Move webaudio to oilpan

Source/modules/webaudio/AudioContext.cpp

 /*
  * 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
@@ skipping 71 matching lines @@
 {
     // FIXME: It would be nice if the minimum sample-rate could be less than 44.1KHz,
     // but that will require some fixes in HRTFPanner::fftSizeForSampleRate(), and some testing there.
     return sampleRate >= 44100 && sampleRate <= 96000;
 }
 
 // Don't allow more than this number of simultaneous AudioContexts talking to hardware.
 const unsigned MaxHardwareContexts = 6;
 unsigned AudioContext::s_hardwareContextCount = 0;
 
-PassRefPtr<AudioContext> AudioContext::create(Document& document, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<AudioContext> AudioContext::create(Document& document, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
     if (s_hardwareContextCount >= MaxHardwareContexts) {
         exceptionState.throwDOMException(
             SyntaxError,
             "number of hardware contexts reached maximum (" + String::number(MaxHardwareContexts) + ").");
         return nullptr;
     }
 
-    RefPtr<AudioContext> audioContext(adoptRef(new AudioContext(&document)));
+    RefPtrWillBeRawPtr<AudioContext> audioContext(adoptRefWillBeThreadSafeRefCountedGarbageCollected(new AudioContext(&document)));
     audioContext->suspendIfNeeded();
     return audioContext.release();
 }
 
 // Constructor for rendering to the audio hardware.
 AudioContext::AudioContext(Document* document)
     : ActiveDOMObject(document)
     , m_isStopScheduled(false)
     , m_isCleared(false)
     , m_isInitialized(false)
@@ skipping 153 matching lines @@
     // FIXME: see if there's a more direct way to handle this issue.
     callOnMainThread(stopDispatch, this);
 }
 
 bool AudioContext::hasPendingActivity() const
 {
     // According to spec AudioContext must die only after page navigates.
     return !m_isCleared;
 }
 
-PassRefPtr<AudioBuffer> AudioContext::createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<AudioBuffer> AudioContext::createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
 {
-    RefPtr<AudioBuffer> audioBuffer = AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate);
+    RefPtrWillBeRawPtr<AudioBuffer> audioBuffer = AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate);
     if (!audioBuffer.get()) {
         if (numberOfChannels > AudioContext::maxNumberOfChannels()) {
             exceptionState.throwDOMException(
                 NotSupportedError,
                 "requested number of channels (" + String::number(numberOfChannels) + ") exceeds maximum (" + String::number(AudioContext::maxNumberOfChannels()) + ")");
         } else if (sampleRate < AudioBuffer::minAllowedSampleRate() || sampleRate > AudioBuffer::maxAllowedSampleRate()) {
             exceptionState.throwDOMException(
                 NotSupportedError,
                 "requested sample rate (" + String::number(sampleRate)
                 + ") does not lie in the allowed range of "
                 + String::number(AudioBuffer::minAllowedSampleRate())
                 + "-" + String::number(AudioBuffer::maxAllowedSampleRate()) + " Hz");
         } else if (!numberOfFrames) {
             exceptionState.throwDOMException(
                 NotSupportedError,
                 "number of frames must be greater than 0.");
         } else {
             exceptionState.throwDOMException(
                 NotSupportedError,
                 "unable to create buffer of " + String::number(numberOfChannels)
                 + " channel(s) of " + String::number(numberOfFrames)
                 + " frames each.");
         }
         return nullptr;
     }
 
     return audioBuffer;

[haraken, 2014/05/07 04:02:16]

audioBuffer.release();

Not related to your CL, but it's better to have .release(). The same comment for
other create methods in this CL.

[keishi, 2014/05/07 07:59:24]

Done.

 }
 
 void AudioContext::decodeAudioData(ArrayBuffer* audioData, PassOwnPtr<AudioBufferCallback> successCallback, PassOwnPtr<AudioBufferCallback> errorCallback, ExceptionState& exceptionState)
 {
     if (!audioData) {
         exceptionState.throwDOMException(
             SyntaxError,
             "invalid ArrayBuffer for audioData.");
         return;
     }
     m_audioDecoder.decodeAsync(audioData, sampleRate(), successCallback, errorCallback);
 }
 
-PassRefPtr<AudioBufferSourceNode> AudioContext::createBufferSource()
+PassRefPtrWillBeRawPtr<AudioBufferSourceNode> AudioContext::createBufferSource()
 {
     ASSERT(isMainThread());
-    RefPtr<AudioBufferSourceNode> node = AudioBufferSourceNode::create(this, m_destinationNode->sampleRate());
+    RefPtrWillBeRawPtr<AudioBufferSourceNode> node = AudioBufferSourceNode::create(this, m_destinationNode->sampleRate());
 
     // Because this is an AudioScheduledSourceNode, the context keeps a reference until it has finished playing.
     // When this happens, AudioScheduledSourceNode::finish() calls AudioContext::notifyNodeFinishedProcessing().
     refNode(node.get());
 
     return node;
 }
 
-PassRefPtr<MediaElementAudioSourceNode> AudioContext::createMediaElementSource(HTMLMediaElement* mediaElement, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<MediaElementAudioSourceNode> AudioContext::createMediaElementSource(HTMLMediaElement* mediaElement, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
     if (!mediaElement) {
         exceptionState.throwDOMException(
             InvalidStateError,
             "invalid HTMLMedialElement.");
         return nullptr;
     }
 
     // First check if this media element already has a source node.
     if (mediaElement->audioSourceNode()) {
         exceptionState.throwDOMException(
             InvalidStateError,
             "invalid HTMLMediaElement.");
         return nullptr;
     }
 
-    RefPtr<MediaElementAudioSourceNode> node = MediaElementAudioSourceNode::create(this, mediaElement);
+    RefPtrWillBeRawPtr<MediaElementAudioSourceNode> node = MediaElementAudioSourceNode::create(this, mediaElement);
 
     mediaElement->setAudioSourceNode(node.get());
 
     refNode(node.get()); // context keeps reference until node is disconnected
     return node;
 }
 
-PassRefPtr<MediaStreamAudioSourceNode> AudioContext::createMediaStreamSource(MediaStream* mediaStream, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<MediaStreamAudioSourceNode> AudioContext::createMediaStreamSource(MediaStream* mediaStream, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
     if (!mediaStream) {
         exceptionState.throwDOMException(
             InvalidStateError,
             "invalid MediaStream source");
         return nullptr;
     }
 
     MediaStreamTrackVector audioTracks = mediaStream->getAudioTracks();
     if (audioTracks.isEmpty()) {
         exceptionState.throwDOMException(
             InvalidStateError,
             "MediaStream has no audio track");
         return nullptr;
     }
 
     // Use the first audio track in the media stream.
     RefPtr<MediaStreamTrack> audioTrack = audioTracks[0];
     OwnPtr<AudioSourceProvider> provider = audioTrack->createWebAudioSource();
-    RefPtr<MediaStreamAudioSourceNode> node = MediaStreamAudioSourceNode::create(this, mediaStream, audioTrack.get(), provider.release());
+    RefPtrWillBeRawPtr<MediaStreamAudioSourceNode> node = MediaStreamAudioSourceNode::create(this, mediaStream, audioTrack.get(), provider.release());
 
     // FIXME: Only stereo streams are supported right now. We should be able to accept multi-channel streams.
     node->setFormat(2, sampleRate());
 
     refNode(node.get()); // context keeps reference until node is disconnected
     return node;
 }
 
-PassRefPtr<MediaStreamAudioDestinationNode> AudioContext::createMediaStreamDestination()
+PassRefPtrWillBeRawPtr<MediaStreamAudioDestinationNode> AudioContext::createMediaStreamDestination()
 {
     // Set number of output channels to stereo by default.
     return MediaStreamAudioDestinationNode::create(this, 2);
 }
 
-PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(ExceptionState& exceptionState)
 {
     // Set number of input/output channels to stereo by default.
     return createScriptProcessor(0, 2, 2, exceptionState);
 }
 
-PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, ExceptionState& exceptionState)
 {
     // Set number of input/output channels to stereo by default.
     return createScriptProcessor(bufferSize, 2, 2, exceptionState);
 }
 
-PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, ExceptionState& exceptionState)
 {
     // Set number of output channels to stereo by default.
     return createScriptProcessor(bufferSize, numberOfInputChannels, 2, exceptionState);
 }
 
-PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, size_t numberOfOutputChannels, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, size_t numberOfOutputChannels, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
-    RefPtr<ScriptProcessorNode> node = ScriptProcessorNode::create(this, m_destinationNode->sampleRate(), bufferSize, numberOfInputChannels, numberOfOutputChannels);
+    RefPtrWillBeRawPtr<ScriptProcessorNode> node = ScriptProcessorNode::create(this, m_destinationNode->sampleRate(), bufferSize, numberOfInputChannels, numberOfOutputChannels);
 
     if (!node.get()) {
         if (!numberOfInputChannels && !numberOfOutputChannels) {
             exceptionState.throwDOMException(
                 IndexSizeError,
                 "number of input channels and output channels cannot both be zero.");
         } else if (numberOfInputChannels > AudioContext::maxNumberOfChannels()) {
             exceptionState.throwDOMException(
                 IndexSizeError,
                 "number of input channels (" + String::number(numberOfInputChannels)
@@ skipping 11 matching lines @@
                 "buffer size (" + String::number(bufferSize)
                 + ") must be a power of two between 256 and 16384.");
         }
         return nullptr;
     }
 
     refNode(node.get()); // context keeps reference until we stop making javascript rendering callbacks
     return node;
 }
 
-PassRefPtr<BiquadFilterNode> AudioContext::createBiquadFilter()
+PassRefPtrWillBeRawPtr<BiquadFilterNode> AudioContext::createBiquadFilter()
 {
     ASSERT(isMainThread());
     return BiquadFilterNode::create(this, m_destinationNode->sampleRate());
 }
 
-PassRefPtr<WaveShaperNode> AudioContext::createWaveShaper()
+PassRefPtrWillBeRawPtr<WaveShaperNode> AudioContext::createWaveShaper()
 {
     ASSERT(isMainThread());
     return WaveShaperNode::create(this);
 }
 
-PassRefPtr<PannerNode> AudioContext::createPanner()
+PassRefPtrWillBeRawPtr<PannerNode> AudioContext::createPanner()
 {
     ASSERT(isMainThread());
     return PannerNode::create(this, m_destinationNode->sampleRate());
 }
 
-PassRefPtr<ConvolverNode> AudioContext::createConvolver()
+PassRefPtrWillBeRawPtr<ConvolverNode> AudioContext::createConvolver()
 {
     ASSERT(isMainThread());
     return ConvolverNode::create(this, m_destinationNode->sampleRate());
 }
 
-PassRefPtr<DynamicsCompressorNode> AudioContext::createDynamicsCompressor()
+PassRefPtrWillBeRawPtr<DynamicsCompressorNode> AudioContext::createDynamicsCompressor()
 {
     ASSERT(isMainThread());
     return DynamicsCompressorNode::create(this, m_destinationNode->sampleRate());
 }
 
-PassRefPtr<AnalyserNode> AudioContext::createAnalyser()
+PassRefPtrWillBeRawPtr<AnalyserNode> AudioContext::createAnalyser()
 {
     ASSERT(isMainThread());
     return AnalyserNode::create(this, m_destinationNode->sampleRate());
 }
 
-PassRefPtr<GainNode> AudioContext::createGain()
+PassRefPtrWillBeRawPtr<GainNode> AudioContext::createGain()
 {
     ASSERT(isMainThread());
     return GainNode::create(this, m_destinationNode->sampleRate());
 }
 
-PassRefPtr<DelayNode> AudioContext::createDelay(ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<DelayNode> AudioContext::createDelay(ExceptionState& exceptionState)
 {
     const double defaultMaxDelayTime = 1;
     return createDelay(defaultMaxDelayTime, exceptionState);
 }
 
-PassRefPtr<DelayNode> AudioContext::createDelay(double maxDelayTime, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<DelayNode> AudioContext::createDelay(double maxDelayTime, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
-    RefPtr<DelayNode> node = DelayNode::create(this, m_destinationNode->sampleRate(), maxDelayTime, exceptionState);
+    RefPtrWillBeRawPtr<DelayNode> node = DelayNode::create(this, m_destinationNode->sampleRate(), maxDelayTime, exceptionState);
     if (exceptionState.hadException())
         return nullptr;
     return node;
 }
 
-PassRefPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(ExceptionState& exceptionState)
 {
     const unsigned ChannelSplitterDefaultNumberOfOutputs = 6;
     return createChannelSplitter(ChannelSplitterDefaultNumberOfOutputs, exceptionState);
 }
 
-PassRefPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(size_t numberOfOutputs, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(size_t numberOfOutputs, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
-    RefPtr<ChannelSplitterNode> node = ChannelSplitterNode::create(this, m_destinationNode->sampleRate(), numberOfOutputs);
+
+    RefPtrWillBeRawPtr<ChannelSplitterNode> node = ChannelSplitterNode::create(this, m_destinationNode->sampleRate(), numberOfOutputs);
 
     if (!node.get()) {
         exceptionState.throwDOMException(
             IndexSizeError,
             "number of outputs (" + String::number(numberOfOutputs)
             + ") must be between 1 and "
             + String::number(AudioContext::maxNumberOfChannels()) + ".");
         return nullptr;
     }
 
     return node;
 }
 
-PassRefPtr<ChannelMergerNode> AudioContext::createChannelMerger(ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ChannelMergerNode> AudioContext::createChannelMerger(ExceptionState& exceptionState)
 {
     const unsigned ChannelMergerDefaultNumberOfInputs = 6;
     return createChannelMerger(ChannelMergerDefaultNumberOfInputs, exceptionState);
 }
 
-PassRefPtr<ChannelMergerNode> AudioContext::createChannelMerger(size_t numberOfInputs, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<ChannelMergerNode> AudioContext::createChannelMerger(size_t numberOfInputs, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
-    RefPtr<ChannelMergerNode> node = ChannelMergerNode::create(this, m_destinationNode->sampleRate(), numberOfInputs);
+
+    RefPtrWillBeRawPtr<ChannelMergerNode> node = ChannelMergerNode::create(this, m_destinationNode->sampleRate(), numberOfInputs);
 
     if (!node.get()) {
         exceptionState.throwDOMException(
             IndexSizeError,
             "number of inputs (" + String::number(numberOfInputs)
             + ") must be between 1 and "
             + String::number(AudioContext::maxNumberOfChannels()) + ".");
         return nullptr;
     }
 
     return node;
 }
 
-PassRefPtr<OscillatorNode> AudioContext::createOscillator()
+PassRefPtrWillBeRawPtr<OscillatorNode> AudioContext::createOscillator()
 {
     ASSERT(isMainThread());
-    RefPtr<OscillatorNode> node = OscillatorNode::create(this, m_destinationNode->sampleRate());
+
+    RefPtrWillBeRawPtr<OscillatorNode> node = OscillatorNode::create(this, m_destinationNode->sampleRate());
 
     // Because this is an AudioScheduledSourceNode, the context keeps a reference until it has finished playing.
     // When this happens, AudioScheduledSourceNode::finish() calls AudioContext::notifyNodeFinishedProcessing().
     refNode(node.get());
 
     return node;
 }
 
-PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Float32Array* imag, ExceptionState& exceptionState)
+PassRefPtrWillBeRawPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Float32Array* imag, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
 
     if (!real) {
         exceptionState.throwDOMException(
             SyntaxError,
             "invalid real array");
         return nullptr;
     }
 
@@ skipping 261 matching lines @@
 
     context->deleteMarkedNodes();
     context->deref();
 }
 
 void AudioContext::deleteMarkedNodes()
 {
     ASSERT(isMainThread());
 
     // Protect this object from being deleted before we release the mutex locked by AutoLocker.
-    RefPtr<AudioContext> protect(this);
+    RefPtrWillBeRawPtr<AudioContext> protect(this);
     {
         AutoLocker locker(this);
 
         while (size_t n = m_nodesToDelete.size()) {
             AudioNode* node = m_nodesToDelete[n - 1];
             m_nodesToDelete.removeLast();
 
             // Before deleting the node, clear out any AudioNodeInputs from m_dirtySummingJunctions.
             unsigned numberOfInputs = node->numberOfInputs();
             for (unsigned i = 0; i < numberOfInputs; ++i)
                 m_dirtySummingJunctions.remove(node->input(i));
 
             // Before deleting the node, clear out any AudioNodeOutputs from m_dirtyAudioNodeOutputs.
             unsigned numberOfOutputs = node->numberOfOutputs();
             for (unsigned i = 0; i < numberOfOutputs; ++i)
                 m_dirtyAudioNodeOutputs.remove(node->output(i));
-
+#if ENABLE(OILPAN)
+            // Finally, clear the keep alive handle that keeps this
+            // object from being collected.
+            node->clearKeepAlive();
+#else
             // Finally, delete it.
             delete node;
+#endif
         }
         m_isDeletionScheduled = false;
     }
 }
 
 void AudioContext::markSummingJunctionDirty(AudioSummingJunction* summingJunction)
 {
     ASSERT(isGraphOwner());
     m_dirtySummingJunctions.add(summingJunction);
 }
 
 void AudioContext::removeMarkedSummingJunction(AudioSummingJunction* summingJunction)
 {
     ASSERT(isMainThread());
     AutoLocker locker(this);
     m_dirtySummingJunctions.remove(summingJunction);
 }
 
 void AudioContext::markAudioNodeOutputDirty(AudioNodeOutput* output)
 {
     ASSERT(isGraphOwner());
     m_dirtyAudioNodeOutputs.add(output);
 }
 
 void AudioContext::handleDirtyAudioSummingJunctions()
 {
     ASSERT(isGraphOwner());
 
-    for (HashSet<AudioSummingJunction*>::iterator i = m_dirtySummingJunctions.begin(); i != m_dirtySummingJunctions.end(); ++i)
+    for (HashSet<AudioSummingJunction* >::iterator i = m_dirtySummingJunctions.begin(); i != m_dirtySummingJunctions.end(); ++i)

[haraken, 2014/05/07 04:02:16]

mis-edit.

[keishi, 2014/05/07 07:59:24]

Done.

         (*i)->updateRenderingState();
 
     m_dirtySummingJunctions.clear();
 }
 
 void AudioContext::handleDirtyAudioNodeOutputs()
 {
     ASSERT(isGraphOwner());
 
     for (HashSet<AudioNodeOutput*>::iterator i = m_dirtyAudioNodeOutputs.begin(); i != m_dirtyAudioNodeOutputs.end(); ++i)
@@ skipping 75 matching lines @@
     if (!renderedBuffer)
         return;
 
     // Avoid firing the event if the document has already gone away.
     if (executionContext()) {
         // Call the offline rendering completion event listener.
         dispatchEvent(OfflineAudioCompletionEvent::create(renderedBuffer));
     }
 }
 
+void AudioContext::trace(Visitor* visitor)
+{
+    visitor->trace(m_renderTarget);
+    visitor->trace(m_destinationNode);
+    visitor->trace(m_listener);
+    visitor->trace(m_dirtySummingJunctions);

[haraken, 2014/05/07 04:02:16]

m_dirtySummingJunctions should not be traced, since it's still off heap.

[keishi, 2014/05/07 07:59:24]

Done.

+}
+
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)