--- a/content/media/webaudio/AudioBufferSourceNode.cpp
+++ b/content/media/webaudio/AudioBufferSourceNode.cpp
@@ -3,64 +3,62 @@
 /* This Source Code Form is subject to the terms of the Mozilla Public
  * License, v. 2.0. If a copy of the MPL was not distributed with this
  * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
 
 #include "AudioBufferSourceNode.h"
 #include "mozilla/dom/AudioBufferSourceNodeBinding.h"
 #include "nsMathUtils.h"
 #include "AudioNodeEngine.h"
+#include "AudioNodeStream.h"
+#include "AudioDestinationNode.h"
 #include "PannerNode.h"
-#include "GainProcessor.h"
 #include "speex/speex_resampler.h"
 #include <limits>
 
 namespace mozilla {
 namespace dom {
 
 NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN(AudioBufferSourceNode)
   NS_IMPL_CYCLE_COLLECTION_UNLINK(mBuffer)
   NS_IMPL_CYCLE_COLLECTION_UNLINK(mPlaybackRate)
-  NS_IMPL_CYCLE_COLLECTION_UNLINK(mGain)
   if (tmp->Context()) {
     // AudioNode's Unlink implementation disconnects us from the graph
     // too, but we need to do this right here to make sure that
     // UnregisterAudioBufferSourceNode can properly untangle us from
     // the possibly connected PannerNodes.
     tmp->DisconnectFromGraph();
     tmp->Context()->UnregisterAudioBufferSourceNode(tmp);
   }
 NS_IMPL_CYCLE_COLLECTION_UNLINK_END_INHERITED(AudioNode)
 
 NS_IMPL_CYCLE_COLLECTION_TRAVERSE_BEGIN_INHERITED(AudioBufferSourceNode, AudioNode)
   NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mBuffer)
   NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mPlaybackRate)
-  NS_IMPL_CYCLE_COLLECTION_TRAVERSE(mGain)
 NS_IMPL_CYCLE_COLLECTION_TRAVERSE_END
 
 NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(AudioBufferSourceNode)
 NS_INTERFACE_MAP_END_INHERITING(AudioNode)
 
 NS_IMPL_ADDREF_INHERITED(AudioBufferSourceNode, AudioNode)
 NS_IMPL_RELEASE_INHERITED(AudioBufferSourceNode, AudioNode)
 
-class AudioBufferSourceNodeEngine : public AudioNodeEngine,
-                                    public GainProcessor
+class AudioBufferSourceNodeEngine : public AudioNodeEngine
 {
 public:
   explicit AudioBufferSourceNodeEngine(AudioNode* aNode,
                                        AudioDestinationNode* aDestination) :
     AudioNodeEngine(aNode),
-    GainProcessor(aDestination),
     mStart(0), mStop(TRACK_TICKS_MAX),
     mResampler(nullptr),
     mOffset(0), mDuration(0),
     mLoopStart(0), mLoopEnd(0),
     mBufferSampleRate(0), mPosition(0), mChannels(0), mPlaybackRate(1.0f),
     mDopplerShift(1.0f),
+    mDestination(static_cast<AudioNodeStream*>(aDestination->Stream())),
     mPlaybackRateTimeline(1.0f), mLoop(false)
   {}
 
   ~AudioBufferSourceNodeEngine()
   {
     if (mResampler) {
       speex_resampler_destroy(mResampler);
     }
@@ -79,19 +77,16 @@ public:
       if (mResampler && mPlaybackRateTimeline.HasSimpleValue() &&
           mPlaybackRateTimeline.GetValue() == 1.0 &&
           mBufferSampleRate == aSampleRate) {
         speex_resampler_destroy(mResampler);
         mResampler = nullptr;
       }
       WebAudioUtils::ConvertAudioParamToTicks(mPlaybackRateTimeline, nullptr, mDestination);
       break;
-    case AudioBufferSourceNode::GAIN:
-      SetGainParameter(aValue);
-      break;
     default:
       NS_ERROR("Bad AudioBufferSourceNodeEngine TimelineParameter");
     }
   }
   virtual void SetStreamTimeParameter(uint32_t aIndex, TrackTicks aParam)
   {
     switch (aIndex) {
     case AudioBufferSourceNode::START: mStart = aParam; break;
@@ -396,31 +391,16 @@ public:
         if (mOffset + t < mDuration) {
           CopyFromBuffer(aStream, aOutput, channels, &written, &currentPosition, mOffset + t, mDuration);
         } else {
           FillWithZeroes(aOutput, channels, &written, &currentPosition, TRACK_TICKS_MAX);
         }
       }
     }
 
-    // Process the gain on the AudioBufferSourceNode
-    if (!aOutput->IsNull()) {
-      if (!mGain.HasSimpleValue() &&
-          aOutput->mBuffer == mBuffer) {
-        // If we have borrowed out buffer, make sure to allocate a new one in case
-        // the gain value is not a simple value.
-        nsTArray<const void*> oldChannels;
-        oldChannels.AppendElements(aOutput->mChannelData);
-        AllocateAudioBlock(channels, aOutput);
-        ProcessGain(aStream, 1.0f, oldChannels, aOutput);
-      } else {
-        ProcessGain(aStream, 1.0f, aOutput->mChannelData, aOutput);
-      }
-    }
-
     // We've finished if we've gone past mStop, or if we're past mDuration when
     // looping is disabled.
     if (currentPosition >= mStop ||
         (!mLoop && currentPosition - mStart + mOffset > mDuration)) {
       *aFinished = true;
     }
   }
 
@@ -432,39 +412,38 @@ public:
   int32_t mDuration;
   int32_t mLoopStart;
   int32_t mLoopEnd;
   int32_t mBufferSampleRate;
   uint32_t mPosition;
   uint32_t mChannels;
   float mPlaybackRate;
   float mDopplerShift;
+  AudioNodeStream* mDestination;
   AudioParamTimeline mPlaybackRateTimeline;
   bool mLoop;
 };
 
 AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* aContext)
   : AudioNode(aContext,
               2,
               ChannelCountMode::Max,
               ChannelInterpretation::Speakers)
   , mLoopStart(0.0)
   , mLoopEnd(0.0)
   , mOffset(0.0)
   , mDuration(std::numeric_limits<double>::min())
   , mPlaybackRate(new AudioParam(this, SendPlaybackRateToStream, 1.0f))
-  , mGain(new AudioParam(this, SendGainToStream, 1.0f))
   , mLoop(false)
   , mStartCalled(false)
   , mStopped(false)
 {
-  AudioBufferSourceNodeEngine* engine =
-      new AudioBufferSourceNodeEngine(this, aContext->Destination());
-  mStream = aContext->Graph()->CreateAudioNodeStream(engine, MediaStreamGraph::INTERNAL_STREAM);
-  engine->SetSourceStream(static_cast<AudioNodeStream*> (mStream.get()));
+  mStream = aContext->Graph()->CreateAudioNodeStream(
+      new AudioBufferSourceNodeEngine(this, aContext->Destination()),
+      MediaStreamGraph::INTERNAL_STREAM);
   mStream->AddMainThreadListener(this);
 }
 
 AudioBufferSourceNode::~AudioBufferSourceNode()
 {
   if (Context()) {
     Context()->UnregisterAudioBufferSourceNode(this);
   }
@@ -638,23 +617,16 @@ AudioBufferSourceNode::NotifyMainThreadS
 void
 AudioBufferSourceNode::SendPlaybackRateToStream(AudioNode* aNode)
 {
   AudioBufferSourceNode* This = static_cast<AudioBufferSourceNode*>(aNode);
   SendTimelineParameterToStream(This, PLAYBACKRATE, *This->mPlaybackRate);
 }
 
 void
-AudioBufferSourceNode::SendGainToStream(AudioNode* aNode)
-{
-  AudioBufferSourceNode* This = static_cast<AudioBufferSourceNode*>(aNode);
-  SendTimelineParameterToStream(This, GAIN, *This->mGain);
-}
-
-void
 AudioBufferSourceNode::SendDopplerShiftToStream(double aDopplerShift)
 {
   SendDoubleParameterToStream(DOPPLERSHIFT, aDopplerShift);
 }
 
 void
 AudioBufferSourceNode::SendLoopParametersToStream()
 {

--- a/content/media/webaudio/AudioBufferSourceNode.h
+++ b/content/media/webaudio/AudioBufferSourceNode.h
@@ -71,20 +71,16 @@ public:
     mBuffer = aBuffer;
     SendBufferParameterToStream(aCx);
     SendLoopParametersToStream();
   }
   AudioParam* PlaybackRate() const
   {
     return mPlaybackRate;
   }
-  AudioParam* Gain() const
-  {
-    return mGain;
-  }
   bool Loop() const
   {
     return mLoop;
   }
   void SetLoop(bool aLoop)
   {
     mLoop = aLoop;
     SendLoopParametersToStream();
@@ -123,36 +119,33 @@ private:
     START,
     STOP,
     OFFSET,
     DURATION,
     LOOP,
     LOOPSTART,
     LOOPEND,
     PLAYBACKRATE,
-    GAIN,
     DOPPLERSHIFT
   };
 
   void SendLoopParametersToStream();
   void SendBufferParameterToStream(JSContext* aCx);
   void SendOffsetAndDurationParametersToStream(AudioNodeStream* aStream,
                                                double aOffset,
                                                double aDuration);
   static void SendPlaybackRateToStream(AudioNode* aNode);
-  static void SendGainToStream(AudioNode* aNode);
 
 private:
   double mLoopStart;
   double mLoopEnd;
   double mOffset;
   double mDuration;
   nsRefPtr<AudioBuffer> mBuffer;
   nsRefPtr<AudioParam> mPlaybackRate;
-  nsRefPtr<AudioParam> mGain;
   SelfReference<AudioBufferSourceNode> mPlayingRef; // a reference to self while playing
   bool mLoop;
   bool mStartCalled;
   bool mStopped;
 };
 
 }
 }

--- a/content/media/webaudio/GainNode.cpp
+++ b/content/media/webaudio/GainNode.cpp
@@ -2,78 +2,108 @@
 /* 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 "AudioNodeEngine.h"
-#include "GainProcessor.h"
+#include "AudioNodeStream.h"
+#include "AudioDestinationNode.h"
+#include "WebAudioUtils.h"
 
 namespace mozilla {
 namespace dom {
 
 NS_IMPL_CYCLE_COLLECTION_INHERITED_1(GainNode, AudioNode,
                                      mGain)
 
 NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION_INHERITED(GainNode)
 NS_INTERFACE_MAP_END_INHERITING(AudioNode)
 
 NS_IMPL_ADDREF_INHERITED(GainNode, AudioNode)
 NS_IMPL_RELEASE_INHERITED(GainNode, AudioNode)
 
-class GainNodeEngine : public AudioNodeEngine,
-                       public GainProcessor
+class GainNodeEngine : public AudioNodeEngine
 {
 public:
   GainNodeEngine(AudioNode* aNode, AudioDestinationNode* aDestination)
     : AudioNodeEngine(aNode)
-    , GainProcessor(aDestination)
+    , mSource(nullptr)
+    , mDestination(static_cast<AudioNodeStream*> (aDestination->Stream()))
+    // Keep the default value in sync with the default value in GainNode::GainNode.
+    , mGain(1.f)
   {
   }
 
+  void SetSourceStream(AudioNodeStream* aSource)
+  {
+    mSource = aSource;
+  }
+
   enum Parameters {
     GAIN
   };
   void SetTimelineParameter(uint32_t aIndex,
                             const AudioParamTimeline& aValue,
                             TrackRate aSampleRate) MOZ_OVERRIDE
   {
     switch (aIndex) {
     case GAIN:
-      SetGainParameter(aValue);
+      MOZ_ASSERT(mSource && mDestination);
+      mGain = aValue;
+      WebAudioUtils::ConvertAudioParamToTicks(mGain, mSource, mDestination);
       break;
     default:
       NS_ERROR("Bad GainNodeEngine TimelineParameter");
     }
   }
 
   virtual void ProduceAudioBlock(AudioNodeStream* aStream,
                                  const AudioChunk& aInput,
                                  AudioChunk* aOutput,
                                  bool* aFinished)
   {
     MOZ_ASSERT(mSource == aStream, "Invalid source stream");
 
     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
+      *aOutput = aInput;
+      aOutput->mVolume *= mGain.GetValue();
     } else {
-      if (mGain.HasSimpleValue()) {
-        // Copy the input chunk to the output chunk, since we will only be
-        // changing the mVolume member.
-        *aOutput = aInput;
-      } else {
-        // Create a new output chunk to avoid modifying the input chunk.
-        AllocateAudioBlock(aInput.mChannelData.Length(), aOutput);
+      // 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.
+      AllocateAudioBlock(aInput.mChannelData.Length(), aOutput);
+
+      // Compute the gain values for the duration of the input AudioChunk
+      // XXX we need to add a method to AudioEventTimeline to compute this buffer directly.
+      float computedGain[WEBAUDIO_BLOCK_SIZE];
+      for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
+        TrackTicks tick = aStream->GetCurrentPosition();
+        computedGain[counter] = mGain.GetValueAtTime(tick, counter) * aInput.mVolume;
       }
-      ProcessGain(aStream, aInput.mVolume, aInput.mChannelData, aOutput);
+
+      // Apply the gain to the output buffer
+      for (size_t channel = 0; channel < aOutput->mChannelData.Length(); ++channel) {
+        const float* inputBuffer = static_cast<const float*> (aInput.mChannelData[channel]);
+        float* buffer = static_cast<float*> (const_cast<void*>
+                          (aOutput->mChannelData[channel]));
+        AudioBlockCopyChannelWithScale(inputBuffer, computedGain, buffer);
+      }
     }
   }
+
+  AudioNodeStream* mSource;
+  AudioNodeStream* mDestination;
+  AudioParamTimeline mGain;
 };
 
 GainNode::GainNode(AudioContext* aContext)
   : AudioNode(aContext,
               2,
               ChannelCountMode::Max,
               ChannelInterpretation::Speakers)
   , mGain(new AudioParam(this, SendGainToStream, 1.0f))

deleted file mode 100644
--- a/content/media/webaudio/GainProcessor.h
+++ /dev/null
@@ -1,85 +0,0 @@
-/* -*- 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 GainProcessor_h_
-#define GainProcessor_h_
-
-#include "AudioNodeStream.h"
-#include "AudioDestinationNode.h"
-#include "WebAudioUtils.h"
-
-namespace mozilla {
-namespace dom {
-
-// This class implements the gain processing logic used by GainNodeEngine
-// and AudioBufferSourceNodeEngine.
-class GainProcessor
-{
-public:
-  explicit GainProcessor(AudioDestinationNode* aDestination)
-    : mSource(nullptr)
-    , mDestination(static_cast<AudioNodeStream*> (aDestination->Stream()))
-    , mGain(1.f)
-  {
-  }
-
-  void SetSourceStream(AudioNodeStream* aSource)
-  {
-    mSource = aSource;
-  }
-
-  void SetGainParameter(const AudioParamTimeline& aValue)
-  {
-    MOZ_ASSERT(mSource && mDestination);
-    mGain = aValue;
-    WebAudioUtils::ConvertAudioParamToTicks(mGain, mSource, mDestination);
-  }
-
-  void ProcessGain(AudioNodeStream* aStream,
-                   float aInputVolume,
-                   const nsTArray<const void*>& aInputChannelData,
-                   AudioChunk* aOutput)
-  {
-    MOZ_ASSERT(mSource == aStream, "Invalid source stream");
-
-    if (mGain.HasSimpleValue()) {
-      // Optimize the case where we only have a single value set as the volume
-      aOutput->mVolume *= mGain.GetValue();
-    } 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.
-
-      // Compute the gain values for the duration of the input AudioChunk
-      // XXX we need to add a method to AudioEventTimeline to compute this buffer directly.
-      float computedGain[WEBAUDIO_BLOCK_SIZE];
-      for (size_t counter = 0; counter < WEBAUDIO_BLOCK_SIZE; ++counter) {
-        TrackTicks tick = aStream->GetCurrentPosition();
-        computedGain[counter] = mGain.GetValueAtTime(tick, counter) * aInputVolume;
-      }
-
-      // Apply the gain to the output buffer
-      MOZ_ASSERT(aInputChannelData.Length() == aOutput->mChannelData.Length());
-      for (size_t channel = 0; channel < aOutput->mChannelData.Length(); ++channel) {
-        const float* inputBuffer = static_cast<const float*> (aInputChannelData[channel]);
-        float* buffer = static_cast<float*> (const_cast<void*>
-                          (aOutput->mChannelData[channel]));
-        AudioBlockCopyChannelWithScale(inputBuffer, computedGain, buffer);
-      }
-    }
-  }
-
-protected:
-  AudioNodeStream* mSource;
-  AudioNodeStream* mDestination;
-  AudioParamTimeline mGain;
-};
-
-}
-}
-
-#endif
-

--- a/content/media/webaudio/test/Makefile.in
+++ b/content/media/webaudio/test/Makefile.in
@@ -35,34 +35,31 @@ MOCHITEST_FILES := \
   test_audioParamLinearRamp.html \
   test_audioParamSetCurveAtTime.html \
   test_audioParamSetCurveAtTimeZeroDuration.html \
   test_audioParamSetTargetAtTime.html \
   test_audioParamSetValueAtTime.html \
   test_audioParamTimelineDestinationOffset.html \
   test_audioBufferSourceNode.html \
   test_audioBufferSourceNodeEnded.html \
-  test_audioBufferSourceNodeGain.html \
-  test_audioBufferSourceNodeGainInLoop.html \
   test_audioBufferSourceNodeLazyLoopParam.html \
   test_audioBufferSourceNodeLoop.html \
   test_audioBufferSourceNodeLoopStartEnd.html \
   test_audioBufferSourceNodeLoopStartEndSame.html \
   test_audioBufferSourceNodeNullBuffer.html \
   test_badConnect.html \
   test_biquadFilterNode.html \
   test_channelMergerNode.html \
   test_channelMergerNodeWithVolume.html \
   test_channelSplitterNode.html \
   test_channelSplitterNodeWithVolume.html \
   test_currentTime.html \
   test_delayNode.html \
   test_delayNodeSmallMaxDelay.html \
   test_delayNodeWithGain.html \
-  test_delayNodeWithGainAlternate.html \
   test_dynamicsCompressorNode.html \
   test_gainNode.html \
   test_gainNodeInLoop.html \
   test_mediaDecoding.html \
   test_mixingRules.html \
   test_nodeToParamConnection.html \
   test_OfflineAudioContext.html \
   test_offlineDestinationChannelCountLess.html \

--- a/content/media/webaudio/test/test_audioBufferSourceNode.html
+++ b/content/media/webaudio/test/test_audioBufferSourceNode.html
@@ -14,18 +14,16 @@ var gTest = {
   length: 4096,
   createGraph: function(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();
-    ok("gain" in source, "AudioBufferSourceNode.gain must exist");
-    is(source.gain.value, 1, "AudioBufferSourceNode.gain's default value must be 1");
 
     var sp = context.createScriptProcessor(2048);
     source.start(0);
     source.buffer = buffer;
     return source;
   },
   createExpectedBuffers: function(context) {
     var buffers = [];

deleted file mode 100644
--- a/content/media/webaudio/test/test_audioBufferSourceNodeGain.html
+++ /dev/null
@@ -1,45 +0,0 @@
-<!DOCTYPE HTML>
-<html>
-<head>
-  <title>Test AudioBufferSourceNode.gain</title>
-  <script type="text/javascript" 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: function(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.gain.value = 0.5;
-
-    source.start(0);
-    return source;
-  },
-  createExpectedBuffers: function(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>

deleted file mode 100644
--- a/content/media/webaudio/test/test_audioBufferSourceNodeGainInLoop.html
+++ /dev/null
@@ -1,46 +0,0 @@
-<!DOCTYPE HTML>
-<html>
-<head>
-  <title>Test AudioBufferSourceNode.gain in presence of loops</title>
-  <script type="text/javascript" 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: function(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);
-
-    // Adjust the gain in a way that we don't just end up modifying AudioChunk::mVolume
-    source.gain.setValueAtTime(0.5, 0);
-    return source;
-  },
-  createExpectedBuffers: function(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>

deleted file mode 100644
--- a/content/media/webaudio/test/test_delayNodeWithGainAlternate.html
+++ /dev/null
@@ -1,51 +0,0 @@
-<!DOCTYPE HTML>
-<html>
-<head>
-  <title>Test DelayNode with an AudioBufferSourceNode.gain value</title>
-  <script type="text/javascript" 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: function(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;
-    source.gain.value = 0.5;
-
-    source.connect(delay);
-
-    // Delay the source stream by 2048 frames
-    delay.delayTime.value = 2048 / context.sampleRate;
-
-    source.start(0);
-    return delay;
-  },
-  createExpectedBuffers: function(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>

--- a/dom/bindings/Bindings.conf
+++ b/dom/bindings/Bindings.conf
@@ -102,17 +102,17 @@ DOMInterfaces = {
 'AudioContext': {
     'implicitJSContext': [ 'createBuffer' ],
     'nativeOwnership': 'refcounted',
     'resultNotAddRefed': [ 'destination', 'listener' ],
 },
 
 'AudioBufferSourceNode': {
     'implicitJSContext': [ 'buffer' ],
-    'resultNotAddRefed': [ 'gain', 'playbackRate' ],
+    'resultNotAddRefed': [ 'playbackRate' ],
 },
 
 'AudioListener' : {
     'nativeOwnership': 'refcounted'
 },
 
 'AudioNode' : {
     'concrete': False,

--- a/dom/webidl/AudioBufferSourceNode.webidl
+++ b/dom/webidl/AudioBufferSourceNode.webidl
@@ -11,17 +11,16 @@
  */
 
 [PrefControlled]
 interface AudioBufferSourceNode : AudioNode {
 
     attribute AudioBuffer? buffer;
 
     readonly attribute AudioParam playbackRate;
-    readonly attribute AudioParam gain;
 
     attribute boolean loop;
     attribute double loopStart;
     attribute double loopEnd;
 
     [Throws]
     void start(optional double when = 0, optional double grainOffset = 0,
                optional double grainDuration);