diff options
Diffstat (limited to 'chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.cpp')
| -rw-r--r-- | chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.cpp | 333 |
1 files changed, 116 insertions, 217 deletions
diff --git a/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.cpp b/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.cpp index 5ae5ffe99f1..a77c9988d43 100644 --- a/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.cpp +++ b/chromium/third_party/WebKit/Source/modules/webaudio/AudioContext.cpp @@ -86,91 +86,67 @@ bool AudioContext::isSampleRateRangeGood(float sampleRate) } // Don't allow more than this number of simultaneous AudioContexts talking to hardware. -const unsigned MaxHardwareContexts = 4; +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 0; + return nullptr; } - RefPtr<AudioContext> audioContext(adoptRef(new AudioContext(&document))); + RefPtrWillBeRawPtr<AudioContext> audioContext(adoptRefWillBeThreadSafeRefCountedGarbageCollected(new AudioContext(&document))); audioContext->suspendIfNeeded(); return audioContext.release(); } -PassRefPtr<AudioContext> AudioContext::create(Document& document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState) -{ - document.addConsoleMessage(JSMessageSource, WarningMessageLevel, "Deprecated AudioContext constructor: use OfflineAudioContext instead"); - return OfflineAudioContext::create(&document, numberOfChannels, numberOfFrames, sampleRate, exceptionState); -} - // Constructor for rendering to the audio hardware. AudioContext::AudioContext(Document* document) : ActiveDOMObject(document) , m_isStopScheduled(false) + , m_isCleared(false) , m_isInitialized(false) - , m_isAudioThreadFinished(false) - , m_destinationNode(0) + , m_destinationNode(nullptr) , m_isDeletionScheduled(false) , m_automaticPullNodesNeedUpdating(false) , m_connectionCount(0) , m_audioThread(0) , m_graphOwnerThread(UndefinedThreadIdentifier) , m_isOfflineContext(false) - , m_activeSourceCount(0) { - constructCommon(); + ScriptWrappable::init(this); m_destinationNode = DefaultAudioDestinationNode::create(this); - // This sets in motion an asynchronous loading mechanism on another thread. - // We can check m_hrtfDatabaseLoader->isLoaded() to find out whether or not it has been fully loaded. - // It's not that useful to have a callback function for this since the audio thread automatically starts rendering on the graph - // when this has finished (see AudioDestinationNode). - m_hrtfDatabaseLoader = HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(sampleRate()); + initialize(); } // Constructor for offline (non-realtime) rendering. AudioContext::AudioContext(Document* document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate) : ActiveDOMObject(document) , m_isStopScheduled(false) + , m_isCleared(false) , m_isInitialized(false) - , m_isAudioThreadFinished(false) - , m_destinationNode(0) + , m_destinationNode(nullptr) + , m_isDeletionScheduled(false) , m_automaticPullNodesNeedUpdating(false) , m_connectionCount(0) , m_audioThread(0) , m_graphOwnerThread(UndefinedThreadIdentifier) , m_isOfflineContext(true) - , m_activeSourceCount(0) { - constructCommon(); - - m_hrtfDatabaseLoader = HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(sampleRate); + ScriptWrappable::init(this); // Create a new destination for offline rendering. m_renderTarget = AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate); - ASSERT(m_renderTarget); - m_destinationNode = OfflineAudioDestinationNode::create(this, m_renderTarget.get()); - ASSERT(m_destinationNode); -} + if (m_renderTarget.get()) + m_destinationNode = OfflineAudioDestinationNode::create(this, m_renderTarget.get()); -void AudioContext::constructCommon() -{ - ScriptWrappable::init(this); - // According to spec AudioContext must die only after page navigate. - // Lets mark it as ActiveDOMObject with pending activity and unmark it in clear method. - setPendingActivity(this); - - FFTFrame::initialize(); - - m_listener = AudioListener::create(); + initialize(); } AudioContext::~AudioContext() @@ -180,7 +156,6 @@ AudioContext::~AudioContext() #endif // AudioNodes keep a reference to their context, so there should be no way to be in the destructor if there are still AudioNodes around. ASSERT(!m_isInitialized); - ASSERT(m_isStopScheduled); ASSERT(!m_nodesToDelete.size()); ASSERT(!m_referencedNodes.size()); ASSERT(!m_finishedNodes.size()); @@ -190,27 +165,27 @@ AudioContext::~AudioContext() ASSERT(!m_renderingAutomaticPullNodes.size()); } -void AudioContext::lazyInitialize() +void AudioContext::initialize() { - if (!m_isInitialized) { - // Don't allow the context to initialize a second time after it's already been explicitly uninitialized. - ASSERT(!m_isAudioThreadFinished); - if (!m_isAudioThreadFinished) { - if (m_destinationNode.get()) { - m_destinationNode->initialize(); + if (isInitialized()) + return; + + FFTFrame::initialize(); + m_listener = AudioListener::create(); - if (!isOfflineContext()) { - // This starts the audio thread. The destination node's provideInput() method will now be called repeatedly to render audio. - // Each time provideInput() is called, a portion of the audio stream is rendered. Let's call this time period a "render quantum". - // NOTE: for now default AudioContext does not need an explicit startRendering() call from JavaScript. - // We may want to consider requiring it for symmetry with OfflineAudioContext. - m_destinationNode->startRendering(); - ++s_hardwareContextCount; - } + if (m_destinationNode.get()) { + m_destinationNode->initialize(); - } - m_isInitialized = true; + if (!isOfflineContext()) { + // This starts the audio thread. The destination node's provideInput() method will now be called repeatedly to render audio. + // Each time provideInput() is called, a portion of the audio stream is rendered. Let's call this time period a "render quantum". + // NOTE: for now default AudioContext does not need an explicit startRendering() call from JavaScript. + // We may want to consider requiring it for symmetry with OfflineAudioContext. + m_destinationNode->startRendering(); + ++s_hardwareContextCount; } + + m_isInitialized = true; } } @@ -222,28 +197,24 @@ void AudioContext::clear() // Audio thread is dead. Nobody will schedule node deletion action. Let's do it ourselves. do { - deleteMarkedNodes(); - m_nodesToDelete.append(m_nodesMarkedForDeletion); + m_nodesToDelete.appendVector(m_nodesMarkedForDeletion); m_nodesMarkedForDeletion.clear(); + deleteMarkedNodes(); } while (m_nodesToDelete.size()); - // It was set in constructCommon. - unsetPendingActivity(this); + m_isCleared = true; } void AudioContext::uninitialize() { ASSERT(isMainThread()); - if (!m_isInitialized) + if (!isInitialized()) return; // This stops the audio thread and all audio rendering. m_destinationNode->uninitialize(); - // Don't allow the context to initialize a second time after it's already been explicitly uninitialized. - m_isAudioThreadFinished = true; - if (!isOfflineContext()) { ASSERT(s_hardwareContextCount); --s_hardwareContextCount; @@ -255,20 +226,6 @@ void AudioContext::uninitialize() m_isInitialized = false; } -bool AudioContext::isInitialized() const -{ - return m_isInitialized; -} - -bool AudioContext::isRunnable() const -{ - if (!isInitialized()) - return false; - - // Check with the HRTF spatialization system to see if it's finished loading. - return m_hrtfDatabaseLoader->isLoaded(); -} - void AudioContext::stopDispatch(void* userData) { AudioContext* context = reinterpret_cast<AudioContext*>(userData); @@ -294,55 +251,15 @@ void AudioContext::stop() callOnMainThread(stopDispatch, this); } -PassRefPtr<AudioBuffer> AudioContext::createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState) +bool AudioContext::hasPendingActivity() const { - RefPtr<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 0; - } - - return audioBuffer; + // According to spec AudioContext must die only after page navigates. + return !m_isCleared; } -PassRefPtr<AudioBuffer> AudioContext::createBuffer(ArrayBuffer* arrayBuffer, bool mixToMono, ExceptionState& exceptionState) +PassRefPtrWillBeRawPtr<AudioBuffer> AudioContext::createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState) { - ASSERT(arrayBuffer); - if (!arrayBuffer) { - exceptionState.throwDOMException( - SyntaxError, - "invalid ArrayBuffer."); - return 0; - } - - RefPtr<AudioBuffer> audioBuffer = AudioBuffer::createFromAudioFileData(arrayBuffer->data(), arrayBuffer->byteLength(), mixToMono, sampleRate()); - if (!audioBuffer.get()) { - exceptionState.throwDOMException( - SyntaxError, - "invalid audio data in ArrayBuffer."); - return 0; - } + RefPtrWillBeRawPtr<AudioBuffer> audioBuffer = AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate, exceptionState); return audioBuffer; } @@ -358,11 +275,10 @@ void AudioContext::decodeAudioData(ArrayBuffer* audioData, PassOwnPtr<AudioBuffe m_audioDecoder.decodeAsync(audioData, sampleRate(), successCallback, errorCallback); } -PassRefPtr<AudioBufferSourceNode> AudioContext::createBufferSource() +PassRefPtrWillBeRawPtr<AudioBufferSourceNode> AudioContext::createBufferSource() { ASSERT(isMainThread()); - lazyInitialize(); - 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(). @@ -371,27 +287,25 @@ PassRefPtr<AudioBufferSourceNode> AudioContext::createBufferSource() 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 0; + return nullptr; } - ASSERT(isMainThread()); - lazyInitialize(); - // First check if this media element already has a source node. if (mediaElement->audioSourceNode()) { exceptionState.throwDOMException( InvalidStateError, "invalid HTMLMediaElement."); - return 0; + return nullptr; } - RefPtr<MediaElementAudioSourceNode> node = MediaElementAudioSourceNode::create(this, mediaElement); + RefPtrWillBeRawPtr<MediaElementAudioSourceNode> node = MediaElementAudioSourceNode::create(this, mediaElement); mediaElement->setAudioSourceNode(node.get()); @@ -399,33 +313,28 @@ PassRefPtr<MediaElementAudioSourceNode> AudioContext::createMediaElementSource(H 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 0; + return nullptr; } - ASSERT(isMainThread()); - lazyInitialize(); - - AudioSourceProvider* provider = 0; - MediaStreamTrackVector audioTracks = mediaStream->getAudioTracks(); - RefPtr<MediaStreamTrack> audioTrack; - - // FIXME: get a provider for non-local MediaStreams (like from a remote peer). - for (size_t i = 0; i < audioTracks.size(); ++i) { - audioTrack = audioTracks[i]; - if (audioTrack->component()->audioSourceProvider()) { - provider = audioTrack->component()->audioSourceProvider(); - break; - } + if (audioTracks.isEmpty()) { + exceptionState.throwDOMException( + InvalidStateError, + "MediaStream has no audio track"); + return nullptr; } - RefPtr<MediaStreamAudioSourceNode> node = MediaStreamAudioSourceNode::create(this, mediaStream, audioTrack.get(), provider); + // Use the first audio track in the media stream. + RefPtrWillBeRawPtr<MediaStreamTrack> audioTrack = audioTracks[0]; + OwnPtr<AudioSourceProvider> provider = audioTrack->createWebAudioSource(); + 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()); @@ -434,36 +343,34 @@ PassRefPtr<MediaStreamAudioSourceNode> AudioContext::createMediaStreamSource(Med return node; } -PassRefPtr<MediaStreamAudioDestinationNode> AudioContext::createMediaStreamDestination() +PassRefPtrWillBeRawPtr<MediaStreamAudioDestinationNode> AudioContext::createMediaStreamDestination() { - // FIXME: Add support for an optional argument which specifies the number of channels. - // FIXME: The default should probably be stereo instead of mono. - return MediaStreamAudioDestinationNode::create(this, 1); + // 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()); - lazyInitialize(); - 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) { @@ -488,90 +395,81 @@ PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t buffe "buffer size (" + String::number(bufferSize) + ") must be a power of two between 256 and 16384."); } - return 0; + 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()); - lazyInitialize(); return BiquadFilterNode::create(this, m_destinationNode->sampleRate()); } -PassRefPtr<WaveShaperNode> AudioContext::createWaveShaper() +PassRefPtrWillBeRawPtr<WaveShaperNode> AudioContext::createWaveShaper() { ASSERT(isMainThread()); - lazyInitialize(); return WaveShaperNode::create(this); } -PassRefPtr<PannerNode> AudioContext::createPanner() +PassRefPtrWillBeRawPtr<PannerNode> AudioContext::createPanner() { ASSERT(isMainThread()); - lazyInitialize(); return PannerNode::create(this, m_destinationNode->sampleRate()); } -PassRefPtr<ConvolverNode> AudioContext::createConvolver() +PassRefPtrWillBeRawPtr<ConvolverNode> AudioContext::createConvolver() { ASSERT(isMainThread()); - lazyInitialize(); return ConvolverNode::create(this, m_destinationNode->sampleRate()); } -PassRefPtr<DynamicsCompressorNode> AudioContext::createDynamicsCompressor() +PassRefPtrWillBeRawPtr<DynamicsCompressorNode> AudioContext::createDynamicsCompressor() { ASSERT(isMainThread()); - lazyInitialize(); return DynamicsCompressorNode::create(this, m_destinationNode->sampleRate()); } -PassRefPtr<AnalyserNode> AudioContext::createAnalyser() +PassRefPtrWillBeRawPtr<AnalyserNode> AudioContext::createAnalyser() { ASSERT(isMainThread()); - lazyInitialize(); return AnalyserNode::create(this, m_destinationNode->sampleRate()); } -PassRefPtr<GainNode> AudioContext::createGain() +PassRefPtrWillBeRawPtr<GainNode> AudioContext::createGain() { ASSERT(isMainThread()); - lazyInitialize(); 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()); - lazyInitialize(); - 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 0; + 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()); - lazyInitialize(); - 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( @@ -579,24 +477,23 @@ PassRefPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(size_t numbe "number of outputs (" + String::number(numberOfOutputs) + ") must be between 1 and " + String::number(AudioContext::maxNumberOfChannels()) + "."); - return 0; + 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()); - lazyInitialize(); - 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( @@ -604,18 +501,17 @@ PassRefPtr<ChannelMergerNode> AudioContext::createChannelMerger(size_t numberOfI "number of inputs (" + String::number(numberOfInputs) + ") must be between 1 and " + String::number(AudioContext::maxNumberOfChannels()) + "."); - return 0; + return nullptr; } return node; } -PassRefPtr<OscillatorNode> AudioContext::createOscillator() +PassRefPtrWillBeRawPtr<OscillatorNode> AudioContext::createOscillator() { ASSERT(isMainThread()); - lazyInitialize(); - 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(). @@ -624,7 +520,7 @@ PassRefPtr<OscillatorNode> AudioContext::createOscillator() return node; } -PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Float32Array* imag, ExceptionState& exceptionState) +PassRefPtrWillBeRawPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Float32Array* imag, ExceptionState& exceptionState) { ASSERT(isMainThread()); @@ -632,14 +528,14 @@ PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Fl exceptionState.throwDOMException( SyntaxError, "invalid real array"); - return 0; + return nullptr; } if (!imag) { exceptionState.throwDOMException( SyntaxError, "invalid imaginary array"); - return 0; + return nullptr; } if (real->length() != imag->length()) { @@ -648,7 +544,7 @@ PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Fl "length of real array (" + String::number(real->length()) + ") and length of imaginary array (" + String::number(imag->length()) + ") must match."); - return 0; + return nullptr; } if (real->length() > 4096) { @@ -656,7 +552,7 @@ PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Fl IndexSizeError, "length of real array (" + String::number(real->length()) + ") exceeds allowed maximum of 4096"); - return 0; + return nullptr; } if (imag->length() > 4096) { @@ -664,10 +560,9 @@ PassRefPtr<PeriodicWave> AudioContext::createPeriodicWave(Float32Array* real, Fl IndexSizeError, "length of imaginary array (" + String::number(imag->length()) + ") exceeds allowed maximum of 4096"); - return 0; + return nullptr; } - lazyInitialize(); return PeriodicWave::create(sampleRate(), real, imag); } @@ -680,7 +575,7 @@ void AudioContext::notifyNodeFinishedProcessing(AudioNode* node) void AudioContext::derefFinishedSourceNodes() { ASSERT(isGraphOwner()); - ASSERT(isAudioThread() || isAudioThreadFinished()); + ASSERT(isAudioThread()); for (unsigned i = 0; i < m_finishedNodes.size(); i++) derefNode(m_finishedNodes[i]); @@ -712,7 +607,7 @@ void AudioContext::derefNode(AudioNode* node) void AudioContext::derefUnfinishedSourceNodes() { - ASSERT(isMainThread() && isAudioThreadFinished()); + ASSERT(isMainThread()); for (unsigned i = 0; i < m_referencedNodes.size(); ++i) m_referencedNodes[i]->deref(AudioNode::RefTypeConnection); @@ -743,7 +638,7 @@ bool AudioContext::tryLock(bool& mustReleaseLock) bool isAudioThread = thisThread == audioThread(); // Try to catch cases of using try lock on main thread - it should use regular lock. - ASSERT(isAudioThread || isAudioThreadFinished()); + ASSERT(isAudioThread); if (!isAudioThread) { // In release build treat tryLock() as lock() (since above ASSERT(isAudioThread) never fires) - this is the best we can do. @@ -858,7 +753,7 @@ void AudioContext::markForDeletion(AudioNode* node) { ASSERT(isGraphOwner()); - if (isAudioThreadFinished()) + if (!isInitialized()) m_nodesToDelete.append(node); else m_nodesMarkedForDeletion.append(node); @@ -872,14 +767,14 @@ void AudioContext::markForDeletion(AudioNode* node) void AudioContext::scheduleNodeDeletion() { - bool isGood = m_isInitialized && isGraphOwner(); + bool isGood = isInitialized() && isGraphOwner(); ASSERT(isGood); if (!isGood) return; // Make sure to call deleteMarkedNodes() on main thread. if (m_nodesMarkedForDeletion.size() && !m_isDeletionScheduled) { - m_nodesToDelete.append(m_nodesMarkedForDeletion); + m_nodesToDelete.appendVector(m_nodesMarkedForDeletion); m_nodesMarkedForDeletion.clear(); m_isDeletionScheduled = true; @@ -907,7 +802,7 @@ 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); @@ -924,9 +819,14 @@ void AudioContext::deleteMarkedNodes() 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; } @@ -955,7 +855,7 @@ 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) (*i)->updateRenderingState(); m_dirtySummingJunctions.clear(); @@ -1051,14 +951,13 @@ void AudioContext::fireCompletionEvent() } } -void AudioContext::incrementActiveSourceCount() -{ - atomicIncrement(&m_activeSourceCount); -} - -void AudioContext::decrementActiveSourceCount() +void AudioContext::trace(Visitor* visitor) { - atomicDecrement(&m_activeSourceCount); + visitor->trace(m_renderTarget); + visitor->trace(m_destinationNode); + visitor->trace(m_listener); + visitor->trace(m_dirtySummingJunctions); + EventTargetWithInlineData::trace(visitor); } } // namespace WebCore |