AudioContext.cpp

root/AudioContext.cpp

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DEFINITIONS

This source file includes following definitions.
isSampleRateRangeGood
create
m_activeSourceCount
m_activeSourceCount
constructCommon
lazyInitialize
clear
uninitialize
isInitialized
isRunnable
stopDispatch
stop
createBuffer
createBuffer
decodeAudioData
createBufferSource
createMediaElementSource
createMediaStreamSource
createMediaStreamDestination
createScriptProcessor
createScriptProcessor
createScriptProcessor
createBiquadFilter
createWaveShaper
createPanner
createConvolver
createDynamicsCompressor
createAnalyser
createGain
createDelay
createDelay
createChannelSplitter
createChannelSplitter
createChannelMerger
createChannelMerger
createOscillator
createPeriodicWave
notifyNodeFinishedProcessing
derefFinishedSourceNodes
refNode
derefNode
derefUnfinishedSourceNodes
lock
tryLock
unlock
isAudioThread
isGraphOwner
addDeferredFinishDeref
handlePreRenderTasks
handlePostRenderTasks
handleDeferredFinishDerefs
markForDeletion
scheduleNodeDeletion
deleteMarkedNodesDispatch
deleteMarkedNodes
markSummingJunctionDirty
removeMarkedSummingJunction
markAudioNodeOutputDirty
handleDirtyAudioSummingJunctions
handleDirtyAudioNodeOutputs
addAutomaticPullNode
removeAutomaticPullNode
updateAutomaticPullNodes
processAutomaticPullNodes
interfaceName
scriptExecutionContext
startRendering
fireCompletionEvent
incrementActiveSourceCount
decrementActiveSourceCount
/*
 * 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 "config.h"

#if ENABLE(WEB_AUDIO)

#include "modules/webaudio/AudioContext.h"

#include "bindings/v8/ExceptionState.h"
#include "core/dom/Document.h"
#include "core/dom/ExceptionCode.h"
#include "core/html/HTMLMediaElement.h"
#include "core/inspector/ScriptCallStack.h"
#include "core/platform/audio/FFTFrame.h"
#include "core/platform/audio/HRTFDatabaseLoader.h"
#include "core/platform/audio/HRTFPanner.h"
#include "modules/mediastream/MediaStream.h"
#include "modules/webaudio/AnalyserNode.h"
#include "modules/webaudio/AsyncAudioDecoder.h"
#include "modules/webaudio/AudioBuffer.h"
#include "modules/webaudio/AudioBufferCallback.h"
#include "modules/webaudio/AudioBufferSourceNode.h"
#include "modules/webaudio/AudioListener.h"
#include "modules/webaudio/AudioNodeInput.h"
#include "modules/webaudio/AudioNodeOutput.h"
#include "modules/webaudio/BiquadFilterNode.h"
#include "modules/webaudio/ChannelMergerNode.h"
#include "modules/webaudio/ChannelSplitterNode.h"
#include "modules/webaudio/ConvolverNode.h"
#include "modules/webaudio/DefaultAudioDestinationNode.h"
#include "modules/webaudio/DelayNode.h"
#include "modules/webaudio/DynamicsCompressorNode.h"
#include "modules/webaudio/GainNode.h"
#include "modules/webaudio/MediaElementAudioSourceNode.h"
#include "modules/webaudio/MediaStreamAudioDestinationNode.h"
#include "modules/webaudio/MediaStreamAudioSourceNode.h"
#include "modules/webaudio/OfflineAudioCompletionEvent.h"
#include "modules/webaudio/OfflineAudioDestinationNode.h"
#include "modules/webaudio/OscillatorNode.h"
#include "modules/webaudio/PannerNode.h"
#include "modules/webaudio/PeriodicWave.h"
#include "modules/webaudio/ScriptProcessorNode.h"
#include "modules/webaudio/WaveShaperNode.h"

#if DEBUG_AUDIONODE_REFERENCES
#include <stdio.h>
#endif

#include "wtf/ArrayBuffer.h"
#include "wtf/Atomics.h"
#include "wtf/MainThread.h"
#include "wtf/OwnPtr.h"
#include "wtf/PassOwnPtr.h"
#include "wtf/RefCounted.h"
#include "wtf/text/WTFString.h"

// FIXME: check the proper way to reference an undefined thread ID
const int UndefinedThreadIdentifier = 0xffffffff;

const unsigned MaxNodesToDeletePerQuantum = 10;

namespace WebCore {

bool AudioContext::isSampleRateRangeGood(float sampleRate)
{
    // 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 = 4;
unsigned AudioContext::s_hardwareContextCount = 0;

PassRefPtr<AudioContext> AudioContext::create(Document* document)
{
    ASSERT(document);
    ASSERT(isMainThread());
    if (s_hardwareContextCount >= MaxHardwareContexts)
        return 0;

    RefPtr<AudioContext> audioContext(adoptRef(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_isInitialized(false)
    , m_isAudioThreadFinished(false)
    , m_destinationNode(0)
    , m_isDeletionScheduled(false)
    , m_automaticPullNodesNeedUpdating(false)
    , m_connectionCount(0)
    , m_audioThread(0)
    , m_graphOwnerThread(UndefinedThreadIdentifier)
    , m_isOfflineContext(false)
    , m_activeSourceCount(0)
{
    constructCommon();

    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());
}

// Constructor for offline (non-realtime) rendering.
AudioContext::AudioContext(Document* document, unsigned numberOfChannels, size_t numberOfFrames, float sampleRate)
    : ActiveDOMObject(document)
    , m_isStopScheduled(false)
    , m_isInitialized(false)
    , m_isAudioThreadFinished(false)
    , m_destinationNode(0)
    , m_automaticPullNodesNeedUpdating(false)
    , m_connectionCount(0)
    , m_audioThread(0)
    , m_graphOwnerThread(UndefinedThreadIdentifier)
    , m_isOfflineContext(true)
    , m_activeSourceCount(0)
{
    constructCommon();

    // FIXME: the passed in sampleRate MUST match the hardware sample-rate since HRTFDatabaseLoader is a singleton.
    m_hrtfDatabaseLoader = HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(sampleRate);

    // Create a new destination for offline rendering.
    m_renderTarget = AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate);
    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();
}

AudioContext::~AudioContext()
{
#if DEBUG_AUDIONODE_REFERENCES
    fprintf(stderr, "%p: AudioContext::~AudioContext()\n", this);
#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());
    ASSERT(!m_automaticPullNodes.size());
    if (m_automaticPullNodesNeedUpdating)
        m_renderingAutomaticPullNodes.resize(m_automaticPullNodes.size());
    ASSERT(!m_renderingAutomaticPullNodes.size());
}

void AudioContext::lazyInitialize()
{
    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 (!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;
        }
    }
}

void AudioContext::clear()
{
    // We have to release our reference to the destination node before the context will ever be deleted since the destination node holds a reference to the context.
    if (m_destinationNode)
        m_destinationNode.clear();

    // Audio thread is dead. Nobody will schedule node deletion action. Let's do it ourselves.
    do {
        deleteMarkedNodes();
        m_nodesToDelete.append(m_nodesMarkedForDeletion);
        m_nodesMarkedForDeletion.clear();
    } while (m_nodesToDelete.size());

    // It was set in constructCommon.
    unsetPendingActivity(this);
}

void AudioContext::uninitialize()
{
    ASSERT(isMainThread());

    if (!m_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;
    }

    // Get rid of the sources which may still be playing.
    derefUnfinishedSourceNodes();

    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);
    ASSERT(context);
    if (!context)
        return;

    context->uninitialize();
    context->clear();
}

void AudioContext::stop()
{
    // Usually ScriptExecutionContext calls stop twice.
    if (m_isStopScheduled)
        return;
    m_isStopScheduled = true;

    // Don't call uninitialize() immediately here because the ScriptExecutionContext is in the middle
    // of dealing with all of its ActiveDOMObjects at this point. uninitialize() can de-reference other
    // ActiveDOMObjects so let's schedule uninitialize() to be called later.
    // FIXME: see if there's a more direct way to handle this issue.
    callOnMainThread(stopDispatch, this);
}

PassRefPtr<AudioBuffer> AudioContext::createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& es)
{
    RefPtr<AudioBuffer> audioBuffer = AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate);
    if (!audioBuffer.get()) {
        es.throwDOMException(SyntaxError);
        return 0;
    }

    return audioBuffer;
}

PassRefPtr<AudioBuffer> AudioContext::createBuffer(ArrayBuffer* arrayBuffer, bool mixToMono, ExceptionState& es)
{
    ASSERT(arrayBuffer);
    if (!arrayBuffer) {
        es.throwDOMException(SyntaxError);
        return 0;
    }

    RefPtr<AudioBuffer> audioBuffer = AudioBuffer::createFromAudioFileData(arrayBuffer->data(), arrayBuffer->byteLength(), mixToMono, sampleRate());
    if (!audioBuffer.get()) {
        es.throwDOMException(SyntaxError);
        return 0;
    }

    return audioBuffer;
}

void AudioContext::decodeAudioData(ArrayBuffer* audioData, PassRefPtr<AudioBufferCallback> successCallback, PassRefPtr<AudioBufferCallback> errorCallback, ExceptionState& es)
{
    if (!audioData) {
        es.throwDOMException(SyntaxError);
        return;
    }
    m_audioDecoder.decodeAsync(audioData, sampleRate(), successCallback, errorCallback);
}

PassRefPtr<AudioBufferSourceNode> AudioContext::createBufferSource()
{
    ASSERT(isMainThread());
    lazyInitialize();
    RefPtr<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& es)
{
    ASSERT(mediaElement);
    if (!mediaElement) {
        es.throwDOMException(InvalidStateError);
        return 0;
    }

    ASSERT(isMainThread());
    lazyInitialize();

    // First check if this media element already has a source node.
    if (mediaElement->audioSourceNode()) {
        es.throwDOMException(InvalidStateError);
        return 0;
    }

    RefPtr<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& es)
{
    ASSERT(mediaStream);
    if (!mediaStream) {
        es.throwDOMException(InvalidStateError);
        return 0;
    }

    ASSERT(isMainThread());
    lazyInitialize();

    AudioSourceProvider* provider = 0;

    MediaStreamTrackVector audioTracks = mediaStream->getAudioTracks();

    // FIXME: get a provider for non-local MediaStreams (like from a remote peer).
    for (size_t i = 0; i < audioTracks.size(); ++i) {
        RefPtr<MediaStreamTrack> localAudio = audioTracks[i];
        MediaStreamSource* source = localAudio->component()->source();
        if (!source->deviceId().isEmpty()) {
            destination()->enableInput(source->deviceId());
            provider = destination()->localAudioInputProvider();
            break;
        }
    }

    RefPtr<MediaStreamAudioSourceNode> node = MediaStreamAudioSourceNode::create(this, mediaStream, provider);

    // 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()
{
    // 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);
}

PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, ExceptionState& es)
{
    // Set number of input/output channels to stereo by default.
    return createScriptProcessor(bufferSize, 2, 2, es);
}

PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, ExceptionState& es)
{
    // Set number of output channels to stereo by default.
    return createScriptProcessor(bufferSize, numberOfInputChannels, 2, es);
}

PassRefPtr<ScriptProcessorNode> AudioContext::createScriptProcessor(size_t bufferSize, size_t numberOfInputChannels, size_t numberOfOutputChannels, ExceptionState& es)
{
    ASSERT(isMainThread());
    lazyInitialize();
    RefPtr<ScriptProcessorNode> node = ScriptProcessorNode::create(this, m_destinationNode->sampleRate(), bufferSize, numberOfInputChannels, numberOfOutputChannels);

    if (!node.get()) {
        es.throwDOMException(SyntaxError);
        return 0;
    }

    refNode(node.get()); // context keeps reference until we stop making javascript rendering callbacks
    return node;
}

PassRefPtr<BiquadFilterNode> AudioContext::createBiquadFilter()
{
    ASSERT(isMainThread());
    lazyInitialize();
    return BiquadFilterNode::create(this, m_destinationNode->sampleRate());
}

PassRefPtr<WaveShaperNode> AudioContext::createWaveShaper()
{
    ASSERT(isMainThread());
    lazyInitialize();
    return WaveShaperNode::create(this);
}

PassRefPtr<PannerNode> AudioContext::createPanner()
{
    ASSERT(isMainThread());
    lazyInitialize();
    return PannerNode::create(this, m_destinationNode->sampleRate());
}

PassRefPtr<ConvolverNode> AudioContext::createConvolver()
{
    ASSERT(isMainThread());
    lazyInitialize();
    return ConvolverNode::create(this, m_destinationNode->sampleRate());
}

PassRefPtr<DynamicsCompressorNode> AudioContext::createDynamicsCompressor()
{
    ASSERT(isMainThread());
    lazyInitialize();
    return DynamicsCompressorNode::create(this, m_destinationNode->sampleRate());
}

PassRefPtr<AnalyserNode> AudioContext::createAnalyser()
{
    ASSERT(isMainThread());
    lazyInitialize();
    return AnalyserNode::create(this, m_destinationNode->sampleRate());
}

PassRefPtr<GainNode> AudioContext::createGain()
{
    ASSERT(isMainThread());
    lazyInitialize();
    return GainNode::create(this, m_destinationNode->sampleRate());
}

PassRefPtr<DelayNode> AudioContext::createDelay(ExceptionState& es)
{
    const double defaultMaxDelayTime = 1;
    return createDelay(defaultMaxDelayTime, es);
}

PassRefPtr<DelayNode> AudioContext::createDelay(double maxDelayTime, ExceptionState& es)
{
    ASSERT(isMainThread());
    lazyInitialize();
    RefPtr<DelayNode> node = DelayNode::create(this, m_destinationNode->sampleRate(), maxDelayTime, es);
    if (es.hadException())
        return 0;
    return node;
}

PassRefPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(ExceptionState& es)
{
    const unsigned ChannelSplitterDefaultNumberOfOutputs = 6;
    return createChannelSplitter(ChannelSplitterDefaultNumberOfOutputs, es);
}

PassRefPtr<ChannelSplitterNode> AudioContext::createChannelSplitter(size_t numberOfOutputs, ExceptionState& es)
{
    ASSERT(isMainThread());
    lazyInitialize();

    RefPtr<ChannelSplitterNode> node = ChannelSplitterNode::create(this, m_destinationNode->sampleRate(), numberOfOutputs);

    if (!node.get()) {
        es.throwDOMException(SyntaxError);
        return 0;
    }

    return node;
}

PassRefPtr<ChannelMergerNode> AudioContext::createChannelMerger(ExceptionState& es)
{
    const unsigned ChannelMergerDefaultNumberOfInputs = 6;
    return createChannelMerger(ChannelMergerDefaultNumberOfInputs, es);
}

PassRefPtr<ChannelMergerNode> AudioContext::createChannelMerger(size_t numberOfInputs, ExceptionState& es)
{
    ASSERT(isMainThread());
    lazyInitialize();

    RefPtr<ChannelMergerNode> node = ChannelMergerNode::create(this, m_destinationNode->sampleRate(), numberOfInputs);

    if (!node.get()) {
        es.throwDOMException(SyntaxError);
        return 0;
    }

    return node;
}

PassRefPtr<OscillatorNode> AudioContext::createOscillator()
{
    ASSERT(isMainThread());
    lazyInitialize();

    RefPtr<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& es)
{
    ASSERT(isMainThread());

    if (!real || !imag || (real->length() != imag->length())) {
        es.throwDOMException(SyntaxError);
        return 0;
    }

    lazyInitialize();
    return PeriodicWave::create(sampleRate(), real, imag);
}

void AudioContext::notifyNodeFinishedProcessing(AudioNode* node)
{
    ASSERT(isAudioThread());
    m_finishedNodes.append(node);
}

void AudioContext::derefFinishedSourceNodes()
{
    ASSERT(isGraphOwner());
    ASSERT(isAudioThread() || isAudioThreadFinished());
    for (unsigned i = 0; i < m_finishedNodes.size(); i++)
        derefNode(m_finishedNodes[i]);

    m_finishedNodes.clear();
}

void AudioContext::refNode(AudioNode* node)
{
    ASSERT(isMainThread());
    AutoLocker locker(this);

    node->ref(AudioNode::RefTypeConnection);
    m_referencedNodes.append(node);
}

void AudioContext::derefNode(AudioNode* node)
{
    ASSERT(isGraphOwner());

    node->deref(AudioNode::RefTypeConnection);

    for (unsigned i = 0; i < m_referencedNodes.size(); ++i) {
        if (node == m_referencedNodes[i]) {
            m_referencedNodes.remove(i);
            break;
        }
    }
}

void AudioContext::derefUnfinishedSourceNodes()
{
    ASSERT(isMainThread() && isAudioThreadFinished());
    for (unsigned i = 0; i < m_referencedNodes.size(); ++i)
        m_referencedNodes[i]->deref(AudioNode::RefTypeConnection);

    m_referencedNodes.clear();
}

void AudioContext::lock(bool& mustReleaseLock)
{
    // Don't allow regular lock in real-time audio thread.
    ASSERT(isMainThread());

    ThreadIdentifier thisThread = currentThread();

    if (thisThread == m_graphOwnerThread) {
        // We already have the lock.
        mustReleaseLock = false;
    } else {
        // Acquire the lock.
        m_contextGraphMutex.lock();
        m_graphOwnerThread = thisThread;
        mustReleaseLock = true;
    }
}

bool AudioContext::tryLock(bool& mustReleaseLock)
{
    ThreadIdentifier thisThread = currentThread();
    bool isAudioThread = thisThread == audioThread();

    // Try to catch cases of using try lock on main thread - it should use regular lock.
    ASSERT(isAudioThread || isAudioThreadFinished());

    if (!isAudioThread) {
        // In release build treat tryLock() as lock() (since above ASSERT(isAudioThread) never fires) - this is the best we can do.
        lock(mustReleaseLock);
        return true;
    }

    bool hasLock;

    if (thisThread == m_graphOwnerThread) {
        // Thread already has the lock.
        hasLock = true;
        mustReleaseLock = false;
    } else {
        // Don't already have the lock - try to acquire it.
        hasLock = m_contextGraphMutex.tryLock();

        if (hasLock)
            m_graphOwnerThread = thisThread;

        mustReleaseLock = hasLock;
    }

    return hasLock;
}

void AudioContext::unlock()
{
    ASSERT(currentThread() == m_graphOwnerThread);

    m_graphOwnerThread = UndefinedThreadIdentifier;
    m_contextGraphMutex.unlock();
}

bool AudioContext::isAudioThread() const
{
    return currentThread() == m_audioThread;
}

bool AudioContext::isGraphOwner() const
{
    return currentThread() == m_graphOwnerThread;
}

void AudioContext::addDeferredFinishDeref(AudioNode* node)
{
    ASSERT(isAudioThread());
    m_deferredFinishDerefList.append(node);
}

void AudioContext::handlePreRenderTasks()
{
    ASSERT(isAudioThread());

    // At the beginning of every render quantum, try to update the internal rendering graph state (from main thread changes).
    // It's OK if the tryLock() fails, we'll just take slightly longer to pick up the changes.
    bool mustReleaseLock;
    if (tryLock(mustReleaseLock)) {
        // Fixup the state of any dirty AudioSummingJunctions and AudioNodeOutputs.
        handleDirtyAudioSummingJunctions();
        handleDirtyAudioNodeOutputs();

        updateAutomaticPullNodes();

        if (mustReleaseLock)
            unlock();
    }
}

void AudioContext::handlePostRenderTasks()
{
    ASSERT(isAudioThread());

    // Must use a tryLock() here too.  Don't worry, the lock will very rarely be contended and this method is called frequently.
    // The worst that can happen is that there will be some nodes which will take slightly longer than usual to be deleted or removed
    // from the render graph (in which case they'll render silence).
    bool mustReleaseLock;
    if (tryLock(mustReleaseLock)) {
        // Take care of finishing any derefs where the tryLock() failed previously.
        handleDeferredFinishDerefs();

        // Dynamically clean up nodes which are no longer needed.
        derefFinishedSourceNodes();

        // Don't delete in the real-time thread. Let the main thread do it.
        // Ref-counted objects held by certain AudioNodes may not be thread-safe.
        scheduleNodeDeletion();

        // Fixup the state of any dirty AudioSummingJunctions and AudioNodeOutputs.
        handleDirtyAudioSummingJunctions();
        handleDirtyAudioNodeOutputs();

        updateAutomaticPullNodes();

        if (mustReleaseLock)
            unlock();
    }
}

void AudioContext::handleDeferredFinishDerefs()
{
    ASSERT(isAudioThread() && isGraphOwner());
    for (unsigned i = 0; i < m_deferredFinishDerefList.size(); ++i) {
        AudioNode* node = m_deferredFinishDerefList[i];
        node->finishDeref(AudioNode::RefTypeConnection);
    }

    m_deferredFinishDerefList.clear();
}

void AudioContext::markForDeletion(AudioNode* node)
{
    ASSERT(isGraphOwner());

    if (isAudioThreadFinished())
        m_nodesToDelete.append(node);
    else
        m_nodesMarkedForDeletion.append(node);

    // This is probably the best time for us to remove the node from automatic pull list,
    // since all connections are gone and we hold the graph lock. Then when handlePostRenderTasks()
    // gets a chance to schedule the deletion work, updateAutomaticPullNodes() also gets a chance to
    // modify m_renderingAutomaticPullNodes.
    removeAutomaticPullNode(node);
}

void AudioContext::scheduleNodeDeletion()
{
    bool isGood = m_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_nodesMarkedForDeletion.clear();

        m_isDeletionScheduled = true;

        // Don't let ourself get deleted before the callback.
        // See matching deref() in deleteMarkedNodesDispatch().
        ref();
        callOnMainThread(deleteMarkedNodesDispatch, this);
    }
}

void AudioContext::deleteMarkedNodesDispatch(void* userData)
{
    AudioContext* context = reinterpret_cast<AudioContext*>(userData);
    ASSERT(context);
    if (!context)
        return;

    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);
    {
        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));

            // Finally, delete it.
            delete node;
        }
        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)
        (*i)->updateRenderingState();

    m_dirtySummingJunctions.clear();
}

void AudioContext::handleDirtyAudioNodeOutputs()
{
    ASSERT(isGraphOwner());

    for (HashSet<AudioNodeOutput*>::iterator i = m_dirtyAudioNodeOutputs.begin(); i != m_dirtyAudioNodeOutputs.end(); ++i)
        (*i)->updateRenderingState();

    m_dirtyAudioNodeOutputs.clear();
}

void AudioContext::addAutomaticPullNode(AudioNode* node)
{
    ASSERT(isGraphOwner());

    if (!m_automaticPullNodes.contains(node)) {
        m_automaticPullNodes.add(node);
        m_automaticPullNodesNeedUpdating = true;
    }
}

void AudioContext::removeAutomaticPullNode(AudioNode* node)
{
    ASSERT(isGraphOwner());

    if (m_automaticPullNodes.contains(node)) {
        m_automaticPullNodes.remove(node);
        m_automaticPullNodesNeedUpdating = true;
    }
}

void AudioContext::updateAutomaticPullNodes()
{
    ASSERT(isGraphOwner());

    if (m_automaticPullNodesNeedUpdating) {
        // Copy from m_automaticPullNodes to m_renderingAutomaticPullNodes.
        m_renderingAutomaticPullNodes.resize(m_automaticPullNodes.size());

        unsigned j = 0;
        for (HashSet<AudioNode*>::iterator i = m_automaticPullNodes.begin(); i != m_automaticPullNodes.end(); ++i, ++j) {
            AudioNode* output = *i;
            m_renderingAutomaticPullNodes[j] = output;
        }

        m_automaticPullNodesNeedUpdating = false;
    }
}

void AudioContext::processAutomaticPullNodes(size_t framesToProcess)
{
    ASSERT(isAudioThread());

    for (unsigned i = 0; i < m_renderingAutomaticPullNodes.size(); ++i)
        m_renderingAutomaticPullNodes[i]->processIfNecessary(framesToProcess);
}

const AtomicString& AudioContext::interfaceName() const
{
    return eventNames().interfaceForAudioContext;
}

ScriptExecutionContext* AudioContext::scriptExecutionContext() const
{
    return m_isStopScheduled ? 0 : ActiveDOMObject::scriptExecutionContext();
}

void AudioContext::startRendering()
{
    destination()->startRendering();
}

void AudioContext::fireCompletionEvent()
{
    ASSERT(isMainThread());
    if (!isMainThread())
        return;

    AudioBuffer* renderedBuffer = m_renderTarget.get();

    ASSERT(renderedBuffer);
    if (!renderedBuffer)
        return;

    // Avoid firing the event if the document has already gone away.
    if (scriptExecutionContext()) {
        // Call the offline rendering completion event listener.
        dispatchEvent(OfflineAudioCompletionEvent::create(renderedBuffer));
    }
}

void AudioContext::incrementActiveSourceCount()
{
    atomicIncrement(&m_activeSourceCount);
}

void AudioContext::decrementActiveSourceCount()
{
    atomicDecrement(&m_activeSourceCount);
}

} // namespace WebCore

#endif // ENABLE(WEB_AUDIO)
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