowers.forceGC();
+SpecialPowers.forceCC();
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ac = new AudioContext();
+  ok(ac.destination.maxChannelCount > 0, "We can query the maximum number of channels");
+
+  var oac = new OfflineAudioContext(2, 1024, 48000);
+  is(oac.destination.maxChannelCount, 2, "This OfflineAudioContext should have 2 max channels.");
+
+  oac = new OfflineAudioContext(6, 1024, 48000);
+  is(oac.destination.maxChannelCount, 6, "This OfflineAudioContext should have 6 max channels.");
+
+  expectException(function() {
+    oac.destination.channelCount = oac.destination.channelCount + 1;
+  }, DOMException.INDEX_SIZE_ERR);
+
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaDecoding.html b/dom/media/webaudio/test/test_mediaDecoding.html
new file mode 100644
index 0000000000..e76a533e4a
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaDecoding.html
@@ -0,0 +1,436 @@
+<!DOCTYPE html>
+<html>
+  <head>
+    <title>Test the decodeAudioData API and Resampling</title>
+    <script src="/tests/SimpleTest/SimpleTest.js"></script>
+    <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+  </head>
+
+  <body>
+    <pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script type="text/javascript">
+
+  // These routines have been copied verbatim from WebKit, and are used in order
+  // to convert a memory buffer into a wave buffer.
+  function writeString(s, a, offset) {
+    for (var i = 0; i < s.length; ++i) {
+      a[offset + i] = s.charCodeAt(i);
+    }
+  }
+
+  function writeInt16(n, a, offset) {
+    n = Math.floor(n);
+
+    var b1 = n & 255;
+    var b2 = (n >> 8) & 255;
+
+    a[offset + 0] = b1;
+    a[offset + 1] = b2;
+  }
+
+  function writeInt32(n, a, offset) {
+    n = Math.floor(n);
+    var b1 = n & 255;
+    var b2 = (n >> 8) & 255;
+    var b3 = (n >> 16) & 255;
+    var b4 = (n >> 24) & 255;
+
+    a[offset + 0] = b1;
+    a[offset + 1] = b2;
+    a[offset + 2] = b3;
+    a[offset + 3] = b4;
+  }
+
+  function writeAudioBuffer(audioBuffer, a, offset) {
+    var n = audioBuffer.length;
+    var channels = audioBuffer.numberOfChannels;
+
+    for (var i = 0; i < n; ++i) {
+      for (var k = 0; k < channels; ++k) {
+        var buffer = audioBuffer.getChannelData(k);
+        var sample = buffer[i] * 32768.0;
+
+        // Clip samples to the limitations of 16-bit.
+        // If we don't do this then we'll get nasty wrap-around distortion.
+        if (sample < -32768)
+          sample = -32768;
+        if (sample > 32767)
+          sample = 32767;
+
+        writeInt16(sample, a, offset);
+        offset += 2;
+      }
+    }
+  }
+
+  function createWaveFileData(audioBuffer) {
+    var frameLength = audioBuffer.length;
+    var numberOfChannels = audioBuffer.numberOfChannels;
+    var sampleRate = audioBuffer.sampleRate;
+    var bitsPerSample = 16;
+    var byteRate = sampleRate * numberOfChannels * bitsPerSample / 8;
+    var blockAlign = numberOfChannels * bitsPerSample / 8;
+    var wavDataByteLength = frameLength * numberOfChannels * 2; // 16-bit audio
+    var headerByteLength = 44;
+    var totalLength = headerByteLength + wavDataByteLength;
+
+    var waveFileData = new Uint8Array(totalLength);
+
+    var subChunk1Size = 16; // for linear PCM
+    var subChunk2Size = wavDataByteLength;
+    var chunkSize = 4 + (8 + subChunk1Size) + (8 + subChunk2Size);
+
+    writeString("RIFF", waveFileData, 0);
+    writeInt32(chunkSize, waveFileData, 4);
+    writeString("WAVE", waveFileData, 8);
+    writeString("fmt ", waveFileData, 12);
+
+    writeInt32(subChunk1Size, waveFileData, 16);      // SubChunk1Size (4)
+    writeInt16(1, waveFileData, 20);                  // AudioFormat (2)
+    writeInt16(numberOfChannels, waveFileData, 22);   // NumChannels (2)
+    writeInt32(sampleRate, waveFileData, 24);         // SampleRate (4)
+    writeInt32(byteRate, waveFileData, 28);           // ByteRate (4)
+    writeInt16(blockAlign, waveFileData, 32);         // BlockAlign (2)
+    writeInt32(bitsPerSample, waveFileData, 34);      // BitsPerSample (4)
+
+    writeString("data", waveFileData, 36);
+    writeInt32(subChunk2Size, waveFileData, 40);      // SubChunk2Size (4)
+
+    // Write actual audio data starting at offset 44.
+    writeAudioBuffer(audioBuffer, waveFileData, 44);
+
+    return waveFileData;
+  }
+
+</script>
+<script class="testbody" type="text/javascript">
+
+  SimpleTest.waitForExplicitFinish();
+
+  // fuzzTolerance and fuzzToleranceMobile are used to determine fuzziness
+  // thresholds.  They're needed to make sure that we can deal with neglibible
+  // differences in the binary buffer caused as a result of resampling the
+  // audio.  fuzzToleranceMobile is typically larger on mobile platforms since
+  // we do fixed-point resampling as opposed to floating-point resampling on
+  // those platforms.
+  // If fuzzMagnitude, is present, is the maximum magnitude difference, per
+  // sample, to consider two samples are identical. It is multiplied by the 
+  // maximum value a sample, in our case INT16_MAX. This allows checking files
+  // that should be identical except one has e.g. a higher quantization noise.
+  var files = [
+    // An ogg file, 44.1khz, mono
+    {
+      url: "ting-44.1k-1ch.ogg",
+      valid: true,
+      expectedUrl: "ting-44.1k-1ch.wav",
+      numberOfChannels: 1,
+      frames: 30592,
+      sampleRate: 44100,
+      duration: 0.693,
+      fuzzTolerance: 5,
+      fuzzToleranceMobile: 1284
+    },
+    // An ogg file, 44.1khz, stereo
+    {
+      url: "ting-44.1k-2ch.ogg",
+      valid: true,
+      expectedUrl: "ting-44.1k-2ch.wav",
+      numberOfChannels: 2,
+      frames: 30592,
+      sampleRate: 44100,
+      duration: 0.693,
+      fuzzTolerance: 6,
+      fuzzToleranceMobile: 2544
+    },
+    // An ogg file, 48khz, mono
+    {
+      url: "ting-48k-1ch.ogg",
+      valid: true,
+      expectedUrl: "ting-48k-1ch.wav",
+      numberOfChannels: 1,
+      frames: 33297,
+      sampleRate: 48000,
+      duration: 0.693,
+      fuzzTolerance: 5,
+      fuzzToleranceMobile: 1388
+    },
+    // An ogg file, 48khz, stereo
+    {
+      url: "ting-48k-2ch.ogg",
+      valid: true,
+      expectedUrl: "ting-48k-2ch.wav",
+      numberOfChannels: 2,
+      frames: 33297,
+      sampleRate: 48000,
+      duration: 0.693,
+      fuzzTolerance: 14,
+      fuzzToleranceMobile: 2752
+    },
+    // Make sure decoding a wave file results in the same buffer (for both the
+    // resampling and non-resampling cases)
+    {
+      url: "ting-44.1k-1ch.wav",
+      valid: true,
+      expectedUrl: "ting-44.1k-1ch.wav",
+      numberOfChannels: 1,
+      frames: 30592,
+      sampleRate: 44100,
+      duration: 0.693,
+      fuzzTolerance: 0,
+      fuzzToleranceMobile: 0
+    },
+    {
+      url: "ting-48k-1ch.wav",
+      valid: true,
+      expectedUrl: "ting-48k-1ch.wav",
+      numberOfChannels: 1,
+      frames: 33297,
+      sampleRate: 48000,
+      duration: 0.693,
+      fuzzTolerance: 0,
+      fuzzToleranceMobile: 0
+    },
+    //  // A wave file
+    //  //{ url: "24bit-44khz.wav", valid: true, expectedUrl: "24bit-44khz-expected.wav" },
+    // A non-audio file
+    { url: "invalid.txt", valid: false, sampleRate: 44100 },
+    // A webm file with no audio
+    { url: "noaudio.webm", valid: false, sampleRate: 48000 },
+    // A video ogg file with audio
+    {
+      url: "audio.ogv",
+      valid: true,
+      expectedUrl: "audio-expected.wav",
+      numberOfChannels: 2,
+      sampleRate: 44100,
+      frames: 47680,
+      duration: 1.0807,
+      fuzzTolerance: 106,
+      fuzzToleranceMobile: 3482
+    },
+    {
+      url: "nil-packet.ogg",
+      expectedUrl: null,
+      valid: true,
+      numberOfChannels: 2,
+      sampleRate: 48000,
+      frames: 18600,
+      duration: 0.3874,
+    },
+    {
+      url: "half-a-second-1ch-44100-mulaw.wav",
+      // It is expected that mulaw and linear are similar enough at 16-bits
+      expectedUrl: "half-a-second-1ch-44100.wav",
+      valid: true,
+      numberOfChannels: 1,
+      sampleRate: 44100,
+      frames: 22050,
+      duration: 0.5,
+      fuzzMagnitude: 0.04,
+    },
+    {
+      url: "half-a-second-1ch-44100-alaw.wav",
+      // It is expected that alaw and linear are similar enough at 16-bits
+      expectedUrl: "half-a-second-1ch-44100.wav",
+      valid: true,
+      numberOfChannels: 1,
+      sampleRate: 44100,
+      frames: 22050,
+      duration: 0.5,
+      fuzzMagnitude: 0.04,
+    },
+    {
+      url: "waveformatextensible.wav",
+      valid: true,
+      numberOfChannels: 1,
+      sampleRate: 44100,
+      frames: 472,
+      duration: 0.01
+    },
+    {
+      // A wav file that has 8 channel, but has a channel mask that doesn't
+      // match the channel count.
+      url: "waveformatextensiblebadmask.wav",
+      valid: true,
+      numberOfChannels: 8,
+      sampleRate: 8000,
+      frames: 80,
+      duration: 0.01
+    }
+  ];
+
+  // Returns true if the memory buffers are less different that |fuzz| bytes
+  function fuzzyMemcmp(buf1, buf2, fuzz) {
+    var difference = 0;
+    is(buf1.length, buf2.length, "same length");
+    for (var i = 0; i < buf1.length; ++i) {
+      if (Math.abs(buf1[i] - buf2[i]) > fuzz.magnitude * (2 << 15)) {
+        ++difference;
+      }
+    }
+    if (difference > fuzz.count) {
+      ok(false, "Expected at most " + fuzz + " bytes difference, found " + difference + " bytes");
+    }
+    console.log(difference, fuzz.count);
+    return difference <= fuzz.count;
+  }
+
+  function getFuzzTolerance(test) {
+    var kIsMobile =
+      navigator.userAgent.includes("Mobile") || // b2g
+      navigator.userAgent.includes("Android");  // android
+    return {
+      magnitude: test.fuzzMagnitude ?? 0,
+      count: kIsMobile ? test.fuzzToleranceMobile ?? 0 : test.fuzzTolerance ?? 0
+    };
+  }
+
+  function bufferIsSilent(buffer) {
+    for (var i = 0; i < buffer.length; ++i) {
+      if (buffer.getChannelData(0)[i] != 0) {
+        return false;
+      }
+    }
+    return true;
+  }
+
+  function checkAudioBuffer(buffer, test) {
+    if (buffer.numberOfChannels != test.numberOfChannels) {
+      is(buffer.numberOfChannels, test.numberOfChannels, "Correct number of channels");
+      return;
+    }
+    ok(Math.abs(buffer.duration - test.duration) < 1e-3, `Correct duration expected ${test.duration} got ${buffer.duration}`);
+    if (Math.abs(buffer.duration - test.duration) >= 1e-3) {
+      ok(false, "got: " + buffer.duration + ", expected: " + test.duration);
+    }
+    is(buffer.sampleRate, test.sampleRate, "Correct sample rate");
+    is(buffer.length, test.frames, "Correct length");
+
+    var wave = createWaveFileData(buffer);
+    if (test.expectedWaveData) {
+      ok(fuzzyMemcmp(wave, test.expectedWaveData, getFuzzTolerance(test)), "Received expected decoded data for " + test.url);
+    }
+  }
+
+  function checkResampledBuffer(buffer, test, callback) {
+    if (buffer.numberOfChannels != test.numberOfChannels) {
+      is(buffer.numberOfChannels, test.numberOfChannels, "Correct number of channels");
+      return;
+    }
+    ok(Math.abs(buffer.duration - test.duration) < 1e-3, "Correct duration");
+    if (Math.abs(buffer.duration - test.duration) >= 1e-3) {
+      ok(false, "got: " + buffer.duration + ", expected: " + test.duration);
+    }
+    // Take into account the resampling when checking the size
+    var expectedLength = test.frames * buffer.sampleRate / test.sampleRate;
+    SimpleTest.ok(
+      Math.abs(buffer.length - expectedLength) < 1.0,
+      "Correct length - got " + buffer.length +
+      ", expected about " + expectedLength
+    );
+
+    // Playback the buffer in the original context, to resample back to the
+    // original rate and compare with the decoded buffer without resampling.
+    let cx = test.nativeContext;
+    var expected = cx.createBufferSource();
+    expected.buffer = test.expectedBuffer;
+    expected.start();
+    var inverse = cx.createGain();
+    inverse.gain.value = -1;
+    expected.connect(inverse);
+    inverse.connect(cx.destination);
+    var resampled = cx.createBufferSource();
+    resampled.buffer = buffer;
+    resampled.start();
+    // This stop should do nothing, but it tests for bug 937475
+    resampled.stop(test.frames / cx.sampleRate);
+    resampled.connect(cx.destination);
+    cx.oncomplete = function (e) {
+      ok(!bufferIsSilent(e.renderedBuffer), "Expect buffer not silent");
+      // Resampling will lose the highest frequency components, so we should
+      // pass the difference through a low pass filter.  However, either the
+      // input files don't have significant high frequency components or the
+      // tolerance in compareBuffers() is too high to detect them.
+      compareBuffers(e.renderedBuffer,
+        cx.createBuffer(test.numberOfChannels,
+          test.frames, test.sampleRate));
+      callback();
+    }
+    cx.startRendering();
+  }
+
+  function runResampling(test, response, callback) {
+    var sampleRate = test.sampleRate == 44100 ? 48000 : 44100;
+    var cx = new OfflineAudioContext(1, 1, sampleRate);
+    cx.decodeAudioData(response, function onSuccess(asyncResult) {
+      is(asyncResult.sampleRate, sampleRate, "Correct sample rate");
+
+      checkResampledBuffer(asyncResult, test, callback);
+    }, function onFailure() {
+      ok(false, "Expected successful decode with resample");
+      callback();
+    });
+  }
+
+  function runTest(test, response, callback) {
+    // We need to copy the array here, because decodeAudioData will detach the
+    // array's buffer.
+    var compressedAudio = response.slice(0);
+    var expectCallback = false;
+    var cx = new OfflineAudioContext(test.numberOfChannels || 1,
+      test.frames || 1, test.sampleRate);
+    cx.decodeAudioData(response, function onSuccess(asyncResult) {
+      ok(expectCallback, "Success callback should fire asynchronously");
+      ok(test.valid, "Did expect success for test " + test.url);
+
+      checkAudioBuffer(asyncResult, test);
+
+      test.expectedBuffer = asyncResult;
+      test.nativeContext = cx;
+      runResampling(test, compressedAudio, callback);
+    }, function onFailure(e) {
+      ok(e instanceof DOMException, "We want to see an exception here");
+      is(e.name, "EncodingError", "Exception name matches");
+      ok(expectCallback, "Failure callback should fire asynchronously");
+      ok(!test.valid, "Did expect failure for test " + test.url);
+      callback();
+    });
+    expectCallback = true;
+  }
+
+  function loadTest(test, callback) {
+    var xhr = new XMLHttpRequest();
+    xhr.open("GET", test.url, true);
+    xhr.responseType = "arraybuffer";
+    xhr.onload = function () {
+      if (!test.expectedUrl) {
+        runTest(test, xhr.response, callback);
+        return;
+      }
+      var getExpected = new XMLHttpRequest();
+      getExpected.open("GET", test.expectedUrl, true);
+      getExpected.responseType = "arraybuffer";
+      getExpected.onload = function () {
+        test.expectedWaveData = new Uint8Array(getExpected.response);
+        runTest(test, xhr.response, callback);
+      };
+      getExpected.send();
+    };
+    xhr.send();
+  }
+
+  function loadNextTest() {
+    if (files.length) {
+      loadTest(files.shift(), loadNextTest);
+    } else {
+      SimpleTest.finish();
+    }
+  }
+
+  loadNextTest();
+
+</script>
+</pre>
+  </body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaElementAudioSourceNode.html b/dom/media/webaudio/test/test_mediaElementAudioSourceNode.html
new file mode 100644
index 0000000000..36e1e9f2cc
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaElementAudioSourceNode.html
@@ -0,0 +1,74 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaElementAudioSourceNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+function test() {
+  var audio = new Audio("small-shot.ogg");
+  var context = new AudioContext();
+  var expectedMinNonzeroSampleCount;
+  var expectedMaxNonzeroSampleCount;
+  var nonzeroSampleCount = 0;
+  var complete = false;
+  var iterationCount = 0;
+
+  // This test ensures we receive at least expectedSampleCount nonzero samples
+  function processSamples(e) {
+    if (complete) {
+      return;
+    }
+
+    if (iterationCount == 0) {
+      // Don't start playing the audio until the AudioContext stuff is connected
+      // and running.
+      audio.play();
+    }
+    ++iterationCount;
+
+    var buf = e.inputBuffer.getChannelData(0);
+    var nonzeroSamplesThisBuffer = 0;
+    for (var i = 0; i < buf.length; ++i) {
+      if (buf[i] != 0) {
+        ++nonzeroSamplesThisBuffer;
+      }
+    }
+    nonzeroSampleCount += nonzeroSamplesThisBuffer;
+    is(e.inputBuffer.numberOfChannels, 1,
+       "Checking data channel count (nonzeroSamplesThisBuffer=" +
+       nonzeroSamplesThisBuffer + ")");
+    ok(nonzeroSampleCount <= expectedMaxNonzeroSampleCount,
+       "Too many nonzero samples (got " + nonzeroSampleCount + ", expected max " + expectedMaxNonzeroSampleCount + ")");
+    if (nonzeroSampleCount >= expectedMinNonzeroSampleCount &&
+        nonzeroSamplesThisBuffer == 0) {
+      ok(true,
+       "Check received enough nonzero samples (got " + nonzeroSampleCount + ", expected min " + expectedMinNonzeroSampleCount + ")");
+      SimpleTest.finish();
+      complete = true;
+    }
+  }
+
+  audio.onloadedmetadata = function() {
+    var node = new MediaElementAudioSourceNode(context, { mediaElement: audio });
+    var sp = context.createScriptProcessor(2048, 1);
+    node.connect(sp);
+    // Use a fuzz factor of 100 to account for samples that just happen to be zero
+    expectedMinNonzeroSampleCount = Math.floor(audio.duration*context.sampleRate) - 100;
+    expectedMaxNonzeroSampleCount = Math.floor(audio.duration*context.sampleRate) + 500;
+    sp.onaudioprocess = processSamples;
+  };
+}
+
+SpecialPowers.pushPrefEnv({"set": [["media.preload.default", 2], ["media.preload.auto", 3]]}, test);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaElementAudioSourceNodeCrossOrigin.html b/dom/media/webaudio/test/test_mediaElementAudioSourceNodeCrossOrigin.html
new file mode 100644
index 0000000000..778658b332
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaElementAudioSourceNodeCrossOrigin.html
@@ -0,0 +1,94 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamAudioSourceNode doesn't get data from cross-origin media resources</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+// Turn off the authentication dialog blocking for this test.
+SpecialPowers.setIntPref("network.auth.subresource-http-auth-allow", 2)
+
+var tests = [
+  // Not the same origin no CORS asked for, should have silence
+  { url: "http://example.org:80/tests/dom/media/webaudio/test/small-shot.ogg",
+    cors: null,
+    expectSilence: true },
+  // Same origin, should have sound
+  { url: "small-shot.ogg",
+    cors: null,
+    expectSilence: false },
+  // Cross-origin but we asked for CORS and the server answered with the right
+  // header, should have
+  { url: "http://example.org:80/tests/dom/media/webaudio/test/corsServer.sjs",
+    cors: "anonymous",
+    expectSilence: false }
+];
+
+var testsRemaining = tests.length;
+
+tests.forEach(function(e) {
+  e.ac = new AudioContext();
+  var a = new Audio();
+  if (e.cors) {
+    a.crossOrigin = e.cors;
+  }
+  a.src = e.url;
+  document.body.appendChild(a);
+
+  a.onloadedmetadata = () => {
+    // Wait for "loadedmetadata" before capturing since tracks are then known
+    // directly. If we set up the capture before "loadedmetadata" we
+    // (internally) have to wait an extra async jump for tracks to become known
+    // to main thread, before setting up audio data forwarding to the node.
+    // As that happens, the audio resource may have already ended on slow test
+    // machines, causing failures.
+    a.onloadedmetadata = null;
+    var measn = e.ac.createMediaElementSource(a);
+    var sp = e.ac.createScriptProcessor(2048, 1);
+    sp.seenSound = false;
+    sp.onaudioprocess = checkBufferSilent;
+
+    measn.connect(sp);
+    a.play();
+  };
+
+  function checkFinished(sp) {
+    if (a.ended) {
+      sp.onaudioprocess = null;
+      var not = e.expectSilence ? "" : "not";
+      is(e.expectSilence, !sp.seenSound,
+         "Buffer is " + not + " silent as expected, for " +
+         e.url + " (cors: " + e.cors + ")");
+      if (--testsRemaining == 0) {
+        SimpleTest.finish();
+      }
+    }
+  }
+
+  function checkBufferSilent(e) {
+    var inputArrayBuffer = e.inputBuffer.getChannelData(0);
+    var silent = true;
+    for (var i = 0; i < inputArrayBuffer.length; i++) {
+      if (inputArrayBuffer[i] != 0.0) {
+        silent = false;
+        break;
+      }
+    }
+    // It is acceptable to find a full buffer of silence here, even if we expect
+    // sound, because Gecko's looping on media elements is not seamless and we
+    // can underrun. We are looking for at least one buffer of non-silent data.
+    e.target.seenSound = !silent || e.target.seenSound;
+    checkFinished(e.target);
+    return silent;
+  }
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaElementAudioSourceNodeFidelity.html b/dom/media/webaudio/test/test_mediaElementAudioSourceNodeFidelity.html
new file mode 100644
index 0000000000..4714172607
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaElementAudioSourceNodeFidelity.html
@@ -0,0 +1,137 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamAudioSourceNode doesn't get data from cross-origin media resources</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+function binIndexForFrequency(frequency, analyser) {
+  return 1 + Math.round(frequency *
+                        analyser.fftSize /
+                        analyser.context.sampleRate);
+}
+
+function debugCanvas(analyser) {
+  var cvs = document.createElement("canvas");
+  document.body.appendChild(cvs);
+
+  // Easy: 1px per bin
+  cvs.width = analyser.frequencyBinCount;
+  cvs.height = 256;
+  cvs.style.border = "1px solid red";
+
+  var c = cvs.getContext('2d');
+  var buf = new Uint8Array(analyser.frequencyBinCount);
+
+  function render() {
+    c.clearRect(0, 0, cvs.width, cvs.height);
+    analyser.getByteFrequencyData(buf);
+    for (var i = 0; i < buf.length; i++) {
+      c.fillRect(i, (256 - (buf[i])), 1, 256);
+    }
+    requestAnimationFrame(render);
+  }
+  requestAnimationFrame(render);
+}
+
+
+function checkFrequency(an) {
+  an.getFloatFrequencyData(frequencyArray);
+  // We should have no energy when checking the data largely outside the index
+  // for 440Hz (the frequency of the sine wave), start checking an octave above,
+  // the Opus compression can add some harmonics to the pure since wave.
+  var maxNoiseIndex = binIndexForFrequency(880, an);
+  for (var i = maxNoiseIndex + 1; i < frequencyArray.length; i++) {
+    if (frequencyArray[i] > frequencyArray[maxNoiseIndex]) {
+       maxNoiseIndex = i;
+    }
+  }
+
+  // On the other hand, we should find a peak at 440Hz. Our sine wave is not
+  // attenuated, we're expecting the peak to reach 0dBFs.
+  var index = binIndexForFrequency(440, an);
+  info("energy at 440: " + frequencyArray[index] +
+       ", threshold " + (an.maxDecibels - 10) +
+       "; max noise at index " + maxNoiseIndex +
+       ": " + frequencyArray[maxNoiseIndex] );
+  if (frequencyArray[index] < (an.maxDecibels - 10)) {
+    return false;
+  }
+  // Let some slack, there might be some noise here because of int -> float
+  // conversion or the Opus encoding.
+  if (frequencyArray[maxNoiseIndex] > an.minDecibels + 40) {
+    return false;
+  }
+
+  return true;
+}
+
+var audioElement = new Audio();
+audioElement.src = 'sine-440-10s.opus'
+audioElement.loop = true;
+var ac = new AudioContext();
+var mediaElementSource = ac.createMediaElementSource(audioElement);
+var an = ac.createAnalyser();
+// Use no smoothing as this would just average with previous
+// getFloatFrequencyData() calls.  Non-seamless looping would introduce noise,
+// and smoothing would spread this into calls after the loop point.
+an.smoothingTimeConstant = 0;
+let frequencyArray = new Float32Array(an.frequencyBinCount);
+
+// Uncomment this to check what the analyser is doing.
+// debugCanvas(an);
+
+mediaElementSource.connect(an)
+
+audioElement.play();
+// We want to check the we have the expected audio for at least two loop of
+// the HTMLMediaElement, piped into an AudioContext. The file is ten seconds,
+// and we use the default FFT size.
+var lastCurrentTime = 0;
+var loopCount = 0;
+audioElement.onplaying = function() {
+  audioElement.ontimeupdate = function() {
+    // We don't run the analysis when close to loop point or at the
+    // beginning, since looping is not seamless, there could be an
+    // unpredictable amount of silence
+    var rv = checkFrequency(an);
+    info("currentTime: " + audioElement.currentTime);
+    if (audioElement.currentTime < 4 ||
+        audioElement.currentTime > 8){
+      return;
+    }
+    if (!rv) {
+      ok(false, "Found unexpected noise during analysis.");
+      audioElement.ontimeupdate = null;
+      audioElement.onplaying = null;
+      ac.close();
+      audioElement.src = '';
+      SimpleTest.finish()
+      return;
+    }
+    ok(true, "Found correct audio signal during analysis");
+    info(lastCurrentTime + " " + audioElement.currentTime);
+    if (lastCurrentTime > audioElement.currentTime) {
+      info("loopCount: " + loopCount);
+      if (loopCount > 1) {
+        audioElement.ontimeupdate = null;
+        audioElement.onplaying = null;
+        ac.close();
+        audioElement.src = '';
+        SimpleTest.finish();
+      }
+      lastCurrentTime = audioElement.currentTime;
+      loopCount++;
+    } else {
+      lastCurrentTime = audioElement.currentTime;
+    }
+  }
+}
+
+</script>
diff --git a/dom/media/webaudio/test/test_mediaElementAudioSourceNodePassThrough.html b/dom/media/webaudio/test/test_mediaElementAudioSourceNodePassThrough.html
new file mode 100644
index 0000000000..5ee12b16ad
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaElementAudioSourceNodePassThrough.html
@@ -0,0 +1,66 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaElementAudioSourceNode with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+function test() {
+  var audio = new Audio("small-shot.ogg");
+  var context = new AudioContext();
+  var node = context.createMediaElementSource(audio);
+  var sp = context.createScriptProcessor(2048, 1);
+  node.connect(sp);
+  var nonzeroSampleCount = 0;
+  var complete = false;
+  var iterationCount = 0;
+
+  var srcWrapped = SpecialPowers.wrap(node);
+  ok("passThrough" in srcWrapped, "MediaElementAudioSourceNode should support the passThrough API");
+  srcWrapped.passThrough = true;
+
+  // This test ensures we receive at least expectedSampleCount nonzero samples
+  function processSamples(e) {
+    if (complete) {
+      return;
+    }
+
+    if (iterationCount == 0) {
+      // Don't start playing the audio until the AudioContext stuff is connected
+      // and running.
+      audio.play();
+    }
+    ++iterationCount;
+
+    var buf = e.inputBuffer.getChannelData(0);
+    var nonzeroSamplesThisBuffer = 0;
+    for (var i = 0; i < buf.length; ++i) {
+      if (buf[i] != 0) {
+        ++nonzeroSamplesThisBuffer;
+      }
+    }
+    nonzeroSampleCount += nonzeroSamplesThisBuffer;
+    if (iterationCount == 10) {
+      is(nonzeroSampleCount, 0, "The input must be silence");
+      SimpleTest.finish();
+      complete = true;
+    }
+  }
+
+  audio.oncanplaythrough = function() {
+    sp.onaudioprocess = processSamples;
+  };
+}
+
+SpecialPowers.pushPrefEnv({"set": [["media.preload.default", 2], ["media.preload.auto", 3]]}, test);
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaElementAudioSourceNodeVideo.html b/dom/media/webaudio/test/test_mediaElementAudioSourceNodeVideo.html
new file mode 100644
index 0000000000..dcb85f12cb
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaElementAudioSourceNodeVideo.html
@@ -0,0 +1,70 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaElementAudioSourceNode before "loadedmetadata"</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+var video = document.createElement("video");
+function test() {
+  video.src = "audiovideo.mp4";
+
+  var context = new AudioContext();
+  var complete = false;
+
+  video.onended = () => {
+    if (complete) {
+      return;
+    }
+
+    complete = true;
+    ok(false, "Video ended without any samples seen");
+    SimpleTest.finish();
+  };
+
+  video.ontimeupdate = () => {
+    info("Timeupdate: " + video.currentTime);
+  };
+
+  var node = context.createMediaElementSource(video);
+  var sp = context.createScriptProcessor(2048, 1);
+  node.connect(sp);
+
+  // This test ensures we receive some nonzero samples when we capture to
+  // WebAudio before "loadedmetadata".
+  sp.onaudioprocess = e => {
+    if (complete) {
+      return;
+    }
+
+    var buf = e.inputBuffer.getChannelData(0);
+    for (var i = 0; i < buf.length; ++i) {
+      if (buf[i] != 0) {
+        complete = true;
+        ok(true, "Got non-zero samples");
+        SimpleTest.finish();
+        return;
+      }
+    }
+  };
+
+  video.play();
+}
+
+if (video.canPlayType("video/mp4")) {
+  test();
+} else {
+  ok(true, "MP4 not supported. Skipping.");
+  SimpleTest.finish();
+}
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaStreamAudioDestinationNode.html b/dom/media/webaudio/test/test_mediaStreamAudioDestinationNode.html
new file mode 100644
index 0000000000..fd0ce8141b
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamAudioDestinationNode.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test MediaStreamAudioDestinationNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<audio id="audioelem"></audio>
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("This test uses a live media element so it needs to wait for the media stack to do some work.");
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var source = context.createBufferSource();
+  source.buffer = buffer;
+
+  var dest = new MediaStreamAudioDestinationNode(context);
+  source.connect(dest);
+
+  var elem = document.getElementById('audioelem');
+  elem.srcObject = dest.stream;
+  elem.onloadedmetadata = function() {
+    ok(true, "got metadata event");
+    setTimeout(function() {
+      is(elem.played.length, 1, "should have a played interval");
+      is(elem.played.start(0), 0, "should have played immediately");
+      isnot(elem.played.end(0), 0, "should have played for a non-zero interval");
+
+      // This will end the media element.
+      dest.stream.getTracks()[0].stop();
+    }, 2000);
+  };
+  elem.onended = function() {
+    ok(true, "media element ended after destination track.stop()");
+    SimpleTest.finish();
+  };
+
+  source.start(0);
+  elem.play();
+});
+</script>
diff --git a/dom/media/webaudio/test/test_mediaStreamAudioSourceNode.html b/dom/media/webaudio/test/test_mediaStreamAudioSourceNode.html
new file mode 100644
index 0000000000..eaa8a564b9
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamAudioSourceNode.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamAudioSourceNode processing is correct</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function createBuffer(context) {
+  var buffer = context.createBuffer(2, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    buffer.getChannelData(1)[i] = -buffer.getChannelData(0)[i];
+  }
+  return buffer;
+}
+
+var gTest = {
+  length: 2048,
+  skipOfflineContextTests: true,
+  createGraph(context) {
+    var sourceGraph = new AudioContext();
+    var source = sourceGraph.createBufferSource();
+    source.buffer = createBuffer(context);
+    var dest = sourceGraph.createMediaStreamDestination();
+    source.connect(dest);
+    source.start(0);
+
+    var mediaStreamSource = new MediaStreamAudioSourceNode(context, { mediaStream: dest.stream });
+    // channelCount and channelCountMode should have no effect
+    mediaStreamSource.channelCount = 1;
+    mediaStreamSource.channelCountMode = "explicit";
+    return mediaStreamSource;
+  },
+  createExpectedBuffers(context) {
+    return createBuffer(context);
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeCrossOrigin.html b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeCrossOrigin.html
new file mode 100644
index 0000000000..aaeaab3159
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeCrossOrigin.html
@@ -0,0 +1,59 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamAudioSourceNode doesn't get data from cross-origin media resources</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+var audio = new Audio("http://example.org:80/tests/dom/media/webaudio/test/small-shot.ogg");
+audio.load();
+var context = new AudioContext();
+audio.onloadedmetadata = function() {
+  var node = context.createMediaStreamSource(audio.mozCaptureStreamUntilEnded());
+  var sp = context.createScriptProcessor(2048, 1);
+  node.connect(sp);
+  var complete = false;
+  var iterationCount = 0;
+
+  // This test ensures we receive at least expectedSampleCount nonzero samples
+  function processSamples(e) {
+    if (complete) {
+      return;
+    }
+
+    if (iterationCount == 0) {
+      // Don't start playing the audio until the AudioContext stuff is connected
+      // and running.
+      audio.play();
+    }
+    ++iterationCount;
+
+    var buf = e.inputBuffer.getChannelData(0);
+    var nonzeroSamplesThisBuffer = 0;
+    for (var i = 0; i < buf.length; ++i) {
+      if (buf[i] != 0) {
+        ++nonzeroSamplesThisBuffer;
+      }
+    }
+    is(nonzeroSamplesThisBuffer, 0,
+       "Checking all samples are zero");
+    if (iterationCount >= 20) {
+      SimpleTest.finish();
+      complete = true;
+    }
+  }
+
+  audio.oncanplaythrough = function() {
+    sp.onaudioprocess = processSamples;
+  };
+}
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeNoGC.html b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeNoGC.html
new file mode 100644
index 0000000000..7920af9f7b
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeNoGC.html
@@ -0,0 +1,116 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test that MediaStreamAudioSourceNode and its input MediaStream stays alive while there are active tracks</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("gUM and WebAudio data is async to main thread. " +
+                               "We need a timeout to see that something does " +
+                               "NOT happen to data.");
+
+let context = new AudioContext();
+let analyser = context.createAnalyser();
+
+function wait(millis, resolveWithThis) {
+  return new Promise(resolve => setTimeout(() => resolve(resolveWithThis), millis));
+}
+
+function binIndexForFrequency(frequency) {
+  return 1 + Math.round(frequency * analyser.fftSize / context.sampleRate);
+}
+
+function waitForAudio(analysisFunction, cancelPromise) {
+  let data = new Uint8Array(analyser.frequencyBinCount);
+  let cancelled = false;
+  let cancelledMsg = "";
+  cancelPromise.then(msg => {
+    cancelled = true;
+    cancelledMsg = msg;
+  });
+  return new Promise((resolve, reject) => {
+    let loop = () => {
+      analyser.getByteFrequencyData(data);
+      if (cancelled) {
+        reject(new Error("waitForAudio cancelled: " + cancelledMsg));
+        return;
+      }
+      if (analysisFunction(data)) {
+        resolve();
+        return;
+      }
+      requestAnimationFrame(loop);
+    };
+    loop();
+  });
+}
+
+async function test(sourceNode) {
+  try {
+    await analyser.connect(context.destination);
+
+    ok(true, "Waiting for audio to pass through the analyser")
+    await waitForAudio(arr => arr[binIndexForFrequency(1000)] > 200,
+                       wait(60000, "Timeout waiting for audio"));
+
+    ok(true, "Audio was detected by the analyser. Forcing CC.");
+    SpecialPowers.forceCC();
+    SpecialPowers.forceGC();
+    SpecialPowers.forceCC();
+    SpecialPowers.forceGC();
+
+    info("Checking that GC didn't destroy the stream or source node");
+    await waitForAudio(arr => arr[binIndexForFrequency(1000)] < 50,
+                       wait(5000, "Timeout waiting for GC (timeout OK)"))
+                  .then(() => Promise.reject("Audio stopped unexpectedly"),
+                        () => Promise.resolve());
+
+    ok(true, "Audio is still flowing");
+  } catch(e) {
+    ok(false, "Error executing test: " + e + (e.stack ? "\n" + e.stack : ""));
+    SimpleTest.finish();
+  }
+}
+
+(async function() {
+  try {
+    await SpecialPowers.pushPrefEnv({
+      set: [
+        // This test expects the fake audio device, specifically for the tones
+        // it outputs. Explicitly disable the audio loopback device and enable
+        // fake streams.
+        ['media.audio_loopback_dev', ''],
+        ['media.navigator.streams.fake', true],
+        ['media.navigator.permission.disabled', true]
+      ]
+    });
+
+    // Test stream source GC
+    let stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    let source = context.createMediaStreamSource(stream);
+    stream = null;
+    source.connect(analyser);
+    await test(source);
+
+    // Test track source GC
+    stream = await navigator.mediaDevices.getUserMedia({audio: true});
+    source = context.createMediaStreamTrackSource(stream.getAudioTracks()[0]);
+    stream = null;
+    source.connect(analyser);
+    await test(source);
+  } catch(e) {
+    ok(false, `Error executing test: ${e}${e.stack ? "\n" + e.stack : ""}`);
+  } finally {
+    context.close();
+    SimpleTest.finish();
+  }
+})();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaStreamAudioSourceNodePassThrough.html b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodePassThrough.html
new file mode 100644
index 0000000000..379bfdbc6a
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodePassThrough.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamAudioSourceNode passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function createBuffer(context, delay) {
+  var buffer = context.createBuffer(2, 2048, context.sampleRate);
+  for (var i = 0; i < 2048 - delay; ++i) {
+    buffer.getChannelData(0)[i + delay] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    buffer.getChannelData(1)[i + delay] = -buffer.getChannelData(0)[i + delay];
+  }
+  return buffer;
+}
+
+var gTest = {
+  length: 2048,
+  skipOfflineContextTests: true,
+  createGraph(context) {
+    var sourceGraph = new AudioContext();
+    var source = sourceGraph.createBufferSource();
+    source.buffer = createBuffer(context, 0);
+    var dest = sourceGraph.createMediaStreamDestination();
+    source.connect(dest);
+    source.start(0);
+
+    var mediaStreamSource = context.createMediaStreamSource(dest.stream);
+    // channelCount and channelCountMode should have no effect
+    mediaStreamSource.channelCount = 1;
+    mediaStreamSource.channelCountMode = "explicit";
+
+    var srcWrapped = SpecialPowers.wrap(mediaStreamSource);
+    ok("passThrough" in srcWrapped, "MediaStreamAudioSourceNode should support the passThrough API");
+    srcWrapped.passThrough = true;
+
+    return mediaStreamSource;
+  },
+  createExpectedBuffers(context) {
+    return context.createBuffer(2, 2048, context.sampleRate);
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeResampling.html b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeResampling.html
new file mode 100644
index 0000000000..efacf1ecc5
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamAudioSourceNodeResampling.html
@@ -0,0 +1,74 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamAudioSourceNode processing is correct</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+function test() {
+  var audio = new Audio("small-shot.ogg");
+  var context = new AudioContext();
+  var expectedMinNonzeroSampleCount;
+  var expectedMaxNonzeroSampleCount;
+  var nonzeroSampleCount = 0;
+  var complete = false;
+  var iterationCount = 0;
+
+  // This test ensures we receive at least expectedSampleCount nonzero samples
+  function processSamples(e) {
+    if (complete) {
+      return;
+    }
+
+    if (iterationCount == 0) {
+      // Don't start playing the audio until the AudioContext stuff is connected
+      // and running.
+      audio.play();
+    }
+    ++iterationCount;
+
+    var buf = e.inputBuffer.getChannelData(0);
+    var nonzeroSamplesThisBuffer = 0;
+    for (var i = 0; i < buf.length; ++i) {
+      if (buf[i] != 0) {
+        ++nonzeroSamplesThisBuffer;
+      }
+    }
+    nonzeroSampleCount += nonzeroSamplesThisBuffer;
+    is(e.inputBuffer.numberOfChannels, 1,
+       "Checking data channel count (nonzeroSamplesThisBuffer=" +
+       nonzeroSamplesThisBuffer + ")");
+    ok(nonzeroSampleCount <= expectedMaxNonzeroSampleCount,
+       "Too many nonzero samples (got " + nonzeroSampleCount + ", expected max " + expectedMaxNonzeroSampleCount + ")");
+    if (nonzeroSampleCount >= expectedMinNonzeroSampleCount &&
+        nonzeroSamplesThisBuffer == 0) {
+      ok(true,
+       "Check received enough nonzero samples (got " + nonzeroSampleCount + ", expected min " + expectedMinNonzeroSampleCount + ")");
+      SimpleTest.finish();
+      complete = true;
+    }
+  }
+
+  audio.onloadedmetadata = function() {
+    var node = context.createMediaStreamSource(audio.mozCaptureStreamUntilEnded());
+    var sp = context.createScriptProcessor(2048, 1, 0);
+    node.connect(sp);
+    // Use a fuzz factor of 100 to account for samples that just happen to be zero
+    expectedMinNonzeroSampleCount = Math.floor(audio.duration*context.sampleRate) - 100;
+    expectedMaxNonzeroSampleCount = Math.floor(audio.duration*context.sampleRate) + 500;
+    sp.onaudioprocess = processSamples;
+  };
+}
+
+SpecialPowers.pushPrefEnv({"set": [["media.preload.default", 2], ["media.preload.auto", 3]]}, test);
+  
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaStreamTrackAudioSourceNode.html b/dom/media/webaudio/test/test_mediaStreamTrackAudioSourceNode.html
new file mode 100644
index 0000000000..350e9e0fab
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamTrackAudioSourceNode.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamTrackAudioSourceNode processing is correct</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function createBuffer(context) {
+  let buffer = context.createBuffer(2, 2048, context.sampleRate);
+  for (let i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    buffer.getChannelData(1)[i] = -buffer.getChannelData(0)[i];
+  }
+  return buffer;
+}
+
+let gTest = {
+  length: 2048,
+  skipOfflineContextTests: true,
+  createGraph(context) {
+    let sourceGraph = new AudioContext();
+    let source = sourceGraph.createBufferSource();
+    source.buffer = createBuffer(context);
+    let dest = sourceGraph.createMediaStreamDestination();
+    source.connect(dest);
+
+    // Extract first audio track from dest.stream
+    let track = dest.stream.getAudioTracks()[0];
+
+    source.start(0);
+
+    let mediaStreamTrackSource = new MediaStreamTrackAudioSourceNode(context, { mediaStreamTrack: track });
+    // channelCount and channelCountMode should have no effect
+    mediaStreamTrackSource.channelCount = 1;
+    mediaStreamTrackSource.channelCountMode = "explicit";
+    return mediaStreamTrackSource;
+  },
+  createExpectedBuffers(context) {
+    return createBuffer(context);
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mediaStreamTrackAudioSourceNodeCrossOrigin.html b/dom/media/webaudio/test/test_mediaStreamTrackAudioSourceNodeCrossOrigin.html
new file mode 100644
index 0000000000..243534de32
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamTrackAudioSourceNodeCrossOrigin.html
@@ -0,0 +1,54 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+<title>Test MediaStreamTrackAudioSourceNode doesn't get data from cross-origin media resources</title>
+<script src="/tests/SimpleTest/SimpleTest.js"></script>
+<link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+const CROSS_ORIGIN_URL = "http://example.org:80/tests/dom/media/webaudio/test/sine-440-10s.opus"
+let iterationCount = 0;
+let context = null;
+let sp;
+
+function processSamples(e) {
+  ++iterationCount;
+
+  let buf = e.inputBuffer.getChannelData(0);
+  let nonzeroSamplesThisBuffer = 0;
+  for (let i = 0; i < buf.length; ++i) {
+    if (buf[i] != 0) {
+      ++nonzeroSamplesThisBuffer;
+    }
+  }
+  is(nonzeroSamplesThisBuffer, 0,
+     "a source that is cross origin cannot be inspected by Web Audio");
+
+  if (iterationCount == 40) {
+    sp.onaudioprocess = null;
+    context.close();
+    SimpleTest.finish();
+  }
+}
+
+let audio = new Audio();
+audio.src = CROSS_ORIGIN_URL;
+audio.onloadedmetadata = function () {
+  context = new AudioContext();
+  let stream = audio.mozCaptureStream();
+  let track = stream.getAudioTracks()[0];
+  let node = context.createMediaStreamTrackSource(track);
+  node.connect(context.destination);
+  sp = context.createScriptProcessor(2048, 1);
+  sp.onaudioprocess = processSamples;
+  node.connect(sp);
+}
+
+</script>
+</pre>
+</body>
diff --git a/dom/media/webaudio/test/test_mediaStreamTrackAudioSourceNodeVideo.html b/dom/media/webaudio/test/test_mediaStreamTrackAudioSourceNodeVideo.html
new file mode 100644
index 0000000000..772b7f8d5f
--- /dev/null
+++ b/dom/media/webaudio/test/test_mediaStreamTrackAudioSourceNodeVideo.html
@@ -0,0 +1,27 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset="utf-8">
+<head>
+  <title>Test MediaStreamTrackAudioSourceNode throw video track</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="/tests/dom/media/webaudio/test/webaudio.js"></script>
+  <script type="text/javascript" src="/tests/dom/media/webrtc/tests/mochitests/head.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+  let context = new AudioContext();
+  let canvas = document.createElement("canvas");
+  canvas.getContext("2d");
+  let track = canvas.captureStream().getTracks()[0];
+
+  expectException(() => {
+    new MediaStreamTrackAudioSourceNode(
+      context,
+      { mediaStreamTrack: track });
+  }, DOMException.INVALID_STATE_ERR);
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_mixingRules.html b/dom/media/webaudio/test/test_mixingRules.html
new file mode 100644
index 0000000000..719175fbfb
--- /dev/null
+++ b/dom/media/webaudio/test/test_mixingRules.html
@@ -0,0 +1,402 @@
+<!DOCTYPE html>
+<html>
+<head>
+  <title>Testcase for AudioNode channel up-mix/down-mix rules</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+
+<body>
+
+<script>
+
+// This test is based on http://src.chromium.org/viewvc/blink/trunk/LayoutTests/webaudio/audionode-channel-rules.html
+
+var context = null;
+var sp = null;
+var renderNumberOfChannels = 8;
+var singleTestFrameLength = 8;
+var testBuffers;
+
+// A list of connections to an AudioNode input, each of which is to be used in one or more specific test cases.
+// Each element in the list is a string, with the number of connections corresponding to the length of the string,
+// and each character in the string is from '1' to '8' representing a 1 to 8 channel connection (from an AudioNode output).
+// For example, the string "128" means 3 connections, having 1, 2, and 8 channels respectively.
+var connectionsList = [];
+for (var i = 1; i <= 8; ++i) {
+  connectionsList.push(i.toString());
+  for (var j = 1; j <= 8; ++j) {
+    connectionsList.push(i.toString() + j.toString());
+  }
+}
+
+// A list of mixing rules, each of which will be tested against all of the connections in connectionsList.
+var mixingRulesList = [
+    {channelCount: 1, channelCountMode: "max", channelInterpretation: "speakers"},
+    {channelCount: 2, channelCountMode: "clamped-max", channelInterpretation: "speakers"},
+    {channelCount: 3, channelCountMode: "clamped-max", channelInterpretation: "speakers"},
+    {channelCount: 4, channelCountMode: "clamped-max", channelInterpretation: "speakers"},
+    {channelCount: 5, channelCountMode: "clamped-max", channelInterpretation: "speakers"},
+    {channelCount: 6, channelCountMode: "clamped-max", channelInterpretation: "speakers"},
+    {channelCount: 7, channelCountMode: "clamped-max", channelInterpretation: "speakers"},
+    {channelCount: 2, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 3, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 4, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 5, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 6, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 7, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 8, channelCountMode: "explicit", channelInterpretation: "speakers"},
+    {channelCount: 1, channelCountMode: "max", channelInterpretation: "discrete"},
+    {channelCount: 2, channelCountMode: "clamped-max", channelInterpretation: "discrete"},
+    {channelCount: 3, channelCountMode: "clamped-max", channelInterpretation: "discrete"},
+    {channelCount: 4, channelCountMode: "clamped-max", channelInterpretation: "discrete"},
+    {channelCount: 5, channelCountMode: "clamped-max", channelInterpretation: "discrete"},
+    {channelCount: 6, channelCountMode: "clamped-max", channelInterpretation: "discrete"},
+    {channelCount: 3, channelCountMode: "explicit", channelInterpretation: "discrete"},
+    {channelCount: 4, channelCountMode: "explicit", channelInterpretation: "discrete"},
+    {channelCount: 5, channelCountMode: "explicit", channelInterpretation: "discrete"},
+    {channelCount: 6, channelCountMode: "explicit", channelInterpretation: "discrete"},
+    {channelCount: 7, channelCountMode: "explicit", channelInterpretation: "discrete"},
+    {channelCount: 8, channelCountMode: "explicit", channelInterpretation: "discrete"},
+];
+
+var numberOfTests = mixingRulesList.length * connectionsList.length;
+
+// Create an n-channel buffer, with all sample data zero except for a shifted impulse.
+// The impulse position depends on the channel index.
+// For example, for a 4-channel buffer:
+// channel0: 1 0 0 0 0 0 0 0
+// channel1: 0 1 0 0 0 0 0 0
+// channel2: 0 0 1 0 0 0 0 0
+// channel3: 0 0 0 1 0 0 0 0
+function createTestBuffer(numberOfChannels) {
+    var buffer = context.createBuffer(numberOfChannels, singleTestFrameLength, context.sampleRate);
+    for (var i = 0; i < numberOfChannels; ++i) {
+        var data = buffer.getChannelData(i);
+        data[i] = 1;
+    }
+    return buffer;
+}
+
+// Discrete channel interpretation mixing:
+// https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/specification.html#UpMix
+// up-mix by filling channels until they run out then ignore remaining dest channels.
+// down-mix by filling as many channels as possible, then dropping remaining source channels.
+function discreteSum(sourceBuffer, destBuffer) {
+    if (sourceBuffer.length != destBuffer.length) {
+        is(sourceBuffer.length, destBuffer.length, "source and destination buffers should have the same length");
+    }
+
+    var numberOfChannels = Math.min(sourceBuffer.numberOfChannels, destBuffer.numberOfChannels);
+    var length = sourceBuffer.length;
+
+    for (var c = 0; c < numberOfChannels; ++c) {
+        var source = sourceBuffer.getChannelData(c);
+        var dest = destBuffer.getChannelData(c);
+        for (var i = 0; i < length; ++i) {
+            dest[i] += source[i];
+        }
+    }
+}
+
+// Speaker channel interpretation mixing:
+// https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/specification.html#UpMix
+// eslint-disable-next-line complexity
+function speakersSum(sourceBuffer, destBuffer)
+{
+    var numberOfSourceChannels = sourceBuffer.numberOfChannels;
+    var numberOfDestinationChannels = destBuffer.numberOfChannels;
+    var length = destBuffer.length;
+
+    if ((numberOfDestinationChannels == 2 && numberOfSourceChannels == 1) ||
+        (numberOfDestinationChannels == 4 && numberOfSourceChannels == 1)) {
+        // Handle mono -> stereo/Quad case (summing mono channel into both left and right).
+        var source = sourceBuffer.getChannelData(0);
+        var destL = destBuffer.getChannelData(0);
+        var destR = destBuffer.getChannelData(1);
+
+        for (var i = 0; i < length; ++i) {
+            destL[i] += source[i];
+            destR[i] += source[i];
+        }
+        } else if ((numberOfDestinationChannels == 4 && numberOfSourceChannels == 2) ||
+                   (numberOfDestinationChannels == 6 && numberOfSourceChannels == 2)) {
+        // Handle stereo -> Quad/5.1 case (summing left and right channels into the output's left and right).
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var destL = destBuffer.getChannelData(0);
+        var destR = destBuffer.getChannelData(1);
+
+        for (var i = 0; i < length; ++i) {
+            destL[i] += sourceL[i];
+            destR[i] += sourceR[i];
+        }
+    } else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 2) {
+        // Handle stereo -> mono case. output += 0.5 * (input.L + input.R).
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var dest = destBuffer.getChannelData(0);
+
+        for (var i = 0; i < length; ++i) {
+            dest[i] += 0.5 * (sourceL[i] + sourceR[i]);
+        }
+    } else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 4) {
+        // Handle Quad -> mono case. output += 0.25 * (input.L + input.R + input.SL + input.SR).
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var sourceSL = sourceBuffer.getChannelData(2);
+        var sourceSR = sourceBuffer.getChannelData(3);
+        var dest = destBuffer.getChannelData(0);
+
+        for (var i = 0; i < length; ++i) {
+            dest[i] += 0.25 * (sourceL[i] + sourceR[i] + sourceSL[i] + sourceSR[i]);
+        }
+    } else if (numberOfDestinationChannels == 2 && numberOfSourceChannels == 4) {
+        // Handle Quad -> stereo case. outputLeft += 0.5 * (input.L + input.SL),
+        //                             outputRight += 0.5 * (input.R + input.SR).
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var sourceSL = sourceBuffer.getChannelData(2);
+        var sourceSR = sourceBuffer.getChannelData(3);
+        var destL = destBuffer.getChannelData(0);
+        var destR = destBuffer.getChannelData(1);
+
+        for (var i = 0; i < length; ++i) {
+            destL[i] += 0.5 * (sourceL[i] + sourceSL[i]);
+            destR[i] += 0.5 * (sourceR[i] + sourceSR[i]);
+        }
+    } else if (numberOfDestinationChannels == 6 && numberOfSourceChannels == 4) {
+        // Handle Quad -> 5.1 case. outputLeft += (inputL, inputR, 0, 0, inputSL, inputSR)
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var sourceSL = sourceBuffer.getChannelData(2);
+        var sourceSR = sourceBuffer.getChannelData(3);
+        var destL = destBuffer.getChannelData(0);
+        var destR = destBuffer.getChannelData(1);
+        var destSL = destBuffer.getChannelData(4);
+        var destSR = destBuffer.getChannelData(5);
+
+        for (var i = 0; i < length; ++i) {
+            destL[i] += sourceL[i];
+            destR[i] += sourceR[i];
+            destSL[i] += sourceSL[i];
+            destSR[i] += sourceSR[i];
+        }
+    } else if (numberOfDestinationChannels == 6 && numberOfSourceChannels == 1) {
+        // Handle mono -> 5.1 case, sum mono channel into center.
+        var source = sourceBuffer.getChannelData(0);
+        var dest = destBuffer.getChannelData(2);
+
+        for (var i = 0; i < length; ++i) {
+            dest[i] += source[i];
+        }
+    } else if (numberOfDestinationChannels == 1 && numberOfSourceChannels == 6) {
+        // Handle 5.1 -> mono.
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var sourceC = sourceBuffer.getChannelData(2);
+        // skip LFE for now, according to current spec.
+        var sourceSL = sourceBuffer.getChannelData(4);
+        var sourceSR = sourceBuffer.getChannelData(5);
+        var dest = destBuffer.getChannelData(0);
+
+        for (var i = 0; i < length; ++i) {
+            dest[i] += 0.7071 * (sourceL[i] + sourceR[i]) + sourceC[i] + 0.5 * (sourceSL[i] + sourceSR[i]);
+        }
+    } else if (numberOfDestinationChannels == 2 && numberOfSourceChannels == 6) {
+        // Handle 5.1 -> stereo.
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var sourceC = sourceBuffer.getChannelData(2);
+        // skip LFE for now, according to current spec.
+        var sourceSL = sourceBuffer.getChannelData(4);
+        var sourceSR = sourceBuffer.getChannelData(5);
+        var destL = destBuffer.getChannelData(0);
+        var destR = destBuffer.getChannelData(1);
+
+        for (var i = 0; i < length; ++i) {
+            destL[i] += sourceL[i] + 0.7071 * (sourceC[i] + sourceSL[i]);
+            destR[i] += sourceR[i] + 0.7071 * (sourceC[i] + sourceSR[i]);
+        }
+    } else if (numberOfDestinationChannels == 4 && numberOfSourceChannels == 6) {
+        // Handle 5.1 -> Quad.
+        var sourceL = sourceBuffer.getChannelData(0);
+        var sourceR = sourceBuffer.getChannelData(1);
+        var sourceC = sourceBuffer.getChannelData(2);
+        // skip LFE for now, according to current spec.
+        var sourceSL = sourceBuffer.getChannelData(4);
+        var sourceSR = sourceBuffer.getChannelData(5);
+        var destL = destBuffer.getChannelData(0);
+        var destR = destBuffer.getChannelData(1);
+        var destSL = destBuffer.getChannelData(2);
+        var destSR = destBuffer.getChannelData(3);
+
+        for (var i = 0; i < length; ++i) {
+            destL[i] += sourceL[i] + 0.7071 * sourceC[i];
+            destR[i] += sourceR[i] + 0.7071 * sourceC[i];
+            destSL[i] += sourceSL[i];
+            destSR[i] += sourceSR[i];
+        }
+    } else {
+        // Fallback for unknown combinations.
+        discreteSum(sourceBuffer, destBuffer);
+    }
+}
+
+function scheduleTest(testNumber, connections, channelCount, channelCountMode, channelInterpretation) {
+    var mixNode = context.createGain();
+    mixNode.channelCount = channelCount;
+    mixNode.channelCountMode = channelCountMode;
+    mixNode.channelInterpretation = channelInterpretation;
+    mixNode.connect(sp);
+
+    for (var i = 0; i < connections.length; ++i) {
+        var connectionNumberOfChannels = connections.charCodeAt(i) - "0".charCodeAt(0);
+
+        var source = context.createBufferSource();
+        // Get a buffer with the right number of channels, converting from 1-based to 0-based index.
+        var buffer = testBuffers[connectionNumberOfChannels - 1];
+        source.buffer = buffer;
+        source.connect(mixNode);
+
+        // Start at the right offset.
+        var sampleFrameOffset = testNumber * singleTestFrameLength;
+        var time = sampleFrameOffset / context.sampleRate;
+        source.start(time);
+    }
+}
+
+function computeNumberOfChannels(connections, channelCount, channelCountMode) {
+    if (channelCountMode == "explicit")
+        return channelCount;
+
+    var computedNumberOfChannels = 1; // Must have at least one channel.
+
+    // Compute "computedNumberOfChannels" based on all the connections.
+    for (var i = 0; i < connections.length; ++i) {
+        var connectionNumberOfChannels = connections.charCodeAt(i) - "0".charCodeAt(0);
+        computedNumberOfChannels = Math.max(computedNumberOfChannels, connectionNumberOfChannels);
+    }
+
+    if (channelCountMode == "clamped-max")
+        computedNumberOfChannels = Math.min(computedNumberOfChannels, channelCount);
+
+    return computedNumberOfChannels;
+}
+
+function checkTestResult(renderedBuffer, testNumber, connections, channelCount, channelCountMode, channelInterpretation) {
+    var computedNumberOfChannels = computeNumberOfChannels(connections, channelCount, channelCountMode);
+
+    // Create a zero-initialized silent AudioBuffer with computedNumberOfChannels.
+    var destBuffer = context.createBuffer(computedNumberOfChannels, singleTestFrameLength, context.sampleRate);
+
+    // Mix all of the connections into the destination buffer.
+    for (var i = 0; i < connections.length; ++i) {
+        var connectionNumberOfChannels = connections.charCodeAt(i) - "0".charCodeAt(0);
+        var sourceBuffer = testBuffers[connectionNumberOfChannels - 1]; // convert from 1-based to 0-based index
+
+        if (channelInterpretation == "speakers") {
+            speakersSum(sourceBuffer, destBuffer);
+        } else if (channelInterpretation == "discrete") {
+            discreteSum(sourceBuffer, destBuffer);
+        } else {
+            ok(false, "Invalid channel interpretation!");
+        }
+    }
+
+    // Validate that destBuffer matches the rendered output.
+    // We need to check the rendered output at a specific sample-frame-offset corresponding
+    // to the specific test case we're checking for based on testNumber.
+
+    var sampleFrameOffset = testNumber * singleTestFrameLength;
+    for (var c = 0; c < renderNumberOfChannels; ++c) {
+        var renderedData = renderedBuffer.getChannelData(c);
+        for (var frame = 0; frame < singleTestFrameLength; ++frame) {
+            var renderedValue = renderedData[frame + sampleFrameOffset];
+
+            var expectedValue = 0;
+            if (c < destBuffer.numberOfChannels) {
+                var expectedData = destBuffer.getChannelData(c);
+                expectedValue = expectedData[frame];
+            }
+
+            if (Math.abs(renderedValue - expectedValue) > 1e-4) {
+                var s = "connections: " + connections + ", " + channelCountMode;
+
+                // channelCount is ignored in "max" mode.
+                if (channelCountMode == "clamped-max" || channelCountMode == "explicit") {
+                    s += "(" + channelCount + ")";
+                }
+
+                s += ", " + channelInterpretation + ". ";
+
+                var message = s + "rendered: " + renderedValue + " expected: " + expectedValue + " channel: " + c + " frame: " + frame;
+                is(renderedValue, expectedValue, message);
+            }
+        }
+    }
+}
+
+function checkResult(event) {
+    var buffer = event.inputBuffer;
+
+    // Sanity check result.
+    ok(buffer.length != numberOfTests * singleTestFrameLength ||
+       buffer.numberOfChannels != renderNumberOfChannels, "Sanity check");
+
+    // Check all the tests.
+    var testNumber = 0;
+    for (var m = 0; m < mixingRulesList.length; ++m) {
+        var mixingRules = mixingRulesList[m];
+        for (var i = 0; i < connectionsList.length; ++i, ++testNumber) {
+            checkTestResult(buffer, testNumber, connectionsList[i], mixingRules.channelCount, mixingRules.channelCountMode, mixingRules.channelInterpretation);
+        }
+    }
+
+    sp.onaudioprocess = null;
+    SimpleTest.finish();
+}
+
+SimpleTest.waitForExplicitFinish();
+function runTest() {
+    // Create 8-channel offline audio context.
+    // Each test will render 8 sample-frames starting at sample-frame position testNumber * 8.
+    var totalFrameLength = numberOfTests * singleTestFrameLength;
+    context = new AudioContext();
+    var nextPowerOfTwo = 256;
+    while (nextPowerOfTwo < totalFrameLength) {
+        nextPowerOfTwo *= 2;
+    }
+    sp = context.createScriptProcessor(nextPowerOfTwo, renderNumberOfChannels);
+
+    // Set destination to discrete mixing.
+    sp.channelCount = renderNumberOfChannels;
+    sp.channelCountMode = "explicit";
+    sp.channelInterpretation = "discrete";
+
+    // Create test buffers from 1 to 8 channels.
+    testBuffers = new Array();
+    for (var i = 0; i < renderNumberOfChannels; ++i) {
+        testBuffers[i] = createTestBuffer(i + 1);
+    }
+
+    // Schedule all the tests.
+    var testNumber = 0;
+    for (var m = 0; m < mixingRulesList.length; ++m) {
+        var mixingRules = mixingRulesList[m];
+        for (var i = 0; i < connectionsList.length; ++i, ++testNumber) {
+            scheduleTest(testNumber, connectionsList[i], mixingRules.channelCount, mixingRules.channelCountMode, mixingRules.channelInterpretation);
+        }
+    }
+
+    // Render then check results.
+    sp.onaudioprocess = checkResult;
+}
+
+runTest();
+
+</script>
+
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_nodeCreationDocumentGone.html b/dom/media/webaudio/test/test_nodeCreationDocumentGone.html
new file mode 100644
index 0000000000..07a4f7a97d
--- /dev/null
+++ b/dom/media/webaudio/test/test_nodeCreationDocumentGone.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.requestCompleteLog();
+SimpleTest.waitForExplicitFinish();
+
+var a = window.open("file_nodeCreationDocumentGone.html");
+a.onbeforeunload = function() {
+  setTimeout(function(){
+    try {
+      a.context.createScriptProcessor(512, 1, 1);
+    } catch(e) {
+      ok (true,"got exception");
+    }
+    setTimeout(function() {
+      ok (true,"no crash");
+      SimpleTest.finish();
+    }, 0);
+  }, 0);
+}
+
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_nodeToParamConnection.html b/dom/media/webaudio/test/test_nodeToParamConnection.html
new file mode 100644
index 0000000000..49098eaffe
--- /dev/null
+++ b/dom/media/webaudio/test/test_nodeToParamConnection.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test connecting an AudioNode to an AudioParam</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  createGraph(context) {
+    var sourceBuffer = context.createBuffer(2, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      sourceBuffer.getChannelData(0)[i] = 1;
+      sourceBuffer.getChannelData(1)[i] = -1;
+    }
+
+    var paramSource = context.createBufferSource();
+    paramSource.buffer = this.buffer;
+
+    var source = context.createBufferSource();
+    source.buffer = sourceBuffer;
+
+    var gain = context.createGain();
+
+    paramSource.connect(gain.gain);
+    source.connect(gain);
+
+    paramSource.start(0);
+    source.start(0);
+    return gain;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(2, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      for (var j = 0; j < 2; ++j) {
+        this.buffer.getChannelData(j)[i] = Math.sin(440 * 2 * (j + 1) * Math.PI * i / context.sampleRate);
+      }
+    }
+    var expectedBuffer = context.createBuffer(2, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = 1 + (this.buffer.getChannelData(0)[i] + this.buffer.getChannelData(1)[i]) / 2;
+      expectedBuffer.getChannelData(1)[i] = -(1 + (this.buffer.getChannelData(0)[i] + this.buffer.getChannelData(1)[i]) / 2);
+    }
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_notAllowedToStartAudioContextGC.html b/dom/media/webaudio/test/test_notAllowedToStartAudioContextGC.html
new file mode 100644
index 0000000000..b8715c1644
--- /dev/null
+++ b/dom/media/webaudio/test/test_notAllowedToStartAudioContextGC.html
@@ -0,0 +1,57 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test GC for not-allow-to-start audio context</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.requestFlakyTimeout(`Checking that something does not happen`);
+
+SimpleTest.waitForExplicitFinish();
+
+var destId;
+
+function observer(subject, topic, data) {
+  let id = parseInt(data);
+  ok(id != destId, "dropping another node, not the context's destination");
+}
+
+SpecialPowers.addAsyncObserver(observer, "webaudio-node-demise", false);
+SimpleTest.registerCleanupFunction(function() {
+  SpecialPowers.removeAsyncObserver(observer, "webaudio-node-demise");
+});
+
+SpecialPowers.pushPrefEnv({"set": [["media.autoplay.default", SpecialPowers.Ci.nsIAutoplay.BLOCKED],
+                                   ["media.autoplay.blocking_policy", 0]]},
+                          startTest);
+
+function startTest() {
+  info("- create audio context -");
+  let ac = new AudioContext();
+
+  info("- get node Id -");
+  destId = SpecialPowers.getPrivilegedProps(ac.destination, "id");
+
+  info("- trigger GCs -");
+  SpecialPowers.forceGC();
+  SpecialPowers.forceCC();
+  SpecialPowers.forceGC();
+
+  info("- after three GCs -");
+
+  // We're doing this async so that we can receive observerservice messages.
+  setTimeout(function() {
+    ok(true, `AudioContext that has been prevented
+              from starting has correctly survived GC`)
+    SimpleTest.finish();
+  }, 1);
+}
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_offlineDestinationChannelCountLess.html b/dom/media/webaudio/test/test_offlineDestinationChannelCountLess.html
new file mode 100644
index 0000000000..8ff1deac4b
--- /dev/null
+++ b/dom/media/webaudio/test/test_offlineDestinationChannelCountLess.html
@@ -0,0 +1,42 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test OfflineAudioContext with a channel count less than the specified number</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ctx = new OfflineAudioContext(2, 100, 22050);
+
+  var buf = ctx.createBuffer(6, 100, ctx.sampleRate);
+  for (var i = 0; i < 6; ++i) {
+    for (var j = 0; j < 100; ++j) {
+      buf.getChannelData(i)[j] = Math.sin(2 * Math.PI * 200 * j / ctx.sampleRate);
+    }
+  }
+
+  var src = ctx.createBufferSource();
+  src.buffer = buf;
+  src.start(0);
+  src.connect(ctx.destination);
+  ctx.destination.channelCountMode = "max";
+  ctx.startRendering();
+  ctx.oncomplete = function(e) {
+    is(e.renderedBuffer.numberOfChannels, 2, "Correct expected number of buffers");
+    compareChannels(e.renderedBuffer.getChannelData(0), buf.getChannelData(0));
+    compareChannels(e.renderedBuffer.getChannelData(1), buf.getChannelData(1));
+
+    SimpleTest.finish();
+  };
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_offlineDestinationChannelCountMore.html b/dom/media/webaudio/test/test_offlineDestinationChannelCountMore.html
new file mode 100644
index 0000000000..fa38114e2b
--- /dev/null
+++ b/dom/media/webaudio/test/test_offlineDestinationChannelCountMore.html
@@ -0,0 +1,46 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test OfflineAudioContext with a channel count less than the specified number</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var ctx = new OfflineAudioContext(6, 100, 22050);
+
+  var buf = ctx.createBuffer(2, 100, ctx.sampleRate);
+  for (var i = 0; i < 2; ++i) {
+    for (var j = 0; j < 100; ++j) {
+      buf.getChannelData(i)[j] = Math.sin(2 * Math.PI * 200 * j / ctx.sampleRate);
+    }
+  }
+  var emptyBuffer = ctx.createBuffer(1, 100, ctx.sampleRate);
+
+  var src = ctx.createBufferSource();
+  src.buffer = buf;
+  src.start(0);
+  src.connect(ctx.destination);
+  ctx.destination.channelCountMode = "max";
+  ctx.startRendering();
+  ctx.oncomplete = function(e) {
+    is(e.renderedBuffer.numberOfChannels, 6, "Correct expected number of buffers");
+    compareChannels(e.renderedBuffer.getChannelData(0), buf.getChannelData(0));
+    compareChannels(e.renderedBuffer.getChannelData(1), buf.getChannelData(1));
+    for (var i = 2; i < 6; ++i) {
+      compareChannels(e.renderedBuffer.getChannelData(i), emptyBuffer.getChannelData(0));
+    }
+
+    SimpleTest.finish();
+  };
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_oscillatorNode.html b/dom/media/webaudio/test/test_oscillatorNode.html
new file mode 100644
index 0000000000..e2a47a4e1e
--- /dev/null
+++ b/dom/media/webaudio/test/test_oscillatorNode.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the OscillatorNode interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+  var context = new AudioContext();
+  var osc = new OscillatorNode(context);
+
+  is(osc.channelCount, 2, "Oscillator node has 2 input channels by default");
+  is(osc.channelCountMode, "max", "Correct channelCountMode for the Oscillator node");
+  is(osc.channelInterpretation, "speakers", "Correct channelCountInterpretation for the Oscillator node");
+  is(osc.type, "sine", "Correct default type");
+  expectException(function() {
+    osc.type = "custom";
+  }, DOMException.INVALID_STATE_ERR);
+  is(osc.type, "sine", "Cannot set the type to custom");
+  is(osc.frequency.value, 440, "Correct default frequency value");
+  is(osc.detune.value, 0, "Correct default detine value");
+
+  // Make sure that we can set all of the valid type values
+  var types = [
+    "sine",
+    "square",
+    "sawtooth",
+    "triangle",
+  ];
+  for (var i = 0; i < types.length; ++i) {
+    osc.type = types[i];
+  }
+
+  // Verify setPeriodicWave()
+  var real = new Float32Array([1.0, 0.5, 0.25, 0.125]);
+  var imag = new Float32Array([1.0, 0.7, -1.0, 0.5]);
+  osc.setPeriodicWave(context.createPeriodicWave(real, imag));
+  is(osc.type, "custom", "Failed to set custom waveform");
+
+  expectNoException(function() {
+    osc.start();
+  });
+  expectNoException(function() {
+    osc.stop();
+  });
+
+  SimpleTest.finish();
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_oscillatorNode2.html b/dom/media/webaudio/test/test_oscillatorNode2.html
new file mode 100644
index 0000000000..69a6655ff1
--- /dev/null
+++ b/dom/media/webaudio/test/test_oscillatorNode2.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test OscillatorNode lifetime and sine phase</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+const signalLength = 2048;
+
+function createOscillator(context) {
+  var osc = context.createOscillator();
+  osc.start(0);
+  osc.stop(signalLength/context.sampleRate);
+  return osc;
+}
+
+function connectUnreferencedOscillator(context, destination) {
+  var osc = createOscillator(context);
+  osc.connect(destination);
+}
+
+var gTest = {
+  length: signalLength,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var blend = context.createGain();
+
+    connectUnreferencedOscillator(context, blend);
+    // Test that the unreferenced oscillator remains alive until it has finished.
+    SpecialPowers.forceGC();
+    SpecialPowers.forceCC();
+
+    // Create another sine wave oscillator in negative time, which should
+    // cancel when mixed with the unreferenced oscillator.
+    var oscillator = createOscillator(context);
+    oscillator.frequency.value = -440;
+    oscillator.connect(blend);
+
+    return blend;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_oscillatorNodeNegativeFrequency.html b/dom/media/webaudio/test/test_oscillatorNodeNegativeFrequency.html
new file mode 100644
index 0000000000..c46c0fea13
--- /dev/null
+++ b/dom/media/webaudio/test/test_oscillatorNodeNegativeFrequency.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the OscillatorNode when the frequency is negative</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+  var types = ["sine",
+               "square",
+               "sawtooth",
+               "triangle"];
+
+  var finished = 0;
+  function finish() {
+    if (++finished == types.length) {
+      SimpleTest.finish();
+    }
+  }
+
+  types.forEach(function(t) {
+    var context = new OfflineAudioContext(1, 256, 44100);
+    var osc = context.createOscillator();
+
+    osc.frequency.value = -440;
+    osc.type = t;
+
+    osc.connect(context.destination);
+    osc.start();
+    context.startRendering().then(function(buffer) {
+      var samples = buffer.getChannelData(0);
+      // This samples the wave form in the middle of the first period, the value
+      // should be negative.
+      ok(samples[Math.floor(44100 / 440 / 4)] < 0., "Phase should be inverted when using a " + t + " waveform");
+      finish();
+    });
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_oscillatorNodePassThrough.html b/dom/media/webaudio/test/test_oscillatorNodePassThrough.html
new file mode 100644
index 0000000000..63c0848d06
--- /dev/null
+++ b/dom/media/webaudio/test/test_oscillatorNodePassThrough.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test Oscillator with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    var source = context.createOscillator();
+
+    var srcWrapped = SpecialPowers.wrap(source);
+    ok("passThrough" in srcWrapped, "OscillatorNode should support the passThrough API");
+    srcWrapped.passThrough = true;
+
+    source.start(0);
+    return source;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+
+    return [expectedBuffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_oscillatorNodeStart.html b/dom/media/webaudio/test/test_oscillatorNodeStart.html
new file mode 100644
index 0000000000..4df129170f
--- /dev/null
+++ b/dom/media/webaudio/test/test_oscillatorNodeStart.html
@@ -0,0 +1,38 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the OscillatorNode interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+
+  var context = new AudioContext();
+  var osc = context.createOscillator();
+  var sp = context.createScriptProcessor(0, 1, 0);
+
+  osc.connect(sp);
+
+  sp.onaudioprocess = function (e) {
+    var input = e.inputBuffer.getChannelData(0);
+    var isSilent = true;
+    for (var i = 0; i < input.length; i++) {
+      if (input[i] != 0.0) {
+        isSilent = false;
+      }
+    }
+    sp.onaudioprocess = null;
+    ok(isSilent, "OscillatorNode should be silent before calling start.");
+    SimpleTest.finish();
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_oscillatorTypeChange.html b/dom/media/webaudio/test/test_oscillatorTypeChange.html
new file mode 100644
index 0000000000..e4b4944703
--- /dev/null
+++ b/dom/media/webaudio/test/test_oscillatorTypeChange.html
@@ -0,0 +1,58 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test OscillatorNode type change after it has started and triangle phase</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+const bufferSize = 1024;
+
+function startTest() {
+  var ctx = new AudioContext();
+
+  var oscillator1 = ctx.createOscillator();
+  oscillator1.connect(ctx.destination);
+  oscillator1.start(0);
+
+  // Assuming the above Web Audio operations have already scheduled an event
+  // to run in stable state and start the graph thread, schedule a subsequent
+  // event to change the type of oscillator1.
+  SimpleTest.executeSoon(function() {
+    oscillator1.type = "triangle";
+
+    // Another triangle wave with -1 gain should cancel the first.  This is
+    // starting at the same time as the type change, assuming that the phase
+    // is reset on type change.  A negative frequency should achieve the same
+    // as the -1 gain but for bug 916285.
+    var oscillator2 = ctx.createOscillator();
+    oscillator2.type = "triangle";
+    oscillator2.start(0);
+
+    var processor = ctx.createScriptProcessor(bufferSize, 1, 0);
+    oscillator1.connect(processor);
+    var gain = ctx.createGain();
+    gain.gain.value = -1;
+    gain.connect(processor);
+    oscillator2.connect(gain);
+
+    processor.onaudioprocess = function(e) {
+      compareChannels(e.inputBuffer.getChannelData(0),
+                      new Float32Array(bufferSize));
+      e.target.onaudioprocess = null;
+      SimpleTest.finish();
+    }
+  });
+};
+
+startTest();
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNode.html b/dom/media/webaudio/test/test_pannerNode.html
new file mode 100644
index 0000000000..7f4d3ea915
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNode.html
@@ -0,0 +1,71 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+function near(a, b, msg) {
+  ok(Math.abs(a - b) < 1e-4, msg);
+}
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var destination = context.destination;
+
+  var source = context.createBufferSource();
+
+  var panner = new PannerNode(context);
+
+  source.buffer = buffer;
+
+  source.connect(panner);
+  panner.connect(destination);
+
+  // Verify default values
+  is(panner.panningModel, "equalpower", "Correct default value for panning model");
+  is(panner.distanceModel, "inverse", "Correct default value for distance model");
+  near(panner.refDistance, 1, "Correct default value for ref distance");
+  near(panner.maxDistance, 10000, "Correct default value for max distance");
+  near(panner.rolloffFactor, 1, "Correct default value for rolloff factor");
+  near(panner.coneInnerAngle, 360, "Correct default value for cone inner angle");
+  near(panner.coneOuterAngle, 360, "Correct default value for cone outer angle");
+  near(panner.coneOuterGain, 0, "Correct default value for cone outer gain");
+  is(panner.channelCount, 2, "panner node has 2 input channels by default");
+  is(panner.channelCountMode, "clamped-max", "Correct channelCountMode for the panner node");
+  is(panner.channelInterpretation, "speakers", "Correct channelCountInterpretation for the panner node");
+
+  panner.setPosition(1, 1, 1);
+  near(panner.positionX.value, 1, "setPosition sets AudioParam properly");
+  near(panner.positionY.value, 1, "setPosition sets AudioParam properly");
+  near(panner.positionZ.value, 1, "setPosition sets AudioParam properly");
+
+  panner.setOrientation(0, 1, 0);
+  near(panner.orientationX.value, 0, "setOrientation sets AudioParam properly");
+  near(panner.orientationY.value, 1, "setOrientation sets AudioParam properly");
+  near(panner.orientationZ.value, 0, "setOrientation sets AudioParam properly");
+
+  source.start(0);
+  SimpleTest.executeSoon(function() {
+    source.stop(0);
+    source.disconnect();
+    panner.disconnect();
+
+    SimpleTest.finish();
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNodeAbove.html b/dom/media/webaudio/test/test_pannerNodeAbove.html
new file mode 100644
index 0000000000..5931fa04de
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNodeAbove.html
@@ -0,0 +1,50 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode directly above</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  numberOfChannels: 2,
+  createGraph(context) {
+    // An up vector will be made perpendicular to the front vector, in the
+    // front-up plane.
+    context.listener.setOrientation(0, 6.311749985202524e+307, 0, 0.1, 1000, 0);
+    // Linearly dependent vectors are ignored.
+    context.listener.setOrientation(0, 0, -6.311749985202524e+307, 0, 0, 6.311749985202524e+307);
+    var panner = context.createPanner();
+    panner.positionX.value = 2; // directly above
+    panner.rolloffFactor = 0; // no distance gain
+    panner.panningModel = "equalpower"; // no effect when directly above
+
+    var source = context.createBufferSource();
+    source.buffer = this.buffer;
+    source.connect(panner);
+    source.start(0);
+
+    return panner;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(2, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      // Different signals in left and right buffers
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+      expectedBuffer.getChannelData(1)[i] = Math.sin(220 * 2 * Math.PI * i / context.sampleRate);
+    }
+    this.buffer = expectedBuffer;
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNodeAtZeroDistance.html b/dom/media/webaudio/test/test_pannerNodeAtZeroDistance.html
new file mode 100644
index 0000000000..b5aff589ee
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNodeAtZeroDistance.html
@@ -0,0 +1,140 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode produces output even when the even when the distance is from the listener is zero</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+const BUF_SIZE = 1024;
+const sampleRate = 44100;
+var types = [
+  "equalpower",
+  "HRTF"
+]
+
+async function testMono(panningModel) {
+  var ac = new OfflineAudioContext(2, BUF_SIZE, sampleRate);
+
+  // A sine to be used to fill the buffers
+  function sine(t) {
+    return Math.sin(440 * 2 * Math.PI * t / ac.sampleRate);
+  }
+
+  var monoBuffer = ac.createBuffer(1, BUF_SIZE, ac.sampleRate);
+  for (var i = 0; i < BUF_SIZE; ++i) {
+    monoBuffer.getChannelData(0)[i] = sine(i);
+  }
+
+  var monoSource = ac.createBufferSource();
+  monoSource.buffer = monoBuffer;
+  monoSource.start(0);
+
+  var panner = new PannerNode(ac, {
+    panningModel,
+    distanceModel: "linear",
+  });
+  monoSource.connect(panner);
+
+  var panner2 = new PannerNode(ac, {
+    panningModel,
+    distanceModel: "inverse",
+  });
+  panner.connect(panner2);
+
+  var panner3 = new PannerNode(ac, {
+    panningModel,
+    distanceModel: "exponential"
+  });
+  panner2.connect(panner3);
+
+  panner3.connect(ac.destination);
+
+  const buffer = await ac.startRendering();
+
+  if (panningModel == "equalpower") {
+    // Use the input buffer to compare the output. According to the spec,
+    // mono input at zero distance will apply gain = cos(0.5 * Math.PI / 2)
+    // https://webaudio.github.io/web-audio-api/#Spatialzation-equal-power-panning
+    const gain = Math.cos(0.5 * Math.PI / 2);
+    for (var i = 0; i < BUF_SIZE; ++i) {
+      monoBuffer.getChannelData(0)[i] = gain * monoBuffer.getChannelData(0)[i];
+    }
+    compareChannels(buffer.getChannelData(0), monoBuffer.getChannelData(0));
+  } else {
+    ok(!isChannelSilent(buffer.getChannelData(0)),
+       "mono panning: expect non-zero left channel");
+  }
+  // Check symmetry
+  compareChannels(buffer.getChannelData(0), buffer.getChannelData(1));
+}
+
+async function testStereo(panningModel) {
+  var ac = new OfflineAudioContext(2, BUF_SIZE, sampleRate);
+
+  // A sine to be used to fill the buffers
+  function sine(t) {
+    return Math.sin(440 * 2 * Math.PI * t / ac.sampleRate);
+  }
+
+  var stereoBuffer = ac.createBuffer(2, BUF_SIZE, ac.sampleRate);
+  for (var i = 0; i < BUF_SIZE; ++i) {
+    stereoBuffer.getChannelData(0)[i] = sine(i);
+    stereoBuffer.getChannelData(1)[i] = sine(i);
+  }
+
+  var stereoSource = ac.createBufferSource();
+  stereoSource.buffer = stereoBuffer;
+  stereoSource.start(0);
+
+  var panner = new PannerNode(ac, {
+    panningModel,
+    distanceModel: "linear",
+  });
+  stereoSource.connect(panner);
+
+  var panner2 = new PannerNode(ac, {
+    panningModel,
+    distanceModel: "inverse",
+  });
+  panner.connect(panner2);
+
+  var panner3 = new PannerNode(ac, {
+    panningModel,
+    distanceModel: "exponential",
+  });
+  panner2.connect(panner3);
+
+  panner3.connect(ac.destination);
+
+  const buffer = await ac.startRendering();
+  if (panningModel == "equalpower") {
+    compareBuffers(buffer, stereoBuffer);
+  } else {
+    ok(!isChannelSilent(buffer.getChannelData(0)),
+       "stereo panning: expect non-zero left channel");
+    // Check symmetry
+    compareChannels(buffer.getChannelData(0), buffer.getChannelData(1));
+  }
+}
+
+async function test(type) {
+  await promiseHRTFReady(sampleRate);
+  await testMono(type)
+  await testStereo(type);
+}
+
+add_task(async function() {
+  for (const panningModel of types) {
+    await test(panningModel);
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNodeChannelCount.html b/dom/media/webaudio/test/test_pannerNodeChannelCount.html
new file mode 100644
index 0000000000..9cb90f32da
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNodeChannelCount.html
@@ -0,0 +1,52 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode directly above</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 2,
+  createGraph(context) {
+    var buffer = context.createBuffer(2, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      var sample = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+      // When mixed into a single channel, this produces silence
+      buffer.getChannelData(0)[i] = sample;
+      buffer.getChannelData(1)[i] = -sample;
+    }
+
+    var panner = context.createPanner();
+    panner.positionX.value = 1;
+    panner.positionY.value = 2;
+    panner.positionZ.value = 3;
+    panner.channelCount = 1;
+    expectException(function() { panner.channelCount = 3; },
+                    DOMException.NOT_SUPPORTED_ERR);
+    panner.channelCountMode = "explicit";
+    expectException(function() { panner.channelCountMode = "max"; },
+                    DOMException.NOT_SUPPORTED_ERR);
+    panner.channelInterpretation = "discrete";
+    panner.channelInterpretation = "speakers";
+
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.connect(panner);
+    source.start(0);
+
+    return panner;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNodeHRTFSymmetry.html b/dom/media/webaudio/test/test_pannerNodeHRTFSymmetry.html
new file mode 100644
index 0000000000..cd7b623e41
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNodeHRTFSymmetry.html
@@ -0,0 +1,76 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test left/right symmetry and block-offset invariance of HRTF panner</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+const blockSize = 128;
+const bufferSize = 4096; // > HRTF panner latency
+
+var ctx = new AudioContext();
+
+function startTest() {
+  var leftPanner = ctx.createPanner();
+  var rightPanner = ctx.createPanner();
+  leftPanner.panningModel = "HRTF";
+  rightPanner.panningModel = "HRTF";
+  leftPanner.positionX.value = -1;
+  rightPanner.positionX.value = 1;
+
+  // Test that PannerNode processes the signal consistently irrespective of
+  // the offset in the processing block.  This is done by inserting a delay of
+  // less than a block size before one panner.
+  const delayTime = 0.7 * blockSize / ctx.sampleRate;
+  var leftDelay = ctx.createDelay(delayTime);
+  leftDelay.delayTime.value = delayTime;
+  leftDelay.connect(leftPanner);
+  // and compensating for the delay after the other.
+  var rightDelay = ctx.createDelay(delayTime);
+  rightDelay.delayTime.value = delayTime;
+  rightPanner.connect(rightDelay);
+
+  // Feed the panners with a signal having some harmonics to fill the spectrum.
+  var oscillator = ctx.createOscillator();
+  oscillator.frequency.value = 110;
+  oscillator.type = "sawtooth";
+  oscillator.connect(leftDelay);
+  oscillator.connect(rightPanner);
+  oscillator.start(0);
+
+  // Switch the channels on one panner output, and it should match the other.
+  var splitter = ctx.createChannelSplitter();
+  leftPanner.connect(splitter);
+  var merger = ctx.createChannelMerger();
+  splitter.connect(merger, 0, 1);
+  splitter.connect(merger, 1, 0);
+
+  // Invert one signal so that mixing with the other will find the difference.
+  var gain = ctx.createGain();
+  gain.gain.value = -1.0;
+  merger.connect(gain);
+
+  var processor = ctx.createScriptProcessor(bufferSize, 2, 0);
+  gain.connect(processor);
+  rightDelay.connect(processor);
+  processor.onaudioprocess =
+    function(e) {
+      compareBuffers(e.inputBuffer,
+                     ctx.createBuffer(2, bufferSize, ctx.sampleRate));
+      e.target.onaudioprocess = null;
+      SimpleTest.finish();
+    }
+}
+
+promiseHRTFReady(ctx.sampleRate).then(startTest);
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNodePassThrough.html b/dom/media/webaudio/test/test_pannerNodePassThrough.html
new file mode 100644
index 0000000000..d8c809a2e2
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNodePassThrough.html
@@ -0,0 +1,53 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+
+    var panner = context.createPanner();
+
+    source.buffer = this.buffer;
+
+    source.connect(panner);
+
+    context.listener.setOrientation(0, 6.311749985202524e+307, 0, 0.1, 1000, 0);
+    context.listener.setOrientation(0, 0, -6.311749985202524e+307, 0, 0, 6.311749985202524e+307);
+    panner.positionX = 2;
+    panner.rolloffFactor = 0;
+    panner.panningModel = "equalpower";
+
+    var pannerWrapped = SpecialPowers.wrap(panner);
+    ok("passThrough" in pannerWrapped, "PannerNode should support the passThrough API");
+    pannerWrapped.passThrough = true;
+
+    source.start(0);
+    return panner;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNodeTail.html b/dom/media/webaudio/test/test_pannerNodeTail.html
new file mode 100644
index 0000000000..7035780bf2
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNodeTail.html
@@ -0,0 +1,200 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test tail time lifetime of PannerNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// This tests that a PannerNode does not release its reference before
+// it finishes emitting sound.
+//
+// The PannerNode tail time is short, so, when a PannerNode is destroyed on
+// the main thread, it is unlikely to notify the graph thread before the tail
+// time expires.  However, by adding DelayNodes downstream from the
+// PannerNodes, the graph thread can have enough time to notice that a
+// DelayNode has been destroyed.
+//
+// In the current implementation, DelayNodes will take a tail-time reference
+// immediately when they receive the first block of sound from an upstream
+// node, so this test connects the downstream DelayNodes while the upstream
+// nodes are finishing, and then runs GC (on the main thread) before the
+// DelayNodes receive any input (on the graph thread).
+//
+// Web Audio doesn't provide a means to precisely time connect()s but we can
+// test that the output of delay nodes matches the output from a reference
+// PannerNode that we know will not be GCed.
+//
+// Another set of delay nodes is added upstream to ensure that the source node
+// has removed its self-reference after dispatching its "ended" event.
+
+SimpleTest.waitForExplicitFinish();
+
+const blockSize = 128;
+// bufferSize should be long enough that to allow an audioprocess event to be
+// sent to the main thread and a connect message to return to the graph
+// thread.
+const bufferSize = 4096;
+const pannerCount = bufferSize / blockSize;
+// sourceDelayBufferCount should be long enough to allow the source node
+// onended to finish and remove the source self-reference.
+const sourceDelayBufferCount = 3;
+var gotEnded = false;
+// ccDelayLength should be long enough to allow CC to run
+var ccDelayBufferCount = 20;
+const ccDelayLength = ccDelayBufferCount * bufferSize;
+
+var ctx = new AudioContext();
+var testPanners = [];
+var referencePanner = new PannerNode(ctx, {panningModel: "HRTF"});
+var referenceProcessCount = 0;
+var referenceOutput = [new Float32Array(bufferSize),
+                       new Float32Array(bufferSize)];
+var testProcessor;
+var testProcessCount = 0;
+
+function onReferenceOutput(e) {
+  switch(referenceProcessCount) {
+
+  case sourceDelayBufferCount - 1:
+    // The panners are about to finish.
+    if (!gotEnded) {
+      todo(false, "Source hasn't ended.  Increase sourceDelayBufferCount?");
+    }
+
+    // Connect each PannerNode output to a downstream DelayNode,
+    // and connect ScriptProcessors to compare test and reference panners.
+    var delayDuration = ccDelayLength / ctx.sampleRate;
+    for (var i = 0; i < pannerCount; ++i) {
+      var delay = ctx.createDelay(delayDuration);
+      delay.delayTime.value = delayDuration;
+      delay.connect(testProcessor);
+      testPanners[i].connect(delay);
+    }
+    testProcessor = null;
+    testPanners = null;
+
+    // The panning effect is linear so only one reference panner is required.
+    // This also checks that the individual panners don't chop their output
+    // too soon.
+    referencePanner.connect(e.target);
+
+    // Assuming the above operations have already scheduled an event to run in
+    // stable state and ask the graph thread to make connections, schedule a
+    // subsequent event to run cycle collection, which should not collect
+    // panners that are still producing sound.
+    SimpleTest.executeSoon(function() {
+      SpecialPowers.forceGC();
+      SpecialPowers.forceCC();
+    });
+
+    break;
+
+  case sourceDelayBufferCount:
+    // Record this buffer during which PannerNode outputs were connected.
+    for (var i = 0; i < 2; ++i) {
+      e.inputBuffer.copyFromChannel(referenceOutput[i], i);
+    }
+    e.target.onaudioprocess = null;
+    e.target.disconnect();
+
+    // If the buffer is silent, there is probably not much point just
+    // increasing the buffer size, because, with the buffer size already
+    // significantly larger than panner tail time, it demonstrates that the
+    // lag between threads is much greater than the tail time.
+    if (isChannelSilent(referenceOutput[0])) {
+      todo(false, "Connections not detected.");
+    }
+  }
+
+  referenceProcessCount++;
+}
+
+function onTestOutput(e) {
+  if (testProcessCount < sourceDelayBufferCount + ccDelayBufferCount) {
+    testProcessCount++;
+    return;
+  }
+
+  for (var i = 0; i < 2; ++i) {
+    compareChannels(e.inputBuffer.getChannelData(i), referenceOutput[i]);
+  }
+  e.target.onaudioprocess = null;
+  e.target.disconnect();
+  SimpleTest.finish();
+}
+
+function startTest() {
+  // 0.002 is MaxDelayTimeSeconds in HRTFpanner.cpp
+  // and 512 is fftSize() at 48 kHz.
+  const expectedPannerTailTime = 0.002 * ctx.sampleRate + 512;
+
+  // Place the listener to the side of the origin, where the panners are
+  // positioned, to maximize delay in one ear.
+  ctx.listener.setPosition(1,0,0);
+
+  // Create some PannerNodes downstream from DelayNodes with delays long
+  // enough for their source to finish, dispatch its "ended" event
+  // and release its playing reference.  The DelayNodes should expire their
+  // tail-time references before the PannerNodes and so only the PannerNode
+  // lifetimes depends on their tail-time references.  Many DelayNodes are
+  // created and timed to finish at different times so that one PannerNode
+  // will be finishing the block processed immediately after the connect is
+  // received.
+  var source = ctx.createBufferSource();
+  // Just short of blockSize here to avoid rounding into the next block
+  var buffer = ctx.createBuffer(1, blockSize - 1, ctx.sampleRate);
+  for (var i = 0; i < buffer.length; ++i) {
+    buffer.getChannelData(0)[i] = Math.cos(Math.PI * i / buffer.length);
+  }
+  source.buffer = buffer;
+  source.start(0);
+  source.onended = function(e) {
+    gotEnded = true;
+  };
+
+  // Time the first test panner to finish just before downstream DelayNodes
+  // are about the be connected.  Note that DelayNode lifetime depends on
+  // maxDelayTime so set that equal to the delay.
+  var delayDuration =
+    (sourceDelayBufferCount * bufferSize
+     - expectedPannerTailTime - 2 * blockSize) / ctx.sampleRate;
+
+  for (var i = 0; i < pannerCount; ++i) {
+    var delay = ctx.createDelay(delayDuration);
+    delay.delayTime.value = delayDuration;
+    source.connect(delay);
+    delay.connect(referencePanner)
+
+    var panner = ctx.createPanner();
+    panner.panningModel = "HRTF";
+    delay.connect(panner);
+    testPanners[i] = panner;
+
+    delayDuration += blockSize / ctx.sampleRate;
+  }
+
+  // Create a ScriptProcessor now to use as a timer to trigger connection of
+  // downstream nodes.  It will also be used to record reference output.
+  var referenceProcessor = ctx.createScriptProcessor(bufferSize, 2, 0);
+  referenceProcessor.onaudioprocess = onReferenceOutput;
+  // Start audioprocess events before source delays are connected.
+  referenceProcessor.connect(ctx.destination);
+
+  // The test ScriptProcessor will record output of testPanners. 
+  // Create it now so that it is synchronized with the referenceProcessor.
+  testProcessor = ctx.createScriptProcessor(bufferSize, 2, 0);
+  testProcessor.onaudioprocess = onTestOutput;
+  // Start audioprocess events before source delays are connected.
+  testProcessor.connect(ctx.destination);
+}
+
+promiseHRTFReady(ctx.sampleRate).then(startTest);
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNode_audioparam_distance.html b/dom/media/webaudio/test/test_pannerNode_audioparam_distance.html
new file mode 100644
index 0000000000..2d955de19d
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNode_audioparam_distance.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Distance effect of a PannerNode with the position set via AudioParams (Bug 1472550)</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+ SimpleTest.waitForExplicitFinish();
+ var o = new OfflineAudioContext(2, 256, 44100);
+
+ // We want a stereo constant source.
+ var b = o.createBuffer(2, 1, 44100);
+ b.getChannelData(0)[0] = 1;
+ b.getChannelData(1)[0] = 1;
+ var c = o.createBufferSource();
+ c.buffer = b;
+ c.loop = true;
+
+ var p = o.createPanner();
+ p.positionY.setValueAtTime(1, 0);
+ p.positionX.setValueAtTime(1, 0);
+ p.positionZ.setValueAtTime(1, 0);
+
+ // Set the listener somewhere far
+ o.listener.setPosition(20, 2, 20);
+
+ c.start();
+ c.connect(p).connect(o.destination);
+
+ o.startRendering().then((ab) => {
+   // Check that the distance attenuates the sound.
+   ok(ab.getChannelData(0)[0] < 0.1, "left channel must be very quiet");
+   ok(ab.getChannelData(1)[0] < 0.1, "right channel must be very quiet");
+   SimpleTest.finish();
+ });
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNode_equalPower.html b/dom/media/webaudio/test/test_pannerNode_equalPower.html
new file mode 100644
index 0000000000..127a87b254
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNode_equalPower.html
@@ -0,0 +1,26 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+<title>Test PannerNode</title>
+<script src="/tests/SimpleTest/SimpleTest.js"></script>
+<script type="text/javascript" src="webaudio.js"></script>
+<script type="text/javascript" src="layouttest-glue.js"></script>
+<script type="text/javascript" src="blink/audio-testing.js"></script>
+<script type="text/javascript" src="blink/panner-model-testing.js"></script>
+<link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  function checkFinished() {
+    SimpleTest.finish();
+  }
+  var ctx = new OfflineAudioContext(2, sampleRate * renderLengthSeconds, sampleRate);
+  createTestAndRun(ctx, nodesToCreate, 2, checkFinished);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_pannerNode_maxDistance.html b/dom/media/webaudio/test/test_pannerNode_maxDistance.html
new file mode 100644
index 0000000000..bb70b0a0ec
--- /dev/null
+++ b/dom/media/webaudio/test/test_pannerNode_maxDistance.html
@@ -0,0 +1,61 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PannerNode outputs silence when the distance is greater than maxDist</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// BUF_SIZE must be greater than HRTF delay.
+const BUF_SIZE = 1024;
+const sampleRate = 44100;
+var types = [
+  "equalpower",
+  "HRTF"
+]
+
+async function test(panningModel) {
+  var ac = new OfflineAudioContext(1, 1024, sampleRate);
+  var osc = ac.createOscillator();
+  var panner = new PannerNode(ac, {
+    panningModel,
+    distanceModel: "linear",
+    maxDistance: 100,
+    positionY: 200,
+  });
+
+  ac.listener.setPosition(0, 0, 0);
+
+  osc.connect(panner);
+  panner.connect(ac.destination);
+
+  osc.start();
+
+  const buffer = await ac.startRendering()
+
+  var silence = true;
+  var array = buffer.getChannelData(0);
+  for (var i = 0; i < buffer.length; i++) {
+    if (array[i] != 0) {
+      ok(false, "Found noise in the buffer.");
+      silence = false;
+    }
+  }
+  ok(silence, "The buffer is silent.");
+}
+
+add_task(async function() {
+  await promiseHRTFReady(sampleRate);
+  for (const panningModel of types) {
+    await test(panningModel);
+  }
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_periodicWave.html b/dom/media/webaudio/test/test_periodicWave.html
new file mode 100644
index 0000000000..7b8a6ab12c
--- /dev/null
+++ b/dom/media/webaudio/test/test_periodicWave.html
@@ -0,0 +1,130 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test the PeriodicWave interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+// real and imag are used in separate PeriodicWaves to make their peak values
+// easy to determine.
+const realMax = 99;
+var real = new Float32Array(realMax + 1);
+real[1] = 2.0; // fundamental
+real[realMax] = 3.0;
+const realPeak = real[1] + real[realMax];
+const realFundamental = 19.0;
+var imag = new Float32Array(4);
+imag[0] = 6.0; // should be ignored.
+imag[3] = 0.5;
+const imagPeak = imag[3];
+const imagFundamental = 551.0;
+
+const testLength = 4096;
+
+addLoadEvent(function() {
+  var ac = new AudioContext();
+  ac.createPeriodicWave(new Float32Array(4096), new Float32Array(4096));
+  expectException(function() {
+    ac.createPeriodicWave(new Float32Array(512), imag);
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    ac.createPeriodicWave(new Float32Array(0), new Float32Array(0));
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    ac.createPeriodicWave(new Float32Array(1), new Float32Array(1));
+  }, DOMException.INDEX_SIZE_ERR);
+  expectNoException(function() {
+    ac.createPeriodicWave(new Float32Array(4097), new Float32Array(4097));
+  });
+
+  expectNoException(function() {
+    new PeriodicWave(ac, {});
+  });
+
+  // real.size == imag.size
+  expectException(function() {
+    new PeriodicWave(ac, {real: new Float32Array(10), imag: new Float32Array(9)});
+  }, DOMException.INDEX_SIZE_ERR);
+
+  // size lower than 2 is not allowed
+  expectException(function() {
+    new PeriodicWave(ac, {real: new Float32Array(0)});
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    new PeriodicWave(ac, {imag: new Float32Array(0)});
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    new PeriodicWave(ac, {real: new Float32Array(1)});
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    new PeriodicWave(ac, {imag: new Float32Array(1)});
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    new PeriodicWave(ac, {real: new Float32Array(0), imag: new Float32Array(0)});
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    new PeriodicWave(ac, {real: new Float32Array(1), imag: new Float32Array(1)});
+  }, DOMException.INDEX_SIZE_ERR);
+
+  new PeriodicWave(ac, {real: new Float32Array(4096), imag: new Float32Array(4096)});
+  new PeriodicWave(ac, {real: new Float32Array(4096) });
+  new PeriodicWave(ac, {imag: new Float32Array(4096) });
+
+  runTest();
+});
+
+var gTest = {
+  createGraph(context) {
+    var merger = context.createChannelMerger();
+
+    var osc0 = context.createOscillator();
+    var osc1 = context.createOscillator();
+
+    osc0.setPeriodicWave(context.
+                         createPeriodicWave(real,
+                                            new Float32Array(real.length)));
+    osc1.setPeriodicWave(context.
+                         createPeriodicWave(new Float32Array(imag.length),
+                                            imag));
+
+    osc0.frequency.value = realFundamental;
+    osc1.frequency.value = imagFundamental;
+
+    osc0.start();
+    osc1.start();
+
+    osc0.connect(merger, 0, 0);
+    osc1.connect(merger, 0, 1);
+
+    return merger;
+  },
+  createExpectedBuffers(context) {
+    var buffer = context.createBuffer(2, testLength, context.sampleRate);
+
+    for (var i = 0; i < buffer.length; ++i) {
+
+      buffer.getChannelData(0)[i] = 1.0 / realPeak *
+        (real[1] * Math.cos(2 * Math.PI * realFundamental * i /
+                            context.sampleRate) +
+         real[realMax] * Math.cos(2 * Math.PI * realMax * realFundamental * i /
+                            context.sampleRate));
+
+      buffer.getChannelData(1)[i] = 1.0 / imagPeak *
+         imag[3] * Math.sin(2 * Math.PI * 3 * imagFundamental * i /
+                            context.sampleRate);
+    }
+    return buffer;
+  },
+};
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_periodicWaveBandLimiting.html b/dom/media/webaudio/test/test_periodicWaveBandLimiting.html
new file mode 100644
index 0000000000..a17328b965
--- /dev/null
+++ b/dom/media/webaudio/test/test_periodicWaveBandLimiting.html
@@ -0,0 +1,86 @@
+<!DOCTYPE html>
+<title>Test effect of band limiting on PeriodicWave signals</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+const sampleRate = 48000;
+const bufferSize = 12800;
+const epsilon = 0.01;
+
+// "All implementations must support arrays up to at least 8192", but the
+// linear interpolation of the current implementation distorts the higher
+// frequency components too much to pass this test.
+const frequencyIndexMax = 200;
+
+// A set of oscillators are created near the Nyquist frequency.
+// These are factors giving each oscillator frequency relative to the Nyquist.
+// The first is an octave below Nyquist and the last is just above.
+const OCTAVE_BELOW = 0;
+const HALF_BELOW = 1;
+const NEAR_BELOW = 2;
+const ABOVE = 3;
+const oscillatorFactors = [0.5, Math.sqrt(0.5), 0.99, 1.01];
+const oscillatorCount = oscillatorFactors.length;
+
+// Return magnitude relative to unit sine wave
+function magnitude(array) {
+  var mag = 0
+  for (var i = 0; i < array.length; ++i) {
+    let sample = array[i];
+    mag += sample * sample;
+  }
+  return Math.sqrt(2 * mag / array.length);
+}
+
+function test_frequency_index(frequencyIndex) {
+
+  var context =
+    new OfflineAudioContext(oscillatorCount, bufferSize, sampleRate);
+
+  var merger = context.createChannelMerger(oscillatorCount);
+  merger.connect(context.destination);
+
+  var real = new Float32Array(frequencyIndex + 1);
+  real[frequencyIndex] = 1;
+  var image = new Float32Array(real.length);
+  var wave = context.createPeriodicWave(real, image);
+
+  for (var i = 0; i < oscillatorCount; ++i) {
+    var oscillator = context.createOscillator();
+    oscillator.frequency.value =
+      oscillatorFactors[i] * sampleRate / (2 * frequencyIndex);
+    oscillator.connect(merger, 0, i);
+    oscillator.setPeriodicWave(wave);
+    oscillator.start(0);
+  }
+
+  return context.startRendering().
+    then((buffer) => {
+      assert_equals(buffer.numberOfChannels, oscillatorCount);
+      var magnitudes = [];
+      for (var i = 0; i < oscillatorCount; ++i) {
+        magnitudes[i] = magnitude(buffer.getChannelData(i));
+      }
+      // Unaffected by band-limiting one octave below Nyquist.
+      assert_approx_equals(magnitudes[OCTAVE_BELOW], 1, epsilon,
+                           "magnitude with frequency octave below Nyquist");
+      // Still at least half the amplitude at half octave below Nyquist.
+      assert_greater_than(magnitudes[HALF_BELOW], 0.5 * (1 - epsilon),
+                          "magnitude with frequency half octave below Nyquist");
+      // Approaching zero or zero near Nyquist.
+      assert_less_than(magnitudes[NEAR_BELOW], 0.1,
+                       "magnitude with frequency near Nyquist");
+      assert_equals(magnitudes[ABOVE], 0,
+                   "magnitude with frequency above Nyquist");
+    });
+}
+
+// The 5/4 ratio with rounding up provides sampling across a range of
+// octaves and offsets within octaves.
+for (var frequencyIndex = 1;
+     frequencyIndex < frequencyIndexMax;
+     frequencyIndex = Math.floor((5 * frequencyIndex + 3) / 4)) {
+  promise_test(test_frequency_index.bind(null, frequencyIndex),
+               "Frequency " + frequencyIndex);
+}
+</script>
diff --git a/dom/media/webaudio/test/test_periodicWaveDisableNormalization.html b/dom/media/webaudio/test/test_periodicWaveDisableNormalization.html
new file mode 100644
index 0000000000..229d48282e
--- /dev/null
+++ b/dom/media/webaudio/test/test_periodicWaveDisableNormalization.html
@@ -0,0 +1,98 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test PeriodicWave disableNormalization Parameter</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+// We create PerodicWave instances containing two tones and compare it to
+// buffers created directly in JavaScript by adding the two waves together.
+// Two of the PeriodicWaves are normalized, the other is not. This test is
+// a modification of test_periodicWave.html.
+//
+// These constants are borrowed from test_periodicWave.html and modified
+// so that the realPeak (which is the normalization factor) will be small
+// enough that the errors are within the bounds for the test.
+const realMax = 99;
+var real = new Float32Array(realMax + 1);
+real[1] = 2.0; // fundamental
+real[realMax] = 0.25;
+
+const realPeak = real[1] + real[realMax];
+const realFundamental = 19.0;
+
+const testLength = 4096;
+
+addLoadEvent(function() {
+  runTest();
+});
+
+var gTest = {
+  createGraph(context) {
+    var merger = context.createChannelMerger();
+
+    var osc0 = context.createOscillator();
+    var osc1 = context.createOscillator();
+    var osc2 = context.createOscillator();
+
+    osc0.setPeriodicWave(context.
+                         createPeriodicWave(real,
+                                            new Float32Array(real.length),
+                                            {disableNormalization: false}));
+    osc1.setPeriodicWave(context.
+                         createPeriodicWave(real,
+                                            new Float32Array(real.length)));
+    osc2.setPeriodicWave(context.
+                         createPeriodicWave(real,
+                                            new Float32Array(real.length),
+                                            {disableNormalization: true}));
+
+    osc0.frequency.value = realFundamental;
+    osc1.frequency.value = realFundamental;
+    osc2.frequency.value = realFundamental;
+
+    osc0.start();
+    osc1.start();
+    osc2.start();
+
+    osc0.connect(merger, 0, 0);
+    osc1.connect(merger, 0, 1);
+    osc2.connect(merger, 0, 2);
+
+    return merger;
+  },
+  createExpectedBuffers(context) {
+    var buffer = context.createBuffer(3, testLength, context.sampleRate);
+
+    for (var i = 0; i < buffer.length; ++i) {
+
+      buffer.getChannelData(0)[i] = 1.0 / realPeak *
+        (real[1] * Math.cos(2 * Math.PI * realFundamental * i /
+                            context.sampleRate) +
+         real[realMax] * Math.cos(2 * Math.PI * realMax * realFundamental * i /
+                            context.sampleRate));
+
+      buffer.getChannelData(1)[i] = buffer.getChannelData(0)[i];
+
+      buffer.getChannelData(2)[i] =
+        (real[1] * Math.cos(2 * Math.PI * realFundamental * i /
+                            context.sampleRate) +
+         real[realMax] * Math.cos(2 * Math.PI * realMax * realFundamental * i /
+                            context.sampleRate));
+    }
+    return buffer;
+  },
+  'numberOfChannels': 3,
+};
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_retrospective-exponentialRampToValueAtTime.html b/dom/media/webaudio/test/test_retrospective-exponentialRampToValueAtTime.html
new file mode 100644
index 0000000000..20d3d59faf
--- /dev/null
+++ b/dom/media/webaudio/test/test_retrospective-exponentialRampToValueAtTime.html
@@ -0,0 +1,51 @@
+<!doctype html>
+<meta charset=utf-8>
+<title>Test exponentialRampToValue with end time in the past</title>
+<script src=/resources/testharness.js></script>
+<script src=/resources/testharnessreport.js></script>
+<script>
+function do_test(t, context) {
+  var source = context.createConstantSource();
+  source.start();
+
+  var test = context.createGain();
+  test.gain.exponentialRampToValueAtTime(0.1, 0.5*context.currentTime);
+  test.gain.exponentialRampToValueAtTime(0.9, 2.0);
+
+  var reference = context.createGain();
+  reference.gain.exponentialRampToValueAtTime(0.1, context.currentTime);
+  reference.gain.exponentialRampToValueAtTime(0.9, 2.0);
+
+  source.connect(test);
+  source.connect(reference);
+
+  var merger = context.createChannelMerger();
+  test.connect(merger, 0, 0);
+  reference.connect(merger, 0, 1);
+
+  var processor = context.createScriptProcessor(0, 2, 0);
+  merger.connect(processor);
+  processor.onaudioprocess =
+    t.step_func_done((e) => {
+      source.stop();
+      processor.onaudioprocess = null;
+
+      var testValue = e.inputBuffer.getChannelData(0)[0];
+      var referenceValue = e.inputBuffer.getChannelData(1)[0];
+
+      assert_equals(testValue, referenceValue,
+                        "value matches expected");
+    });
+}
+
+async_test(function(t) {
+  var context = new AudioContext;
+  (function waitForTimeAdvance() {
+    if (context.currentTime == 0) {
+      t.step_timeout(waitForTimeAdvance, 0);
+    } else {
+      do_test(t, context);
+    }
+  })();
+});
+</script>
diff --git a/dom/media/webaudio/test/test_retrospective-linearRampToValueAtTime.html b/dom/media/webaudio/test/test_retrospective-linearRampToValueAtTime.html
new file mode 100644
index 0000000000..1594a30bd1
--- /dev/null
+++ b/dom/media/webaudio/test/test_retrospective-linearRampToValueAtTime.html
@@ -0,0 +1,51 @@
+<!doctype html>
+<meta charset=utf-8>
+<title>Test linearRampToValue with end time in the past</title>
+<script src=/resources/testharness.js></script>
+<script src=/resources/testharnessreport.js></script>
+<script>
+function do_test(t, context) {
+  var source = context.createConstantSource();
+  source.start();
+
+  var test = context.createGain();
+  test.gain.linearRampToValueAtTime(0.1, 0.5*context.currentTime);
+  test.gain.linearRampToValueAtTime(0.9, 2.0);
+
+  var reference = context.createGain();
+  reference.gain.linearRampToValueAtTime(0.1, context.currentTime);
+  reference.gain.linearRampToValueAtTime(0.9, 2.0);
+
+  source.connect(test);
+  source.connect(reference);
+
+  var merger = context.createChannelMerger();
+  test.connect(merger, 0, 0);
+  reference.connect(merger, 0, 1);
+
+  var processor = context.createScriptProcessor(0, 2, 0);
+  merger.connect(processor);
+  processor.onaudioprocess =
+    t.step_func_done((e) => {
+      source.stop();
+      processor.onaudioprocess = null;
+
+      var testValue = e.inputBuffer.getChannelData(0)[0];
+      var referenceValue = e.inputBuffer.getChannelData(1)[0];
+
+      assert_equals(testValue, referenceValue,
+                        "value matches expected");
+    });
+}
+
+async_test(function(t) {
+  var context = new AudioContext;
+  (function waitForTimeAdvance() {
+    if (context.currentTime == 0) {
+      t.step_timeout(waitForTimeAdvance, 0);
+    } else {
+      do_test(t, context);
+    }
+  })();
+});
+</script>
diff --git a/dom/media/webaudio/test/test_retrospective-setTargetAtTime.html b/dom/media/webaudio/test/test_retrospective-setTargetAtTime.html
new file mode 100644
index 0000000000..9b04fe22bb
--- /dev/null
+++ b/dom/media/webaudio/test/test_retrospective-setTargetAtTime.html
@@ -0,0 +1,51 @@
+<!doctype html>
+<meta charset=utf-8>
+<title>Test setTargetAtTime with start time in the past</title>
+<script src=/resources/testharness.js></script>
+<script src=/resources/testharnessreport.js></script>
+<script>
+function do_test(t, context) {
+  var source = context.createConstantSource();
+  source.start();
+
+  var test = context.createGain();
+  test.gain.setTargetAtTime(0.1, 0.5*context.currentTime, 0.1);
+  test.gain.linearRampToValueAtTime(0.9, 2.0);
+
+  var reference = context.createGain();
+  reference.gain.setTargetAtTime(0.1, context.currentTime, 0.1);
+  reference.gain.linearRampToValueAtTime(0.9, 2.0);
+
+  source.connect(test);
+  source.connect(reference);
+
+  var merger = context.createChannelMerger();
+  test.connect(merger, 0, 0);
+  reference.connect(merger, 0, 1);
+
+  var processor = context.createScriptProcessor(0, 2, 0);
+  merger.connect(processor);
+  processor.onaudioprocess =
+    t.step_func_done((e) => {
+      source.stop();
+      processor.onaudioprocess = null;
+
+      var testValue = e.inputBuffer.getChannelData(0)[0];
+      var referenceValue = e.inputBuffer.getChannelData(1)[0];
+
+      assert_equals(testValue, referenceValue,
+                        "value matches expected");
+    });
+}
+
+async_test(function(t) {
+  var context = new AudioContext;
+  (function waitForTimeAdvance() {
+    if (context.currentTime == 0) {
+      t.step_timeout(waitForTimeAdvance, 0);
+    } else {
+      do_test(t, context);
+    }
+  })();
+});
+</script>
diff --git a/dom/media/webaudio/test/test_retrospective-setValueAtTime.html b/dom/media/webaudio/test/test_retrospective-setValueAtTime.html
new file mode 100644
index 0000000000..b9657ef211
--- /dev/null
+++ b/dom/media/webaudio/test/test_retrospective-setValueAtTime.html
@@ -0,0 +1,54 @@
+<!DOCTYPE html>
+<title>Test setValueAtTime with startTime in the past</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+function do_test(t, context) {
+  var source = context.createConstantSource();
+  source.start();
+
+  // Use a ramp of slope 1/sample to measure time.
+  // The end value is the extent of exact precision in single precision float.
+  const rampEnd = Math.pow(2, 24);
+  const rampEndSeconds = rampEnd / context.sampleRate;
+  var test = context.createGain();
+  test.gain.setValueAtTime(0.0, 0.5*context.currentTime);
+  test.gain.linearRampToValueAtTime(rampEnd, rampEndSeconds);
+
+  var reference = context.createGain();
+  reference.gain.setValueAtTime(0.0, context.currentTime);
+  reference.gain.linearRampToValueAtTime(rampEnd, rampEndSeconds);
+
+  source.connect(test);
+  source.connect(reference);
+
+  var merger = context.createChannelMerger();
+  test.connect(merger, 0, 0);
+  reference.connect(merger, 0, 1);
+
+  var processor = context.createScriptProcessor(0, 2, 0);
+  merger.connect(processor);
+  processor.onaudioprocess =
+    t.step_func_done((e) => {
+      source.stop();
+      processor.onaudioprocess = null;
+
+      var testValue = e.inputBuffer.getChannelData(0)[0];
+      var referenceValue = e.inputBuffer.getChannelData(1)[0];
+
+      assert_equals(testValue, referenceValue,
+                        "ramp value matches expected");
+    });
+}
+
+async_test(function(t) {
+  var context = new AudioContext;
+  (function waitForTimeAdvance() {
+    if (context.currentTime == 0) {
+      t.step_timeout(waitForTimeAdvance, 0);
+    } else {
+      do_test(t, context);
+    }
+  })();
+});
+</script>
diff --git a/dom/media/webaudio/test/test_retrospective-setValueCurveAtTime.html b/dom/media/webaudio/test/test_retrospective-setValueCurveAtTime.html
new file mode 100644
index 0000000000..008b240129
--- /dev/null
+++ b/dom/media/webaudio/test/test_retrospective-setValueCurveAtTime.html
@@ -0,0 +1,49 @@
+<!doctype html>
+<meta charset=utf-8>
+<title>Test SetValueCurve with start time in the past</title>
+<script src=/resources/testharness.js></script>
+<script src=/resources/testharnessreport.js></script>
+<script>
+function do_test(t, context) {
+  var source = context.createConstantSource();
+  source.start();
+
+  var test = context.createGain();
+  test.gain.setValueCurveAtTime(new Float32Array([1.0, 0.1]), 0.0, 1.0);
+
+  var reference = context.createGain();
+  reference.gain.setValueCurveAtTime(new Float32Array([1.0, 0.1]), 0.5*context.currentTime, 1.0);
+
+  source.connect(test);
+  source.connect(reference);
+
+  var merger = context.createChannelMerger();
+  test.connect(merger, 0, 0);
+  reference.connect(merger, 0, 1);
+
+  var processor = context.createScriptProcessor(0, 2, 0);
+  merger.connect(processor);
+  processor.onaudioprocess =
+    t.step_func_done((e) => {
+      source.stop();
+      processor.onaudioprocess = null;
+
+      var testValue = e.inputBuffer.getChannelData(0)[0];
+      var referenceValue = e.inputBuffer.getChannelData(1)[0];
+
+      assert_equals(testValue, referenceValue,
+                        "value matches expected");
+    });
+}
+
+async_test(function(t) {
+  var context = new AudioContext;
+  (function waitForTimeAdvance() {
+    if (context.currentTime == 0) {
+      t.step_timeout(waitForTimeAdvance, 0);
+    } else {
+      do_test(t, context);
+    }
+  })();
+});
+</script>
diff --git a/dom/media/webaudio/test/test_scriptProcessorNode.html b/dom/media/webaudio/test/test_scriptProcessorNode.html
new file mode 100644
index 0000000000..ec263755cb
--- /dev/null
+++ b/dom/media/webaudio/test/test_scriptProcessorNode.html
@@ -0,0 +1,132 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ScriptProcessorNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// We do not use our generic graph test framework here because
+// the testing logic here is sort of complicated, and would
+// not be easy to map to OfflineAudioContext, as ScriptProcessorNodes
+// can experience delays.
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = null;
+
+  var sourceSP = context.createScriptProcessor(2048);
+  sourceSP.addEventListener("audioprocess", function(e) {
+    // generate the audio
+    for (var i = 0; i < 2048; ++i) {
+      // Make sure our first sample won't be zero
+      e.outputBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * (i + 1) / context.sampleRate);
+      e.outputBuffer.getChannelData(1)[i] = Math.sin(880 * 2 * Math.PI * (i + 1) / context.sampleRate);
+    }
+    // Remember our generated audio
+    buffer = e.outputBuffer;
+
+    sourceSP.removeEventListener("audioprocess", arguments.callee);
+  });
+
+  expectException(function() {
+    context.createScriptProcessor(1);
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    context.createScriptProcessor(2);
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    context.createScriptProcessor(128);
+  }, DOMException.INDEX_SIZE_ERR);
+  expectException(function() {
+    context.createScriptProcessor(255);
+  }, DOMException.INDEX_SIZE_ERR);
+
+  is(sourceSP.channelCount, 2, "script processor node has 2 input channels by default");
+  is(sourceSP.channelCountMode, "explicit", "Correct channelCountMode for the script processor node");
+  is(sourceSP.channelInterpretation, "speakers", "Correct channelCountInterpretation for the script processor node");
+
+  function findFirstNonZeroSample(buffer) {
+    for (var i = 0; i < buffer.length; ++i) {
+      if (buffer.getChannelData(0)[i] != 0) {
+        return i;
+      }
+    }
+    return buffer.length;
+  }
+
+  var sp = context.createScriptProcessor(2048);
+  sourceSP.connect(sp);
+  sp.connect(context.destination);
+  var lastPlaybackTime = 0;
+
+  var emptyBuffer = context.createBuffer(1, 2048, context.sampleRate);
+
+  function checkAudioProcessingEvent(e) {
+    is(e.target, sp, "Correct event target");
+    ok(e.playbackTime > lastPlaybackTime, "playbackTime correctly set");
+    lastPlaybackTime = e.playbackTime;
+    is(e.inputBuffer.numberOfChannels, 2, "Correct number of channels for the input buffer");
+    is(e.inputBuffer.length, 2048, "Correct length for the input buffer");
+    is(e.inputBuffer.sampleRate, context.sampleRate, "Correct sample rate for the input buffer");
+    is(e.outputBuffer.numberOfChannels, 2, "Correct number of channels for the output buffer");
+    is(e.outputBuffer.length, 2048, "Correct length for the output buffer");
+    is(e.outputBuffer.sampleRate, context.sampleRate, "Correct sample rate for the output buffer");
+
+    compareChannels(e.outputBuffer.getChannelData(0), emptyBuffer.getChannelData(0));
+    compareChannels(e.outputBuffer.getChannelData(1), emptyBuffer.getChannelData(0));
+  }
+
+  sp.onaudioprocess = function(e) {
+    isnot(buffer, null, "The audioprocess handler for sourceSP must be run at this point");
+    checkAudioProcessingEvent(e);
+
+    // Because of the initial latency added by the second script processor node,
+    // we will never see any generated audio frames in the first callback.
+    compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0));
+    compareChannels(e.inputBuffer.getChannelData(1), emptyBuffer.getChannelData(0));
+
+    sp.onaudioprocess = function(e) {
+      checkAudioProcessingEvent(e);
+
+      var firstNonZero = findFirstNonZeroSample(e.inputBuffer);
+      ok(firstNonZero <= 2048, "First non-zero sample within range");
+
+      compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0), firstNonZero);
+      compareChannels(e.inputBuffer.getChannelData(1), emptyBuffer.getChannelData(0), firstNonZero);
+      compareChannels(e.inputBuffer.getChannelData(0), buffer.getChannelData(0), 2048 - firstNonZero, firstNonZero, 0);
+      compareChannels(e.inputBuffer.getChannelData(1), buffer.getChannelData(1), 2048 - firstNonZero, firstNonZero, 0);
+
+      if (firstNonZero == 0) {
+        // If we did not experience any delays, the test is done!
+        sp.onaudioprocess = null;
+
+        SimpleTest.finish();
+      } else if (firstNonZero != 2048) {
+        // In case we just saw a zero buffer this time, wait one more round
+        sp.onaudioprocess = function(e) {
+          checkAudioProcessingEvent(e);
+
+          compareChannels(e.inputBuffer.getChannelData(0), buffer.getChannelData(0), firstNonZero, 0, 2048 - firstNonZero);
+          compareChannels(e.inputBuffer.getChannelData(1), buffer.getChannelData(1), firstNonZero, 0, 2048 - firstNonZero);
+          compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0), undefined, firstNonZero);
+          compareChannels(e.inputBuffer.getChannelData(1), emptyBuffer.getChannelData(0), undefined, firstNonZero);
+
+          sp.onaudioprocess = null;
+
+          SimpleTest.finish();
+        };
+      }
+    };
+  };
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_scriptProcessorNodeChannelCount.html b/dom/media/webaudio/test/test_scriptProcessorNodeChannelCount.html
new file mode 100644
index 0000000000..5e9a9960b7
--- /dev/null
+++ b/dom/media/webaudio/test/test_scriptProcessorNodeChannelCount.html
@@ -0,0 +1,80 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// We do not use our generic graph test framework here because
+// the testing logic here is sort of complicated, and would
+// not be easy to map to OfflineAudioContext, as ScriptProcessorNodes
+// can experience delays.
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(6, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    for (var j = 0; j < 6; ++j) {
+      buffer.getChannelData(0)[i] = Math.sin(440 * j * Math.PI * i / context.sampleRate);
+    }
+  }
+
+  var monoBuffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    monoBuffer.getChannelData(0)[i] = 1;
+  }
+
+  var source = context.createBufferSource();
+
+  var sp = context.createScriptProcessor(2048, 3);
+  expectException(function() { sp.channelCount = 2; },
+                  DOMException.NOT_SUPPORTED_ERR);
+  sp.channelCountMode = "explicit";
+  expectException(function() { sp.channelCountMode = "max"; },
+                  DOMException.NOT_SUPPORTED_ERR);
+  expectException(function() { sp.channelCountMode = "clamped-max"; },
+                  DOMException.NOT_SUPPORTED_ERR);
+  sp.channelInterpretation = "discrete";
+  source.start(0);
+  source.buffer = buffer;
+  source.connect(sp);
+  sp.connect(context.destination);
+
+  var monoSource = context.createBufferSource();
+  monoSource.buffer = monoBuffer;
+  monoSource.connect(sp);
+  monoSource.start(2048 / context.sampleRate);
+
+  sp.onaudioprocess = function(e) {
+    is(e.inputBuffer.numberOfChannels, 3, "Should be correctly down-mixed to three channels");
+    for (var i = 0; i < 3; ++i) {
+      compareChannels(e.inputBuffer.getChannelData(i), buffer.getChannelData(i));
+    }
+
+    // On the next iteration, we'll get a silence buffer
+    sp.onaudioprocess = function(e) {
+      var emptyBuffer = context.createBuffer(1, 2048, context.sampleRate);
+      is(e.inputBuffer.numberOfChannels, 3, "Should be correctly up-mixed to three channels");
+      compareChannels(e.inputBuffer.getChannelData(0), monoBuffer.getChannelData(0));
+      for (var i = 1; i < 3; ++i) {
+        compareChannels(e.inputBuffer.getChannelData(i), emptyBuffer.getChannelData(0));
+      }
+
+      sp.onaudioprocess = null;
+      sp.disconnect(context.destination);
+
+      SimpleTest.finish();
+    };
+  };
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_scriptProcessorNodeNotConnected.html b/dom/media/webaudio/test/test_scriptProcessorNodeNotConnected.html
new file mode 100644
index 0000000000..fb45895380
--- /dev/null
+++ b/dom/media/webaudio/test/test_scriptProcessorNodeNotConnected.html
@@ -0,0 +1,34 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode: should not fire audioprocess if not connected.</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("This test needs to wait a while to ensure that a given event does not happen.");
+addLoadEvent(function() {
+  var context = new AudioContext();
+
+  var sp = context.createScriptProcessor(2048, 2, 2);
+  sp.onaudioprocess = function(e) {
+    ok(false, "Should not call onaudioprocess if the node is not connected.");
+    sp.onaudioprocess = null;
+    SimpleTest.finish();
+  };
+  setTimeout(function() {
+    console.log(sp.onaudioprocess);
+    if (sp.onaudioprocess) {
+      ok(true, "onaudioprocess not fired.");
+      SimpleTest.finish();
+    }
+  }, 4000);
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_scriptProcessorNodePassThrough.html b/dom/media/webaudio/test/test_scriptProcessorNodePassThrough.html
new file mode 100644
index 0000000000..5d2d8170e2
--- /dev/null
+++ b/dom/media/webaudio/test/test_scriptProcessorNodePassThrough.html
@@ -0,0 +1,103 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ScriptProcessorNode with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+// We do not use our generic graph test framework here because
+// the testing logic here is sort of complicated, and would
+// not be easy to map to OfflineAudioContext, as ScriptProcessorNodes
+// can experience delays.
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = null;
+
+  var sourceSP = context.createScriptProcessor(2048);
+  sourceSP.addEventListener("audioprocess", function(e) {
+    // generate the audio
+    for (var i = 0; i < 2048; ++i) {
+      // Make sure our first sample won't be zero
+      e.outputBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * (i + 1) / context.sampleRate);
+      e.outputBuffer.getChannelData(1)[i] = Math.sin(880 * 2 * Math.PI * (i + 1) / context.sampleRate);
+    }
+    // Remember our generated audio
+    buffer = e.outputBuffer;
+
+    sourceSP.removeEventListener("audioprocess", arguments.callee);
+  });
+
+  function findFirstNonZeroSample(buffer) {
+    for (var i = 0; i < buffer.length; ++i) {
+      if (buffer.getChannelData(0)[i] != 0) {
+        return i;
+      }
+    }
+    return buffer.length;
+  }
+
+  var sp = context.createScriptProcessor(2048);
+  sourceSP.connect(sp);
+
+  var spWrapped = SpecialPowers.wrap(sp);
+  ok("passThrough" in spWrapped, "ScriptProcessorNode should support the passThrough API");
+  spWrapped.passThrough = true;
+
+  sp.onaudioprocess = function() {
+    ok(false, "The audioprocess event must never be dispatched on the passthrough ScriptProcessorNode");
+  };
+
+  var sp2 = context.createScriptProcessor(2048);
+  sp.connect(sp2);
+  sp2.connect(context.destination);
+
+  var emptyBuffer = context.createBuffer(1, 2048, context.sampleRate);
+
+  sp2.onaudioprocess = function(e) {
+    // Because of the initial latency added by the second script processor node,
+    // we will never see any generated audio frames in the first callback.
+    compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0));
+    compareChannels(e.inputBuffer.getChannelData(1), emptyBuffer.getChannelData(0));
+
+    sp2.onaudioprocess = function(e) {
+      var firstNonZero = findFirstNonZeroSample(e.inputBuffer);
+      ok(firstNonZero <= 2048, "First non-zero sample within range");
+
+      compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0), firstNonZero);
+      compareChannels(e.inputBuffer.getChannelData(1), emptyBuffer.getChannelData(0), firstNonZero);
+      compareChannels(e.inputBuffer.getChannelData(0), buffer.getChannelData(0), 2048 - firstNonZero, firstNonZero, 0);
+      compareChannels(e.inputBuffer.getChannelData(1), buffer.getChannelData(1), 2048 - firstNonZero, firstNonZero, 0);
+
+      if (firstNonZero == 0) {
+        // If we did not experience any delays, the test is done!
+        sp2.onaudioprocess = null;
+
+        SimpleTest.finish();
+      } else if (firstNonZero != 2048) {
+        // In case we just saw a zero buffer this time, wait one more round
+        sp2.onaudioprocess = function(e) {
+          compareChannels(e.inputBuffer.getChannelData(0), buffer.getChannelData(0), firstNonZero, 0, 2048 - firstNonZero);
+          compareChannels(e.inputBuffer.getChannelData(1), buffer.getChannelData(1), firstNonZero, 0, 2048 - firstNonZero);
+          compareChannels(e.inputBuffer.getChannelData(0), emptyBuffer.getChannelData(0), undefined, firstNonZero);
+          compareChannels(e.inputBuffer.getChannelData(1), emptyBuffer.getChannelData(0), undefined, firstNonZero);
+
+          sp2.onaudioprocess = null;
+
+          SimpleTest.finish();
+        };
+      }
+    };
+  };
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_scriptProcessorNodeZeroInputOutput.html b/dom/media/webaudio/test/test_scriptProcessorNodeZeroInputOutput.html
new file mode 100644
index 0000000000..f4b25d49dd
--- /dev/null
+++ b/dom/media/webaudio/test/test_scriptProcessorNodeZeroInputOutput.html
@@ -0,0 +1,39 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+
+  var sp = context.createScriptProcessor(2048, 0, 2);
+  sp.onaudioprocess = function(e) {
+    is(e.inputBuffer.numberOfChannels, 0, "Should have 0 input channels");
+    is(e.outputBuffer.numberOfChannels, 2, "Should have 2 output channels");
+    sp.onaudioprocess = null;
+
+    sp = context.createScriptProcessor(2048, 2, 0);
+    sp.onaudioprocess = function(e) {
+      is(e.inputBuffer.numberOfChannels, 2, "Should have 2 input channels");
+      is(e.outputBuffer.numberOfChannels, 0, "Should have 0 output channels");
+      sp.onaudioprocess = null;
+
+      SimpleTest.finish();
+    };
+    sp.connect(context.destination);
+  };
+  sp.connect(context.destination);
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_scriptProcessorNode_playbackTime1.html b/dom/media/webaudio/test/test_scriptProcessorNode_playbackTime1.html
new file mode 100644
index 0000000000..ec695f952b
--- /dev/null
+++ b/dom/media/webaudio/test/test_scriptProcessorNode_playbackTime1.html
@@ -0,0 +1,52 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test ScriptProcessorNode playbackTime for bug 970773</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+
+var context = new AudioContext();
+const delay = 0.1;
+
+function doTest() {
+  const processorBufferLength = 256;
+  // |currentTime| may include double precision floating point
+  // rounding errors, so round to nearest integer sample to ignore these.
+  var minimumPlaybackSample =
+    Math.round(context.currentTime * context.sampleRate) +
+    processorBufferLength;
+  var sp = context.createScriptProcessor(processorBufferLength);
+  sp.connect(context.destination);
+  sp.onaudioprocess =
+    function(e) {
+      is(e.inputBuffer.length, processorBufferLength,
+         "expected buffer length");
+      var playbackSample = Math.round(e.playbackTime * context.sampleRate)
+      ok(playbackSample >= minimumPlaybackSample,
+         "playbackSample " + playbackSample +
+         " beyond expected minimum " + minimumPlaybackSample);
+      sp.onaudioprocess = null;
+      SimpleTest.finish();
+    };
+}
+
+// Wait until AudioDestinationNode has accumulated enough 'extra' time so that
+// a failure would be easily detected.
+(function waitForExtraTime() {
+  if (context.currentTime < delay) {
+    SimpleTest.executeSoon(waitForExtraTime);
+  } else {
+    doTest();
+  }
+})();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_sequentialBufferSourceWithResampling.html b/dom/media/webaudio/test/test_sequentialBufferSourceWithResampling.html
new file mode 100644
index 0000000000..218bf462c5
--- /dev/null
+++ b/dom/media/webaudio/test/test_sequentialBufferSourceWithResampling.html
@@ -0,0 +1,72 @@
+<!DOCTYPE html>
+<title>Test seamless playback of a series of resampled buffers</title>
+<script src="/resources/testharness.js"></script>
+<script src="/resources/testharnessreport.js"></script>
+<script>
+// Permitting some accumulation of rounding to int16_t.
+// 64/2^15 would be only just small enough to detect off-by-one-subsample
+// scheduling errors with the frequencies here.
+const EPSILON = 4.0 / Math.pow(2, 15);
+// Offsets test for rounding to nearest rather than up or down.
+const OFFSETS = [EPSILON, 1.0 - EPSILON];
+// The ratio of resampling is 147:160, so 256 start points is enough to cover
+// every fractional offset.
+const LENGTH = 256;
+
+function do_test(context_rate, buffer_rate, start_offset) {
+
+  var context =
+    new OfflineAudioContext(2, LENGTH, context_rate);
+
+  var merger = context.createChannelMerger(context.destination.channelCount);
+  merger.connect(context.destination);
+
+  // Create an audio signal that will be repeated
+  var repeating_signal = context.createBuffer(1, 1, buffer_rate);
+  repeating_signal.getChannelData(0)[0] = 0.5;
+
+  // Schedule a series of nodes to repeat the signal.
+  for (var i = 0; i < LENGTH; ++i) {
+    var source = context.createBufferSource();
+    source.buffer = repeating_signal;
+    source.connect(merger, 0, 0);
+    source.start((i + start_offset) / buffer_rate);
+  }
+
+  // A single long signal should produce the same result.
+  var long_signal = context.createBuffer(1, LENGTH, buffer_rate);
+  var c = long_signal.getChannelData(0);
+  for (var i = 0; i < c.length; ++i) {
+    c[i] = 0.5;
+  }
+
+  var source = context.createBufferSource();
+  source.buffer = long_signal;
+  source.connect(merger, 0, 1);
+  source.start(start_offset / buffer_rate);
+
+  return context.startRendering().
+    then((buffer) => {
+      let series_output = buffer.getChannelData(0);
+      let expected = buffer.getChannelData(1);
+
+      for (var i = 0; i < buffer.length; ++i) {
+        assert_approx_equals(series_output[i], expected[i], EPSILON,
+                             "series output at " + i);
+      }
+    });
+}
+
+function start_tests(context_rate, buffer_rate) {
+  OFFSETS.forEach((start_offset) => {
+    promise_test(() => do_test(context_rate, buffer_rate, start_offset),
+                 "" + context_rate + " context, "
+                 + buffer_rate + " buffer, "
+                 + start_offset + " start");
+  });
+}
+
+start_tests(48000, 44100);
+start_tests(44100, 48000);
+
+</script>
diff --git a/dom/media/webaudio/test/test_setValueCurveWithNonFiniteElements.html b/dom/media/webaudio/test/test_setValueCurveWithNonFiniteElements.html
new file mode 100644
index 0000000000..28829e1ec2
--- /dev/null
+++ b/dom/media/webaudio/test/test_setValueCurveWithNonFiniteElements.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset=utf-8>
+<head>
+  <title>Bug 1308437 - setValueCurve should throw on non-finite elements</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+
+function testInfiniteElement(audioContext, audioParam) {
+  // create value curve with infinite element
+  var arr = new Float32Array(5);
+  arr[0] = 0.5;
+  arr[1] = 1;
+  arr[2] = Infinity;
+  arr[3] = 1;
+  arr[4] = 0.5;
+
+  try {
+    audioParam.setValueCurveAtTime(arr, audioContext.currentTime(), 2);
+    ok(false, "We shouldn't be able to call setValueCurve with Infinity but we can");
+  } catch(e) {
+    ok(e instanceof TypeError, "TypeError is thrown");
+  }
+};
+
+function testNanElement(audioContext, audioParam) {
+  // create value curve with infinite element
+  var arr = new Float32Array(5);
+  arr[0] = 0.5;
+  arr[1] = 1;
+  arr[2] = NaN;
+  arr[3] = 1;
+  arr[4] = 0.5;
+
+  try {
+    audioParam.setValueCurveAtTime(arr, audioContext.currentTime(), 2);
+    ok(false, "We shouldn't be able to call setValueCurve with NaN but we can");
+  } catch(e) {
+    ok(e instanceof TypeError, "TypeError is thrown");
+  }
+};
+
+addLoadEvent(function() {
+  var audioContext = new AudioContext();
+  var gainNode = audioContext.createGain();
+
+  testInfiniteElement(audioContext, gainNode.gain);
+  testNanElement(audioContext, gainNode.gain);
+
+  SimpleTest.finish();
+});
+</script>
+</pre>
+</body>
+</html>
\ No newline at end of file
diff --git a/dom/media/webaudio/test/test_singleSourceDest.html b/dom/media/webaudio/test/test_singleSourceDest.html
new file mode 100644
index 0000000000..fd4de50f5d
--- /dev/null
+++ b/dom/media/webaudio/test/test_singleSourceDest.html
@@ -0,0 +1,70 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test whether we can create an AudioContext interface</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  var context = new AudioContext();
+  var buffer = context.createBuffer(1, 2048, context.sampleRate);
+  for (var i = 0; i < 2048; ++i) {
+    buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+  }
+
+  var destination = context.destination;
+  is(destination.context, context, "Destination node has proper context");
+  is(destination.context, context, "Destination node has proper context");
+  is(destination.numberOfInputs, 1, "Destination node has 1 inputs");
+  is(destination.numberOfOutputs, 0, "Destination node has 0 outputs");
+  is(destination.channelCount, 2, "Destination node has 2 input channels by default");
+  is(destination.channelCountMode, "explicit", "Correct channelCountMode for the destination node");
+  is(destination.channelInterpretation, "speakers", "Correct channelCountInterpretation for the destination node");
+  ok(destination instanceof EventTarget, "AudioNodes must be EventTargets");
+
+  var source = context.createBufferSource();
+  is(source.context, context, "Source node has proper context");
+  is(source.numberOfInputs, 0, "Source node has 0 inputs");
+  is(source.numberOfOutputs, 1, "Source node has 1 outputs");
+  is(source.loop, false, "Source node is not looping");
+  is(source.loopStart, 0, "Correct default value for loopStart");
+  is(source.loopEnd, 0, "Correct default value for loopEnd");
+  ok(!source.buffer, "Source node should not have a buffer when it's created");
+  is(source.channelCount, 2, "source node has 2 input channels by default");
+  is(source.channelCountMode, "max", "Correct channelCountMode for the source node");
+  is(source.channelInterpretation, "speakers", "Correct channelCountInterpretation for the source node");
+
+  expectException(function() {
+    source.channelCount = 0;
+  }, DOMException.NOT_SUPPORTED_ERR);
+
+  source.buffer = buffer;
+  ok(source.buffer, "Source node should have a buffer now");
+
+  source.connect(destination);
+
+  is(source.numberOfInputs, 0, "Source node has 0 inputs");
+  is(source.numberOfOutputs, 1, "Source node has 0 outputs");
+  is(destination.numberOfInputs, 1, "Destination node has 0 inputs");
+  is(destination.numberOfOutputs, 0, "Destination node has 0 outputs");
+
+  source.start(0);
+  SimpleTest.executeSoon(function() {
+    source.stop(0);
+    source.disconnect();
+
+    SpecialPowers.clearUserPref("media.webaudio.enabled");
+    SimpleTest.finish();
+  });
+});
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_slowStart.html b/dom/media/webaudio/test/test_slowStart.html
new file mode 100644
index 0000000000..17de7351c1
--- /dev/null
+++ b/dom/media/webaudio/test/test_slowStart.html
@@ -0,0 +1,48 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioContext.currentTime</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+SimpleTest.waitForExplicitFinish();
+SimpleTest.requestFlakyTimeout("This test needs to periodically query the AudioContext's position.");
+const CUBEB_INIT_DELAY = 5000;
+// Delay audio stream start by a good 5 seconds
+SpecialPowers.pushPrefEnv({"set": [["media.cubeb.slow_stream_init_ms",
+    CUBEB_INIT_DELAY]]}, runTest);
+
+
+function runTest() {
+  let ac = new AudioContext();
+  let notStartedYetCount = 0;
+  let startWallClockTime = performance.now();
+  is(ac.currentTime, 0, "AudioContext.currentTime should be 0 initially");
+  is(ac.state, "suspended", "AudioContext.currentTime is initially suspended");
+  let intervalHandle = setInterval(function() {
+    if (ac.state == "running" || ac.currentTime > 0) {
+      clearInterval(intervalHandle);
+      return;
+    }
+    is(ac.currentTime, 0, "AudioContext.currentTime is still 0");
+    is(ac.state, "suspended", "AudioContext.currentTime is still suspended");
+    notStartedYetCount++;
+  });
+  ac.onstatechange = function() {
+    is(ac.state, "running", "The AudioContext eventually started.");
+    var startDuration = performance.now() - startWallClockTime;
+    info(`AudioContext start time with a delay of ${CUBEB_INIT_DELAY}): ${startDuration}`);
+    ok(notStartedYetCount > 0, "We should have observed the AudioContext in \"suspended\" state");
+    ok(startDuration >= CUBEB_INIT_DELAY, "The AudioContext state transition was correct.");
+    SimpleTest.finish();
+  }
+}
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_stereoPannerNode.html b/dom/media/webaudio/test/test_stereoPannerNode.html
new file mode 100644
index 0000000000..d08e1640b2
--- /dev/null
+++ b/dom/media/webaudio/test/test_stereoPannerNode.html
@@ -0,0 +1,295 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test StereoPannerNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var SR = 44100;
+var BUF_SIZE = 128;
+var PANNING = 0.1;
+var GAIN = 0.5;
+
+// Cheap reimplementation of some bits of the spec
+function gainForPanningMonoToStereo(panning) {
+  panning += 1;
+  panning /= 2;
+  return [ Math.cos(0.5 * Math.PI * panning),
+           Math.sin(0.5 * Math.PI * panning) ];
+}
+
+function gainForPanningStereoToStereo(panning) {
+  if (panning <= 0) {
+    panning += 1.;
+  }
+  return [ Math.cos(0.5 * Math.PI * panning),
+           Math.sin(0.5 * Math.PI * panning) ];
+}
+
+function applyStereoToStereoPanning(l, r, panningValues, panning) {
+  var outL, outR;
+  if (panning <= 0) {
+    outL = l + r * panningValues[0];
+    outR = r * panningValues[1];
+  } else {
+    outL = l * panningValues[0];
+    outR = r + l * panningValues[1];
+  }
+  return [outL,outR];
+}
+
+function applyMonoToStereoPanning(c, panning) {
+  return [c * panning[0], c * panning[1]];
+}
+
+// Test the DOM interface
+var context = new OfflineAudioContext(1, 1, SR);
+var stereoPanner = new StereoPannerNode(context);
+ok(stereoPanner.pan, "The AudioParam member must exist");
+is(stereoPanner.pan.value, 0.0, "Correct initial value");
+is(stereoPanner.pan.defaultValue, 0.0, "Correct default value");
+is(stereoPanner.channelCount, 2, "StereoPannerNode node has 2 input channels by default");
+is(stereoPanner.channelCountMode, "clamped-max", "Correct channelCountMode for the StereoPannerNode");
+is(stereoPanner.channelInterpretation, "speakers", "Correct channelCountInterpretation for the StereoPannerNode");
+expectException(function() {
+  stereoPanner.channelCount = 3;
+}, DOMException.NOT_SUPPORTED_ERR);
+expectException(function() {
+  stereoPanner.channelCountMode = "max";
+}, DOMException.NOT_SUPPORTED_ERR);
+
+// A sine to be used to fill the buffers
+function sine(t) {
+  return Math.sin(440 * 2 * Math.PI * t / context.sampleRate);
+}
+
+// A couple mono and stereo buffers: the StereoPannerNode equation is different
+// if the input is mono or stereo
+var stereoBuffer = new AudioBuffer({ numberOfChannels: 2,
+                                     length: BUF_SIZE,
+                                     sampleRate: context.sampleRate });
+var monoBuffer = new AudioBuffer({ numberOfChannels: 1,
+                                   length: BUF_SIZE,
+                                   sampleRate: context.sampleRate });
+for (var i = 0; i < BUF_SIZE; ++i) {
+  monoBuffer.getChannelData(0)[i]   =
+  stereoBuffer.getChannelData(0)[i] =
+  stereoBuffer.getChannelData(1)[i] = sine(i);
+}
+
+// Expected test vectors
+function expectedBufferNoop(gain) {
+  gain = gain || 1.0;
+  var expectedBuffer = new AudioBuffer({ numberOfChannels: 2,
+                                         length: BUF_SIZE,
+                                         sampleRate: SR });
+  for (var i = 0; i < BUF_SIZE; i++) {
+    expectedBuffer.getChannelData(0)[i] = gain * sine(i);
+    expectedBuffer.getChannelData(1)[i] = gain * sine(i);
+  }
+  return expectedBuffer;
+}
+
+function expectedBufferForStereo(panning, gain) {
+  gain = gain || 1.0;
+  var expectedBuffer = new AudioBuffer({ numberOfChannels: 2,
+                                         length: BUF_SIZE,
+                                         sampleRate: SR });
+  var gainPanning = gainForPanningStereoToStereo(panning);
+  for (var i = 0; i < BUF_SIZE; i++) {
+    var values = [ gain * sine(i), gain * sine(i) ];
+    var processed = applyStereoToStereoPanning(values[0], values[1], gainPanning, PANNING);
+    expectedBuffer.getChannelData(0)[i] = processed[0];
+    expectedBuffer.getChannelData(1)[i] = processed[1];
+  }
+  return expectedBuffer;
+}
+
+function expectedBufferForMono(panning, gain) {
+  gain = gain || 1.0;
+  var expectedBuffer = new AudioBuffer({ numberOfChannels: 2,
+                                         length: BUF_SIZE,
+                                         sampleRate: SR });
+  var gainPanning = gainForPanningMonoToStereo(panning);
+  gainPanning[0] *= gain;
+  gainPanning[1] *= gain;
+  for (var i = 0; i < BUF_SIZE; i++) {
+    var value = sine(i);
+    var processed = applyMonoToStereoPanning(value, gainPanning);
+    expectedBuffer.getChannelData(0)[i] = processed[0];
+    expectedBuffer.getChannelData(1)[i] = processed[1];
+  }
+  return expectedBuffer;
+}
+
+// Actual test cases
+var tests = [
+  function monoPanningNoop(ctx, panner) {
+    var monoSource = ctx.createBufferSource();
+    monoSource.connect(panner);
+    monoSource.buffer = monoBuffer;
+    monoSource.start(0);
+    return expectedBufferForMono(0);
+  },
+  function stereoPanningNoop(ctx, panner) {
+    var stereoSource = ctx.createBufferSource();
+    stereoSource.connect(panner);
+    stereoSource.buffer = stereoBuffer;
+    stereoSource.start(0);
+    return expectedBufferNoop();
+  },
+  function monoPanningNoopWithGain(ctx, panner) {
+    var monoSource = ctx.createBufferSource();
+    var gain = ctx.createGain();
+    gain.gain.value = GAIN;
+    monoSource.connect(gain);
+    gain.connect(panner);
+    monoSource.buffer = monoBuffer;
+    monoSource.start(0);
+    return expectedBufferForMono(0, GAIN);
+  },
+  function stereoPanningNoopWithGain(ctx, panner) {
+    var stereoSource = ctx.createBufferSource();
+    var gain = ctx.createGain();
+    gain.gain.value = GAIN;
+    stereoSource.connect(gain);
+    gain.connect(panner);
+    stereoSource.buffer = stereoBuffer;
+    stereoSource.start(0);
+    return expectedBufferNoop(GAIN);
+  },
+  function stereoPanningAutomation(ctx, panner) {
+    var stereoSource = ctx.createBufferSource();
+    stereoSource.connect(panner);
+    stereoSource.buffer = stereoBuffer;
+    panner.pan.setValueAtTime(0.1, 0.0);
+    stereoSource.start(0);
+    return expectedBufferForStereo(PANNING);
+  },
+  function stereoPanning(ctx, panner) {
+    var stereoSource = ctx.createBufferSource();
+    stereoSource.buffer = stereoBuffer;
+    stereoSource.connect(panner);
+    panner.pan.value = 0.1;
+    stereoSource.start(0);
+    return expectedBufferForStereo(PANNING);
+  },
+  function monoPanningAutomation(ctx, panner) {
+    var monoSource = ctx.createBufferSource();
+    monoSource.connect(panner);
+    monoSource.buffer = monoBuffer;
+    panner.pan.setValueAtTime(PANNING, 0.0);
+    monoSource.start(0);
+    return expectedBufferForMono(PANNING);
+  },
+  function monoPanning(ctx, panner) {
+    var monoSource = ctx.createBufferSource();
+    monoSource.connect(panner);
+    monoSource.buffer = monoBuffer;
+    panner.pan.value = 0.1;
+    monoSource.start(0);
+    return expectedBufferForMono(PANNING);
+  },
+  function monoPanningWithGain(ctx, panner) {
+    var monoSource = ctx.createBufferSource();
+    var gain = ctx.createGain();
+    gain.gain.value = GAIN;
+    monoSource.connect(gain);
+    gain.connect(panner);
+    monoSource.buffer = monoBuffer;
+    panner.pan.value = 0.1;
+    monoSource.start(0);
+    return expectedBufferForMono(PANNING, GAIN);
+  },
+  function stereoPanningWithGain(ctx, panner) {
+    var stereoSource = ctx.createBufferSource();
+    var gain = ctx.createGain();
+    gain.gain.value = GAIN;
+    stereoSource.connect(gain);
+    gain.connect(panner);
+    stereoSource.buffer = stereoBuffer;
+    panner.pan.value = 0.1;
+    stereoSource.start(0);
+    return expectedBufferForStereo(PANNING, GAIN);
+  },
+  function monoPanningWithGainAndAutomation(ctx, panner) {
+    var monoSource = ctx.createBufferSource();
+    var gain = ctx.createGain();
+    gain.gain.value = GAIN;
+    monoSource.connect(gain);
+    gain.connect(panner);
+    monoSource.buffer = monoBuffer;
+    panner.pan.setValueAtTime(PANNING, 0);
+    monoSource.start(0);
+    return expectedBufferForMono(PANNING, GAIN);
+  },
+  function stereoPanningWithGainAndAutomation(ctx, panner) {
+    var stereoSource = ctx.createBufferSource();
+    var gain = ctx.createGain();
+    gain.gain.value = GAIN;
+    stereoSource.connect(gain);
+    gain.connect(panner);
+    stereoSource.buffer = stereoBuffer;
+    panner.pan.setValueAtTime(PANNING, 0);
+    stereoSource.start(0);
+    return expectedBufferForStereo(PANNING, GAIN);
+  },
+  function bug_1783181(ctx, panner) {
+    const length = 128;
+    const buffer = new AudioBuffer({ length, numberOfChannels: 2, sampleRate: ctx.sampleRate });
+
+    buffer.copyToChannel(new Float32Array([1, 0.5, 0, -0.5, -1]), 0);
+    buffer.copyToChannel(new Float32Array([-0.5, -0.25, 0, 0.25, 0.5]), 1);
+
+    const audioBufferSourceNode = new AudioBufferSourceNode(ctx, { buffer });
+
+    audioBufferSourceNode.connect(panner);
+
+    panner.pan.setValueAtTime(0.5, 0);
+    panner.pan.setValueAtTime(0, 2 / ctx.sampleRate);
+    panner.pan.linearRampToValueAtTime(1, 5 / ctx.sampleRate);
+    panner.pan.cancelScheduledValues(3 / ctx.sampleRate);
+
+    audioBufferSourceNode.start(0);
+
+    const expected = new AudioBuffer({ length, numberOfChannels: 2, sampleRate: ctx.sampleRate });
+    expected.copyToChannel(new Float32Array([ 0.7071067690849304, 0.3535533845424652, 0, -0.5, -1 ]), 0);
+    expected.copyToChannel(new Float32Array([ 0.20710676908493042, 0.10355338454246521, 0, 0.25, 0.5 ]), 1);
+
+    return expected;
+  }
+];
+
+var finished = 0;
+function finish() {
+  if (++finished == tests.length) {
+    SimpleTest.finish();
+  }
+}
+
+tests.forEach(function(f) {
+  var ac = new OfflineAudioContext(2, BUF_SIZE, SR);
+  var panner = ac.createStereoPanner();
+  panner.connect(ac.destination);
+  var expected = f(ac, panner);
+  ac.oncomplete = function(e) {
+    info(f.name);
+    compareBuffers(e.renderedBuffer, expected);
+    finish();
+  };
+  ac.startRendering()
+});
+
+SimpleTest.waitForExplicitFinish();
+
+</script>
+</pre>
+<pre id=dump>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_stereoPannerNodePassThrough.html b/dom/media/webaudio/test/test_stereoPannerNodePassThrough.html
new file mode 100644
index 0000000000..2d774d366b
--- /dev/null
+++ b/dom/media/webaudio/test/test_stereoPannerNodePassThrough.html
@@ -0,0 +1,47 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test StereoPanerNode with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+
+    var stereoPanner = context.createStereoPanner();
+
+    source.buffer = this.buffer;
+
+    source.connect(stereoPanner);
+
+    var stereoPannerWrapped = SpecialPowers.wrap(stereoPanner);
+    ok("passThrough" in stereoPannerWrapped, "StereoPannerNode should support the passThrough API");
+    stereoPannerWrapped.passThrough = true;
+
+    source.start(0);
+    return stereoPanner;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      this.buffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_stereoPanningWithGain.html b/dom/media/webaudio/test/test_stereoPanningWithGain.html
new file mode 100644
index 0000000000..94dfa3bb92
--- /dev/null
+++ b/dom/media/webaudio/test/test_stereoPanningWithGain.html
@@ -0,0 +1,49 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test stereo equalpower panning with a GainNode</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script src="webaudio.js" type="text/javascript"></script>
+<script class="testbody" type="text/javascript">
+
+const size = 256;
+
+var gTest = {
+  numberOfChannels: 2,
+  createGraph(context) {
+    var panner = context.createPanner();
+    panner.setPosition(1.0, 0.0, 0.0); // reference distance the right
+    panner.panningModel = "equalpower";
+
+    var gain = context.createGain();
+    gain.gain.value = -0.5;
+    gain.connect(panner);
+
+    var buffer = context.createBuffer(2, 2, context.sampleRate);
+    buffer.getChannelData(0)[0] = 1.0;
+    buffer.getChannelData(1)[1] = 1.0;
+    var source = context.createBufferSource();
+    source.buffer = buffer;
+    source.connect(gain);
+    source.start(0);
+
+    return panner;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(2, size, context.sampleRate);
+    expectedBuffer.getChannelData(1)[0] = -0.5;
+    expectedBuffer.getChannelData(1)[1] = -0.5;
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_waveDecoder.html b/dom/media/webaudio/test/test_waveDecoder.html
new file mode 100644
index 0000000000..65c429f2ba
--- /dev/null
+++ b/dom/media/webaudio/test/test_waveDecoder.html
@@ -0,0 +1,69 @@
+<!DOCTYPE HTML>
+<html>
+<meta charset=utf-8>
+<head>
+  <title>Test that we decode uint8 and sint16 wave files with correct conversion to float64</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+var testsDone = 0;
+var tests = ["UklGRjUrAABXQVZFZm10IBAAAAABAAEAESsAABErAAABAAgAZGF0YQMAAAD/AIA=",
+             "UklGRkZWAABXQVZFZm10IBAAAAABAAEAESsAACJWAAACABAAZGF0YQYAAAD/fwCAAAA="];
+
+SimpleTest.waitForExplicitFinish();
+
+function base64ToUint8Buffer(b64) {
+  var str = atob(b64)
+  var u8 = new Uint8Array(str.length);
+  for (var i = 0; i < str.length; ++i) {
+    u8[i] = str.charCodeAt(i);
+  }
+  return u8;
+}
+
+function fixupBufferSampleRate(u8, rate) {
+  u8[24] = (rate & 0x000000ff) >> 0;
+  u8[25] = (rate & 0x0000ff00) >> 8;
+  u8[26] = (rate & 0x00ff0000) >> 16;
+  u8[27] = (rate & 0xff000000) >> 24;
+}
+
+function finishTest() {
+  testsDone += 1;
+  if (testsDone == tests.length) {
+    SimpleTest.finish();
+  }
+}
+
+function decodeComplete(b) {
+  ok(true, "Decoding succeeded.");
+  is(b.numberOfChannels, 1, "Should have 1 channel.");
+  is(b.length, 3, "Should have three samples.");
+  var samples = b.getChannelData(0);
+  ok(samples[0] >  0.99 && samples[0] <  1.01, "Check near  1.0. Got " + samples[0]);
+  ok(samples[1] > -1.01 && samples[1] < -0.99, "Check near -1.0. Got " + samples[1]);
+  ok(samples[2] > -0.01 && samples[2] <  0.01, "Check near  0.0. Got " + samples[2]);
+  finishTest();
+}
+
+function decodeFailed() {
+  ok(false, "Decoding failed.");
+  finishTest();
+}
+
+addLoadEvent(function() {
+  var context = new AudioContext();
+
+  for (var i = 0; i < tests.length; ++i) {
+    var u8 = base64ToUint8Buffer(tests[i]);
+    fixupBufferSampleRate(u8, context.sampleRate);
+    context.decodeAudioData(u8.buffer, decodeComplete, decodeFailed);
+  }
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_waveShaper.html b/dom/media/webaudio/test/test_waveShaper.html
new file mode 100644
index 0000000000..9d2f1b3fa2
--- /dev/null
+++ b/dom/media/webaudio/test/test_waveShaper.html
@@ -0,0 +1,60 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test WaveShaperNode with no curve</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+    source.buffer = this.buffer;
+
+    var shaper = new WaveShaperNode(context);
+    shaper.curve = this.curve;
+
+    source.connect(shaper);
+
+    source.start(0);
+    return shaper;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(1, 4096, context.sampleRate);
+    for (var i = 1; i < 4095; ++i) {
+      this.buffer.getChannelData(0)[i] = 2 * (i / 4096) - 1;
+    }
+    // Two out of range values
+    this.buffer.getChannelData(0)[0] = -2;
+    this.buffer.getChannelData(0)[4095] = 2;
+
+    this.curve = new Float32Array(2048);
+    for (var i = 0; i < 2048; ++i) {
+      this.curve[i] = Math.sin(100 * Math.PI * (i + 1) / context.sampleRate);
+    }
+
+    var expectedBuffer = context.createBuffer(1, 4096, context.sampleRate);
+    for (var i = 1; i < 4095; ++i) {
+      var input = this.buffer.getChannelData(0)[i];
+      var index = Math.floor(this.curve.length * (input + 1) / 2);
+      index = Math.max(0, Math.min(this.curve.length - 1, index));
+      expectedBuffer.getChannelData(0)[i] = this.curve[index];
+    }
+    expectedBuffer.getChannelData(0)[0] = this.curve[0];
+    expectedBuffer.getChannelData(0)[4095] = this.curve[2047];
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_waveShaperGain.html b/dom/media/webaudio/test/test_waveShaperGain.html
new file mode 100644
index 0000000000..45411eca02
--- /dev/null
+++ b/dom/media/webaudio/test/test_waveShaperGain.html
@@ -0,0 +1,73 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+<meta charset="utf-8">
+  <title>Test that WaveShaperNode doesn't corrupt its inputs when the gain is !=
+    1.0 (bug 1203616)</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre>
+</pre>
+<script class="testbody" type="text/javascript">
+SimpleTest.waitForExplicitFinish();
+var samplerate = 44100;
+var context = new OfflineAudioContext(1, 44100, samplerate);
+
+var dc = context.createBufferSource();
+
+var buffer = context.createBuffer(1, 1, samplerate);
+buffer.getChannelData(0)[0] = 1.0;
+dc.buffer = buffer;
+
+var gain = context.createGain();
+var ws2 = context.createWaveShaper();
+var ws = [];
+
+// No-op waveshaper curves.
+for (var i = 0; i < 2; i++) {
+  ws[i] = context.createWaveShaper();
+  var curve = new Float32Array(2);
+  curve[0] = -1.0;
+  curve[1] = 1.0;
+  ws[i].curve = curve;
+  ws[i].connect(context.destination);
+  gain.connect(ws[i]);
+}
+
+dc.connect(gain);
+dc.start();
+
+gain.gain.value = 0.5;
+
+context.startRendering().then(buffer => {
+  document.querySelector("pre").innerHTML = buffer.getChannelData(0)[0];
+  ok(buffer.getChannelData(0)[0] == 1.0, "Volume was handled properly");
+
+  context = new OfflineAudioContext(1, 100, samplerate);
+  var oscillator = context.createOscillator();
+  var gain = context.createGain();
+  var waveShaper = context.createWaveShaper();
+
+  oscillator.start(0);
+  oscillator.connect(gain);
+
+  // to silence
+  gain.gain.value = 0;
+  gain.connect(waveShaper);
+
+  // convert all signal into 1.0. The non unity values are to detect the use
+  // of uninitialized buffers (see Bug 1283910).
+  waveShaper.curve = new Float32Array([ 0.5, 0.5, 0.5, 0.5, 0.5, 1, 1, 0.5, 0.5, 0.5, 0.5, 0.5 ]);
+  waveShaper.connect(context.destination);
+
+  context.startRendering().then((buffer) => {
+    var result = buffer.getChannelData(0);
+    ok(result.every(x => x === 1), "WaveShaper handles zero gain properly");
+    SimpleTest.finish();
+  });
+});
+</script>
+</body>
+
diff --git a/dom/media/webaudio/test/test_waveShaperInvalidLengthCurve.html b/dom/media/webaudio/test/test_waveShaperInvalidLengthCurve.html
new file mode 100644
index 0000000000..0901521a7b
--- /dev/null
+++ b/dom/media/webaudio/test/test_waveShaperInvalidLengthCurve.html
@@ -0,0 +1,66 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test WaveShaperNode with an invalid curve</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+    source.buffer = this.buffer;
+
+    var shaper = context.createWaveShaper();
+
+    expectException(() => {
+        shaper.curve = new Float32Array(0);
+    }, DOMException.INVALID_STATE_ERR);
+
+    is(shaper.curve, null, "The curve mustn't have been set");
+
+    expectException(() => {
+        shaper.curve = new Float32Array(1);
+    }, DOMException.INVALID_STATE_ERR);
+
+    is(shaper.curve, null, "The curve mustn't have been set");
+
+    expectNoException(() => {
+        shaper.curve = new Float32Array(2);
+    });
+
+    isnot(shaper.curve, null, "The curve must have been set");
+
+    expectNoException(() => {
+        shaper.curve = null;
+    });
+
+    is(shaper.curve, null, "The curve must be null by default");
+
+    source.connect(shaper);
+
+    source.start(0);
+    return shaper;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    this.buffer = expectedBuffer;
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_waveShaperNoCurve.html b/dom/media/webaudio/test/test_waveShaperNoCurve.html
new file mode 100644
index 0000000000..2da0b511af
--- /dev/null
+++ b/dom/media/webaudio/test/test_waveShaperNoCurve.html
@@ -0,0 +1,43 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test WaveShaperNode with no curve</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 2048,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+    source.buffer = this.buffer;
+
+    var shaper = context.createWaveShaper();
+    is(shaper.curve, null, "The shaper curve must be null by default");
+
+    source.connect(shaper);
+
+    source.start(0);
+    return shaper;
+  },
+  createExpectedBuffers(context) {
+    var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate);
+    for (var i = 0; i < 2048; ++i) {
+      expectedBuffer.getChannelData(0)[i] = Math.sin(440 * 2 * Math.PI * i / context.sampleRate);
+    }
+    this.buffer = expectedBuffer;
+    return expectedBuffer;
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_waveShaperPassThrough.html b/dom/media/webaudio/test/test_waveShaperPassThrough.html
new file mode 100644
index 0000000000..d34add9c90
--- /dev/null
+++ b/dom/media/webaudio/test/test_waveShaperPassThrough.html
@@ -0,0 +1,55 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test WaveShaperNode with passthrough</title>
+  <script src="/tests/SimpleTest/SimpleTest.js"></script>
+  <script type="text/javascript" src="webaudio.js"></script>
+  <link rel="stylesheet" type="text/css" href="/tests/SimpleTest/test.css" />
+</head>
+<body>
+<pre id="test">
+<script class="testbody" type="text/javascript">
+
+var gTest = {
+  length: 4096,
+  numberOfChannels: 1,
+  createGraph(context) {
+    var source = context.createBufferSource();
+    source.buffer = this.buffer;
+
+    var shaper = context.createWaveShaper();
+    shaper.curve = this.curve;
+
+    var shaperWrapped = SpecialPowers.wrap(shaper);
+    ok("passThrough" in shaperWrapped, "WaveShaperNode should support the passThrough API");
+    shaperWrapped.passThrough = true;
+
+    source.connect(shaper);
+
+    source.start(0);
+    return shaper;
+  },
+  createExpectedBuffers(context) {
+    this.buffer = context.createBuffer(1, 4096, context.sampleRate);
+    for (var i = 1; i < 4095; ++i) {
+      this.buffer.getChannelData(0)[i] = 2 * (i / 4096) - 1;
+    }
+    // Two out of range values
+    this.buffer.getChannelData(0)[0] = -2;
+    this.buffer.getChannelData(0)[4095] = 2;
+
+    this.curve = new Float32Array(2048);
+    for (var i = 0; i < 2048; ++i) {
+      this.curve[i] = Math.sin(100 * Math.PI * (i + 1) / context.sampleRate);
+    }
+
+    return [this.buffer];
+  },
+};
+
+runTest();
+
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/test_webAudio_muteTab.html b/dom/media/webaudio/test/test_webAudio_muteTab.html
new file mode 100644
index 0000000000..3a3d60eea7
--- /dev/null
+++ b/dom/media/webaudio/test/test_webAudio_muteTab.html
@@ -0,0 +1,93 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <script type="application/javascript" src="mediaStreamPlayback.js"></script>
+</head>
+<body>
+<pre id="test">
+
+<script>
+/* import-globals-from ../../webrtc/tests/mochitests/mediaStreamPlayback.js */
+createHTML({
+  title: "Check tab muting when the tab plays audio via the Web Audio API",
+  bug: "1346880",
+  visible: false
+});
+
+/**
+ * Check that muting a tab results in no audible audio: mute a tab, in which
+ * an OscillatorNode is playing. The default audio output device is a
+ * pulseaudio null-sink. Simulateously, record the other side of the null
+ * sink, and check that no audio has been written to the sink, because the tab
+ * was muted. Then, umute the tab and check that audio is being sent to the
+ * null-sink. */
+runTest(async () => {
+  if (!SpecialPowers.getCharPref("media.audio_loopback_dev", "")) {
+    todo(false, "No loopback device set by framework. Try --use-test-media-devices");
+    return;
+  }
+
+  // Mute the tab
+  await SpecialPowers.toggleMuteState(true, window.top);
+
+  const stream = await getUserMedia({audio: {
+      noiseSuppression: false,
+      echoCancellation: false,
+      autoGainControl: false,
+  }});
+  try {
+    const ac = new AudioContext();
+    const osc = new OscillatorNode(ac);
+    osc.connect(ac.destination);
+    osc.start();
+
+    const analyser = new AudioStreamAnalyser(ac, stream);
+    // Wait for some time, checking there is only ever silent audio in the
+    // loopback stream. `waitForAnalysisSuccess` runs off requestAnimationFrame
+    let silenceFor = 3 / (1 / 60);
+    await analyser.waitForAnalysisSuccess(array => {
+        // `array` has values between 0 and 255, 0 being silence.
+        const sum = array.reduce((acc, v) => { return acc + v; });
+        if (sum == 0) {
+          silenceFor--;
+        } else {
+          info(`Sum of the array values ${sum}`);
+          ok(false, `Found non-silent data in the loopback stream while the tab was muted.`);
+          return true;
+        }
+        if (silenceFor == 0) {
+          ok(true, "Muting the tab was effective");
+        }
+        return silenceFor == 0;
+    });
+
+    // Unmute the tab
+    await SpecialPowers.toggleMuteState(false, window.top);
+
+    await analyser.waitForAnalysisSuccess(array => {
+        // `array` has values between 0 and 255, 0 being silence.
+        const sum = array.reduce((acc, v) => { return acc + v; });
+        if (sum != 0) {
+          info(`Sum after unmuting ${sum}`);
+          ok(true, "Unmuting the tab was effective");
+          return true;
+        } else {
+          // Increment again if we find silence.
+          silenceFor++;
+          if (silenceFor > 100) {
+            ok(false, "Unmuting wasn't effective")
+            return true;
+          }
+          return false;
+        }
+    });
+  } finally {
+    for (let t of stream.getTracks()) {
+      t.stop();
+    }
+  }
+});
+</script>
+</pre>
+</body>
+</html>
diff --git a/dom/media/webaudio/test/ting-44.1k-1ch.ogg b/dom/media/webaudio/test/ting-44.1k-1ch.ogg
new file mode 100644
index 0000000000..a11aaf1cbf
--- /dev/null
+++ b/dom/media/webaudio/test/ting-44.1k-1ch.ogg
diff --git a/dom/media/webaudio/test/ting-44.1k-1ch.wav b/dom/media/webaudio/test/ting-44.1k-1ch.wav
new file mode 100644
index 0000000000..dd731e354c
--- /dev/null
+++ b/dom/media/webaudio/test/ting-44.1k-1ch.wav
diff --git a/dom/media/webaudio/test/ting-44.1k-2ch.ogg b/dom/media/webaudio/test/ting-44.1k-2ch.ogg
new file mode 100644
index 0000000000..94e0014858
--- /dev/null
+++ b/dom/media/webaudio/test/ting-44.1k-2ch.ogg
diff --git a/dom/media/webaudio/test/ting-44.1k-2ch.wav b/dom/media/webaudio/test/ting-44.1k-2ch.wav
new file mode 100644
index 0000000000..b2ef11ecbe
--- /dev/null
+++ b/dom/media/webaudio/test/ting-44.1k-2ch.wav
diff --git a/dom/media/webaudio/test/ting-48k-1ch.ogg b/dom/media/webaudio/test/ting-48k-1ch.ogg
new file mode 100644
index 0000000000..f45ce33a58
--- /dev/null
+++ b/dom/media/webaudio/test/ting-48k-1ch.ogg
diff --git a/dom/media/webaudio/test/ting-48k-1ch.wav b/dom/media/webaudio/test/ting-48k-1ch.wav
new file mode 100644
index 0000000000..91874dfbb2
--- /dev/null
+++ b/dom/media/webaudio/test/ting-48k-1ch.wav
diff --git a/dom/media/webaudio/test/ting-48k-2ch.ogg b/dom/media/webaudio/test/ting-48k-2ch.ogg
new file mode 100644
index 0000000000..e4c564abbd
--- /dev/null
+++ b/dom/media/webaudio/test/ting-48k-2ch.ogg
diff --git a/dom/media/webaudio/test/ting-48k-2ch.wav b/dom/media/webaudio/test/ting-48k-2ch.wav
new file mode 100644
index 0000000000..6adb8d3e5d
--- /dev/null
+++ b/dom/media/webaudio/test/ting-48k-2ch.wav
diff --git a/dom/media/webaudio/test/ting-dualchannel44.1.wav b/dom/media/webaudio/test/ting-dualchannel44.1.wav
new file mode 100644
index 0000000000..62954394d3
--- /dev/null
+++ b/dom/media/webaudio/test/ting-dualchannel44.1.wav
diff --git a/dom/media/webaudio/test/ting-dualchannel48.wav b/dom/media/webaudio/test/ting-dualchannel48.wav
new file mode 100644
index 0000000000..a0b8247888
--- /dev/null
+++ b/dom/media/webaudio/test/ting-dualchannel48.wav
diff --git a/dom/media/webaudio/test/waveformatextensible.wav b/dom/media/webaudio/test/waveformatextensible.wav
new file mode 100644
index 0000000000..f16e588021
--- /dev/null
+++ b/dom/media/webaudio/test/waveformatextensible.wav
diff --git a/dom/media/webaudio/test/waveformatextensiblebadmask.wav b/dom/media/webaudio/test/waveformatextensiblebadmask.wav
new file mode 100644
index 0000000000..3aa55df98e
--- /dev/null
+++ b/dom/media/webaudio/test/waveformatextensiblebadmask.wav
diff --git a/dom/media/webaudio/test/webaudio.js b/dom/media/webaudio/test/webaudio.js
new file mode 100644
index 0000000000..049e0e5af3
--- /dev/null
+++ b/dom/media/webaudio/test/webaudio.js
@@ -0,0 +1,367 @@
+// Helpers for Web Audio tests
+
+// It is expected that the test defines this.
+/* global gTest */
+
+function expectException(func, exceptionCode) {
+  var threw = false;
+  try {
+    func();
+  } catch (ex) {
+    threw = true;
+    is(ex.constructor.name, "DOMException", "Expect a DOM exception");
+    is(ex.code, exceptionCode, "Expect the correct exception code");
+  }
+  ok(threw, "The exception was thrown");
+}
+
+function expectNoException(func) {
+  var threw = false;
+  try {
+    func();
+  } catch (ex) {
+    threw = true;
+  }
+  ok(!threw, "An exception was not thrown");
+}
+
+function expectTypeError(func) {
+  var threw = false;
+  try {
+    func();
+  } catch (ex) {
+    threw = true;
+    ok(ex instanceof TypeError, "Expect a TypeError");
+  }
+  ok(threw, "The exception was thrown");
+}
+
+function expectRejectedPromise(that, func, exceptionName) {
+  var promise = that[func]();
+
+  ok(promise instanceof Promise, "Expect a Promise");
+
+  promise
+    .then(function (res) {
+      ok(false, "Promise resolved when it should have been rejected.");
+    })
+    .catch(function (err) {
+      is(
+        err.name,
+        exceptionName,
+        "Promise correctly reject with " + exceptionName
+      );
+    });
+}
+
+function fuzzyCompare(a, b) {
+  return Math.abs(a - b) < 9e-3;
+}
+
+function compareChannels(
+  buf1,
+  buf2,
+  /*optional*/ length,
+  /*optional*/ sourceOffset,
+  /*optional*/ destOffset,
+  /*optional*/ skipLengthCheck
+) {
+  if (!skipLengthCheck) {
+    is(buf1.length, buf2.length, "Channels must have the same length");
+  }
+  sourceOffset = sourceOffset || 0;
+  destOffset = destOffset || 0;
+  if (length == undefined) {
+    length = buf1.length - sourceOffset;
+  }
+  var difference = 0;
+  var maxDifference = 0;
+  var firstBadIndex = -1;
+  for (var i = 0; i < length; ++i) {
+    if (!fuzzyCompare(buf1[i + sourceOffset], buf2[i + destOffset])) {
+      difference++;
+      maxDifference = Math.max(
+        maxDifference,
+        Math.abs(buf1[i + sourceOffset] - buf2[i + destOffset])
+      );
+      if (firstBadIndex == -1) {
+        firstBadIndex = i;
+      }
+    }
+  }
+
+  is(
+    difference,
+    0,
+    "maxDifference: " +
+      maxDifference +
+      ", first bad index: " +
+      firstBadIndex +
+      " with test-data offset " +
+      sourceOffset +
+      " and expected-data offset " +
+      destOffset +
+      "; corresponding values " +
+      buf1[firstBadIndex + sourceOffset] +
+      " and " +
+      buf2[firstBadIndex + destOffset] +
+      " --- differences"
+  );
+}
+
+function compareBuffers(got, expected) {
+  if (got.numberOfChannels != expected.numberOfChannels) {
+    is(
+      got.numberOfChannels,
+      expected.numberOfChannels,
+      "Correct number of buffer channels"
+    );
+    return;
+  }
+  if (got.length != expected.length) {
+    is(got.length, expected.length, "Correct buffer length");
+    return;
+  }
+  if (got.sampleRate != expected.sampleRate) {
+    is(got.sampleRate, expected.sampleRate, "Correct sample rate");
+    return;
+  }
+
+  for (var i = 0; i < got.numberOfChannels; ++i) {
+    compareChannels(
+      got.getChannelData(i),
+      expected.getChannelData(i),
+      got.length,
+      0,
+      0,
+      true
+    );
+  }
+}
+
+/**
+ * Compute the root mean square (RMS,
+ * <http://en.wikipedia.org/wiki/Root_mean_square>) of a channel of a slice
+ * (defined by `start` and `end`) of an AudioBuffer.
+ *
+ * This is useful to detect that a buffer is noisy or silent.
+ */
+function rms(audiobuffer, channel = 0, start = 0, end = audiobuffer.length) {
+  var buffer = audiobuffer.getChannelData(channel);
+  var rms = 0;
+  for (var i = start; i < end; i++) {
+    rms += buffer[i] * buffer[i];
+  }
+
+  rms /= buffer.length;
+  rms = Math.sqrt(rms);
+  return rms;
+}
+
+function getEmptyBuffer(context, length) {
+  return context.createBuffer(
+    gTest.numberOfChannels,
+    length,
+    context.sampleRate
+  );
+}
+
+function isChannelSilent(channel) {
+  for (var i = 0; i < channel.length; ++i) {
+    if (channel[i] != 0.0) {
+      return false;
+    }
+  }
+  return true;
+}
+
+const HRTFPannersByRate = new Map();
+/**
+ * Return a promise that resolves when PannerNodes with HRTF panningModel in
+ * an AudioContext of the specified sample rate will be ready to produce
+ * non-zero output.  Before the HRIR database is loaded, such PannerNodes
+ * produce zero output.
+ */
+async function promiseHRTFReady(sampleRate) {
+  if (HRTFPannersByRate.has(sampleRate)) {
+    return;
+  }
+
+  const ctx = new AudioContext({ sampleRate });
+  const processor = ctx.createScriptProcessor(4096, 2, 0);
+  const panner = new PannerNode(ctx, { panningModel: "HRTF" });
+  panner.connect(processor);
+  const oscillator = ctx.createOscillator();
+  oscillator.connect(panner);
+  oscillator.start(0);
+
+  await new Promise(r => {
+    processor.onaudioprocess = e => {
+      if (!isChannelSilent(e.inputBuffer.getChannelData(0))) {
+        r();
+      }
+    };
+  });
+
+  ctx.suspend();
+  oscillator.disconnect();
+  panner.disconnect();
+  processor.onaudioprocess = null;
+  // Keep a reference to the panner so that the database is not unloaded.
+  HRTFPannersByRate.set(sampleRate, panner);
+}
+
+/**
+ * This function assumes that the test file defines a single gTest variable with
+ * the following properties and methods:
+ *
+ * + numberOfChannels: optional property which specifies the number of channels
+ *                     in the output.  The default value is 2.
+ * + createGraph: mandatory method which takes a context object and does
+ *                everything needed in order to set up the Web Audio graph.
+ *                This function returns the node to be inspected.
+ * + createGraphAsync: async version of createGraph.  This function takes
+ *                     a callback which should be called with an argument
+ *                     set to the node to be inspected when the callee is
+ *                     ready to proceed with the test.  Either this function
+ *                     or createGraph must be provided.
+ * + createExpectedBuffers: optional method which takes a context object and
+ *                          returns either one expected buffer or an array of
+ *                          them, designating what is expected to be observed
+ *                          in the output.  If omitted, the output is expected
+ *                          to be silence.  All buffers must have the same
+ *                          length, which must be a bufferSize supported by
+ *                          ScriptProcessorNode.  This function is guaranteed
+ *                          to be called before createGraph.
+ * + length: property equal to the total number of frames which we are waiting
+ *           to see in the output, mandatory if createExpectedBuffers is not
+ *           provided, in which case it must be a bufferSize supported by
+ *           ScriptProcessorNode (256, 512, 1024, 2048, 4096, 8192, or 16384).
+ *           If createExpectedBuffers is provided then this must be equal to
+ *           the number of expected buffers * the expected buffer length.
+ *
+ * + skipOfflineContextTests: optional. when true, skips running tests on an offline
+ *                            context by circumventing testOnOfflineContext.
+ */
+function runTest() {
+  function done() {
+    SimpleTest.finish();
+  }
+
+  SimpleTest.waitForExplicitFinish();
+  function runTestFunction() {
+    if (!gTest.numberOfChannels) {
+      gTest.numberOfChannels = 2; // default
+    }
+
+    var testLength;
+
+    function runTestOnContext(context, callback, testOutput) {
+      if (!gTest.createExpectedBuffers) {
+        // Assume that the output is silence
+        var expectedBuffers = getEmptyBuffer(context, gTest.length);
+      } else {
+        var expectedBuffers = gTest.createExpectedBuffers(context);
+      }
+      if (!(expectedBuffers instanceof Array)) {
+        expectedBuffers = [expectedBuffers];
+      }
+      var expectedFrames = 0;
+      for (var i = 0; i < expectedBuffers.length; ++i) {
+        is(
+          expectedBuffers[i].numberOfChannels,
+          gTest.numberOfChannels,
+          "Correct number of channels for expected buffer " + i
+        );
+        expectedFrames += expectedBuffers[i].length;
+      }
+      if (gTest.length && gTest.createExpectedBuffers) {
+        is(expectedFrames, gTest.length, "Correct number of expected frames");
+      }
+
+      if (gTest.createGraphAsync) {
+        gTest.createGraphAsync(context, function (nodeToInspect) {
+          testOutput(nodeToInspect, expectedBuffers, callback);
+        });
+      } else {
+        testOutput(gTest.createGraph(context), expectedBuffers, callback);
+      }
+    }
+
+    function testOnNormalContext(callback) {
+      function testOutput(nodeToInspect, expectedBuffers, callback) {
+        testLength = 0;
+        var sp = context.createScriptProcessor(
+          expectedBuffers[0].length,
+          gTest.numberOfChannels,
+          0
+        );
+        nodeToInspect.connect(sp);
+        sp.onaudioprocess = function (e) {
+          var expectedBuffer = expectedBuffers.shift();
+          testLength += expectedBuffer.length;
+          compareBuffers(e.inputBuffer, expectedBuffer);
+          if (!expectedBuffers.length) {
+            sp.onaudioprocess = null;
+            callback();
+          }
+        };
+      }
+      var context = new AudioContext();
+      runTestOnContext(context, callback, testOutput);
+    }
+
+    function testOnOfflineContext(callback, sampleRate) {
+      function testOutput(nodeToInspect, expectedBuffers, callback) {
+        nodeToInspect.connect(context.destination);
+        context.oncomplete = function (e) {
+          var samplesSeen = 0;
+          while (expectedBuffers.length) {
+            var expectedBuffer = expectedBuffers.shift();
+            is(
+              e.renderedBuffer.numberOfChannels,
+              expectedBuffer.numberOfChannels,
+              "Correct number of input buffer channels"
+            );
+            for (var i = 0; i < e.renderedBuffer.numberOfChannels; ++i) {
+              compareChannels(
+                e.renderedBuffer.getChannelData(i),
+                expectedBuffer.getChannelData(i),
+                expectedBuffer.length,
+                samplesSeen,
+                undefined,
+                true
+              );
+            }
+            samplesSeen += expectedBuffer.length;
+          }
+          callback();
+        };
+        context.startRendering();
+      }
+
+      var context = new OfflineAudioContext(
+        gTest.numberOfChannels,
+        testLength,
+        sampleRate
+      );
+      runTestOnContext(context, callback, testOutput);
+    }
+
+    testOnNormalContext(function () {
+      if (!gTest.skipOfflineContextTests) {
+        testOnOfflineContext(function () {
+          testOnOfflineContext(done, 44100);
+        }, 48000);
+      } else {
+        done();
+      }
+    });
+  }
+
+  if (document.readyState !== "complete") {
+    addLoadEvent(runTestFunction);
+  } else {
+    runTestFunction();
+  }
+}