Bug 864164 - Part 2: Send the AudioBufferSourceNode buffer parameter changes to the stream; r=padenot

content/media/webaudio/AudioBufferSourceNode.cpp

@@ -12,6 +12,7 @@
 #include "AudioDestinationNode.h"
 #include "PannerNode.h"
 #include "speex/speex_resampler.h"
+#include <limits>
 
 namespace mozilla {
 namespace dom {
@@ -400,10 +401,13 @@ AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* aContext)
   : AudioNode(aContext)
   , mLoopStart(0.0)
   , mLoopEnd(0.0)
+  , mOffset(0.0)
+  , mDuration(std::numeric_limits<double>::min())
   , mPlaybackRate(new AudioParam(this, SendPlaybackRateToStream, 1.0f))
   , mPannerNode(nullptr)
   , mLoop(false)
   , mStartCalled(false)
+  , mOffsetAndDurationRemembered(false)
 {
   mStream = aContext->Graph()->CreateAudioNodeStream(
       new AudioBufferSourceNodeEngine(this, aContext->Destination()),
@@ -425,7 +429,7 @@ AudioBufferSourceNode::WrapObject(JSContext* aCx, JSObject* aScope)
 }
 
 void
-AudioBufferSourceNode::Start(JSContext* aCx, double aWhen, double aOffset,
+AudioBufferSourceNode::Start(double aWhen, double aOffset,
                              const Optional<double>& aDuration, ErrorResult& aRv)
 {
   if (mStartCalled) {
@@ -435,40 +439,77 @@ AudioBufferSourceNode::Start(JSContext* aCx, double aWhen, double aOffset,
   mStartCalled = true;
 
   AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
-  if (!mBuffer || !ns) {
+  if (!ns) {
     // Nothing to play, or we're already dead for some reason
     return;
   }
 
-  float rate = mBuffer->SampleRate();
-  int32_t lengthSamples = mBuffer->Length();
-  nsRefPtr<ThreadSharedFloatArrayBufferList> data =
-    mBuffer->GetThreadSharedChannelsForRate(aCx);
+  if (mBuffer) {
+    double duration = aDuration.WasPassed() ?
+                      aDuration.Value() :
+                      std::numeric_limits<double>::min();
+    SendOffsetAndDurationParametersToStream(ns, aOffset, duration);
+  } else {
+    // Remember our argument so that we can use them once we have a buffer
+    mOffset = aOffset;
+    mDuration = aDuration.WasPassed() ?
+                aDuration.Value() :
+                std::numeric_limits<double>::min();
+    mOffsetAndDurationRemembered = true;
+  }
+
+  // Don't set parameter unnecessarily
+  if (aWhen > 0.0) {
+    ns->SetStreamTimeParameter(START, Context()->DestinationStream(), aWhen);
+  }
+
+  MOZ_ASSERT(!mPlayingRef, "We can only accept a successful start() call once");
+  mPlayingRef.Take(this);
+}
+
+void
+AudioBufferSourceNode::SendBufferParameterToStream(JSContext* aCx)
+{
+  AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
+  MOZ_ASSERT(ns, "Why don't we have a stream here?");
+
+  if (mBuffer) {
+    float rate = mBuffer->SampleRate();
+    nsRefPtr<ThreadSharedFloatArrayBufferList> data =
+      mBuffer->GetThreadSharedChannelsForRate(aCx);
+    ns->SetBuffer(data.forget());
+    ns->SetInt32Parameter(SAMPLE_RATE, rate);
+  } else {
+    ns->SetBuffer(nullptr);
+  }
+
+  if (mOffsetAndDurationRemembered) {
+    SendOffsetAndDurationParametersToStream(ns, mOffset, mDuration);
+  }
+}
+
+void
+AudioBufferSourceNode::SendOffsetAndDurationParametersToStream(AudioNodeStream* aStream,
+                                                               double aOffset,
+                                                               double aDuration)
+{
+  float rate = mBuffer ? mBuffer->SampleRate() : Context()->SampleRate();
+  int32_t lengthSamples = mBuffer ? mBuffer->Length() : 0;
   double length = double(lengthSamples) / rate;
   double offset = std::max(0.0, aOffset);
-  double endOffset = aDuration.WasPassed() ?
-      std::min(aOffset + aDuration.Value(), length) : length;
+  double endOffset = aDuration == std::numeric_limits<double>::min() ?
+                     length : std::min(aOffset + aDuration, length);
 
   if (offset >= endOffset) {
     return;
   }
 
-  ns->SetBuffer(data.forget());
-  // Don't set parameter unnecessarily
-  if (aWhen > 0.0) {
-    ns->SetStreamTimeParameter(START, Context()->DestinationStream(), aWhen);
-  }
   int32_t offsetTicks = NS_lround(offset*rate);
   // Don't set parameter unnecessarily
   if (offsetTicks > 0) {
-    ns->SetInt32Parameter(OFFSET, offsetTicks);
+    aStream->SetInt32Parameter(OFFSET, offsetTicks);
   }
-  ns->SetInt32Parameter(DURATION,
-      NS_lround(endOffset*rate) - offsetTicks);
-  ns->SetInt32Parameter(SAMPLE_RATE, rate);
-
-  MOZ_ASSERT(!mPlayingRef, "We can only accept a successful start() call once");
-  mPlayingRef.Take(this);
+  aStream->SetInt32Parameter(DURATION, NS_lround(endOffset*rate) - offsetTicks);
 }
 
 void
@@ -479,6 +520,12 @@ AudioBufferSourceNode::Stop(double aWhen, ErrorResult& aRv)
     return;
   }
 
+  if (!mBuffer) {
+    // We don't have a buffer, so the stream is never marked as finished.
+    // Therefore we need to drop our playing ref right now.
+    mPlayingRef.Drop(this);
+  }
+
   AudioNodeStream* ns = static_cast<AudioNodeStream*>(mStream.get());
   if (!ns) {
     // We've already stopped and had our stream shut down

content/media/webaudio/AudioBufferSourceNode.h

@@ -55,18 +55,18 @@ public:
 
   virtual JSObject* WrapObject(JSContext* aCx, JSObject* aScope);
 
-  void Start(JSContext* aCx, double aWhen, double aOffset,
+  void Start(double aWhen, double aOffset,
              const Optional<double>& aDuration, ErrorResult& aRv);
-  void NoteOn(JSContext* aCx, double aWhen, ErrorResult& aRv)
+  void NoteOn(double aWhen, ErrorResult& aRv)
   {
-    Start(aCx, aWhen, 0.0, Optional<double>(), aRv);
+    Start(aWhen, 0.0, Optional<double>(), aRv);
   }
-  void NoteGrainOn(JSContext* aCx, double aWhen, double aOffset,
+  void NoteGrainOn(double aWhen, double aOffset,
                    double aDuration, ErrorResult& aRv)
   {
     Optional<double> duration;
     duration.Construct(aDuration);
-    Start(aCx, aWhen, aOffset, duration, aRv);
+    Start(aWhen, aOffset, duration, aRv);
   }
   void Stop(double aWhen, ErrorResult& aRv);
   void NoteOff(double aWhen, ErrorResult& aRv)
@@ -74,13 +74,14 @@ public:
     Stop(aWhen, aRv);
   }
 
-  AudioBuffer* GetBuffer() const
+  AudioBuffer* GetBuffer(JSContext* aCx) const
   {
     return mBuffer;
   }
-  void SetBuffer(AudioBuffer* aBuffer)
+  void SetBuffer(JSContext* aCx, AudioBuffer* aBuffer)
   {
     mBuffer = aBuffer;
+    SendBufferParameterToStream(aCx);
   }
   AudioParam* PlaybackRate() const
   {
@@ -136,17 +137,24 @@ private:
   };
 
   void SendLoopParametersToStream();
+  void SendBufferParameterToStream(JSContext* aCx);
+  void SendOffsetAndDurationParametersToStream(AudioNodeStream* aStream,
+                                               double aOffset,
+                                               double aDuration);
   static void SendPlaybackRateToStream(AudioNode* aNode);
 
 private:
   double mLoopStart;
   double mLoopEnd;
+  double mOffset;
+  double mDuration;
   nsRefPtr<AudioBuffer> mBuffer;
   nsRefPtr<AudioParam> mPlaybackRate;
   PannerNode* mPannerNode;
   SelfReference<AudioBufferSourceNode> mPlayingRef; // a reference to self while playing
   bool mLoop;
   bool mStartCalled;
+  bool mOffsetAndDurationRemembered;
 };
 
 }

content/media/webaudio/test/Makefile.in

@@ -25,6 +25,7 @@ MOCHITEST_FILES := \
   test_audioBufferSourceNode.html \
   test_audioBufferSourceNodeLoop.html \
   test_audioBufferSourceNodeLoopStartEnd.html \
+  test_audioBufferSourceNodeNullBuffer.html \
   test_badConnect.html \
   test_biquadFilterNode.html \
   test_currentTime.html \

content/media/webaudio/test/test_audioBufferSourceNode.html

@@ -21,10 +21,10 @@ addLoadEvent(function() {
   }
 
   var source = context.createBufferSource();
-  source.buffer = buffer;
 
   var sp = context.createScriptProcessor(2048);
   source.start(0);
+  source.buffer = buffer;
   source.connect(sp);
   sp.connect(context.destination);
   sp.onaudioprocess = function(e) {

content/media/webaudio/test/test_audioBufferSourceNodeNullBuffer.html

@@ -0,0 +1,42 @@
+<!DOCTYPE HTML>
+<html>
+<head>
+  <title>Test AudioBufferSourceNode</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">
+
+SimpleTest.waitForExplicitFinish();
+addLoadEvent(function() {
+  SpecialPowers.setBoolPref("media.webaudio.enabled", true);
+
+  var context = new AudioContext();
+  var expectedBuffer = context.createBuffer(1, 2048, context.sampleRate); // silence
+
+  var source = context.createBufferSource();
+
+  var sp = context.createScriptProcessor(2048);
+  source.start(0);
+  source.buffer = null;
+  is(source.buffer, null, "Try playing back a null buffer");
+  source.connect(sp);
+  sp.connect(context.destination);
+  sp.onaudioprocess = function(e) {
+    compareBuffers(e.inputBuffer.getChannelData(0), expectedBuffer.getChannelData(0));
+    compareBuffers(e.inputBuffer.getChannelData(1), expectedBuffer.getChannelData(0));
+
+    sp.onaudioprocess = null;
+
+    SpecialPowers.clearUserPref("media.webaudio.enabled");
+    SimpleTest.finish();
+  };
+});
+
+</script>
+</pre>
+</body>
+</html>

dom/bindings/Bindings.conf

@@ -106,7 +106,7 @@ DOMInterfaces = {
 },
 
 'AudioBufferSourceNode': {
-    'implicitJSContext': [ 'start', 'noteOn', 'noteGrainOn' ],
+    'implicitJSContext': [ 'buffer' ],
     'resultNotAddRefed': [ 'playbackRate' ],
 },