Merge WebKit at r75315: Initial merge by git.

Change-Id: I570314b346ce101c935ed22a626b48c2af266b84

74 files changed, 0 insertions, 7568 deletions

diff --git a/WebCore/webaudio/AudioBasicProcessorNode.cpp b/WebCore/webaudio/AudioBasicProcessorNode.cpp
deleted file mode 100644
index 828062e..0000000
--- a/WebCore/webaudio/AudioBasicProcessorNode.cpp
+++ /dev/null
@@ -1,149 +0,0 @@
-/*
- * 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 "AudioBasicProcessorNode.h"
-
-#include "AudioBus.h"
-#include "AudioContext.h"
-#include "AudioNodeInput.h"
-#include "AudioNodeOutput.h"
-#include "AudioProcessor.h"
-
-namespace WebCore {
-
-AudioBasicProcessorNode::AudioBasicProcessorNode(AudioContext* context, double sampleRate)
-    : AudioNode(context, sampleRate)
-{
-    addInput(adoptPtr(new AudioNodeInput(this)));
-    addOutput(adoptPtr(new AudioNodeOutput(this, 0)));
-
-    // The subclass must create m_processor.
-}
-
-void AudioBasicProcessorNode::initialize()
-{
-    if (isInitialized())
-        return;
-
-    ASSERT(processor());
-    processor()->initialize();
-
-    AudioNode::initialize();
-}
-
-void AudioBasicProcessorNode::uninitialize()
-{
-    if (!isInitialized())
-        return;
-
-    ASSERT(processor());
-    processor()->uninitialize();
-
-    AudioNode::uninitialize();
-}
-
-void AudioBasicProcessorNode::process(size_t framesToProcess)
-{
-    AudioBus* destinationBus = output(0)->bus();
-    
-    // The realtime thread can't block on this lock, so we call tryLock() instead.
-    if (m_processLock.tryLock()) {
-        if (!isInitialized() || !processor())
-            destinationBus->zero();
-        else {
-            AudioBus* sourceBus = input(0)->bus();
-
-            // FIXME: if we take "tail time" into account, then we can avoid calling processor()->process() once the tail dies down.
-            if (!input(0)->isConnected())
-                sourceBus->zero();
-            
-            processor()->process(sourceBus, destinationBus, framesToProcess);  
-        }
-
-        m_processLock.unlock();
-    } else {
-        // Too bad - the tryLock() failed.  We must be in the middle of re-connecting and were already outputting silence anyway...
-        destinationBus->zero();
-    }
-}
-
-// Nice optimization in the very common case allowing for "in-place" processing
-void AudioBasicProcessorNode::pullInputs(size_t framesToProcess)
-{
-    // Render input stream - suggest to the input to render directly into output bus for in-place processing in process() if possible.
-    input(0)->pull(output(0)->bus(), framesToProcess);
-}
-
-void AudioBasicProcessorNode::reset()
-{
-    if (processor())
-        processor()->reset();
-}
-
-// As soon as we know the channel count of our input, we can lazily initialize.
-// Sometimes this may be called more than once with different channel counts, in which case we must safely
-// uninitialize and then re-initialize with the new channel count.
-void AudioBasicProcessorNode::checkNumberOfChannelsForInput(AudioNodeInput* input)
-{
-    ASSERT(context()->isAudioThread() && context()->isGraphOwner());
-    
-    ASSERT(input == this->input(0));
-    if (input != this->input(0))
-        return;
-
-    ASSERT(processor());
-    if (!processor())
-        return;
-
-    unsigned numberOfChannels = input->numberOfChannels();
-    
-    if (isInitialized() && numberOfChannels != output(0)->numberOfChannels()) {
-        // We're already initialized but the channel count has changed.
-        // We need to be careful since we may be actively processing right now, so synchronize with process().
-        MutexLocker locker(m_processLock);
-        uninitialize();
-    }
-    
-    if (!isInitialized()) {
-        // This will propagate the channel count to any nodes connected further down the chain...
-        output(0)->setNumberOfChannels(numberOfChannels);
-
-        // Re-initialize the processor with the new channel count.
-        processor()->setNumberOfChannels(numberOfChannels);
-        initialize();
-    }
-}
-
-unsigned AudioBasicProcessorNode::numberOfChannels()
-{
-    return output(0)->numberOfChannels();
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioBasicProcessorNode.h b/WebCore/webaudio/AudioBasicProcessorNode.h
deleted file mode 100644
index 38bfd3b..0000000
--- a/WebCore/webaudio/AudioBasicProcessorNode.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef AudioBasicProcessorNode_h
-#define AudioBasicProcessorNode_h
-
-#include "AudioNode.h"
-#include <wtf/PassRefPtr.h>
-#include <wtf/RefCounted.h>
-#include <wtf/Threading.h>
-
-namespace WebCore {
-
-class AudioBus;
-class AudioNodeInput;
-class AudioProcessor;
-    
-// AudioBasicProcessorNode is an AudioNode with one input and one output where the input and output have the same number of channels.
-class AudioBasicProcessorNode : public AudioNode {
-public:
-    AudioBasicProcessorNode(AudioContext*, double sampleRate);
-
-    // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void pullInputs(size_t framesToProcess);
-    virtual void reset();
-    virtual void initialize();
-    virtual void uninitialize();
-
-    // Called in the main thread when the number of channels for the input may have changed.
-    virtual void checkNumberOfChannelsForInput(AudioNodeInput*);
-
-    // Returns the number of channels for both the input and the output.
-    unsigned numberOfChannels();
-
-protected:
-    AudioProcessor* processor() { return m_processor.get(); }
-    OwnPtr<AudioProcessor> m_processor;
-
-private:
-    // This synchronizes live channel count changes which require an uninitialization / re-initialization.
-    mutable Mutex m_processLock;
-};
-
-} // namespace WebCore
-
-#endif // AudioBasicProcessorNode_h
diff --git a/WebCore/webaudio/AudioBuffer.cpp b/WebCore/webaudio/AudioBuffer.cpp
deleted file mode 100644
index f46d153..0000000
--- a/WebCore/webaudio/AudioBuffer.cpp
+++ /dev/null
@@ -1,110 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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) & ENABLE(3D_CANVAS)
-
-#include "AudioBuffer.h"
-
-#include "AudioBus.h"
-#include "AudioFileReader.h"
-#include "ExceptionCode.h"
-#include <wtf/OwnPtr.h>
-
-namespace WebCore {
-
-PassRefPtr<AudioBuffer> AudioBuffer::create(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate)
-{
-    return adoptRef(new AudioBuffer(numberOfChannels, numberOfFrames, sampleRate));
-}
-
-PassRefPtr<AudioBuffer> AudioBuffer::createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, double sampleRate)
-{
-    OwnPtr<AudioBus> bus = createBusFromInMemoryAudioFile(data, dataSize, mixToMono, sampleRate);
-    if (bus.get())
-        return adoptRef(new AudioBuffer(bus.get()));
-
-    return 0;
-}
-
-AudioBuffer::AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate)
-    : m_gain(1.0)
-    , m_sampleRate(sampleRate)
-    , m_length(numberOfFrames)
-{
-    m_channels.reserveCapacity(numberOfChannels);
-
-    for (unsigned i = 0; i < numberOfChannels; ++i) {
-        RefPtr<Float32Array> channelDataArray = Float32Array::create(m_length);
-        m_channels.append(channelDataArray);
-    }
-}
-
-AudioBuffer::AudioBuffer(AudioBus* bus)
-    : m_gain(1.0)
-    , m_sampleRate(bus->sampleRate())
-    , m_length(bus->length())
-{
-    // Copy audio data from the bus to the Float32Arrays we manage.
-    unsigned numberOfChannels = bus->numberOfChannels();
-    m_channels.reserveCapacity(numberOfChannels);
-    for (unsigned i = 0; i < numberOfChannels; ++i) {
-        RefPtr<Float32Array> channelDataArray = Float32Array::create(m_length);
-        ExceptionCode ec;
-        channelDataArray->setRange(bus->channel(i)->data(), m_length, 0, ec);
-        m_channels.append(channelDataArray);
-    }
-}
-
-void AudioBuffer::releaseMemory()
-{
-    m_channels.clear();
-}
-
-Float32Array* AudioBuffer::getChannelData(unsigned channelIndex)
-{
-    if (channelIndex >= m_channels.size())
-        return 0;
-
-    return m_channels[channelIndex].get();
-}
-
-void AudioBuffer::zero()
-{
-    for (unsigned i = 0; i < m_channels.size(); ++i) {
-        if (getChannelData(i)) {
-            ExceptionCode ec;
-            getChannelData(i)->zeroRange(0, length(), ec);
-        }
-    }
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO) & ENABLE(3D_CANVAS)
diff --git a/WebCore/webaudio/AudioBuffer.h b/WebCore/webaudio/AudioBuffer.h
deleted file mode 100644
index b11a20e..0000000
--- a/WebCore/webaudio/AudioBuffer.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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.
- */
-
-#ifndef AudioBuffer_h
-#define AudioBuffer_h
-
-#include "Float32Array.h"
-#include <wtf/PassRefPtr.h>
-#include <wtf/RefCounted.h>
-#include <wtf/RefPtr.h>
-#include <wtf/Vector.h>
-
-namespace WebCore {
-
-class AudioBus;
-    
-class AudioBuffer : public RefCounted<AudioBuffer> {
-public:   
-    static PassRefPtr<AudioBuffer> create(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate);
-
-    // Returns 0 if data is not a valid audio file.
-    static PassRefPtr<AudioBuffer> createFromAudioFileData(const void* data, size_t dataSize, bool mixToMono, double sampleRate);
-
-    // Format
-    size_t length() const { return m_length; }
-    double duration() const { return length() / sampleRate(); }
-    double sampleRate() const { return m_sampleRate; }
-
-    // Channel data access
-    unsigned numberOfChannels() const { return m_channels.size(); }
-    Float32Array* getChannelData(unsigned channelIndex);
-    void zero();
-
-    // Scalar gain
-    double gain() const { return m_gain; }
-    void setGain(double gain) { m_gain = gain; }
-
-    // Because an AudioBuffer has a JavaScript wrapper, which will be garbage collected, it may take awhile for this object to be deleted.
-    // releaseMemory() can be called when the AudioContext goes away, so we can release the memory earlier than when the garbage collection happens.
-    // Careful! Only call this when the page unloads, after the AudioContext is no longer processing.
-    void releaseMemory();
-    
-protected:
-    AudioBuffer(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate);
-    AudioBuffer(AudioBus* bus);
-
-    double m_gain; // scalar gain
-    double m_sampleRate;
-    size_t m_length;
-
-    Vector<RefPtr<Float32Array> > m_channels;
-};
-
-} // namespace WebCore
-
-#endif // AudioBuffer_h
diff --git a/WebCore/webaudio/AudioBuffer.idl b/WebCore/webaudio/AudioBuffer.idl
deleted file mode 100644
index e7353bf..0000000
--- a/WebCore/webaudio/AudioBuffer.idl
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO & 3D_CANVAS
-    ] AudioBuffer {
-        readonly attribute long length; // in sample-frames
-        readonly attribute float duration; // in seconds
-        readonly attribute float sampleRate; // in sample-frames per second
-
-        attribute float gain; // linear gain (default 1.0)
-
-        // Channel access
-        readonly attribute unsigned long numberOfChannels;
-        Float32Array getChannelData(in unsigned long channelIndex);
-    };
-}
diff --git a/WebCore/webaudio/AudioBufferSourceNode.cpp b/WebCore/webaudio/AudioBufferSourceNode.cpp
deleted file mode 100644
index 05abed8..0000000
--- a/WebCore/webaudio/AudioBufferSourceNode.cpp
+++ /dev/null
@@ -1,455 +0,0 @@
-/*
- * 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 "AudioBufferSourceNode.h"
-
-#include "AudioContext.h"
-#include "AudioNodeOutput.h"
-#include <algorithm>
-#include <wtf/MathExtras.h>
-
-using namespace std;
-
-namespace WebCore {
-
-const double DefaultGrainDuration = 0.020; // 20ms
-
-PassRefPtr<AudioBufferSourceNode> AudioBufferSourceNode::create(AudioContext* context, double sampleRate)
-{
-    return adoptRef(new AudioBufferSourceNode(context, sampleRate));
-}
-
-AudioBufferSourceNode::AudioBufferSourceNode(AudioContext* context, double sampleRate)
-    : AudioSourceNode(context, sampleRate)
-    , m_buffer(0)
-    , m_isPlaying(false)
-    , m_isLooping(false)
-    , m_hasFinished(false)
-    , m_startTime(0.0)
-    , m_schedulingFrameDelay(0)
-    , m_readIndex(0)
-    , m_isGrain(false)
-    , m_grainOffset(0.0)
-    , m_grainDuration(DefaultGrainDuration)
-    , m_grainFrameCount(0)
-    , m_lastGain(1.0)
-    , m_pannerNode(0)
-{
-    setType(NodeTypeAudioBufferSource);
-
-    m_gain = AudioGain::create("gain", 1.0, 0.0, 1.0);
-    m_playbackRate = AudioParam::create("playbackRate", 1.0, 0.0, AudioResampler::MaxRate);
-
-    // Default to mono.  A call to setBuffer() will set the number of output channels to that of the buffer.
-    addOutput(adoptPtr(new AudioNodeOutput(this, 1)));
-
-    initialize();
-}
-
-AudioBufferSourceNode::~AudioBufferSourceNode()
-{
-    uninitialize();
-}
-
-void AudioBufferSourceNode::process(size_t framesToProcess)
-{
-    AudioBus* outputBus = output(0)->bus();
-
-    if (!isInitialized()) {
-        outputBus->zero();
-        return;
-    }
-
-    // The audio thread can't block on this lock, so we call tryLock() instead.
-    // Careful - this is a tryLock() and not an autolocker, so we must unlock() before every return.
-    if (m_processLock.tryLock()) {
-        // Check if it's time to start playing.
-        double sampleRate = this->sampleRate();
-        double pitchRate = totalPitchRate();
-        double quantumStartTime = context()->currentTime();
-        double quantumEndTime = quantumStartTime + framesToProcess / sampleRate;
-
-        if (!m_isPlaying || m_hasFinished || !buffer() || m_startTime >= quantumEndTime) {
-            // FIXME: can optimize here by propagating silent hint instead of forcing the whole chain to process silence.
-            outputBus->zero();
-            m_processLock.unlock();
-            return;
-        }
-
-        // Handle sample-accurate scheduling so that buffer playback will happen at a very precise time.
-        m_schedulingFrameDelay = 0;
-        if (m_startTime >= quantumStartTime) {
-            // m_schedulingFrameDelay is set here only the very first render quantum (because of above check: m_startTime >= quantumEndTime)
-            // So: quantumStartTime <= m_startTime < quantumEndTime
-            ASSERT(m_startTime < quantumEndTime);
-            
-            double startTimeInQuantum = m_startTime - quantumStartTime;
-            double startFrameInQuantum = startTimeInQuantum * sampleRate;
-            
-            // m_schedulingFrameDelay is used in provideInput(), so factor in the current playback pitch rate.
-            m_schedulingFrameDelay = static_cast<int>(pitchRate * startFrameInQuantum);
-        }
-
-        // FIXME: optimization opportunity:
-        // With a bit of work, it should be possible to avoid going through the resampler completely when the pitchRate == 1,
-        // especially if the pitchRate has never deviated from 1 in the past.
-
-        // Read the samples through the pitch resampler.  Our provideInput() method will be called by the resampler.
-        m_resampler.setRate(pitchRate);
-        m_resampler.process(this, outputBus, framesToProcess);
-
-        // Apply the gain (in-place) to the output bus.
-        double totalGain = gain()->value() * m_buffer->gain();
-        outputBus->copyWithGainFrom(*outputBus, &m_lastGain, totalGain);
-
-        m_processLock.unlock();
-    } else {
-        // Too bad - the tryLock() failed.  We must be in the middle of changing buffers and were already outputting silence anyway.
-        outputBus->zero();
-    }
-}
-
-// The resampler calls us back here to get the input samples from our buffer.
-void AudioBufferSourceNode::provideInput(AudioBus* bus, size_t numberOfFrames)
-{
-    ASSERT(context()->isAudioThread());
-    
-    // Basic sanity checking
-    ASSERT(bus);
-    ASSERT(buffer());
-    if (!bus || !buffer())
-        return;
-
-    unsigned numberOfChannels = this->numberOfChannels();
-    unsigned busNumberOfChannels = bus->numberOfChannels();
-
-    // FIXME: we can add support for sources with more than two channels, but this is not a common case.
-    bool channelCountGood = numberOfChannels == busNumberOfChannels && (numberOfChannels == 1 || numberOfChannels == 2);
-    ASSERT(channelCountGood);
-    if (!channelCountGood)
-        return;
-
-    // Get the destination pointers.
-    float* destinationL = bus->channel(0)->data();
-    ASSERT(destinationL);
-    if (!destinationL)
-        return;
-    float* destinationR = (numberOfChannels < 2) ? 0 : bus->channel(1)->data();
-
-    size_t bufferLength = buffer()->length();
-    double bufferSampleRate = buffer()->sampleRate();
-
-    // Calculate the start and end frames in our buffer that we want to play.
-    // If m_isGrain is true, then we will be playing a portion of the total buffer.
-    unsigned startFrame = m_isGrain ? static_cast<unsigned>(m_grainOffset * bufferSampleRate) : 0;
-    unsigned endFrame = m_isGrain ? static_cast<unsigned>(startFrame + m_grainDuration * bufferSampleRate) : bufferLength;
-
-    // This is a HACK to allow for HRTF tail-time - avoids glitch at end.
-    // FIXME: implement tailTime for each AudioNode for a more general solution to this problem.
-    if (m_isGrain)
-        endFrame += 512;
-
-    // Do some sanity checking.
-    if (startFrame >= bufferLength)
-        startFrame = !bufferLength ? 0 : bufferLength - 1;
-    if (endFrame > bufferLength)
-        endFrame = bufferLength;
-    if (m_readIndex >= endFrame)
-        m_readIndex = startFrame; // reset to start
-    
-    int framesToProcess = numberOfFrames;
-
-    // Handle sample-accurate scheduling so that we play the buffer at a very precise time.
-    // m_schedulingFrameDelay will only be non-zero the very first time that provideInput() is called, which corresponds
-    // with the very start of the buffer playback.
-    if (m_schedulingFrameDelay > 0) {
-        ASSERT(m_schedulingFrameDelay <= framesToProcess);
-        if (m_schedulingFrameDelay <= framesToProcess) {
-            // Generate silence for the initial portion of the destination.
-            memset(destinationL, 0, sizeof(float) * m_schedulingFrameDelay);
-            destinationL += m_schedulingFrameDelay;
-            if (destinationR) {
-                memset(destinationR, 0, sizeof(float) * m_schedulingFrameDelay);
-                destinationR += m_schedulingFrameDelay;
-            }
-
-            // Since we just generated silence for the initial portion, we have fewer frames to provide.
-            framesToProcess -= m_schedulingFrameDelay;
-        }
-    }
-    
-    // We have to generate a certain number of output sample-frames, but we need to handle the case where we wrap around
-    // from the end of the buffer to the start if playing back with looping and also the case where we simply reach the
-    // end of the sample data, but haven't yet rendered numberOfFrames worth of output.
-    while (framesToProcess > 0) {
-        ASSERT(m_readIndex <= endFrame);
-        if (m_readIndex > endFrame)
-            return;
-            
-        // Figure out how many frames we can process this time.
-        int framesAvailable = endFrame - m_readIndex;
-        int framesThisTime = min(framesToProcess, framesAvailable);
-        
-        // Create the destination bus for the part of the destination we're processing this time.
-        AudioBus currentDestinationBus(busNumberOfChannels, framesThisTime, false);
-        currentDestinationBus.setChannelMemory(0, destinationL, framesThisTime);
-        if (busNumberOfChannels > 1)
-            currentDestinationBus.setChannelMemory(1, destinationR, framesThisTime);
-
-        // Generate output from the buffer.
-        readFromBuffer(&currentDestinationBus, framesThisTime);
-
-        // Update the destination pointers.
-        destinationL += framesThisTime;
-        if (busNumberOfChannels > 1)
-            destinationR += framesThisTime;
-
-        framesToProcess -= framesThisTime;
-
-        // Handle the case where we reach the end of the part of the sample data we're supposed to play for the buffer.
-        if (m_readIndex >= endFrame) {
-            m_readIndex = startFrame;
-            m_grainFrameCount = 0;
-            
-            if (!looping()) {
-                // If we're not looping, then stop playing when we get to the end.
-                m_isPlaying = false;
-
-                if (framesToProcess > 0) {
-                    // We're not looping and we've reached the end of the sample data, but we still need to provide more output,
-                    // so generate silence for the remaining.
-                    memset(destinationL, 0, sizeof(float) * framesToProcess);
-
-                    if (destinationR)
-                        memset(destinationR, 0, sizeof(float) * framesToProcess);
-                }
-
-                if (!m_hasFinished) {
-                    // Let the context dereference this AudioNode.
-                    context()->notifyNodeFinishedProcessing(this);
-                    m_hasFinished = true;
-                }
-                return;
-            }
-        }
-    }
-}
-
-void AudioBufferSourceNode::readFromBuffer(AudioBus* destinationBus, size_t framesToProcess)
-{
-    bool isBusGood = destinationBus && destinationBus->length() == framesToProcess && destinationBus->numberOfChannels() == numberOfChannels();
-    ASSERT(isBusGood);
-    if (!isBusGood)
-        return;
-    
-    unsigned numberOfChannels = this->numberOfChannels();
-    // FIXME: we can add support for sources with more than two channels, but this is not a common case.
-    bool channelCountGood = numberOfChannels == 1 || numberOfChannels == 2;
-    ASSERT(channelCountGood);
-    if (!channelCountGood)
-        return;
-            
-    // Get pointers to the start of the sample buffer.
-    float* sourceL = m_buffer->getChannelData(0)->data();
-    float* sourceR = m_buffer->numberOfChannels() == 2 ? m_buffer->getChannelData(1)->data() : 0;
-
-    // Sanity check buffer access.
-    bool isSourceGood = sourceL && (numberOfChannels == 1 || sourceR) && m_readIndex + framesToProcess <= m_buffer->length();
-    ASSERT(isSourceGood);
-    if (!isSourceGood)
-        return;
-
-    // Offset the pointers to the current read position in the sample buffer.
-    sourceL += m_readIndex;
-    sourceR += m_readIndex;
-
-    // Get pointers to the destination.
-    float* destinationL = destinationBus->channel(0)->data();
-    float* destinationR = numberOfChannels == 2 ? destinationBus->channel(1)->data() : 0;
-    bool isDestinationGood = destinationL && (numberOfChannels == 1 || destinationR);
-    ASSERT(isDestinationGood);
-    if (!isDestinationGood)
-        return;
-
-    if (m_isGrain)
-        readFromBufferWithGrainEnvelope(sourceL, sourceR, destinationL, destinationR, framesToProcess);
-    else {
-        // Simply copy the data from the source buffer to the destination.
-        memcpy(destinationL, sourceL, sizeof(float) * framesToProcess);
-        if (numberOfChannels == 2)
-            memcpy(destinationR, sourceR, sizeof(float) * framesToProcess);
-    }
-
-    // Advance the buffer's read index.
-    m_readIndex += framesToProcess;
-}
-
-void AudioBufferSourceNode::readFromBufferWithGrainEnvelope(float* sourceL, float* sourceR, float* destinationL, float* destinationR, size_t framesToProcess)
-{
-    ASSERT(sourceL && destinationL);
-    if (!sourceL || !destinationL)
-        return;
-        
-    int grainFrameLength = static_cast<int>(m_grainDuration * m_buffer->sampleRate());
-    bool isStereo = sourceR && destinationR;
-    
-    int n = framesToProcess;
-    while (n--) {
-        // Apply the grain envelope.
-        float x = static_cast<float>(m_grainFrameCount) / static_cast<float>(grainFrameLength);
-        m_grainFrameCount++;
-
-        x = min(1.0f, x);
-        float grainEnvelope = sinf(piFloat * x);
-        
-        *destinationL++ = grainEnvelope * *sourceL++;
-
-        if (isStereo)
-            *destinationR++ = grainEnvelope * *sourceR++;
-    }
-}
-
-void AudioBufferSourceNode::reset()
-{
-    m_resampler.reset();
-    m_readIndex = 0;
-    m_grainFrameCount = 0;
-    m_lastGain = gain()->value();
-}
-
-void AudioBufferSourceNode::setBuffer(AudioBuffer* buffer)
-{
-    ASSERT(isMainThread());
-    
-    // The context must be locked since changing the buffer can re-configure the number of channels that are output.
-    AudioContext::AutoLocker contextLocker(context());
-    
-    // This synchronizes with process().
-    MutexLocker processLocker(m_processLock);
-    
-    if (buffer) {
-        // Do any necesssary re-configuration to the buffer's number of channels.
-        unsigned numberOfChannels = buffer->numberOfChannels();
-        m_resampler.configureChannels(numberOfChannels);
-        output(0)->setNumberOfChannels(numberOfChannels);
-    }
-
-    m_readIndex = 0;
-    m_buffer = buffer;
-}
-
-unsigned AudioBufferSourceNode::numberOfChannels()
-{
-    return output(0)->numberOfChannels();
-}
-
-void AudioBufferSourceNode::noteOn(double when)
-{
-    ASSERT(isMainThread());
-    if (m_isPlaying)
-        return;
-
-    m_isGrain = false;
-    m_startTime = when;
-    m_readIndex = 0;
-    m_isPlaying = true;
-}
-
-void AudioBufferSourceNode::noteGrainOn(double when, double grainOffset, double grainDuration)
-{
-    ASSERT(isMainThread());
-    if (m_isPlaying)
-        return;
-
-    if (!buffer())
-        return;
-        
-    // Do sanity checking of grain parameters versus buffer size.
-    double bufferDuration = buffer()->duration();
-
-    if (grainDuration > bufferDuration)
-        return; // FIXME: maybe should throw exception - consider in specification.
-    
-    double maxGrainOffset = bufferDuration - grainDuration;
-    maxGrainOffset = max(0.0, maxGrainOffset);
-
-    grainOffset = max(0.0, grainOffset);
-    grainOffset = min(maxGrainOffset, grainOffset);    
-    m_grainOffset = grainOffset;
-
-    m_grainDuration = grainDuration;
-    m_grainFrameCount = 0;
-    
-    m_isGrain = true;
-    m_startTime = when;
-    m_readIndex = static_cast<int>(m_grainOffset * buffer()->sampleRate());
-    m_isPlaying = true;
-}
-
-void AudioBufferSourceNode::noteOff(double)
-{
-    ASSERT(isMainThread());
-    if (!m_isPlaying)
-        return;
-        
-    // FIXME: the "when" argument to this method is ignored.
-    m_isPlaying = false;
-    m_readIndex = 0;
-}
-
-double AudioBufferSourceNode::totalPitchRate()
-{
-    double dopplerRate = 1.0;
-    if (m_pannerNode.get())
-        dopplerRate = m_pannerNode->dopplerRate();
-    
-    // Incorporate buffer's sample-rate versus AudioContext's sample-rate.
-    // Normally it's not an issue because buffers are loaded at the AudioContext's sample-rate, but we can handle it in any case.
-    double sampleRateFactor = 1.0;
-    if (buffer())
-        sampleRateFactor = buffer()->sampleRate() / sampleRate();
-    
-    double basePitchRate = playbackRate()->value();
-
-    double totalRate = dopplerRate * sampleRateFactor * basePitchRate;
-
-    // Sanity check the total rate.  It's very important that the resampler not get any bad rate values.
-    totalRate = max(0.0, totalRate);
-    totalRate = min(AudioResampler::MaxRate, totalRate);
-    
-    bool isTotalRateValid = !isnan(totalRate) && !isinf(totalRate);
-    ASSERT(isTotalRateValid);
-    if (!isTotalRateValid)
-        totalRate = 1.0;
-    
-    return totalRate;
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioBufferSourceNode.h b/WebCore/webaudio/AudioBufferSourceNode.h
deleted file mode 100644
index 40b8555..0000000
--- a/WebCore/webaudio/AudioBufferSourceNode.h
+++ /dev/null
@@ -1,147 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef AudioBufferSourceNode_h
-#define AudioBufferSourceNode_h
-
-#include "AudioBuffer.h"
-#include "AudioBus.h"
-#include "AudioGain.h"
-#include "AudioPannerNode.h"
-#include "AudioResampler.h"
-#include "AudioSourceNode.h"
-#include "AudioSourceProvider.h"
-#include <wtf/PassRefPtr.h>
-#include <wtf/RefPtr.h>
-#include <wtf/Threading.h>
-
-namespace WebCore {
-
-class AudioContext;
-
-// AudioBufferSourceNode is an AudioNode representing an audio source from an in-memory audio asset represented by an AudioBuffer.
-// It generally will be used for short sounds which require a high degree of scheduling flexibility (can playback in rhythmically perfect ways).
-
-class AudioBufferSourceNode : public AudioSourceNode, public AudioSourceProvider {
-public:
-    static PassRefPtr<AudioBufferSourceNode> create(AudioContext*, double sampleRate);
-
-    virtual ~AudioBufferSourceNode();
-    
-    // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
-
-    // AudioSourceProvider
-    // When process() is called, the resampler calls provideInput (in the audio thread) to gets its input stream.
-    virtual void provideInput(AudioBus*, size_t numberOfFrames);
-    
-    // setBuffer() is called on the main thread.  This is the buffer we use for playback.
-    void setBuffer(AudioBuffer*);
-    AudioBuffer* buffer() { return m_buffer.get(); }
-                    
-    // numberOfChannels() returns the number of output channels.  This value equals the number of channels from the buffer.
-    // If a new buffer is set with a different number of channels, then this value will dynamically change.
-    unsigned numberOfChannels();
-                    
-    // Play-state
-    // noteOn(), noteGrainOn(), and noteOff() must all be called from the main thread.
-    void noteOn(double when);
-    void noteGrainOn(double when, double grainOffset, double grainDuration);
-    void noteOff(double when);
-
-    bool looping() const { return m_isLooping; }
-    void setLooping(bool looping) { m_isLooping = looping; }
-    
-    AudioGain* gain() { return m_gain.get(); }                                        
-    AudioParam* playbackRate() { return m_playbackRate.get(); }
-
-    // If a panner node is set, then we can incorporate doppler shift into the playback pitch rate.
-    void setPannerNode(PassRefPtr<AudioPannerNode> pannerNode) { m_pannerNode = pannerNode; }
-
-private:
-    AudioBufferSourceNode(AudioContext*, double sampleRate);
-
-    // m_buffer holds the sample data which this node outputs.
-    RefPtr<AudioBuffer> m_buffer;
-
-    // Used for the "gain" and "playbackRate" attributes.
-    RefPtr<AudioGain> m_gain;
-    RefPtr<AudioParam> m_playbackRate;
-
-    // m_isPlaying is set to true when noteOn() or noteGrainOn() is called.
-    bool m_isPlaying;
-
-    // If m_isLooping is false, then this node will be done playing and become inactive after it reaches the end of the sample data in the buffer.
-    // If true, it will wrap around to the start of the buffer each time it reaches the end.
-    bool m_isLooping;
-
-    // This node is considered finished when it reaches the end of the buffer's sample data after noteOn() has been called.
-    // This will only be set to true if m_isLooping == false.
-    bool m_hasFinished;
-
-    // m_startTime is the time to start playing based on the context's timeline (0.0 or a time less than the context's current time means "now").
-    double m_startTime; // in seconds
-
-    // m_schedulingFrameDelay is the sample-accurate scheduling offset.
-    // It's used so that we start rendering audio samples at a very precise point in time.
-    // It will only be a non-zero value the very first render quantum that we render from the buffer.
-    int m_schedulingFrameDelay;
-
-    // m_readIndex is a sample-frame index into our buffer representing the current playback position.
-    unsigned m_readIndex;
-
-    // Granular playback
-    bool m_isGrain;
-    double m_grainOffset; // in seconds
-    double m_grainDuration; // in seconds
-    int m_grainFrameCount; // keeps track of which frame in the grain we're currently rendering
-
-    // totalPitchRate() returns the instantaneous pitch rate (non-time preserving).
-    // It incorporates the base pitch rate, any sample-rate conversion factor from the buffer, and any doppler shift from an associated panner node.
-    double totalPitchRate();
-
-    // m_resampler performs the pitch rate changes to the buffer playback.
-    AudioResampler m_resampler;
-
-    // m_lastGain provides continuity when we dynamically adjust the gain.
-    double m_lastGain;
-    
-    // We optionally keep track of a panner node which has a doppler shift that is incorporated into the pitch rate.
-    RefPtr<AudioPannerNode> m_pannerNode;
-
-    // This synchronizes process() with setBuffer() which can cause dynamic channel count changes.
-    mutable Mutex m_processLock;
-
-    // Reads the next framesToProcess sample-frames from the AudioBuffer into destinationBus.
-    // A grain envelope will be applied if m_isGrain is set to true.
-    void readFromBuffer(AudioBus* destinationBus, size_t framesToProcess);
-
-    // readFromBufferWithGrainEnvelope() is a low-level blitter which reads from the AudioBuffer and applies a grain envelope.
-    void readFromBufferWithGrainEnvelope(float* sourceL, float* sourceR, float* destinationL, float* destinationR, size_t framesToProcess);
-};
-
-} // namespace WebCore
-
-#endif // AudioBufferSourceNode_h
diff --git a/WebCore/webaudio/AudioBufferSourceNode.idl b/WebCore/webaudio/AudioBufferSourceNode.idl
deleted file mode 100644
index dec7461..0000000
--- a/WebCore/webaudio/AudioBufferSourceNode.idl
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    // A cached (non-streamed), memory-resident audio source
-    interface [
-        Conditional=WEB_AUDIO,
-        GenerateToJS
-    ] AudioBufferSourceNode : AudioSourceNode {
-        attribute [JSCCustomSetter] AudioBuffer buffer;
-
-        readonly attribute AudioGain gain;
-        readonly attribute AudioParam playbackRate;
-        attribute boolean looping; // FIXME: change name to 'loop' once samples are updated
-
-        void noteOn(in float when);
-        void noteGrainOn(in float when, in float grainOffset, in float grainDuration);
-        void noteOff(in float when);
-    };
-}
diff --git a/WebCore/webaudio/AudioChannelMerger.cpp b/WebCore/webaudio/AudioChannelMerger.cpp
deleted file mode 100644
index c418a61..0000000
--- a/WebCore/webaudio/AudioChannelMerger.cpp
+++ /dev/null
@@ -1,102 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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 "AudioChannelMerger.h"
-
-#include "AudioNodeInput.h"
-#include "AudioNodeOutput.h"
-
-namespace WebCore {
-    
-// This is considering that 5.1 (6 channels) is the largest we'll ever deal with.
-// It can easily be increased to support more if the web audio specification is updated.
-const unsigned NumberOfInputs = 6;
-
-AudioChannelMerger::AudioChannelMerger(AudioContext* context, double sampleRate)
-    : AudioNode(context, sampleRate)
-{
-    // Create a fixed number of inputs (able to handle the maximum number of channels we deal with).
-    for (unsigned i = 0; i < NumberOfInputs; ++i)
-        addInput(adoptPtr(new AudioNodeInput(this)));
-
-    addOutput(adoptPtr(new AudioNodeOutput(this, 1)));
-    
-    setType(NodeTypeChannelMerger);
-    
-    initialize();
-}
-
-void AudioChannelMerger::process(size_t framesToProcess)
-{
-    AudioNodeOutput* output = this->output(0);
-    ASSERT(output);
-    ASSERT_UNUSED(framesToProcess, framesToProcess == output->bus()->length());    
-    
-    // Count how many channels we have all together from all of the inputs.
-    unsigned numberOfOutputChannels = 0;
-    for (unsigned i = 0; i < numberOfInputs(); ++i) {
-        AudioNodeInput* input = this->input(i);
-        if (input->isConnected())
-            numberOfOutputChannels += input->bus()->numberOfChannels();
-    }
-
-    // Set the correct number of channels on the output
-    output->setNumberOfChannels(numberOfOutputChannels);
-    
-    // Now merge the channels back into one output.
-    unsigned outputChannelIndex = 0;
-    for (unsigned i = 0; i < numberOfInputs(); ++i) {
-        AudioNodeInput* input = this->input(i);
-        if (input->isConnected()) {
-            unsigned numberOfInputChannels = input->bus()->numberOfChannels();
-            
-            // Merge channels from this particular input.
-            for (unsigned j = 0; j < numberOfInputChannels; ++j) {
-                AudioChannel* inputChannel = input->bus()->channel(j);
-                AudioChannel* outputChannel = output->bus()->channel(outputChannelIndex);
-                outputChannel->copyFrom(inputChannel);
-                
-                ++outputChannelIndex;
-            }
-        }
-    }
-    
-    ASSERT(outputChannelIndex == numberOfOutputChannels);
-}
-
-void AudioChannelMerger::reset()
-{
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioChannelMerger.h b/WebCore/webaudio/AudioChannelMerger.h
deleted file mode 100644
index 20a9628..0000000
--- a/WebCore/webaudio/AudioChannelMerger.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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.
- */
-
-#ifndef AudioChannelMerger_h
-#define AudioChannelMerger_h
-
-#include "AudioNode.h"
-#include <wtf/PassRefPtr.h>
-
-namespace WebCore {
-
-class AudioContext;
-    
-class AudioChannelMerger : public AudioNode {
-public:
-    static PassRefPtr<AudioChannelMerger> create(AudioContext* context, double sampleRate)
-    {
-        return adoptRef(new AudioChannelMerger(context, sampleRate));      
-    }
-
-    // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
-
-private:
-    AudioChannelMerger(AudioContext*, double sampleRate);
-};
-
-} // namespace WebCore
-
-#endif // AudioChannelMerger_h
diff --git a/WebCore/webaudio/AudioChannelMerger.idl b/WebCore/webaudio/AudioChannelMerger.idl
deleted file mode 100644
index 3862af9..0000000
--- a/WebCore/webaudio/AudioChannelMerger.idl
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO
-    ] AudioChannelMerger : AudioNode {
-    };
-}
diff --git a/WebCore/webaudio/AudioChannelSplitter.cpp b/WebCore/webaudio/AudioChannelSplitter.cpp
deleted file mode 100644
index f4fa041..0000000
--- a/WebCore/webaudio/AudioChannelSplitter.cpp
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- * 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 "AudioChannelSplitter.h"
-
-#include "AudioNodeInput.h"
-#include "AudioNodeOutput.h"
-
-namespace WebCore {
-    
-// This is considering that 5.1 (6 channels) is the largest we'll ever deal with.
-// It can easily be increased to support more if the web audio specification is updated.
-const unsigned NumberOfOutputs = 6;
-
-AudioChannelSplitter::AudioChannelSplitter(AudioContext* context, double sampleRate)
-    : AudioNode(context, sampleRate)
-{
-    addInput(adoptPtr(new AudioNodeInput(this)));
-
-    // Create a fixed number of outputs (able to handle the maximum number of channels fed to an input).
-    for (unsigned i = 0; i < NumberOfOutputs; ++i)
-        addOutput(adoptPtr(new AudioNodeOutput(this, 1)));
-    
-    setType(NodeTypeChannelSplitter);
-    
-    initialize();
-}
-
-void AudioChannelSplitter::process(size_t framesToProcess)
-{
-    AudioBus* source = input(0)->bus();
-    ASSERT(source);
-    ASSERT_UNUSED(framesToProcess, framesToProcess == source->length());
-    
-    unsigned numberOfSourceChannels = source->numberOfChannels();
-    
-    ASSERT(numberOfOutputs() == NumberOfOutputs);
-    for (unsigned i = 0; i < NumberOfOutputs; ++i) {
-        AudioBus* destination = output(i)->bus();
-        ASSERT(destination);
-        
-        if (i < numberOfSourceChannels) {
-            // Split the channel out if it exists in the source.
-            // It would be nice to avoid the copy and simply pass along pointers, but this becomes extremely difficult with fanout and fanin.
-            destination->channel(0)->copyFrom(source->channel(i));
-        } else if (output(i)->renderingFanOutCount() > 0) {
-            // Only bother zeroing out the destination if it's connected to anything
-            destination->zero();
-        }
-    }
-}
-
-void AudioChannelSplitter::reset()
-{
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioChannelSplitter.h b/WebCore/webaudio/AudioChannelSplitter.h
deleted file mode 100644
index 7dadac5..0000000
--- a/WebCore/webaudio/AudioChannelSplitter.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef AudioChannelSplitter_h
-#define AudioChannelSplitter_h
-
-#include "AudioNode.h"
-#include <wtf/PassRefPtr.h>
-
-namespace WebCore {
-
-class AudioContext;
-    
-class AudioChannelSplitter : public AudioNode {
-public:
-    static PassRefPtr<AudioChannelSplitter> create(AudioContext* context, double sampleRate)
-    {
-        return adoptRef(new AudioChannelSplitter(context, sampleRate));      
-    }
-
-    // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
-
-private:
-    AudioChannelSplitter(AudioContext*, double sampleRate);
-};
-
-} // namespace WebCore
-
-#endif // AudioChannelSplitter_h
diff --git a/WebCore/webaudio/AudioChannelSplitter.idl b/WebCore/webaudio/AudioChannelSplitter.idl
deleted file mode 100644
index 076c051..0000000
--- a/WebCore/webaudio/AudioChannelSplitter.idl
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO
-    ] AudioChannelSplitter : AudioNode {
-    };
-}
diff --git a/WebCore/webaudio/AudioContext.cpp b/WebCore/webaudio/AudioContext.cpp
deleted file mode 100644
index a452775..0000000
--- a/WebCore/webaudio/AudioContext.cpp
+++ /dev/null
@@ -1,529 +0,0 @@
-/*
- * 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 "AudioContext.h"
-
-#include "ArrayBuffer.h"
-#include "AudioBuffer.h"
-#include "AudioBufferSourceNode.h"
-#include "AudioChannelMerger.h"
-#include "AudioChannelSplitter.h"
-#include "AudioGainNode.h"
-#include "AudioListener.h"
-#include "AudioNodeInput.h"
-#include "AudioNodeOutput.h"
-#include "AudioPannerNode.h"
-#include "ConvolverNode.h"
-#include "DelayNode.h"
-#include "Document.h"
-#include "HRTFDatabaseLoader.h"
-#include "HRTFPanner.h"
-#include "HighPass2FilterNode.h"
-#include "JavaScriptAudioNode.h"
-#include "LowPass2FilterNode.h"
-#include "PlatformString.h"
-#include "RealtimeAnalyserNode.h"
-
-#include <wtf/OwnPtr.h>
-#include <wtf/PassOwnPtr.h>
-#include <wtf/RefCounted.h>
-
-// FIXME: check the proper way to reference an undefined thread ID
-const int UndefinedThreadIdentifier = 0xffffffff;
-
-const unsigned MaxNodesToDeletePerQuantum = 10;
-
-namespace WebCore {
-
-PassRefPtr<AudioContext> AudioContext::create(Document* document)
-{
-    return adoptRef(new AudioContext(document));
-}
-
-AudioContext::AudioContext(Document* document)
-    : ActiveDOMObject(document, this)
-    , m_isInitialized(false)
-    , m_isAudioThreadFinished(false)
-    , m_document(document)
-    , m_destinationNode(0)
-    , m_connectionCount(0)
-    , m_audioThread(0)
-    , m_graphOwnerThread(UndefinedThreadIdentifier)
-{
-    // Note: because adoptRef() won't be called until we leave this constructor, but code in this constructor needs to reference this context,
-    // relax the check.
-    relaxAdoptionRequirement();
-    
-    m_destinationNode = AudioDestinationNode::create(this);
-    m_listener = AudioListener::create();
-    m_temporaryMonoBus = adoptPtr(new AudioBus(1, AudioNode::ProcessingSizeInFrames));
-    m_temporaryStereoBus = adoptPtr(new AudioBus(2, AudioNode::ProcessingSizeInFrames));
-
-    // 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());
-}
-
-AudioContext::~AudioContext()
-{
-#if DEBUG_AUDIONODE_REFERENCES
-    printf("%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_nodesToDelete.size());
-    ASSERT(!m_referencedNodes.size());
-    ASSERT(!m_finishedNodes.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()) {
-                // 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".
-                m_destinationNode->initialize();
-            }
-            m_isInitialized = true;
-        }
-    }
-}
-
-void AudioContext::uninitialize()
-{
-    if (m_isInitialized) {    
-        // 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;
-
-        // 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.
-        m_destinationNode.clear();
-        
-        // Get rid of the sources which may still be playing.
-        derefUnfinishedSourceNodes();
-        
-        // Because the AudioBuffers are garbage collected, we can't delete them here.
-        // Instead, at least release the potentially large amount of allocated memory for the audio data.
-        // Note that we do this *after* the context is uninitialized and stops processing audio.
-        for (unsigned i = 0; i < m_allocatedBuffers.size(); ++i)
-            m_allocatedBuffers[i]->releaseMemory();
-        m_allocatedBuffers.clear();
-    
-        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::stop()
-{
-    m_document = 0; // document is going away
-    uninitialize();
-}
-
-Document* AudioContext::document()
-{
-    ASSERT(m_document);
-    return m_document;
-}
-
-bool AudioContext::hasDocument()
-{
-    return m_document;
-}
-
-void AudioContext::refBuffer(PassRefPtr<AudioBuffer> buffer)
-{
-    m_allocatedBuffers.append(buffer);
-}
-
-PassRefPtr<AudioBuffer> AudioContext::createBuffer(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate)
-{
-    return AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate);
-}
-
-PassRefPtr<AudioBuffer> AudioContext::createBuffer(ArrayBuffer* arrayBuffer, bool mixToMono)
-{
-    ASSERT(arrayBuffer);
-    if (!arrayBuffer)
-        return 0;
-    
-    return AudioBuffer::createFromAudioFileData(arrayBuffer->data(), arrayBuffer->byteLength(), mixToMono, sampleRate());
-}
-
-PassRefPtr<AudioBufferSourceNode> AudioContext::createBufferSource()
-{
-    ASSERT(isMainThread());
-    lazyInitialize();
-    RefPtr<AudioBufferSourceNode> node = AudioBufferSourceNode::create(this, m_destinationNode->sampleRate());
-
-    refNode(node.get()); // context keeps reference until source has finished playing
-    return node;
-}
-
-PassRefPtr<JavaScriptAudioNode> AudioContext::createJavaScriptNode(size_t bufferSize)
-{
-    ASSERT(isMainThread());
-    lazyInitialize();
-    RefPtr<JavaScriptAudioNode> node = JavaScriptAudioNode::create(this, m_destinationNode->sampleRate(), bufferSize);
-
-    refNode(node.get()); // context keeps reference until we stop making javascript rendering callbacks
-    return node;
-}
-
-PassRefPtr<LowPass2FilterNode> AudioContext::createLowPass2Filter()
-{
-    ASSERT(isMainThread());
-    lazyInitialize();
-    return LowPass2FilterNode::create(this, m_destinationNode->sampleRate());
-}
-
-PassRefPtr<HighPass2FilterNode> AudioContext::createHighPass2Filter()
-{
-    ASSERT(isMainThread());
-    lazyInitialize();
-    return HighPass2FilterNode::create(this, m_destinationNode->sampleRate());
-}
-
-PassRefPtr<AudioPannerNode> AudioContext::createPanner()
-{
-    ASSERT(isMainThread());
-    lazyInitialize();
-    return AudioPannerNode::create(this, m_destinationNode->sampleRate());
-}
-
-PassRefPtr<ConvolverNode> AudioContext::createConvolver()
-{
-    ASSERT(isMainThread());
-    lazyInitialize();
-    return ConvolverNode::create(this, m_destinationNode->sampleRate());
-}
-
-PassRefPtr<RealtimeAnalyserNode> AudioContext::createAnalyser()
-{
-    ASSERT(isMainThread());
-    lazyInitialize();
-    return RealtimeAnalyserNode::create(this, m_destinationNode->sampleRate());
-}
-
-PassRefPtr<AudioGainNode> AudioContext::createGainNode()
-{
-    ASSERT(isMainThread());
-    lazyInitialize();
-    return AudioGainNode::create(this, m_destinationNode->sampleRate());
-}
-
-PassRefPtr<DelayNode> AudioContext::createDelayNode()
-{
-    ASSERT(isMainThread());
-    lazyInitialize();
-    return DelayNode::create(this, m_destinationNode->sampleRate());
-}
-
-PassRefPtr<AudioChannelSplitter> AudioContext::createChannelSplitter()
-{
-    ASSERT(isMainThread());
-    lazyInitialize();
-    return AudioChannelSplitter::create(this, m_destinationNode->sampleRate());
-}
-
-PassRefPtr<AudioChannelMerger> AudioContext::createChannelMerger()
-{
-    ASSERT(isMainThread());
-    lazyInitialize();
-    return AudioChannelMerger::create(this, m_destinationNode->sampleRate());
-}
-
-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, AudioNode::RefType refType)
-{
-    ASSERT(isAudioThread());
-    m_deferredFinishDerefList.append(AudioContext::RefInfo(node, refType));
-}
-
-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 AudioNodeInputs and AudioNodeOutputs.
-        handleDirtyAudioNodeInputs();
-        handleDirtyAudioNodeOutputs();
-        
-        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();
-
-        // Finally actually delete.
-        deleteMarkedNodes();
-
-        // Fixup the state of any dirty AudioNodeInputs and AudioNodeOutputs.
-        handleDirtyAudioNodeInputs();
-        handleDirtyAudioNodeOutputs();
-        
-        if (mustReleaseLock)
-            unlock();
-    }
-}
-
-void AudioContext::handleDeferredFinishDerefs()
-{
-    ASSERT(isAudioThread() && isGraphOwner());
-    for (unsigned i = 0; i < m_deferredFinishDerefList.size(); ++i) {
-        AudioNode* node = m_deferredFinishDerefList[i].m_node;
-        AudioNode::RefType refType = m_deferredFinishDerefList[i].m_refType;
-        node->finishDeref(refType);
-    }
-    
-    m_deferredFinishDerefList.clear();
-}
-
-void AudioContext::markForDeletion(AudioNode* node)
-{
-    ASSERT(isGraphOwner());
-    m_nodesToDelete.append(node);
-}
-
-void AudioContext::deleteMarkedNodes()
-{
-    ASSERT(isGraphOwner() || isAudioThreadFinished());
-
-    // Note: deleting an AudioNode can cause m_nodesToDelete to grow.
-    size_t nodesDeleted = 0;
-    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_dirtyAudioNodeInputs.
-        unsigned numberOfInputs = node->numberOfInputs();
-        for (unsigned i = 0; i < numberOfInputs; ++i)
-            m_dirtyAudioNodeInputs.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;
-
-        // Don't delete too many nodes per render quantum since we don't want to do too much work in the realtime audio thread.
-        if (++nodesDeleted > MaxNodesToDeletePerQuantum)
-            break;
-    }
-}
-
-void AudioContext::markAudioNodeInputDirty(AudioNodeInput* input)
-{
-    ASSERT(isGraphOwner());    
-    m_dirtyAudioNodeInputs.add(input);
-}
-
-void AudioContext::markAudioNodeOutputDirty(AudioNodeOutput* output)
-{
-    ASSERT(isGraphOwner());    
-    m_dirtyAudioNodeOutputs.add(output);
-}
-
-void AudioContext::handleDirtyAudioNodeInputs()
-{
-    ASSERT(isGraphOwner());    
-
-    for (HashSet<AudioNodeInput*>::iterator i = m_dirtyAudioNodeInputs.begin(); i != m_dirtyAudioNodeInputs.end(); ++i)
-        (*i)->updateRenderingState();
-
-    m_dirtyAudioNodeInputs.clear();
-}
-
-void AudioContext::handleDirtyAudioNodeOutputs()
-{
-    ASSERT(isGraphOwner());    
-
-    for (HashSet<AudioNodeOutput*>::iterator i = m_dirtyAudioNodeOutputs.begin(); i != m_dirtyAudioNodeOutputs.end(); ++i)
-        (*i)->updateRenderingState();
-
-    m_dirtyAudioNodeOutputs.clear();
-}
-
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioContext.h b/WebCore/webaudio/AudioContext.h
deleted file mode 100644
index ddd474c..0000000
--- a/WebCore/webaudio/AudioContext.h
+++ /dev/null
@@ -1,259 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef AudioContext_h
-#define AudioContext_h
-
-#include "ActiveDOMObject.h"
-#include "AudioBus.h"
-#include "AudioDestinationNode.h"
-#include "HRTFDatabaseLoader.h"
-#include <wtf/HashSet.h>
-#include <wtf/OwnPtr.h>
-#include <wtf/PassRefPtr.h>
-#include <wtf/RefCounted.h>
-#include <wtf/RefPtr.h>
-#include <wtf/Threading.h>
-#include <wtf/Vector.h>
-#include <wtf/text/AtomicStringHash.h>
-
-namespace WebCore {
-
-class ArrayBuffer;
-class AudioBuffer;
-class AudioBufferSourceNode;
-class AudioChannelMerger;
-class AudioChannelSplitter;
-class AudioGainNode;
-class AudioPannerNode;
-class AudioListener;
-class DelayNode;
-class Document;
-class LowPass2FilterNode;
-class HighPass2FilterNode;
-class ConvolverNode;
-class RealtimeAnalyserNode;
-class JavaScriptAudioNode;
-
-// AudioContext is the cornerstone of the web audio API and all AudioNodes are created from it.
-// For thread safety between the audio thread and the main thread, it has a rendering graph locking mechanism. 
-
-class AudioContext : public ActiveDOMObject, public RefCounted<AudioContext> {
-public:
-    static PassRefPtr<AudioContext> create(Document*);
-
-    virtual ~AudioContext();
-
-    bool isInitialized() const;
-
-    // Returns true when initialize() was called AND all asynchronous initialization has completed.
-    bool isRunnable() const;
-
-    // Document notification
-    virtual void stop();
-
-    Document* document(); // ASSERTs if document no longer exists.
-    bool hasDocument();
-
-    AudioDestinationNode* destination() { return m_destinationNode.get(); }
-    double currentTime() { return m_destinationNode->currentTime(); }
-    double sampleRate() { return m_destinationNode->sampleRate(); }
-
-    PassRefPtr<AudioBuffer> createBuffer(unsigned numberOfChannels, size_t numberOfFrames, double sampleRate);
-    PassRefPtr<AudioBuffer> createBuffer(ArrayBuffer* arrayBuffer, bool mixToMono);
-
-    // Keep track of this buffer so we can release memory after the context is shut down...
-    void refBuffer(PassRefPtr<AudioBuffer> buffer);
-
-    AudioListener* listener() { return m_listener.get(); }
-
-    // The AudioNode create methods are called on the main thread (from JavaScript).
-    PassRefPtr<AudioBufferSourceNode> createBufferSource();
-    PassRefPtr<AudioGainNode> createGainNode();
-    PassRefPtr<DelayNode> createDelayNode();
-    PassRefPtr<LowPass2FilterNode> createLowPass2Filter();
-    PassRefPtr<HighPass2FilterNode> createHighPass2Filter();
-    PassRefPtr<AudioPannerNode> createPanner();
-    PassRefPtr<ConvolverNode> createConvolver();
-    PassRefPtr<RealtimeAnalyserNode> createAnalyser();
-    PassRefPtr<JavaScriptAudioNode> createJavaScriptNode(size_t bufferSize);
-    PassRefPtr<AudioChannelSplitter> createChannelSplitter();
-    PassRefPtr<AudioChannelMerger> createChannelMerger();
-
-    AudioBus* temporaryMonoBus() { return m_temporaryMonoBus.get(); }
-    AudioBus* temporaryStereoBus() { return m_temporaryStereoBus.get(); }
-
-    // When a source node has no more processing to do (has finished playing), then it tells the context to dereference it.
-    void notifyNodeFinishedProcessing(AudioNode*);
-
-    // Called at the start of each render quantum.
-    void handlePreRenderTasks();
-
-    // Called at the end of each render quantum.
-    void handlePostRenderTasks();
-
-    // Called periodically at the end of each render quantum to dereference finished source nodes.
-    void derefFinishedSourceNodes();
-
-    // We reap all marked nodes at the end of each realtime render quantum in deleteMarkedNodes().
-    void markForDeletion(AudioNode*);
-    void deleteMarkedNodes();
-
-    // Keeps track of the number of connections made.
-    void incrementConnectionCount()
-    {
-        ASSERT(isMainThread());
-        m_connectionCount++;
-    }
-
-    unsigned connectionCount() const { return m_connectionCount; }
-
-    //
-    // Thread Safety and Graph Locking:
-    //
-    
-    void setAudioThread(ThreadIdentifier thread) { m_audioThread = thread; } // FIXME: check either not initialized or the same
-    ThreadIdentifier audioThread() const { return m_audioThread; }
-    bool isAudioThread() const;
-
-    // Returns true only after the audio thread has been started and then shutdown.
-    bool isAudioThreadFinished() { return m_isAudioThreadFinished; }
-    
-    // mustReleaseLock is set to true if we acquired the lock in this method call and caller must unlock(), false if it was previously acquired.
-    void lock(bool& mustReleaseLock);
-
-    // Returns true if we own the lock.
-    // mustReleaseLock is set to true if we acquired the lock in this method call and caller must unlock(), false if it was previously acquired.
-    bool tryLock(bool& mustReleaseLock);
-
-    void unlock();
-
-    // Returns true if this thread owns the context's lock.
-    bool isGraphOwner() const;
-
-    class AutoLocker {
-    public:
-        AutoLocker(AudioContext* context)
-            : m_context(context)
-        {
-            ASSERT(context);
-            context->lock(m_mustReleaseLock);
-        }
-        
-        ~AutoLocker()
-        {
-            if (m_mustReleaseLock)
-                m_context->unlock();
-        }
-    private:
-        AudioContext* m_context;
-        bool m_mustReleaseLock;
-    };
-    
-    // In AudioNode::deref() a tryLock() is used for calling finishDeref(), but if it fails keep track here.
-    void addDeferredFinishDeref(AudioNode*, AudioNode::RefType);
-
-    // In the audio thread at the start of each render cycle, we'll call handleDeferredFinishDerefs().
-    void handleDeferredFinishDerefs();
-
-    // Only accessed when the graph lock is held.
-    void markAudioNodeInputDirty(AudioNodeInput*);
-    void markAudioNodeOutputDirty(AudioNodeOutput*);
-    
-private:
-    AudioContext(Document*);
-    void lazyInitialize();
-    void uninitialize();
-    
-    bool m_isInitialized;
-    bool m_isAudioThreadFinished;
-    bool m_isAudioThreadShutdown;
-
-    Document* m_document;
-
-    // The context itself keeps a reference to all source nodes.  The source nodes, then reference all nodes they're connected to.
-    // In turn, these nodes reference all nodes they're connected to.  All nodes are ultimately connected to the AudioDestinationNode.
-    // When the context dereferences a source node, it will be deactivated from the rendering graph along with all other nodes it is
-    // uniquely connected to.  See the AudioNode::ref() and AudioNode::deref() methods for more details.
-    void refNode(AudioNode*);
-    void derefNode(AudioNode*);
-
-    // When the context goes away, there might still be some sources which haven't finished playing.
-    // Make sure to dereference them here.
-    void derefUnfinishedSourceNodes();
-
-    RefPtr<AudioDestinationNode> m_destinationNode;
-    RefPtr<AudioListener> m_listener;
-
-    // Only accessed in the main thread.
-    Vector<RefPtr<AudioBuffer> > m_allocatedBuffers;
-
-    // Only accessed in the audio thread.
-    Vector<AudioNode*> m_finishedNodes;
-
-    // We don't use RefPtr<AudioNode> here because AudioNode has a more complex ref() / deref() implementation
-    // with an optional argument for refType.  We need to use the special refType: RefTypeConnection
-    // Either accessed when the graph lock is held, or on the main thread when the audio thread has finished.
-    Vector<AudioNode*> m_referencedNodes;
-
-    // Accumulate nodes which need to be deleted at the end of a render cycle (in realtime thread) here.
-    Vector<AudioNode*> m_nodesToDelete;
-
-    // Only accessed when the graph lock is held.
-    HashSet<AudioNodeInput*> m_dirtyAudioNodeInputs;
-    HashSet<AudioNodeOutput*> m_dirtyAudioNodeOutputs;
-    void handleDirtyAudioNodeInputs();
-    void handleDirtyAudioNodeOutputs();
-
-    OwnPtr<AudioBus> m_temporaryMonoBus;
-    OwnPtr<AudioBus> m_temporaryStereoBus;
-
-    unsigned m_connectionCount;
-
-    // Graph locking.
-    Mutex m_contextGraphMutex;
-    volatile ThreadIdentifier m_audioThread;
-    volatile ThreadIdentifier m_graphOwnerThread; // if the lock is held then this is the thread which owns it, otherwise == UndefinedThreadIdentifier
-    
-    // Deferred de-referencing.
-    struct RefInfo {
-        RefInfo(AudioNode* node, AudioNode::RefType refType)
-            : m_node(node)
-            , m_refType(refType)
-        {
-        }
-        AudioNode* m_node;
-        AudioNode::RefType m_refType;
-    };    
-
-    // Only accessed in the audio thread.
-    Vector<RefInfo> m_deferredFinishDerefList;
-    
-    // HRTF Database loader
-    RefPtr<HRTFDatabaseLoader> m_hrtfDatabaseLoader;
-};
-
-} // WebCore
-
-#endif // AudioContext_h
diff --git a/WebCore/webaudio/AudioContext.idl b/WebCore/webaudio/AudioContext.idl
deleted file mode 100644
index 9f0f49c..0000000
--- a/WebCore/webaudio/AudioContext.idl
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * 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.
- */
-
-module webaudio {
-    interface [
-        Conditional=WEB_AUDIO,
-        CanBeConstructed,
-        CustomConstructFunction,
-        V8CustomConstructor
-    ] AudioContext {
-        // All rendered audio ultimately connects to destination, which represents the audio hardware.
-        readonly attribute AudioDestinationNode destination;
-
-        // All scheduled times are relative to this time in seconds.
-        readonly attribute float currentTime;
-
-        // All AudioNodes in the context run at this sample-rate (in sample-frames per second).
-        readonly attribute float sampleRate;
-
-        // All panning is relative to this listener.
-        readonly attribute AudioListener listener;
-
-        AudioBuffer createBuffer(in unsigned long numberOfChannels, in unsigned long numberOfFrames, in float sampleRate);
-        AudioBuffer createBuffer(in ArrayBuffer buffer, in boolean mixToMono);
-
-        // Source
-        AudioBufferSourceNode createBufferSource();
-
-        // Processing nodes
-        AudioGainNode createGainNode();
-        DelayNode createDelayNode();
-        LowPass2FilterNode createLowPass2Filter();
-        HighPass2FilterNode createHighPass2Filter();
-        AudioPannerNode createPanner();
-        ConvolverNode createConvolver();
-        RealtimeAnalyserNode createAnalyser();
-        JavaScriptAudioNode createJavaScriptNode(in unsigned long bufferSize);
-
-        // Channel splitting and merging
-        AudioChannelSplitter createChannelSplitter();
-        AudioChannelMerger createChannelMerger();
-    };
-}
diff --git a/WebCore/webaudio/AudioDestinationNode.cpp b/WebCore/webaudio/AudioDestinationNode.cpp
deleted file mode 100644
index d2f4928..0000000
--- a/WebCore/webaudio/AudioDestinationNode.cpp
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * 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 "AudioDestinationNode.h"
-
-#include "AudioBus.h"
-#include "AudioContext.h"
-#include "AudioNodeInput.h"
-#include "AudioNodeOutput.h"
-#include <wtf/Threading.h>
-
-namespace WebCore {
-
-AudioDestinationNode::AudioDestinationNode(AudioContext* context)
-    : AudioNode(context, AudioDestination::hardwareSampleRate())
-    , m_currentTime(0.0)
-{
-    addInput(adoptPtr(new AudioNodeInput(this)));
-    
-    setType(NodeTypeDestination);
-    
-    initialize();
-}
-
-AudioDestinationNode::~AudioDestinationNode()
-{
-    uninitialize();
-}
-
-void AudioDestinationNode::initialize()
-{
-    if (isInitialized())
-        return;
-
-    double hardwareSampleRate = AudioDestination::hardwareSampleRate();
-#ifndef NDEBUG    
-    fprintf(stderr, ">>>> hardwareSampleRate = %f\n", hardwareSampleRate);
-#endif
-    
-    m_destination = AudioDestination::create(*this, hardwareSampleRate);
-    m_destination->start();
-    
-    AudioNode::initialize();
-}
-
-void AudioDestinationNode::uninitialize()
-{
-    if (!isInitialized())
-        return;
-
-    m_destination->stop();
-
-    AudioNode::uninitialize();
-}
-
-// The audio hardware calls us back here to gets its input stream.
-void AudioDestinationNode::provideInput(AudioBus* destinationBus, size_t numberOfFrames)
-{
-    context()->setAudioThread(currentThread());
-    
-    if (!context()->isRunnable()) {
-        destinationBus->zero();
-        return;
-    }
-
-    // Let the context take care of any business at the start of each render quantum.
-    context()->handlePreRenderTasks();
-
-    // This will cause the node(s) connected to us to process, which in turn will pull on their input(s),
-    // all the way backwards through the rendering graph.
-    AudioBus* renderedBus = input(0)->pull(destinationBus, numberOfFrames);
-    
-    if (!renderedBus)
-        destinationBus->zero();
-    else if (renderedBus != destinationBus) {
-        // in-place processing was not possible - so copy
-        destinationBus->copyFrom(*renderedBus);
-    }
-
-    // Let the context take care of any business at the end of each render quantum.
-    context()->handlePostRenderTasks();
-    
-    // Advance current time.
-    m_currentTime += numberOfFrames / sampleRate();
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioDestinationNode.h b/WebCore/webaudio/AudioDestinationNode.h
deleted file mode 100644
index 4c21bb8..0000000
--- a/WebCore/webaudio/AudioDestinationNode.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef AudioDestinationNode_h
-#define AudioDestinationNode_h
-
-#include "AudioDestination.h"
-#include "AudioNode.h"
-#include "AudioSourceProvider.h"
-#include <wtf/OwnPtr.h>
-#include <wtf/PassRefPtr.h>
-
-namespace WebCore {
-
-class AudioBus;
-class AudioContext;
-    
-class AudioDestinationNode : public AudioNode, public AudioSourceProvider {
-public:
-    static PassRefPtr<AudioDestinationNode> create(AudioContext* context)
-    {
-        return adoptRef(new AudioDestinationNode(context));        
-    }
-
-    virtual ~AudioDestinationNode();
-    
-    // AudioNode   
-    virtual void process(size_t) { }; // we're pulled by hardware so this is never called
-    virtual void reset() { m_currentTime = 0.0; };
-    virtual void initialize();
-    virtual void uninitialize();
-    
-    // The audio hardware calls here periodically to gets its input stream.
-    virtual void provideInput(AudioBus*, size_t numberOfFrames);
-
-    double currentTime() { return m_currentTime; }
-
-    double sampleRate() const { return m_destination->sampleRate(); }
-
-    unsigned numberOfChannels() const { return 2; } // FIXME: update when multi-channel (more than stereo) is supported
-    
-private:
-    AudioDestinationNode(AudioContext*);
-
-    OwnPtr<AudioDestination> m_destination;
-    double m_currentTime;
-};
-
-} // namespace WebCore
-
-#endif // AudioDestinationNode_h
diff --git a/WebCore/webaudio/AudioDestinationNode.idl b/WebCore/webaudio/AudioDestinationNode.idl
deleted file mode 100644
index d7bf09f..0000000
--- a/WebCore/webaudio/AudioDestinationNode.idl
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO,
-        GenerateToJS
-    ] AudioDestinationNode : AudioNode {
-        readonly attribute long numberOfChannels;
-    };
-}
diff --git a/WebCore/webaudio/AudioGain.h b/WebCore/webaudio/AudioGain.h
deleted file mode 100644
index eb3c52d..0000000
--- a/WebCore/webaudio/AudioGain.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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.
- */
-
-#ifndef AudioGain_h
-#define AudioGain_h
-
-#include "AudioParam.h"
-#include <wtf/PassRefPtr.h>
-
-namespace WebCore {
-
-class AudioGain : public AudioParam {
-public:   
-    static PassRefPtr<AudioGain> create(const char* name, double defaultValue, double minValue, double maxValue)
-    {
-        return adoptRef(new AudioGain(name, defaultValue, minValue, maxValue));
-    }
-
-private:
-    AudioGain(const char* name, double defaultValue, double minValue, double maxValue)
-        : AudioParam(name, defaultValue, minValue, maxValue)
-    {
-    }
-};
-
-} // namespace WebCore
-
-#endif // AudioParam_h
diff --git a/WebCore/webaudio/AudioGain.idl b/WebCore/webaudio/AudioGain.idl
deleted file mode 100644
index ead7c9a..0000000
--- a/WebCore/webaudio/AudioGain.idl
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO,
-        GenerateToJS
-    ] AudioGain : AudioParam {
-    };
-}
diff --git a/WebCore/webaudio/AudioGainNode.cpp b/WebCore/webaudio/AudioGainNode.cpp
deleted file mode 100644
index 5b9af07..0000000
--- a/WebCore/webaudio/AudioGainNode.cpp
+++ /dev/null
@@ -1,113 +0,0 @@
-/*
- * 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 "AudioGainNode.h"
-
-#include "AudioBus.h"
-#include "AudioNodeInput.h"
-#include "AudioNodeOutput.h"
-
-namespace WebCore {
-
-AudioGainNode::AudioGainNode(AudioContext* context, double sampleRate)
-    : AudioNode(context, sampleRate)
-    , m_lastGain(1.0)
-{
-    m_gain = AudioGain::create("gain", 1.0, 0.0, 1.0);
-
-    addInput(adoptPtr(new AudioNodeInput(this)));
-    addOutput(adoptPtr(new AudioNodeOutput(this, 1)));
-    
-    setType(NodeTypeGain);
-    
-    initialize();
-}
-
-void AudioGainNode::process(size_t /*framesToProcess*/)
-{
-    // FIXME: there is a nice optimization to avoid processing here, and let the gain change
-    // happen in the summing junction input of the AudioNode we're connected to.
-    // Then we can avoid all of the following:
-
-    AudioBus* outputBus = output(0)->bus();
-    ASSERT(outputBus);
-
-    // The realtime thread can't block on this lock, so we call tryLock() instead.
-    if (m_processLock.tryLock()) {
-        if (!isInitialized() || !input(0)->isConnected())
-            outputBus->zero();
-        else {
-            AudioBus* inputBus = input(0)->bus();
-
-            // Apply the gain with de-zippering into the output bus.
-            outputBus->copyWithGainFrom(*inputBus, &m_lastGain, gain()->value());
-        }
-
-        m_processLock.unlock();
-    } else {
-        // Too bad - the tryLock() failed.  We must be in the middle of re-connecting and were already outputting silence anyway...
-        outputBus->zero();
-    }
-}
-
-void AudioGainNode::reset()
-{
-    // Snap directly to desired gain.
-    m_lastGain = gain()->value();
-}
-
-// FIXME: this can go away when we do mixing with gain directly in summing junction of AudioNodeInput
-//
-// As soon as we know the channel count of our input, we can lazily initialize.
-// Sometimes this may be called more than once with different channel counts, in which case we must safely
-// uninitialize and then re-initialize with the new channel count.
-void AudioGainNode::checkNumberOfChannelsForInput(AudioNodeInput* input)
-{
-    ASSERT(input && input == this->input(0));
-    if (input != this->input(0))
-        return;
-        
-    unsigned numberOfChannels = input->numberOfChannels();    
-
-    if (isInitialized() && numberOfChannels != output(0)->numberOfChannels()) {
-        // We're already initialized but the channel count has changed.
-        // We need to be careful since we may be actively processing right now, so synchronize with process().
-        MutexLocker locker(m_processLock);
-        uninitialize();
-    }
-
-    if (!isInitialized()) {
-        // This will propagate the channel count to any nodes connected further downstream in the graph.
-        output(0)->setNumberOfChannels(numberOfChannels);
-        initialize();
-    }
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioGainNode.h b/WebCore/webaudio/AudioGainNode.h
deleted file mode 100644
index 3710472..0000000
--- a/WebCore/webaudio/AudioGainNode.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef AudioGainNode_h
-#define AudioGainNode_h
-
-#include "AudioGain.h"
-#include "AudioNode.h"
-#include <wtf/PassRefPtr.h>
-#include <wtf/Threading.h>
-
-namespace WebCore {
-
-class AudioContext;
-    
-// AudioGainNode is an AudioNode with one input and one output which applies a gain (volume) change to the audio signal.
-// De-zippering (smoothing) is applied when the gain value is changed dynamically.
-
-class AudioGainNode : public AudioNode {
-public:
-    static PassRefPtr<AudioGainNode> create(AudioContext* context, double sampleRate)
-    {
-        return adoptRef(new AudioGainNode(context, sampleRate));      
-    }
-    
-    // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
-
-    // Called in the main thread when the number of channels for the input may have changed.
-    virtual void checkNumberOfChannelsForInput(AudioNodeInput*);
-
-    // JavaScript interface
-    AudioGain* gain() { return m_gain.get(); }                                   
-    
-private:
-    AudioGainNode(AudioContext*, double sampleRate);
-
-    double m_lastGain; // for de-zippering
-    RefPtr<AudioGain> m_gain;
-
-    // This synchronizes live channel count changes which require an uninitialization / re-initialization.
-    // FIXME: this can go away when we implement optimization for mixing with gain directly in summing junction of AudioNodeInput.
-    mutable Mutex m_processLock;
-};
-
-} // namespace WebCore
-
-#endif // AudioGainNode_h
diff --git a/WebCore/webaudio/AudioGainNode.idl b/WebCore/webaudio/AudioGainNode.idl
deleted file mode 100644
index 3d4f40f..0000000
--- a/WebCore/webaudio/AudioGainNode.idl
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO,
-        GenerateToJS
-    ] AudioGainNode : AudioNode {
-        // FIXME: eventually it will be interesting to remove the readonly restriction, but need to properly deal with thread safety here.
-        readonly attribute AudioGain gain;
-    };
-}
diff --git a/WebCore/webaudio/AudioListener.cpp b/WebCore/webaudio/AudioListener.cpp
deleted file mode 100644
index 44fb02c..0000000
--- a/WebCore/webaudio/AudioListener.cpp
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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 "AudioListener.h"
-
-#include "AudioBus.h"
-
-namespace WebCore {
-
-AudioListener::AudioListener()
-    : m_position(0, 0, 0)
-    , m_orientation(0.0, 0.0, -1.0)
-    , m_upVector(0.0, 1.0, 0.0)
-    , m_velocity(0, 0, 0)
-    , m_dopplerFactor(1.0)
-    , m_speedOfSound(343.3)
-{
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioListener.h b/WebCore/webaudio/AudioListener.h
deleted file mode 100644
index 5281a89..0000000
--- a/WebCore/webaudio/AudioListener.h
+++ /dev/null
@@ -1,94 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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.
- */
-
-#ifndef AudioListener_h
-#define AudioListener_h
-
-#include "FloatPoint3D.h"
-#include <wtf/PassRefPtr.h>
-#include <wtf/RefCounted.h>
-
-namespace WebCore {
-
-// AudioListener maintains the state of the listener in the audio scene as defined in the OpenAL specification.
-
-class AudioListener : public RefCounted<AudioListener> {
-public:
-    static PassRefPtr<AudioListener> create()
-    {
-        return adoptRef(new AudioListener());
-    }
-
-    // Position
-    void setPosition(double x, double y, double z) { setPosition(FloatPoint3D(x, y, z)); }
-    void setPosition(const FloatPoint3D &position) { m_position = position; }
-    const FloatPoint3D& position() const { return m_position; }
-
-    // Orientation
-    void setOrientation(double x, double y, double z, double upX, double upY, double upZ)
-    {
-        setOrientation(FloatPoint3D(x, y, z));
-        setUpVector(FloatPoint3D(upX, upY, upZ));
-    }
-    void setOrientation(const FloatPoint3D &orientation) { m_orientation = orientation; }
-    const FloatPoint3D& orientation() const { return m_orientation; }
-
-    // Up-vector
-    void setUpVector(const FloatPoint3D &upVector) { m_upVector = upVector; }
-    const FloatPoint3D& upVector() const { return m_upVector; }
-
-    // Velocity
-    void setVelocity(double x, double y, double z) { setVelocity(FloatPoint3D(x, y, z)); }
-    void setVelocity(const FloatPoint3D &velocity) { m_velocity = velocity; }
-    const FloatPoint3D& velocity() const { return m_velocity; }
-
-    // Doppler factor
-    void setDopplerFactor(double dopplerFactor) { m_dopplerFactor = dopplerFactor; }
-    double dopplerFactor() const { return m_dopplerFactor; }
-
-    // Speed of sound
-    void setSpeedOfSound(double speedOfSound) { m_speedOfSound = speedOfSound; }
-    double speedOfSound() const { return m_speedOfSound; }
-
-private:
-    AudioListener();
-
-    // Position / Orientation
-    FloatPoint3D m_position;
-    FloatPoint3D m_orientation;
-    FloatPoint3D m_upVector;
-
-    FloatPoint3D m_velocity;
-
-    double m_dopplerFactor;
-    double m_speedOfSound;
-};
-
-} // WebCore
-
-#endif // AudioListener_h
diff --git a/WebCore/webaudio/AudioListener.idl b/WebCore/webaudio/AudioListener.idl
deleted file mode 100644
index cf6d8cf..0000000
--- a/WebCore/webaudio/AudioListener.idl
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO
-    ] AudioListener {
-        attribute float dopplerFactor;  // same as OpenAL (default 1.0)
-        attribute float speedOfSound;   // in meters / second (default 343.3)
-
-        void setPosition(in float x, in float y, in float z);
-        void setOrientation(in float x, in float y, in float z, in float xUp, in float yUp, in float zUp);
-        void setVelocity(in float x, in float y, in float z);
-    };
-}
diff --git a/WebCore/webaudio/AudioNode.cpp b/WebCore/webaudio/AudioNode.cpp
deleted file mode 100644
index 18ddd3b..0000000
--- a/WebCore/webaudio/AudioNode.cpp
+++ /dev/null
@@ -1,317 +0,0 @@
-/*
- * 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 "AudioNode.h"
-
-#include "AudioContext.h"
-#include "AudioNodeInput.h"
-#include "AudioNodeOutput.h"
-#include <wtf/Atomics.h>
-
-namespace WebCore {
-
-AudioNode::AudioNode(AudioContext* context, double sampleRate)
-    : m_isInitialized(false)
-    , m_type(NodeTypeUnknown)
-    , m_context(context)
-    , m_sampleRate(sampleRate)
-    , m_lastProcessingTime(-1.0)
-    , m_normalRefCount(1) // start out with normal refCount == 1 (like WTF::RefCounted class)
-    , m_connectionRefCount(0)
-    , m_disabledRefCount(0)
-    , m_isMarkedForDeletion(false)
-    , m_isDisabled(false)
-{
-#if DEBUG_AUDIONODE_REFERENCES
-    if (!s_isNodeCountInitialized) {
-        s_isNodeCountInitialized = true;
-        atexit(AudioNode::printNodeCounts);
-    }
-#endif
-}
-
-AudioNode::~AudioNode()
-{
-#if DEBUG_AUDIONODE_REFERENCES
-    --s_nodeCount[type()];
-    printf("%p: %d: AudioNode::~AudioNode() %d %d %d\n", this, type(), m_normalRefCount, m_connectionRefCount, m_disabledRefCount);
-#endif
-}
-
-void AudioNode::initialize()
-{
-    m_isInitialized = true;
-}
-
-void AudioNode::uninitialize()
-{
-    m_isInitialized = false;
-}
-
-void AudioNode::setType(NodeType type)
-{
-    m_type = type;
-
-#if DEBUG_AUDIONODE_REFERENCES
-    ++s_nodeCount[type];
-#endif
-}
-
-void AudioNode::lazyInitialize()
-{
-    if (!isInitialized())
-        initialize();
-}
-
-void AudioNode::addInput(PassOwnPtr<AudioNodeInput> input)
-{
-    m_inputs.append(input);
-}
-
-void AudioNode::addOutput(PassOwnPtr<AudioNodeOutput> output)
-{
-    m_outputs.append(output);
-}
-
-AudioNodeInput* AudioNode::input(unsigned i)
-{
-    return m_inputs[i].get();
-}
-
-AudioNodeOutput* AudioNode::output(unsigned i)
-{
-    return m_outputs[i].get();
-}
-
-bool AudioNode::connect(AudioNode* destination, unsigned outputIndex, unsigned inputIndex)
-{
-    ASSERT(isMainThread()); 
-    AudioContext::AutoLocker locker(context());
-    
-    // Sanity check input and output indices.
-    if (outputIndex >= numberOfOutputs())
-        return false;
-    if (destination && inputIndex >= destination->numberOfInputs())
-        return false;
-
-    AudioNodeOutput* output = this->output(outputIndex);
-    if (!destination) {
-        // Disconnect output from any inputs it may be currently connected to.
-        output->disconnectAllInputs();
-        return true;
-    }
-
-    AudioNodeInput* input = destination->input(inputIndex);
-    input->connect(output);
-
-    // Let context know that a connection has been made.
-    context()->incrementConnectionCount();
-
-    return true;
-}
-
-bool AudioNode::disconnect(unsigned outputIndex)
-{
-    ASSERT(isMainThread());
-    AudioContext::AutoLocker locker(context());
-    
-    return connect(0, outputIndex);
-}
-
-void AudioNode::processIfNecessary(size_t framesToProcess)
-{
-    ASSERT(context()->isAudioThread());
-    
-    if (!isInitialized())
-        return;
-
-    // Ensure that we only process once per rendering quantum.
-    // This handles the "fanout" problem where an output is connected to multiple inputs.
-    // The first time we're called during this time slice we process, but after that we don't want to re-process,
-    // instead our output(s) will already have the results cached in their bus;
-    double currentTime = context()->currentTime();
-    if (m_lastProcessingTime != currentTime) {
-        m_lastProcessingTime = currentTime; // important to first update this time because of feedback loops in the rendering graph
-        pullInputs(framesToProcess);
-        process(framesToProcess);
-    }
-}
-
-void AudioNode::pullInputs(size_t framesToProcess)
-{
-    ASSERT(context()->isAudioThread());
-    
-    // Process all of the AudioNodes connected to our inputs.
-    for (unsigned i = 0; i < m_inputs.size(); ++i)
-        input(i)->pull(0, framesToProcess);
-}
-
-void AudioNode::ref(RefType refType)
-{
-    switch (refType) {
-    case RefTypeNormal:
-        atomicIncrement(&m_normalRefCount);
-        break;
-    case RefTypeConnection:
-        atomicIncrement(&m_connectionRefCount);
-        break;
-    case RefTypeDisabled:
-        atomicIncrement(&m_disabledRefCount);
-        break;
-    default:
-        ASSERT_NOT_REACHED();
-    }
-
-#if DEBUG_AUDIONODE_REFERENCES
-    printf("%p: %d: AudioNode::ref(%d) %d %d %d\n", this, type(), refType, m_normalRefCount, m_connectionRefCount, m_disabledRefCount);
-#endif
-
-    if (m_connectionRefCount == 1 && refType == RefTypeConnection) {
-        // FIXME: implement wake-up - this is an advanced feature and is not necessary in a simple implementation.
-        // We should not be "actively" connected to anything, but now we're "waking up"
-        // For example, a note which has finished playing, but is now being played again.
-        // Note that if this is considered a worthwhile feature to add, then an evaluation of the locking considerations must be made.
-    }
-}
-
-void AudioNode::deref(RefType refType)
-{
-    // The actually work for deref happens completely within the audio context's graph lock.
-    // In the case of the audio thread, we must use a tryLock to avoid glitches.
-    bool hasLock = false;
-    bool mustReleaseLock = false;
-    
-    if (context()->isAudioThread()) {
-        // Real-time audio thread must not contend lock (to avoid glitches).
-        hasLock = context()->tryLock(mustReleaseLock);
-    } else {
-        context()->lock(mustReleaseLock);
-        hasLock = true;
-    }
-    
-    if (hasLock) {
-        // This is where the real deref work happens.
-        finishDeref(refType);
-
-        if (mustReleaseLock)
-            context()->unlock();
-    } else {
-        // We were unable to get the lock, so put this in a list to finish up later.
-        ASSERT(context()->isAudioThread());
-        context()->addDeferredFinishDeref(this, refType);
-    }
-
-    // Once AudioContext::uninitialize() is called there's no more chances for deleteMarkedNodes() to get called, so we call here.
-    // We can't call in AudioContext::~AudioContext() since it will never be called as long as any AudioNode is alive
-    // because AudioNodes keep a reference to the context.
-    if (context()->isAudioThreadFinished())
-        context()->deleteMarkedNodes();
-}
-
-void AudioNode::finishDeref(RefType refType)
-{
-    ASSERT(context()->isGraphOwner());
-    
-    switch (refType) {
-    case RefTypeNormal:
-        ASSERT(m_normalRefCount > 0);
-        atomicDecrement(&m_normalRefCount);
-        break;
-    case RefTypeConnection:
-        ASSERT(m_connectionRefCount > 0);
-        atomicDecrement(&m_connectionRefCount);
-        break;
-    case RefTypeDisabled:
-        ASSERT(m_disabledRefCount > 0);
-        atomicDecrement(&m_disabledRefCount);
-        break;
-    default:
-        ASSERT_NOT_REACHED();
-    }
-    
-#if DEBUG_AUDIONODE_REFERENCES
-    printf("%p: %d: AudioNode::deref(%d) %d %d %d\n", this, type(), refType, m_normalRefCount, m_connectionRefCount, m_disabledRefCount);
-#endif
-
-    if (!m_connectionRefCount) {
-        if (!m_normalRefCount && !m_disabledRefCount) {
-            if (!m_isMarkedForDeletion) {
-                // All references are gone - we need to go away.
-                for (unsigned i = 0; i < m_outputs.size(); ++i)
-                    output(i)->disconnectAllInputs(); // this will deref() nodes we're connected to...
-
-                // Mark for deletion at end of each render quantum or when context shuts down.
-                context()->markForDeletion(this);
-                m_isMarkedForDeletion = true;
-            }
-        } else if (refType == RefTypeConnection) {
-            if (!m_isDisabled) {
-                // Still may have JavaScript references, but no more "active" connection references, so put all of our outputs in a "dormant" disabled state.
-                // Garbage collection may take a very long time after this time, so the "dormant" disabled nodes should not bog down the rendering...
-
-                // As far as JavaScript is concerned, our outputs must still appear to be connected.
-                // But internally our outputs should be disabled from the inputs they're connected to.
-                // disable() can recursively deref connections (and call disable()) down a whole chain of connected nodes.
-
-                // FIXME: we special case the convolver and delay since they have a significant tail-time and shouldn't be disconnected simply
-                // because they no longer have any input connections.  This needs to be handled more generally where AudioNodes have
-                // a tailTime attribute.  Then the AudioNode only needs to remain "active" for tailTime seconds after there are no
-                // longer any active connections.
-                if (type() != NodeTypeConvolver && type() != NodeTypeDelay) {
-                    m_isDisabled = true;
-                    for (unsigned i = 0; i < m_outputs.size(); ++i)
-                        output(i)->disable();
-                }
-            }
-        }
-    }
-}
-
-#if DEBUG_AUDIONODE_REFERENCES
-
-bool AudioNode::s_isNodeCountInitialized = false;
-int AudioNode::s_nodeCount[NodeTypeEnd];
-
-void AudioNode::printNodeCounts()
-{
-    printf("\n\n");
-    printf("===========================\n");
-    printf("AudioNode: reference counts\n");
-    printf("===========================\n");
-
-    for (unsigned i = 0; i < NodeTypeEnd; ++i)
-        printf("%d: %d\n", i, s_nodeCount[i]);
-
-    printf("===========================\n\n\n");
-}
-
-#endif // DEBUG_AUDIONODE_REFERENCES
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioNode.h b/WebCore/webaudio/AudioNode.h
deleted file mode 100644
index 069407d..0000000
--- a/WebCore/webaudio/AudioNode.h
+++ /dev/null
@@ -1,171 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef AudioNode_h
-#define AudioNode_h
-
-#include <wtf/OwnPtr.h>
-#include <wtf/PassOwnPtr.h>
-#include <wtf/RefPtr.h>
-#include <wtf/Vector.h>
-
-#define DEBUG_AUDIONODE_REFERENCES 0
-
-namespace WebCore {
-
-class AudioContext;
-class AudioNodeInput;
-class AudioNodeOutput;
-
-// An AudioNode is the basic building block for handling audio within an AudioContext.
-// It may be an audio source, an intermediate processing module, or an audio destination.
-// Each AudioNode can have inputs and/or outputs. An AudioSourceNode has no inputs and a single output.
-// An AudioDestinationNode has one input and no outputs and represents the final destination to the audio hardware.
-// Most processing nodes such as filters will have one input and one output, although multiple inputs and outputs are possible.
-
-class AudioNode {
-public:
-    enum { ProcessingSizeInFrames = 128 };
-
-    AudioNode(AudioContext*, double sampleRate);
-    virtual ~AudioNode();
-
-    AudioContext* context() { return m_context.get(); }
-
-    enum NodeType {
-        NodeTypeUnknown,
-        NodeTypeDestination,
-        NodeTypeAudioBufferSource,
-        NodeTypeJavaScript,
-        NodeTypeLowPass2Filter,
-        NodeTypeHighPass2Filter,
-        NodeTypePanner,
-        NodeTypeConvolver,
-        NodeTypeDelay,
-        NodeTypeGain,
-        NodeTypeChannelSplitter,
-        NodeTypeChannelMerger,
-        NodeTypeAnalyser,
-        NodeTypeEnd
-    };
-
-    NodeType type() const { return m_type; }
-    void setType(NodeType);
-
-    // We handle our own ref-counting because of the threading issues and subtle nature of
-    // how AudioNodes can continue processing (playing one-shot sound) after there are no more
-    // JavaScript references to the object.
-    enum RefType { RefTypeNormal, RefTypeConnection, RefTypeDisabled };
-
-    // Can be called from main thread or context's audio thread.
-    void ref(RefType refType = RefTypeNormal);
-    void deref(RefType refType = RefTypeNormal);
-
-    // Can be called from main thread or context's audio thread.  It must be called while the context's graph lock is held.
-    void finishDeref(RefType refType);
-
-    // The AudioNodeInput(s) (if any) will already have their input data available when process() is called.
-    // Subclasses will take this input data and put the results in the AudioBus(s) of its AudioNodeOutput(s) (if any).
-    // Called from context's audio thread.
-    virtual void process(size_t framesToProcess) = 0;
-
-    // Resets DSP processing state (clears delay lines, filter memory, etc.)
-    // Called from context's audio thread.
-    virtual void reset() = 0;
-
-    // No significant resources should be allocated until initialize() is called.
-    // Processing may not occur until a node is initialized.
-    virtual void initialize();
-    virtual void uninitialize();
-
-    bool isInitialized() const { return m_isInitialized; }
-    void lazyInitialize();
-
-    unsigned numberOfInputs() const { return m_inputs.size(); }
-    unsigned numberOfOutputs() const { return m_outputs.size(); }
-
-    AudioNodeInput* input(unsigned);
-    AudioNodeOutput* output(unsigned);
-
-    // connect() / disconnect() return true on success.
-    // Called from main thread by corresponding JavaScript methods.
-    bool connect(AudioNode* destination, unsigned outputIndex = 0, unsigned inputIndex = 0);
-    bool disconnect(unsigned outputIndex = 0);
-
-    double sampleRate() const { return m_sampleRate; }
-
-    // processIfNecessary() is called by our output(s) when the rendering graph needs this AudioNode to process.
-    // This method ensures that the AudioNode will only process once per rendering time quantum even if it's called repeatedly.
-    // This handles the case of "fanout" where an output is connected to multiple AudioNode inputs.
-    // Called from context's audio thread.
-    void processIfNecessary(size_t framesToProcess);
-
-    // Called when a new connection has been made to one of our inputs or the connection number of channels has changed.
-    // This potentially gives us enough information to perform a lazy initialization or, if necessary, a re-initialization.
-    // Called from main thread.
-    virtual void checkNumberOfChannelsForInput(AudioNodeInput*) { }
-
-#if DEBUG_AUDIONODE_REFERENCES
-    static void printNodeCounts();
-#endif
-
-    bool isMarkedForDeletion() const { return m_isMarkedForDeletion; }
-
-protected:
-    // Inputs and outputs must be created before the AudioNode is initialized.
-    void addInput(PassOwnPtr<AudioNodeInput>);
-    void addOutput(PassOwnPtr<AudioNodeOutput>);
-    
-    // Called by processIfNecessary() to cause all parts of the rendering graph connected to us to process.
-    // Each rendering quantum, the audio data for each of the AudioNode's inputs will be available after this method is called.
-    // Called from context's audio thread.
-    virtual void pullInputs(size_t framesToProcess);
-
-private:
-    volatile bool m_isInitialized;
-    NodeType m_type;
-    RefPtr<AudioContext> m_context;
-    double m_sampleRate;
-    Vector<OwnPtr<AudioNodeInput> > m_inputs;
-    Vector<OwnPtr<AudioNodeOutput> > m_outputs;
-
-    double m_lastProcessingTime;
-
-    // Ref-counting
-    volatile int m_normalRefCount;
-    volatile int m_connectionRefCount;
-    volatile int m_disabledRefCount;
-    
-    bool m_isMarkedForDeletion;
-    bool m_isDisabled;
-    
-#if DEBUG_AUDIONODE_REFERENCES
-    static bool s_isNodeCountInitialized;
-    static int s_nodeCount[NodeTypeEnd];
-#endif
-};
-
-} // namespace WebCore
-
-#endif // AudioNode_h
diff --git a/WebCore/webaudio/AudioNode.idl b/WebCore/webaudio/AudioNode.idl
deleted file mode 100644
index 8d903e2..0000000
--- a/WebCore/webaudio/AudioNode.idl
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO
-    ] AudioNode {
-        readonly attribute AudioContext context;
-        readonly attribute unsigned long numberOfInputs;
-        readonly attribute unsigned long numberOfOutputs;
-
-        [Custom] void connect(in AudioNode destination, in unsigned long output, in unsigned long input)
-            raises(DOMException);
-
-        [Custom] void disconnect(in unsigned long output)
-            raises(DOMException);
-    };
-}
diff --git a/WebCore/webaudio/AudioNodeInput.cpp b/WebCore/webaudio/AudioNodeInput.cpp
deleted file mode 100644
index 9fd1852..0000000
--- a/WebCore/webaudio/AudioNodeInput.cpp
+++ /dev/null
@@ -1,270 +0,0 @@
-/*
- * 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 "AudioNodeInput.h"
-
-#include "AudioContext.h"
-#include "AudioNode.h"
-#include "AudioNodeOutput.h"
-#include <algorithm>
-
-using namespace std;
- 
-namespace WebCore {
-
-AudioNodeInput::AudioNodeInput(AudioNode* node)
-    : m_node(node)
-    , m_renderingStateNeedUpdating(false)
-{
-    m_monoSummingBus = adoptPtr(new AudioBus(1, AudioNode::ProcessingSizeInFrames));
-    m_stereoSummingBus = adoptPtr(new AudioBus(2, AudioNode::ProcessingSizeInFrames));
-}
-
-void AudioNodeInput::connect(AudioNodeOutput* output)
-{
-    ASSERT(context()->isGraphOwner());
-    
-    ASSERT(output && node());
-    if (!output || !node())
-        return;
-
-    // Check if we're already connected to this output.
-    if (m_outputs.contains(output))
-        return;
-        
-    output->addInput(this);
-    m_outputs.add(output);
-    changedOutputs();
-
-    // Sombody has just connected to us, so count it as a reference.
-    node()->ref(AudioNode::RefTypeConnection);
-}
-
-void AudioNodeInput::disconnect(AudioNodeOutput* output)
-{
-    ASSERT(context()->isGraphOwner());
-
-    ASSERT(output && node());
-    if (!output || !node())
-        return;
-
-    // First try to disconnect from "active" connections.
-    if (m_outputs.contains(output)) {
-        m_outputs.remove(output);
-        changedOutputs();
-        output->removeInput(this);
-        node()->deref(AudioNode::RefTypeConnection); // Note: it's important to return immediately after all deref() calls since the node may be deleted.
-        return;
-    }
-    
-    // Otherwise, try to disconnect from disabled connections.
-    if (m_disabledOutputs.contains(output)) {
-        m_disabledOutputs.remove(output);
-        output->removeInput(this);
-        node()->deref(AudioNode::RefTypeDisabled); // Note: it's important to return immediately after all deref() calls since the node may be deleted.
-        return;
-    }
-
-    ASSERT_NOT_REACHED();
-}
-
-void AudioNodeInput::disable(AudioNodeOutput* output)
-{
-    ASSERT(context()->isGraphOwner());
-
-    ASSERT(output && node());
-    if (!output || !node())
-        return;
-
-    ASSERT(m_outputs.contains(output));
-    
-    m_disabledOutputs.add(output);
-    m_outputs.remove(output);
-    changedOutputs();
-
-    node()->ref(AudioNode::RefTypeDisabled);
-    node()->deref(AudioNode::RefTypeConnection); // Note: it's important to return immediately after all deref() calls since the node may be deleted.
-}
-
-void AudioNodeInput::enable(AudioNodeOutput* output)
-{
-    ASSERT(context()->isGraphOwner());
-
-    ASSERT(output && node());
-    if (!output || !node())
-        return;
-
-    ASSERT(m_disabledOutputs.contains(output));
-
-    // Move output from disabled list to active list.
-    m_outputs.add(output);
-    m_disabledOutputs.remove(output);
-    changedOutputs();
-
-    node()->ref(AudioNode::RefTypeConnection);
-    node()->deref(AudioNode::RefTypeDisabled); // Note: it's important to return immediately after all deref() calls since the node may be deleted.
-}
-
-void AudioNodeInput::changedOutputs()
-{
-    ASSERT(context()->isGraphOwner());
-    if (!m_renderingStateNeedUpdating && !node()->isMarkedForDeletion()) {    
-        context()->markAudioNodeInputDirty(this);
-        m_renderingStateNeedUpdating = true;
-    }
-}
-
-void AudioNodeInput::updateRenderingState()
-{
-    ASSERT(context()->isAudioThread() && context()->isGraphOwner());
-    
-    if (m_renderingStateNeedUpdating && !node()->isMarkedForDeletion()) {
-        // Copy from m_outputs to m_renderingOutputs.
-        m_renderingOutputs.resize(m_outputs.size());
-        unsigned j = 0;
-        for (HashSet<AudioNodeOutput*>::iterator i = m_outputs.begin(); i != m_outputs.end(); ++i, ++j) {
-            AudioNodeOutput* output = *i;
-            m_renderingOutputs[j] = output;
-            output->updateRenderingState();
-        }
-
-        node()->checkNumberOfChannelsForInput(this);
-        
-        m_renderingStateNeedUpdating = false;
-    }
-}
-
-unsigned AudioNodeInput::numberOfChannels() const
-{
-    // Find the number of channels of the connection with the largest number of channels.
-    unsigned maxChannels = 1; // one channel is the minimum allowed
-
-    for (HashSet<AudioNodeOutput*>::iterator i = m_outputs.begin(); i != m_outputs.end(); ++i) {
-        AudioNodeOutput* output = *i;
-        maxChannels = max(maxChannels, output->bus()->numberOfChannels());
-    }
-    
-    return maxChannels;
-}
-
-unsigned AudioNodeInput::numberOfRenderingChannels()
-{
-    ASSERT(context()->isAudioThread());
-
-    // Find the number of channels of the rendering connection with the largest number of channels.
-    unsigned maxChannels = 1; // one channel is the minimum allowed
-
-    for (unsigned i = 0; i < numberOfRenderingConnections(); ++i)
-        maxChannels = max(maxChannels, renderingOutput(i)->bus()->numberOfChannels());
-    
-    return maxChannels;
-}
-
-AudioBus* AudioNodeInput::bus()
-{
-    ASSERT(context()->isAudioThread());
-
-    // Handle single connection specially to allow for in-place processing.
-    if (numberOfRenderingConnections() == 1)
-        return renderingOutput(0)->bus();
-
-    // Multiple connections case (or no connections).
-    return internalSummingBus();
-}
-
-AudioBus* AudioNodeInput::internalSummingBus()
-{
-    ASSERT(context()->isAudioThread());
-
-    // We must pick a summing bus which is the right size to handle the largest connection.
-    switch (numberOfRenderingChannels()) {
-    case 1:
-        return m_monoSummingBus.get();
-    case 2:
-        return m_stereoSummingBus.get();
-    // FIXME: could implement more than just mono and stereo mixing in the future
-    }
-    
-    ASSERT_NOT_REACHED();
-    return 0;
-}
-
-void AudioNodeInput::sumAllConnections(AudioBus* summingBus, size_t framesToProcess)
-{
-    ASSERT(context()->isAudioThread());
-
-    // We shouldn't be calling this method if there's only one connection, since it's less efficient.
-    ASSERT(numberOfRenderingConnections() > 1);
-
-    ASSERT(summingBus);
-    if (!summingBus)
-        return;
-        
-    summingBus->zero();
-
-    for (unsigned i = 0; i < numberOfRenderingConnections(); ++i) {
-        AudioNodeOutput* output = renderingOutput(i);
-        ASSERT(output);
-
-        // Render audio from this output.
-        AudioBus* connectionBus = output->pull(0, framesToProcess);
-
-        // Sum, with unity-gain.
-        summingBus->sumFrom(*connectionBus);
-    }
-}
-
-AudioBus* AudioNodeInput::pull(AudioBus* inPlaceBus, size_t framesToProcess)
-{
-    ASSERT(context()->isAudioThread());
-
-    // Handle single connection case.
-    if (numberOfRenderingConnections() == 1) {
-        // The output will optimize processing using inPlaceBus if it's able.
-        AudioNodeOutput* output = this->renderingOutput(0);
-        return output->pull(inPlaceBus, framesToProcess);
-    }
-
-    AudioBus* internalSummingBus = this->internalSummingBus();
-
-    if (!numberOfRenderingConnections()) {
-        // At least, generate silence if we're not connected to anything.
-        // FIXME: if we wanted to get fancy, we could propagate a 'silent hint' here to optimize the downstream graph processing.
-        internalSummingBus->zero();
-        return internalSummingBus;
-    }
-
-    // Handle multiple connections case.
-    sumAllConnections(internalSummingBus, framesToProcess);
-    
-    return internalSummingBus;
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioNodeInput.h b/WebCore/webaudio/AudioNodeInput.h
deleted file mode 100644
index 1d90986..0000000
--- a/WebCore/webaudio/AudioNodeInput.h
+++ /dev/null
@@ -1,125 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef AudioNodeInput_h
-#define AudioNodeInput_h
-
-#include "AudioBus.h"
-#include "AudioNode.h"
-#include <wtf/HashSet.h>
-#include <wtf/Vector.h>
-
-namespace WebCore {
-
-class AudioNode;
-class AudioNodeOutput;
-
-// An AudioNodeInput represents an input to an AudioNode and can be connected from one or more AudioNodeOutputs.
-// In the case of multiple connections, the input will act as a unity-gain summing junction, mixing all the outputs.
-// The number of channels of the input's bus is the maximum of the number of channels of all its connections.
-
-class AudioNodeInput {
-public:
-    AudioNodeInput(AudioNode*);
-
-    // Can be called from any thread.
-    AudioNode* node() const { return m_node; }
-    AudioContext* context() { return m_node->context(); }
-
-    // Must be called with the context's graph lock.
-    void connect(AudioNodeOutput*);
-    void disconnect(AudioNodeOutput*);
-
-    // disable() will take the output out of the active connections list and set aside in a disabled list.
-    // enable() will put the output back into the active connections list.
-    // Must be called with the context's graph lock.
-    void enable(AudioNodeOutput*);
-    void disable(AudioNodeOutput*);
-
-    // pull() processes all of the AudioNodes connected to us.
-    // In the case of multiple connections it sums the result into an internal summing bus.
-    // In the single connection case, it allows in-place processing where possible using inPlaceBus.
-    // It returns the bus which it rendered into, returning inPlaceBus if in-place processing was performed.
-    // Called from context's audio thread.
-    AudioBus* pull(AudioBus* inPlaceBus, size_t framesToProcess);
-
-    // bus() contains the rendered audio after pull() has been called for each time quantum.
-    // Called from context's audio thread.
-    AudioBus* bus();
-    
-    // This copies m_outputs to m_renderingOutputs.  Please see comments for these lists below.
-    // This must be called when we own the context's graph lock in the audio thread at the very start or end of the render quantum.
-    void updateRenderingState();
-
-    // Rendering code accesses its version of the current connections here.
-    unsigned numberOfRenderingConnections() const { return m_renderingOutputs.size(); }
-    AudioNodeOutput* renderingOutput(unsigned i) { return m_renderingOutputs[i]; }
-    const AudioNodeOutput* renderingOutput(unsigned i) const { return m_renderingOutputs[i]; }
-    bool isConnected() const { return numberOfRenderingConnections() > 0; }
-
-    // The number of channels of the connection with the largest number of channels.
-    unsigned numberOfChannels() const;        
-    
-private:
-    AudioNode* m_node;
-
-    // m_outputs contains the AudioNodeOutputs representing current connections which are not disabled.
-    // The rendering code should never use this directly, but instead uses m_renderingOutputs.
-    HashSet<AudioNodeOutput*> m_outputs;
-
-    // numberOfConnections() should never be called from the audio rendering thread.
-    // Instead numberOfRenderingConnections() and renderingOutput() should be used.
-    unsigned numberOfConnections() const { return m_outputs.size(); }
-
-    // This must be called whenever we modify m_outputs.
-    void changedOutputs();
-    
-    // m_renderingOutputs is a copy of m_outputs which will never be modified during the graph rendering on the audio thread.
-    // This is the list which is used by the rendering code.
-    // Whenever m_outputs is modified, the context is told so it can later update m_renderingOutputs from m_outputs at a safe time.
-    // Most of the time, m_renderingOutputs is identical to m_outputs.
-    Vector<AudioNodeOutput*> m_renderingOutputs;
-
-    // m_renderingStateNeedUpdating keeps track if m_outputs is modified.
-    bool m_renderingStateNeedUpdating;
-
-    // The number of channels of the rendering connection with the largest number of channels.
-    unsigned numberOfRenderingChannels();
-
-    // m_disabledOutputs contains the AudioNodeOutputs which are disabled (will not be processed) by the audio graph rendering.
-    // But, from JavaScript's perspective, these outputs are still connected to us.
-    // Generally, these represent disabled connections from "notes" which have finished playing but are not yet garbage collected.
-    HashSet<AudioNodeOutput*> m_disabledOutputs;
-
-    // Called from context's audio thread.
-    AudioBus* internalSummingBus();
-    void sumAllConnections(AudioBus* summingBus, size_t framesToProcess);
-
-    OwnPtr<AudioBus> m_monoSummingBus;
-    OwnPtr<AudioBus> m_stereoSummingBus;
-};
-
-} // namespace WebCore
-
-#endif // AudioNodeInput_h
diff --git a/WebCore/webaudio/AudioNodeOutput.cpp b/WebCore/webaudio/AudioNodeOutput.cpp
deleted file mode 100644
index 4c777e6..0000000
--- a/WebCore/webaudio/AudioNodeOutput.cpp
+++ /dev/null
@@ -1,216 +0,0 @@
-/*
- * 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 "AudioNodeOutput.h"
-
-#include "AudioBus.h"
-#include "AudioContext.h"
-#include "AudioNodeInput.h"
-#include <wtf/Threading.h>
-
-namespace WebCore {
-
-AudioNodeOutput::AudioNodeOutput(AudioNode* node, unsigned numberOfChannels)
-    : m_node(node)
-    , m_numberOfChannels(numberOfChannels)
-    , m_desiredNumberOfChannels(numberOfChannels)
-    , m_internalOutputBus(0)
-    , m_actualDestinationBus(0)
-    , m_isEnabled(true)
-    , m_renderingFanOutCount(0)
-{
-    m_monoInternalBus = adoptPtr(new AudioBus(1, AudioNode::ProcessingSizeInFrames));
-    m_stereoInternalBus = adoptPtr(new AudioBus(2, AudioNode::ProcessingSizeInFrames));
-    setInternalBus();
-}
-
-void AudioNodeOutput::setNumberOfChannels(unsigned numberOfChannels)
-{
-    ASSERT(context()->isGraphOwner());
-    
-    m_desiredNumberOfChannels = numberOfChannels;
-
-    if (context()->isAudioThread()) {
-        // If we're in the audio thread then we can take care of it right away (we should be at the very start or end of a rendering quantum).
-        updateNumberOfChannels();
-    } else {
-        // Let the context take care of it in the audio thread in the pre and post render tasks.
-        context()->markAudioNodeOutputDirty(this);
-    }
-}
-
-void AudioNodeOutput::setInternalBus()
-{
-    switch (m_numberOfChannels) {
-    case 0:
-    case 1:
-        m_internalOutputBus = m_monoInternalBus.get();
-        break;
-    case 2:
-        m_internalOutputBus = m_stereoInternalBus.get();
-        break;
-    default:
-        // FIXME: later we can fully implement more than stereo, 5.1, etc.
-        ASSERT_NOT_REACHED();            
-    }
-
-    // This may later be changed in pull() to point to an in-place bus with the same number of channels.
-    m_actualDestinationBus = m_internalOutputBus;
-}
-
-void AudioNodeOutput::updateRenderingState()
-{
-    updateNumberOfChannels();
-    m_renderingFanOutCount = fanOutCount();
-}
-
-void AudioNodeOutput::updateNumberOfChannels()
-{
-    ASSERT(context()->isAudioThread() && context()->isGraphOwner());
-
-    if (m_numberOfChannels != m_desiredNumberOfChannels) {
-        m_numberOfChannels = m_desiredNumberOfChannels;
-        setInternalBus();
-        propagateChannelCount();
-    }
-}
-
-void AudioNodeOutput::propagateChannelCount()
-{
-    ASSERT(context()->isAudioThread() && context()->isGraphOwner());
-    
-    if (isChannelCountKnown()) {
-        // Announce to any nodes we're connected to that we changed our channel count for its input.
-        for (InputsIterator i = m_inputs.begin(); i != m_inputs.end(); ++i) {
-            AudioNodeInput* input = *i;
-            AudioNode* connectionNode = input->node();
-            connectionNode->checkNumberOfChannelsForInput(input);
-        }
-    }
-}
-
-AudioBus* AudioNodeOutput::pull(AudioBus* inPlaceBus, size_t framesToProcess)
-{
-    ASSERT(context()->isAudioThread());
-    ASSERT(m_renderingFanOutCount > 0);
-    
-    // Causes our AudioNode to process if it hasn't already for this render quantum.
-    // We try to do in-place processing (using inPlaceBus) if at all possible,
-    // but we can't process in-place if we're connected to more than one input (fan-out > 1).
-    // In this case pull() is called multiple times per rendering quantum, and the processIfNecessary() call below will
-    // cause our node to process() only the first time, caching the output in m_internalOutputBus for subsequent calls.    
-    
-    bool isInPlace = inPlaceBus && inPlaceBus->numberOfChannels() == numberOfChannels() && m_renderingFanOutCount == 1;
-
-    // Setup the actual destination bus for processing when our node's process() method gets called in processIfNecessary() below.
-    m_actualDestinationBus = isInPlace ? inPlaceBus : m_internalOutputBus;
-
-    node()->processIfNecessary(framesToProcess);
-    return m_actualDestinationBus;
-}
-
-AudioBus* AudioNodeOutput::bus() const
-{
-    ASSERT(const_cast<AudioNodeOutput*>(this)->context()->isAudioThread());
-    ASSERT(m_actualDestinationBus);
-    return m_actualDestinationBus;
-}
-
-unsigned AudioNodeOutput::renderingFanOutCount() const
-{
-    return m_renderingFanOutCount;
-}
-
-unsigned AudioNodeOutput::fanOutCount()
-{
-    ASSERT(context()->isGraphOwner());
-    return m_inputs.size();
-}
-
-void AudioNodeOutput::addInput(AudioNodeInput* input)
-{
-    ASSERT(context()->isGraphOwner());
-
-    ASSERT(input);
-    if (!input)
-        return;
-
-    m_inputs.add(input);
-}
-
-void AudioNodeOutput::removeInput(AudioNodeInput* input)
-{
-    ASSERT(context()->isGraphOwner());
-
-    ASSERT(input);
-    if (!input)
-        return;
-
-    m_inputs.remove(input);
-}
-
-void AudioNodeOutput::disconnectAllInputs()
-{
-    ASSERT(context()->isGraphOwner());
-    
-    // AudioNodeInput::disconnect() changes m_inputs by calling removeInput().
-    while (!m_inputs.isEmpty()) {
-        AudioNodeInput* input = *m_inputs.begin();
-        input->disconnect(this);
-    }
-}
-
-void AudioNodeOutput::disable()
-{
-    ASSERT(context()->isGraphOwner());
-
-    if (m_isEnabled) {
-        for (InputsIterator i = m_inputs.begin(); i != m_inputs.end(); ++i) {
-            AudioNodeInput* input = *i;
-            input->disable(this);
-        }
-        m_isEnabled = false;
-    }
-}
-
-void AudioNodeOutput::enable()
-{
-    ASSERT(context()->isGraphOwner());
-
-    if (!m_isEnabled) {
-        for (InputsIterator i = m_inputs.begin(); i != m_inputs.end(); ++i) {
-            AudioNodeInput* input = *i;
-            input->enable(this);
-        }
-        m_isEnabled = true;
-    }
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioNodeOutput.h b/WebCore/webaudio/AudioNodeOutput.h
deleted file mode 100644
index 7114b38..0000000
--- a/WebCore/webaudio/AudioNodeOutput.h
+++ /dev/null
@@ -1,134 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef AudioNodeOutput_h
-#define AudioNodeOutput_h
-
-#include "AudioBus.h"
-#include "AudioNode.h"
-#include <wtf/HashSet.h>
-#include <wtf/OwnPtr.h>
-#include <wtf/Vector.h>
-
-namespace WebCore {
-
-class AudioContext;
-class AudioNodeInput;
-
-// AudioNodeOutput represents a single output for an AudioNode.
-// It may be connected to one or more AudioNodeInputs.
-
-class AudioNodeOutput {
-public:
-    // It's OK to pass 0 for numberOfChannels in which case setNumberOfChannels() must be called later on.
-    AudioNodeOutput(AudioNode*, unsigned numberOfChannels);
-
-    // Can be called from any thread.
-    AudioNode* node() const { return m_node; }
-    AudioContext* context() { return m_node->context(); }
-    
-    // Causes our AudioNode to process if it hasn't already for this render quantum.
-    // It returns the bus containing the processed audio for this output, returning inPlaceBus if in-place processing was possible.
-    // Called from context's audio thread.
-    AudioBus* pull(AudioBus* inPlaceBus, size_t framesToProcess);
-
-    // bus() will contain the rendered audio after pull() is called for each rendering time quantum.
-    // Called from context's audio thread.
-    AudioBus* bus() const;
-
-    // fanOutCount() is the number of AudioNodeInputs that we're connected to.
-    // This function should not be called in audio thread rendering code, instead renderingFanOutCount() should be used.
-    // It must be called with the context's graph lock.
-    unsigned fanOutCount();
-
-    // renderingFanOutCount() is the number of AudioNodeInputs that we're connected to during rendering.
-    // Unlike fanOutCount() it will not change during the course of a render quantum.
-    unsigned renderingFanOutCount() const;
-
-    // It must be called with the context's graph lock.
-    void disconnectAllInputs();
-
-    void setNumberOfChannels(unsigned);
-    unsigned numberOfChannels() const { return m_numberOfChannels; }
-    bool isChannelCountKnown() const { return numberOfChannels() > 0; }
-
-    // Disable/Enable happens when there are still JavaScript references to a node, but it has otherwise "finished" its work.
-    // For example, when a note has finished playing.  It is kept around, because it may be played again at a later time.
-    // They must be called with the context's graph lock.
-    void disable();
-    void enable();
-
-    // updateRenderingState() is called in the audio thread at the start or end of the render quantum to handle any recent changes to the graph state.
-    // It must be called with the context's graph lock.
-    void updateRenderingState();
-    
-private:
-    AudioNode* m_node;
-
-    friend class AudioNodeInput;
-    
-    // These are called from AudioNodeInput.
-    // They must be called with the context's graph lock.
-    void addInput(AudioNodeInput*);
-    void removeInput(AudioNodeInput*);
-
-    // setInternalBus() sets m_internalOutputBus appropriately for the number of channels.
-    // It is called in the constructor or in the audio thread with the context's graph lock.
-    void setInternalBus();
-
-    // Announce to any nodes we're connected to that we changed our channel count for its input.
-    // It must be called in the audio thread with the context's graph lock.
-    void propagateChannelCount();
-
-    // updateNumberOfChannels() is called in the audio thread at the start or end of the render quantum to pick up channel changes.
-    // It must be called with the context's graph lock.
-    void updateNumberOfChannels();
-
-    // m_numberOfChannels will only be changed in the audio thread.
-    // The main thread sets m_desiredNumberOfChannels which will later get picked up in the audio thread in updateNumberOfChannels().
-    unsigned m_numberOfChannels;
-    unsigned m_desiredNumberOfChannels;
-    
-    // m_internalOutputBus will point to either m_monoInternalBus or m_stereoInternalBus.
-    // It must only be changed in the audio thread (or constructor).
-    AudioBus* m_internalOutputBus;
-    OwnPtr<AudioBus> m_monoInternalBus;
-    OwnPtr<AudioBus> m_stereoInternalBus;
-
-    // m_actualDestinationBus is set in pull() and will either point to one of our internal busses or to the in-place bus.
-    // It must only be changed in the audio thread (or constructor).
-    AudioBus* m_actualDestinationBus;
-
-    HashSet<AudioNodeInput*> m_inputs;
-    typedef HashSet<AudioNodeInput*>::iterator InputsIterator;
-    bool m_isEnabled;
-    
-    // For the purposes of rendering, keeps track of the number of inputs we're connected to.
-    // This value should only be changed at the very start or end of the rendering quantum.
-    unsigned m_renderingFanOutCount;
-};
-
-} // namespace WebCore
-
-#endif // AudioNodeOutput_h
diff --git a/WebCore/webaudio/AudioPannerNode.cpp b/WebCore/webaudio/AudioPannerNode.cpp
deleted file mode 100644
index 5c94763..0000000
--- a/WebCore/webaudio/AudioPannerNode.cpp
+++ /dev/null
@@ -1,317 +0,0 @@
-/*
- * 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 "AudioPannerNode.h"
-
-#include "AudioBufferSourceNode.h"
-#include "AudioBus.h"
-#include "AudioContext.h"
-#include "AudioNodeInput.h"
-#include "AudioNodeOutput.h"
-#include "HRTFPanner.h"
-#include <wtf/MathExtras.h>
-
-using namespace std;
-
-namespace WebCore {
-
-static void fixNANs(double &x)
-{
-    if (isnan(x) || isinf(x))
-        x = 0.0;
-}
-
-AudioPannerNode::AudioPannerNode(AudioContext* context, double sampleRate)
-    : AudioNode(context, sampleRate)
-    , m_panningModel(Panner::PanningModelHRTF)
-    , m_lastGain(-1.0)
-    , m_connectionCount(0)
-{
-    addInput(adoptPtr(new AudioNodeInput(this)));
-    addOutput(adoptPtr(new AudioNodeOutput(this, 2)));
-    
-    m_distanceGain = AudioGain::create("distanceGain", 1.0, 0.0, 1.0);
-    m_coneGain = AudioGain::create("coneGain", 1.0, 0.0, 1.0);
-
-    m_position = FloatPoint3D(0, 0, 0);
-    m_orientation = FloatPoint3D(1, 0, 0);
-    m_velocity = FloatPoint3D(0, 0, 0);
-    
-    setType(NodeTypePanner);
-
-    initialize();
-}
-
-AudioPannerNode::~AudioPannerNode()
-{
-    uninitialize();
-}
-
-void AudioPannerNode::pullInputs(size_t framesToProcess)
-{
-    // We override pullInputs(), so we can detect new AudioSourceNodes which have connected to us when new connections are made.
-    // These AudioSourceNodes need to be made aware of our existence in order to handle doppler shift pitch changes.
-    if (m_connectionCount != context()->connectionCount()) {
-        m_connectionCount = context()->connectionCount();
-
-        // Recursively go through all nodes connected to us.
-        notifyAudioSourcesConnectedToNode(this);
-    }
-    
-    AudioNode::pullInputs(framesToProcess);
-}
-
-void AudioPannerNode::process(size_t framesToProcess)
-{
-    AudioBus* destination = output(0)->bus();
-
-    if (!isInitialized() || !input(0)->isConnected() || !m_panner.get()) {
-        destination->zero();
-        return;
-    }
-
-    AudioBus* source = input(0)->bus();
-
-    if (!source) {
-        destination->zero();
-        return;
-    }
-
-    // Apply the panning effect.
-    double azimuth;
-    double elevation;
-    getAzimuthElevation(&azimuth, &elevation);
-    m_panner->pan(azimuth, elevation, source, destination, framesToProcess);
-
-    // Get the distance and cone gain.
-    double totalGain = distanceConeGain();
-
-    // Snap to desired gain at the beginning.
-    if (m_lastGain == -1.0)
-        m_lastGain = totalGain;
-        
-    // Apply gain in-place with de-zippering.
-    destination->copyWithGainFrom(*destination, &m_lastGain, totalGain);
-}
-
-void AudioPannerNode::reset()
-{
-    m_lastGain = -1.0; // force to snap to initial gain
-    if (m_panner.get())
-        m_panner->reset();
-}
-
-void AudioPannerNode::initialize()
-{
-    if (isInitialized())
-        return;
-        
-    m_panner = Panner::create(m_panningModel, sampleRate());
-
-    AudioNode::initialize();
-}
-
-void AudioPannerNode::uninitialize()
-{
-    if (!isInitialized())
-        return;
-        
-    m_panner.clear();
-    AudioNode::uninitialize();
-}
-
-AudioListener* AudioPannerNode::listener()
-{
-    return context()->listener();
-}
-
-void AudioPannerNode::setPanningModel(unsigned short model)
-{
-    if (!m_panner.get() || model != m_panningModel) {
-        OwnPtr<Panner> newPanner = Panner::create(model, sampleRate());
-        m_panner = newPanner.release();
-    }
-}
-
-void AudioPannerNode::getAzimuthElevation(double* outAzimuth, double* outElevation)
-{
-    // FIXME: we should cache azimuth and elevation (if possible), so we only re-calculate if a change has been made.
-
-    double azimuth = 0.0;
-
-    // Calculate the source-listener vector
-    FloatPoint3D listenerPosition = listener()->position();
-    FloatPoint3D sourceListener = m_position - listenerPosition;
-
-    if (sourceListener.isZero()) {
-        // degenerate case if source and listener are at the same point
-        *outAzimuth = 0.0;
-        *outElevation = 0.0;
-        return;
-    }
-
-    sourceListener.normalize();
-
-    // Align axes
-    FloatPoint3D listenerFront = listener()->orientation();
-    FloatPoint3D listenerUp = listener()->upVector();
-    FloatPoint3D listenerRight = listenerFront.cross(listenerUp);
-    listenerRight.normalize();
-
-    FloatPoint3D listenerFrontNorm = listenerFront;
-    listenerFrontNorm.normalize();
-
-    FloatPoint3D up = listenerRight.cross(listenerFrontNorm);
-
-    double upProjection = sourceListener.dot(up);
-
-    FloatPoint3D projectedSource = sourceListener - upProjection * up;
-    projectedSource.normalize();
-
-    azimuth = 180.0 * acos(projectedSource.dot(listenerRight)) / piDouble;
-    fixNANs(azimuth); // avoid illegal values
-
-    // Source  in front or behind the listener
-    double frontBack = projectedSource.dot(listenerFrontNorm);
-    if (frontBack < 0.0)
-        azimuth = 360.0 - azimuth;
-
-    // Make azimuth relative to "front" and not "right" listener vector
-    if ((azimuth >= 0.0) && (azimuth <= 270.0))
-        azimuth = 90.0 - azimuth;
-    else
-        azimuth = 450.0 - azimuth;
-
-    // Elevation
-    double elevation = 90.0 - 180.0 * acos(sourceListener.dot(up)) / piDouble;
-    fixNANs(azimuth); // avoid illegal values
-
-    if (elevation > 90.0)
-        elevation = 180.0 - elevation;
-    else if (elevation < -90.0)
-        elevation = -180.0 - elevation;
-
-    if (outAzimuth)
-        *outAzimuth = azimuth;
-    if (outElevation)
-        *outElevation = elevation;
-}
-
-float AudioPannerNode::dopplerRate()
-{
-    double dopplerShift = 1.0;
-
-    // FIXME: optimize for case when neither source nor listener has changed...
-    double dopplerFactor = listener()->dopplerFactor();
-
-    if (dopplerFactor > 0.0) {
-        double speedOfSound = listener()->speedOfSound();
-
-        const FloatPoint3D &sourceVelocity = m_velocity;
-        const FloatPoint3D &listenerVelocity = listener()->velocity();
-
-        // Don't bother if both source and listener have no velocity
-        bool sourceHasVelocity = !sourceVelocity.isZero();
-        bool listenerHasVelocity = !listenerVelocity.isZero();
-
-        if (sourceHasVelocity || listenerHasVelocity) {
-            // Calculate the source to listener vector
-            FloatPoint3D listenerPosition = listener()->position();
-            FloatPoint3D sourceToListener = m_position - listenerPosition;
-
-            double sourceListenerMagnitude = sourceToListener.length();
-
-            double listenerProjection = sourceToListener.dot(listenerVelocity) / sourceListenerMagnitude;
-            double sourceProjection = sourceToListener.dot(sourceVelocity) / sourceListenerMagnitude;
-
-            listenerProjection = -listenerProjection;
-            sourceProjection = -sourceProjection;
-
-            double scaledSpeedOfSound = speedOfSound / dopplerFactor;
-            listenerProjection = min(listenerProjection, scaledSpeedOfSound);
-            sourceProjection = min(sourceProjection, scaledSpeedOfSound);
-
-            dopplerShift = ((speedOfSound - dopplerFactor * listenerProjection) / (speedOfSound - dopplerFactor * sourceProjection));
-            fixNANs(dopplerShift); // avoid illegal values
-
-            // Limit the pitch shifting to 4 octaves up and 3 octaves down.
-            if (dopplerShift > 16.0)
-                dopplerShift = 16.0;
-            else if (dopplerShift < 0.125)
-                dopplerShift = 0.125;   
-        }
-    }
-
-    return static_cast<float>(dopplerShift);
-}
-
-float AudioPannerNode::distanceConeGain()
-{
-    FloatPoint3D listenerPosition = listener()->position();
-
-    double listenerDistance = m_position.distanceTo(listenerPosition);
-    double distanceGain = m_distanceEffect.gain(listenerDistance);
-    
-    m_distanceGain->setValue(static_cast<float>(distanceGain));
-
-    // FIXME: could optimize by caching coneGain
-    double coneGain = m_coneEffect.gain(m_position, m_orientation, listenerPosition);
-    
-    m_coneGain->setValue(static_cast<float>(coneGain));
-
-    return float(distanceGain * coneGain);
-}
-
-void AudioPannerNode::notifyAudioSourcesConnectedToNode(AudioNode* node)
-{
-    ASSERT(node);
-    if (!node)
-        return;
-        
-    // First check if this node is an AudioBufferSourceNode.  If so, let it know about us so that doppler shift pitch can be taken into account.
-    if (node->type() == NodeTypeAudioBufferSource) {
-        AudioBufferSourceNode* bufferSourceNode = reinterpret_cast<AudioBufferSourceNode*>(node);
-        bufferSourceNode->setPannerNode(this);
-    } else {    
-        // Go through all inputs to this node.
-        for (unsigned i = 0; i < node->numberOfInputs(); ++i) {
-            AudioNodeInput* input = node->input(i);
-
-            // For each input, go through all of its connections, looking for AudioBufferSourceNodes.
-            for (unsigned j = 0; j < input->numberOfRenderingConnections(); ++j) {
-                AudioNodeOutput* connectedOutput = input->renderingOutput(j);
-                AudioNode* connectedNode = connectedOutput->node();
-                notifyAudioSourcesConnectedToNode(connectedNode); // recurse
-            }
-        }
-    }
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioPannerNode.h b/WebCore/webaudio/AudioPannerNode.h
deleted file mode 100644
index 61e34a9..0000000
--- a/WebCore/webaudio/AudioPannerNode.h
+++ /dev/null
@@ -1,148 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef AudioPannerNode_h
-#define AudioPannerNode_h
-
-#include "AudioBus.h"
-#include "AudioGain.h"
-#include "AudioListener.h"
-#include "AudioNode.h"
-#include "Cone.h"
-#include "Distance.h"
-#include "FloatPoint3D.h"
-#include "Panner.h"
-#include <wtf/OwnPtr.h>
-
-namespace WebCore {
-
-// AudioPannerNode is an AudioNode with one input and one output.
-// It positions a sound in 3D space, with the exact effect dependent on the panning model.
-// It has a position and an orientation in 3D space which is relative to the position and orientation of the context's AudioListener.
-// A distance effect will attenuate the gain as the position moves away from the listener.
-// A cone effect will attenuate the gain as the orientation moves away from the listener.
-// All of these effects follow the OpenAL specification very closely.
-
-class AudioPannerNode : public AudioNode {
-public:
-    // These must be defined as in the .idl file and must match those in the Panner class.
-    enum {
-        EQUALPOWER = 0,
-        HRTF = 1,
-        SOUNDFIELD = 2,
-    };
-
-    static PassRefPtr<AudioPannerNode> create(AudioContext* context, double sampleRate)
-    {
-        return adoptRef(new AudioPannerNode(context, sampleRate));
-    }
-
-    virtual ~AudioPannerNode();
-
-    // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void pullInputs(size_t framesToProcess);
-    virtual void reset();
-    virtual void initialize();
-    virtual void uninitialize();
-
-    // Listener
-    AudioListener* listener();
-
-    // Panning model
-    unsigned short panningModel() const { return m_panningModel; }
-    void setPanningModel(unsigned short);
-
-    // Position
-    FloatPoint3D position() const { return m_position; }
-    void setPosition(float x, float y, float z) { m_position = FloatPoint3D(x, y, z); }
-
-    // Orientation
-    FloatPoint3D orientation() const { return m_position; }
-    void setOrientation(float x, float y, float z) { m_orientation = FloatPoint3D(x, y, z); }
-
-    // Velocity
-    FloatPoint3D velocity() const { return m_velocity; }
-    void setVelocity(float x, float y, float z) { m_velocity = FloatPoint3D(x, y, z); }
-
-    // Distance parameters
-    unsigned short distanceModel() { return m_distanceEffect.model(); }
-    void setDistanceModel(unsigned short model) { m_distanceEffect.setModel(static_cast<DistanceEffect::ModelType>(model), true); }
-    
-    float refDistance() { return static_cast<float>(m_distanceEffect.refDistance()); }
-    void setRefDistance(float refDistance) { m_distanceEffect.setRefDistance(refDistance); }
-
-    float maxDistance() { return static_cast<float>(m_distanceEffect.maxDistance()); }
-    void setMaxDistance(float maxDistance) { m_distanceEffect.setMaxDistance(maxDistance); }
-
-    float rolloffFactor() { return static_cast<float>(m_distanceEffect.rolloffFactor()); }
-    void setRolloffFactor(float rolloffFactor) { m_distanceEffect.setRolloffFactor(rolloffFactor); }
-
-    // Sound cones - angles in degrees
-    float coneInnerAngle() const { return static_cast<float>(m_coneEffect.innerAngle()); }
-    void setConeInnerAngle(float angle) { m_coneEffect.setInnerAngle(angle); }
-
-    float coneOuterAngle() const { return static_cast<float>(m_coneEffect.outerAngle()); }
-    void setConeOuterAngle(float angle) { m_coneEffect.setOuterAngle(angle); }
-
-    float coneOuterGain() const { return static_cast<float>(m_coneEffect.outerGain()); }
-    void setConeOuterGain(float angle) { m_coneEffect.setOuterGain(angle); }
-
-    void getAzimuthElevation(double* outAzimuth, double* outElevation);
-    float dopplerRate();
-
-    // Accessors for dynamically calculated gain values.
-    AudioGain* distanceGain() { return m_distanceGain.get(); }                                        
-    AudioGain* coneGain() { return m_coneGain.get(); }                                        
-
-private:
-    AudioPannerNode(AudioContext*, double sampleRate);
-
-    // Returns the combined distance and cone gain attenuation.
-    float distanceConeGain();
-
-    // Notifies any AudioBufferSourceNodes connected to us either directly or indirectly about our existence.
-    // This is in order to handle the pitch change necessary for the doppler shift.
-    void notifyAudioSourcesConnectedToNode(AudioNode*);
-
-    OwnPtr<Panner> m_panner;
-    unsigned m_panningModel;
-
-    FloatPoint3D m_position;
-    FloatPoint3D m_orientation;
-    FloatPoint3D m_velocity;
-
-    // Gain
-    RefPtr<AudioGain> m_distanceGain;
-    RefPtr<AudioGain> m_coneGain;
-    DistanceEffect m_distanceEffect;
-    ConeEffect m_coneEffect;
-    double m_lastGain;
-
-    unsigned m_connectionCount;
-};
-
-} // namespace WebCore
-
-#endif // AudioPannerNode_h
diff --git a/WebCore/webaudio/AudioPannerNode.idl b/WebCore/webaudio/AudioPannerNode.idl
deleted file mode 100644
index 2db093d..0000000
--- a/WebCore/webaudio/AudioPannerNode.idl
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO,
-        GenerateConstructor,
-        GenerateToJS
-    ] AudioPannerNode : AudioNode {
-        // Panning model
-        const unsigned short EQUALPOWER = 0;
-        const unsigned short HRTF = 1;
-        const unsigned short SOUNDFIELD = 2;
-
-        // Default model for stereo is HRTF 
-        attribute unsigned long panningModel; // FIXME: use unsigned short when glue generation supports it
-
-        // Uses a 3D cartesian coordinate system 
-        void setPosition(in float x, in float y, in float z);
-        void setOrientation(in float x, in float y, in float z);
-        void setVelocity(in float x, in float y, in float z);
-
-        // Distance model
-        attribute unsigned long distanceModel; // FIXME: use unsigned short when glue generation supports it
-        attribute float refDistance;
-        attribute float maxDistance;
-        attribute float rolloffFactor;
-
-        // Directional sound cone
-        attribute float coneInnerAngle;
-        attribute float coneOuterAngle;
-        attribute float coneOuterGain;
-
-        // Dynamically calculated gain values          
-        readonly attribute AudioGain coneGain;
-        readonly attribute AudioGain distanceGain;
-    };
-}
diff --git a/WebCore/webaudio/AudioParam.cpp b/WebCore/webaudio/AudioParam.cpp
deleted file mode 100644
index dcf918f..0000000
--- a/WebCore/webaudio/AudioParam.cpp
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * 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 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 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 "AudioParam.h"
-
-#include <wtf/MathExtras.h>
-
-namespace WebCore {
-
-const double AudioParam::DefaultSmoothingConstant = 0.05;
-const double AudioParam::SnapThreshold = 0.001;
-
-void AudioParam::setValue(float value)
-{
-    // Check against JavaScript giving us bogus floating-point values.
-    // Don't ASSERT, since this can happen if somebody writes bad JS.
-    if (!isnan(value) && !isinf(value))
-        m_value = value;
-}
-
-bool AudioParam::smooth()
-{
-    if (m_smoothedValue == m_value) {
-        // Smoothed value has already approached and snapped to value.
-        return true;
-    }
-
-    // Exponential approach
-    m_smoothedValue += (m_value - m_smoothedValue) * m_smoothingConstant;
-
-    // If we get close enough then snap to actual value.
-    if (fabs(m_smoothedValue - m_value) < SnapThreshold) // FIXME: the threshold needs to be adjustable depending on range - but this is OK general purpose value.
-        m_smoothedValue = m_value;
-
-    return false;
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioParam.h b/WebCore/webaudio/AudioParam.h
deleted file mode 100644
index 88b7615..0000000
--- a/WebCore/webaudio/AudioParam.h
+++ /dev/null
@@ -1,100 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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.
- */
-
-#ifndef AudioParam_h
-#define AudioParam_h
-
-#include "PlatformString.h"
-#include <sys/types.h>
-#include <wtf/PassRefPtr.h>
-#include <wtf/RefCounted.h>
-
-namespace WebCore {
-
-class AudioParam : public RefCounted<AudioParam> {
-public:
-    static const double DefaultSmoothingConstant;
-    static const double SnapThreshold;
-
-    static PassRefPtr<AudioParam> create(const String& name, double defaultValue, double minValue, double maxValue, unsigned units = 0)
-    {
-        return adoptRef(new AudioParam(name, defaultValue, minValue, maxValue, units));
-    }
-
-    AudioParam(const String& name, double defaultValue, double minValue, double maxValue, unsigned units = 0)
-        : m_name(name)
-        , m_value(defaultValue)
-        , m_defaultValue(defaultValue)
-        , m_minValue(minValue)
-        , m_maxValue(maxValue)
-        , m_units(units)
-        , m_smoothedValue(defaultValue)
-        , m_smoothingConstant(DefaultSmoothingConstant)
-    {
-    }
-
-    float value() const { return static_cast<float>(m_value); }
-    
-    void setValue(float);
-
-    String name() const { return m_name; }
-
-    float minValue() const { return static_cast<float>(m_minValue); }
-    float maxValue() const { return static_cast<float>(m_maxValue); }
-    float defaultValue() const { return static_cast<float>(m_defaultValue); }
-    unsigned units() const { return m_units; }
-
-    // Value smoothing:
-
-    // When a new value is set with setValue(), in our internal use of the parameter we don't immediately jump to it.
-    // Instead we smoothly approach this value to avoid glitching.
-    float smoothedValue() const { return static_cast<float>(m_smoothedValue); }
-
-    // Smoothly exponentially approaches to (de-zippers) the desired value.
-    // Returns true if smoothed value has already snapped exactly to value.
-    bool smooth();
-
-    void resetSmoothedValue() { m_smoothedValue = m_value; }
-    void setSmoothingConstant(double k) { m_smoothingConstant = k; }
-
-private:
-    String m_name;
-    double m_value;
-    double m_defaultValue;
-    double m_minValue;
-    double m_maxValue;
-    unsigned m_units;
-
-    // Smoothing (de-zippering)
-    double m_smoothedValue;
-    double m_smoothingConstant;
-};
-
-} // namespace WebCore
-
-#endif // AudioParam_h
diff --git a/WebCore/webaudio/AudioParam.idl b/WebCore/webaudio/AudioParam.idl
deleted file mode 100644
index ff2598e..0000000
--- a/WebCore/webaudio/AudioParam.idl
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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.
- */
-
-module webaudio {
-    interface [
-        Conditional=WEB_AUDIO
-    ] AudioParam {
-        attribute float value;
-        readonly attribute float minValue;
-        readonly attribute float maxValue;
-        readonly attribute float defaultValue;
-        
-        readonly attribute DOMString name;
-
-        // FIXME: Could define units constants here (seconds, decibels, cents, etc.)...
-        readonly attribute unsigned short units;
-    };
-}
diff --git a/WebCore/webaudio/AudioProcessingEvent.cpp b/WebCore/webaudio/AudioProcessingEvent.cpp
deleted file mode 100644
index 54ce521..0000000
--- a/WebCore/webaudio/AudioProcessingEvent.cpp
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * 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 "AudioProcessingEvent.h"
-
-#include "AudioBuffer.h"
-#include "EventNames.h"
-
-namespace WebCore {
-
-PassRefPtr<AudioProcessingEvent> AudioProcessingEvent::create(PassRefPtr<AudioBuffer> inputBuffer, PassRefPtr<AudioBuffer> outputBuffer)
-{
-    return adoptRef(new AudioProcessingEvent(inputBuffer, outputBuffer));
-}
-
-AudioProcessingEvent::AudioProcessingEvent(PassRefPtr<AudioBuffer> inputBuffer, PassRefPtr<AudioBuffer> outputBuffer)
-    : Event(eventNames().audioprocessEvent, true, false)
-    , m_inputBuffer(inputBuffer)
-    , m_outputBuffer(outputBuffer)
-{
-}
-
-AudioProcessingEvent::~AudioProcessingEvent()
-{
-}
-
-bool AudioProcessingEvent::isAudioProcessingEvent() const
-{
-    return true;
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/AudioProcessingEvent.h b/WebCore/webaudio/AudioProcessingEvent.h
deleted file mode 100644
index a88669c..0000000
--- a/WebCore/webaudio/AudioProcessingEvent.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef AudioProcessingEvent_h
-#define AudioProcessingEvent_h
-
-#include "AudioBuffer.h"
-#include "Event.h"
-#include <wtf/PassRefPtr.h>
-#include <wtf/RefPtr.h>
-
-namespace WebCore {
-
-class AudioBuffer;
-    
-class AudioProcessingEvent : public Event {
-public:
-    static PassRefPtr<AudioProcessingEvent> create(PassRefPtr<AudioBuffer> inputBuffer, PassRefPtr<AudioBuffer> outputBuffer);
-    
-    virtual ~AudioProcessingEvent();
-
-    virtual bool isAudioProcessingEvent() const;
-
-    AudioBuffer* inputBuffer() { return m_inputBuffer.get(); }
-    AudioBuffer* outputBuffer() { return m_outputBuffer.get(); }
-
-private:
-    AudioProcessingEvent(PassRefPtr<AudioBuffer> inputBuffer, PassRefPtr<AudioBuffer> outputBuffer);
-
-    RefPtr<AudioBuffer> m_inputBuffer;
-    RefPtr<AudioBuffer> m_outputBuffer;
-};
-
-} // namespace WebCore
-
-#endif // AudioProcessingEvent_h
diff --git a/WebCore/webaudio/AudioProcessingEvent.idl b/WebCore/webaudio/AudioProcessingEvent.idl
deleted file mode 100644
index c2f8a83..0000000
--- a/WebCore/webaudio/AudioProcessingEvent.idl
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO,
-        GenerateToJS
-    ] AudioProcessingEvent : Event {
-        readonly attribute AudioBuffer inputBuffer;
-        readonly attribute AudioBuffer outputBuffer; 
-    };
-}
diff --git a/WebCore/webaudio/AudioSourceNode.h b/WebCore/webaudio/AudioSourceNode.h
deleted file mode 100644
index 6091371..0000000
--- a/WebCore/webaudio/AudioSourceNode.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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.
- */
-
-#ifndef AudioSourceNode_h
-#define AudioSourceNode_h
-
-#include "AudioNode.h"
-
-namespace WebCore {
-
-class AudioSourceNode : public AudioNode {
-public:
-    AudioSourceNode(AudioContext* context, double sampleRate)
-        : AudioNode(context, sampleRate)
-    {
-    }
-};
-
-} // namespace WebCore
-
-#endif // AudioSourceNode_h
diff --git a/WebCore/webaudio/AudioSourceNode.idl b/WebCore/webaudio/AudioSourceNode.idl
deleted file mode 100644
index ec3c356..0000000
--- a/WebCore/webaudio/AudioSourceNode.idl
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * 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.
- * 3.  Neither the name of Apple Computer, Inc. ("Apple") nor the names of
- *     its contributors may be used to endorse or promote products derived
- *     from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY APPLE 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 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO
-    ] AudioSourceNode : AudioNode {
-    };
-}
diff --git a/WebCore/webaudio/BiquadDSPKernel.cpp b/WebCore/webaudio/BiquadDSPKernel.cpp
deleted file mode 100644
index a4b28be..0000000
--- a/WebCore/webaudio/BiquadDSPKernel.cpp
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * 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 "BiquadDSPKernel.h"
-
-#include "BiquadProcessor.h"
-
-namespace WebCore {
-
-void BiquadDSPKernel::process(const float* source, float* destination, size_t framesToProcess)
-{
-    ASSERT(source && destination && biquadProcessor());
-    
-    // Recompute filter coefficients if any of the parameters have changed.
-    // FIXME: as an optimization, implement a way that a Biquad object can simply copy its internal filter coefficients from another Biquad object.
-    // Then re-factor this code to only run for the first BiquadDSPKernel of each BiquadProcessor.
-    if (biquadProcessor()->filterCoefficientsDirty()) {
-        double value1 = biquadProcessor()->parameter1()->smoothedValue();
-        double value2 = biquadProcessor()->parameter2()->smoothedValue();
-        
-        // Convert from Hertz to normalized frequency 0 -> 1.
-        double nyquist = this->nyquist();
-        double normalizedValue1 = value1 / nyquist;
-
-        // Configure the biquad with the new filter parameters for the appropriate type of filter.
-        switch (biquadProcessor()->type()) {
-        case BiquadProcessor::LowPass2:
-            m_biquad.setLowpassParams(normalizedValue1, value2);
-            break;
-
-        case BiquadProcessor::HighPass2:
-            m_biquad.setHighpassParams(normalizedValue1, value2);
-            break;
-
-        case BiquadProcessor::LowShelf:
-            m_biquad.setLowShelfParams(normalizedValue1, value2);
-            break;
-
-        // FIXME: add other biquad filter types...
-        case BiquadProcessor::Peaking:
-        case BiquadProcessor::Allpass:
-        case BiquadProcessor::HighShelf:
-            break;
-        }
-    }
-
-    m_biquad.process(source, destination, framesToProcess);
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/BiquadDSPKernel.h b/WebCore/webaudio/BiquadDSPKernel.h
deleted file mode 100644
index 47d0f34..0000000
--- a/WebCore/webaudio/BiquadDSPKernel.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef BiquadDSPKernel_h
-#define BiquadDSPKernel_h
-
-#include "AudioDSPKernel.h"
-#include "Biquad.h"
-#include "BiquadProcessor.h"
-
-namespace WebCore {
-
-class BiquadProcessor;
-
-// BiquadDSPKernel is an AudioDSPKernel and is responsible for filtering one channel of a BiquadProcessor using a Biquad object.
-
-class BiquadDSPKernel : public AudioDSPKernel {
-public:  
-    BiquadDSPKernel(BiquadProcessor* processor)
-    : AudioDSPKernel(processor)
-    {
-    }
-    
-    // AudioDSPKernel
-    virtual void process(const float* source, float* dest, size_t framesToProcess);
-    virtual void reset() { m_biquad.reset(); }
-    
-protected:
-    Biquad m_biquad;
-    BiquadProcessor* biquadProcessor() { return static_cast<BiquadProcessor*>(processor()); }
-};
-
-} // namespace WebCore
-
-#endif // BiquadDSPKernel_h
diff --git a/WebCore/webaudio/BiquadProcessor.cpp b/WebCore/webaudio/BiquadProcessor.cpp
deleted file mode 100644
index 97a480e..0000000
--- a/WebCore/webaudio/BiquadProcessor.cpp
+++ /dev/null
@@ -1,125 +0,0 @@
-/*
- * 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 "BiquadProcessor.h"
-
-#include "BiquadDSPKernel.h"
-
-namespace WebCore {
-
-BiquadProcessor::BiquadProcessor(FilterType type, double sampleRate, size_t numberOfChannels, bool autoInitialize)
-    : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
-    , m_type(type)
-    , m_parameter1(0)
-    , m_parameter2(0)
-    , m_parameter3(0)
-    , m_filterCoefficientsDirty(true)
-{
-    double nyquist = 0.5 * this->sampleRate();
-    
-    switch (type) {
-    // Highpass and lowpass share the same parameters and only differ in filter type.
-    case LowPass2:
-    case HighPass2:
-        m_parameter1 = AudioParam::create("frequency", 350.0, 20.0, nyquist);
-        m_parameter2 = AudioParam::create("resonance", 0.0, -20.0, 20.0);
-        m_parameter3 = AudioParam::create("unused", 0.0, 0.0, 1.0);
-        break;
-
-    case Peaking:
-        m_parameter1 = AudioParam::create("frequency", 2500.0, 20.0, nyquist);
-        m_parameter2 = AudioParam::create("gain", 0.0, -20.0, 20.0);
-        m_parameter3 = AudioParam::create("Q", 0.5, 0.0, 1000.0);
-        break;
-    case Allpass:
-        m_parameter1 = AudioParam::create("frequency", 2500.0, 20.0, nyquist);
-        m_parameter2 = AudioParam::create("Q", 0.5, 0.0, 1000.0);
-        m_parameter3 = AudioParam::create("unused", 0.0, 0.0, 1.0);
-        break;
-    case LowShelf:
-        m_parameter1 = AudioParam::create("frequency", 80.0, 20.0, nyquist);
-        m_parameter2 = AudioParam::create("gain", 0.0, 0.0, 1.0);
-        m_parameter3 = AudioParam::create("unused", 0.0, 0.0, 1.0);
-        break;
-    case HighShelf:
-        m_parameter1 = AudioParam::create("frequency", 10000.0, 20.0, nyquist);
-        m_parameter2 = AudioParam::create("gain", 0.0, 0.0, 1.0);
-        m_parameter3 = AudioParam::create("unused", 0.0, 0.0, 1.0);
-        break;
-    }
-
-    if (autoInitialize)
-        initialize();
-}
-
-BiquadProcessor::~BiquadProcessor()
-{
-    if (isInitialized())
-        uninitialize();
-}
-
-PassOwnPtr<AudioDSPKernel> BiquadProcessor::createKernel()
-{
-    return adoptPtr(new BiquadDSPKernel(this));
-}
-
-void BiquadProcessor::process(AudioBus* source, AudioBus* destination, size_t framesToProcess)
-{
-    if (!isInitialized()) {
-        destination->zero();
-        return;
-    }
-        
-    // Deal with smoothing / de-zippering.  Start out assuming filter parameters are not changing.
-    // The BiquadDSPKernel objects rely on this value to see if they need to re-compute their internal filter coefficients.
-    m_filterCoefficientsDirty = false;
-    
-    if (m_hasJustReset) {
-        // Snap to exact values first time after reset, then smooth for subsequent changes.
-        m_parameter1->resetSmoothedValue();
-        m_parameter2->resetSmoothedValue();
-        m_parameter3->resetSmoothedValue();
-        m_filterCoefficientsDirty = true;
-        m_hasJustReset = false;
-    } else {
-        // Smooth all of the filter parameters.  If they haven't yet converged to their target value then mark coefficients as dirty.
-        bool isStable1 = m_parameter1->smooth();
-        bool isStable2 = m_parameter2->smooth();
-        bool isStable3 = m_parameter3->smooth();
-        if (!(isStable1 && isStable2 && isStable3))
-            m_filterCoefficientsDirty = true;
-    }
-        
-    // For each channel of our input, process using the corresponding BiquadDSPKernel into the output channel.
-    for (unsigned i = 0; i < m_kernels.size(); ++i)
-        m_kernels[i]->process(source->channel(i)->data(), destination->channel(i)->data(), framesToProcess);
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/BiquadProcessor.h b/WebCore/webaudio/BiquadProcessor.h
deleted file mode 100644
index 55dca33..0000000
--- a/WebCore/webaudio/BiquadProcessor.h
+++ /dev/null
@@ -1,78 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef BiquadProcessor_h
-#define BiquadProcessor_h
-
-#include "AudioDSPKernel.h"
-#include "AudioDSPKernelProcessor.h"
-#include "AudioNode.h"
-#include "AudioParam.h"
-#include "Biquad.h"
-#include <wtf/RefPtr.h>
-
-namespace WebCore {
-
-// BiquadProcessor is an AudioDSPKernelProcessor which uses Biquad objects to implement several common filters.
-
-class BiquadProcessor : public AudioDSPKernelProcessor {
-public:
-    enum FilterType {
-        LowPass2,
-        HighPass2,
-        Peaking,
-        Allpass,
-        LowShelf,
-        HighShelf
-    };
-    
-    BiquadProcessor(FilterType, double sampleRate, size_t numberOfChannels, bool autoInitialize = true);
-    virtual ~BiquadProcessor();
-    
-    virtual PassOwnPtr<AudioDSPKernel> createKernel();
-        
-    virtual void process(AudioBus* source, AudioBus* destination, size_t framesToProcess);
-
-    bool filterCoefficientsDirty() const { return m_filterCoefficientsDirty; }
-
-    AudioParam* parameter1() { return m_parameter1.get(); }
-    AudioParam* parameter2() { return m_parameter2.get(); }
-    AudioParam* parameter3() { return m_parameter3.get(); }
-
-    FilterType type() const { return m_type; }
-
-private:
-    FilterType m_type;
-
-    RefPtr<AudioParam> m_parameter1;
-    RefPtr<AudioParam> m_parameter2;
-    RefPtr<AudioParam> m_parameter3;
-
-    // so DSP kernels know when to re-compute coefficients
-    bool m_filterCoefficientsDirty;
-};
-
-} // namespace WebCore
-
-#endif // BiquadProcessor_h
diff --git a/WebCore/webaudio/ConvolverNode.cpp b/WebCore/webaudio/ConvolverNode.cpp
deleted file mode 100644
index c778a41..0000000
--- a/WebCore/webaudio/ConvolverNode.cpp
+++ /dev/null
@@ -1,152 +0,0 @@
-/*
- * 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 "ConvolverNode.h"
-
-#include "AudioBuffer.h"
-#include "AudioNodeInput.h"
-#include "AudioNodeOutput.h"
-#include "Reverb.h"
-
-// Note about empirical tuning:
-// The maximum FFT size affects reverb performance and accuracy.
-// If the reverb is single-threaded and processes entirely in the real-time audio thread,
-// it's important not to make this too high.  In this case 8192 is a good value.
-// But, the Reverb object is multi-threaded, so we want this as high as possible without losing too much accuracy.
-// Very large FFTs will have worse phase errors.  Given these constraints 16384 is a good compromise.
-const size_t MaxFFTSize = 16384;
-
-namespace WebCore {
-
-ConvolverNode::ConvolverNode(AudioContext* context, double sampleRate)
-    : AudioNode(context, sampleRate)
-{
-    addInput(adoptPtr(new AudioNodeInput(this)));
-    addOutput(adoptPtr(new AudioNodeOutput(this, 2)));
-    
-    setType(NodeTypeConvolver);
-    
-    initialize();
-}
-
-ConvolverNode::~ConvolverNode()
-{
-    uninitialize();
-}
-
-void ConvolverNode::process(size_t framesToProcess)
-{
-    AudioBus* outputBus = output(0)->bus();
-    ASSERT(outputBus);
-
-    // Synchronize with possible dynamic changes to the impulse response.
-    if (m_processLock.tryLock()) {
-        if (!isInitialized() || !m_reverb.get())
-            outputBus->zero();
-        else {
-            // Process using the convolution engine.
-            // Note that we can handle the case where nothing is connected to the input, in which case we'll just feed silence into the convolver.
-            // FIXME:  If we wanted to get fancy we could try to factor in the 'tail time' and stop processing once the tail dies down if
-            // we keep getting fed silence.
-            m_reverb->process(input(0)->bus(), outputBus, framesToProcess);
-        }
-        
-        m_processLock.unlock();
-    } else {
-        // Too bad - the tryLock() failed.  We must be in the middle of setting a new impulse response.
-        outputBus->zero();
-    }
-}
-
-void ConvolverNode::reset()
-{
-    MutexLocker locker(m_processLock);
-    if (m_reverb.get())
-        m_reverb->reset();
-}
-
-void ConvolverNode::initialize()
-{
-    if (isInitialized())
-        return;
-        
-    AudioNode::initialize();
-}
-
-void ConvolverNode::uninitialize()
-{
-    if (!isInitialized())
-        return;
-
-    m_reverb.clear();
-    AudioNode::uninitialize();
-}
-
-void ConvolverNode::setBuffer(AudioBuffer* buffer)
-{
-    ASSERT(isMainThread());
-    
-    ASSERT(buffer);
-    if (!buffer)
-        return;
-
-    unsigned numberOfChannels = buffer->numberOfChannels();
-    size_t bufferLength = buffer->length();
-
-    // The current implementation supports up to four channel impulse responses, which are interpreted as true-stereo (see Reverb class).
-    bool isBufferGood = numberOfChannels > 0 && numberOfChannels <= 4 && bufferLength;
-    ASSERT(isBufferGood);
-    if (!isBufferGood)
-        return;
-
-    // Wrap the AudioBuffer by an AudioBus. It's an efficient pointer set and not a memcpy().
-    // This memory is simply used in the Reverb constructor and no reference to it is kept for later use in that class.
-    AudioBus bufferBus(numberOfChannels, bufferLength, false);
-    for (unsigned i = 0; i < numberOfChannels; ++i)
-        bufferBus.setChannelMemory(i, buffer->getChannelData(i)->data(), bufferLength);
-    
-    // Create the reverb with the given impulse response.
-    OwnPtr<Reverb> reverb = adoptPtr(new Reverb(&bufferBus, AudioNode::ProcessingSizeInFrames, MaxFFTSize, 2, true));
-
-    {
-        // Synchronize with process().
-        MutexLocker locker(m_processLock);
-        m_reverb = reverb.release();
-        m_buffer = buffer;
-    }
-}
-
-AudioBuffer* ConvolverNode::buffer()
-{
-    ASSERT(isMainThread());
-    return m_buffer.get();
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/ConvolverNode.h b/WebCore/webaudio/ConvolverNode.h
deleted file mode 100644
index 7b71ba9..0000000
--- a/WebCore/webaudio/ConvolverNode.h
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef ConvolverNode_h
-#define ConvolverNode_h
-
-#include "AudioNode.h"
-#include <wtf/OwnPtr.h>
-#include <wtf/RefPtr.h>
-#include <wtf/Threading.h>
-
-namespace WebCore {
-
-class AudioBuffer;
-class Reverb;
-    
-class ConvolverNode : public AudioNode {
-public:
-    static PassRefPtr<ConvolverNode> create(AudioContext* context, double sampleRate)
-    {
-        return adoptRef(new ConvolverNode(context, sampleRate));      
-    }
-    
-    virtual ~ConvolverNode();
-    
-    // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
-    virtual void initialize();
-    virtual void uninitialize();
-
-    // Impulse responses
-    void setBuffer(AudioBuffer*);
-    AudioBuffer* buffer();
-
-private:
-    ConvolverNode(AudioContext*, double sampleRate);
-
-    OwnPtr<Reverb> m_reverb;
-    RefPtr<AudioBuffer> m_buffer;
-
-    // This synchronizes dynamic changes to the convolution impulse response with process().
-    mutable Mutex m_processLock;
-};
-
-} // namespace WebCore
-
-#endif // ConvolverNode_h
diff --git a/WebCore/webaudio/ConvolverNode.idl b/WebCore/webaudio/ConvolverNode.idl
deleted file mode 100644
index d3eb475..0000000
--- a/WebCore/webaudio/ConvolverNode.idl
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    // A linear convolution effect
-    interface [
-        Conditional=WEB_AUDIO,
-        GenerateToJS
-    ] ConvolverNode : AudioNode {
-        attribute [JSCCustomSetter] AudioBuffer buffer;
-    };
-}
diff --git a/WebCore/webaudio/DelayDSPKernel.cpp b/WebCore/webaudio/DelayDSPKernel.cpp
deleted file mode 100644
index 9cb0450..0000000
--- a/WebCore/webaudio/DelayDSPKernel.cpp
+++ /dev/null
@@ -1,140 +0,0 @@
-/*
- * 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 "DelayDSPKernel.h"
-
-#include "AudioUtilities.h"
-#include <algorithm>
-
-using namespace std;
-  
-const double DefaultMaxDelayTime = 1.0;
-const double SmoothingTimeConstant = 0.020; // 20ms
-  
-namespace WebCore {
-
-DelayDSPKernel::DelayDSPKernel(DelayProcessor* processor)
-    : AudioDSPKernel(processor)
-    , m_maxDelayTime(DefaultMaxDelayTime)
-    , m_writeIndex(0)
-    , m_firstTime(true)
-{
-    ASSERT(processor && processor->sampleRate() > 0);
-    if (!processor)
-        return;
-
-    m_buffer.resize(static_cast<size_t>(processor->sampleRate() * DefaultMaxDelayTime));
-    m_buffer.zero();
-
-    m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, processor->sampleRate());
-}
-
-DelayDSPKernel::DelayDSPKernel(double maxDelayTime, double sampleRate)
-    : AudioDSPKernel(sampleRate)
-    , m_maxDelayTime(maxDelayTime)
-    , m_writeIndex(0)
-    , m_firstTime(true)
-{
-    ASSERT(maxDelayTime > 0.0);
-    if (maxDelayTime <= 0.0)
-        return;
-        
-    size_t bufferLength = static_cast<size_t>(sampleRate * maxDelayTime);
-    ASSERT(bufferLength);
-    if (!bufferLength)
-        return;
-    
-    m_buffer.resize(bufferLength);
-    m_buffer.zero();
-
-    m_smoothingRate = AudioUtilities::discreteTimeConstantForSampleRate(SmoothingTimeConstant, sampleRate);
-}
-
-void DelayDSPKernel::process(const float* source, float* destination, size_t framesToProcess)
-{
-    size_t bufferLength = m_buffer.size();
-    float* buffer = m_buffer.data();
-
-    ASSERT(bufferLength);
-    if (!bufferLength)
-        return;
-        
-    ASSERT(source && destination);
-    if (!source || !destination)
-        return;
-        
-    double sampleRate = this->sampleRate();
-    double delayTime = delayProcessor() ? delayProcessor()->delayTime()->value() : m_desiredDelayFrames / sampleRate;
-
-    // Make sure the delay time is in a valid range.
-    delayTime = min(maxDelayTime(), delayTime);
-    delayTime = max(0.0, delayTime);
-
-    if (m_firstTime) {
-        m_currentDelayTime = delayTime;
-        m_firstTime = false;
-    }
-    
-    int n = framesToProcess;
-    while (n--) {
-        // Approach desired delay time.
-        m_currentDelayTime += (delayTime - m_currentDelayTime) * m_smoothingRate;
-
-        double desiredDelayFrames = m_currentDelayTime * sampleRate;
-
-        double readPosition = m_writeIndex + bufferLength - desiredDelayFrames;
-        if (readPosition > bufferLength)
-            readPosition -= bufferLength;
-
-        // Linearly interpolate in-between delay times.
-        int readIndex1 = static_cast<int>(readPosition);
-        int readIndex2 = (readIndex1 + 1) % bufferLength;
-        double interpolationFactor = readPosition - readIndex1;
-
-        double input = static_cast<float>(*source++);
-        buffer[m_writeIndex] = static_cast<float>(input);
-        m_writeIndex = (m_writeIndex + 1) % bufferLength;        
-        
-        double sample1 = buffer[readIndex1];
-        double sample2 = buffer[readIndex2];
-        
-        double output = (1.0 - interpolationFactor) * sample1 + interpolationFactor * sample2;
-
-        *destination++ = static_cast<float>(output);
-    }        
-}
-
-void DelayDSPKernel::reset()
-{
-    m_firstTime = true;
-    m_buffer.zero();
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/DelayDSPKernel.h b/WebCore/webaudio/DelayDSPKernel.h
deleted file mode 100644
index 2ae36cb..0000000
--- a/WebCore/webaudio/DelayDSPKernel.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef DelayDSPKernel_h
-#define DelayDSPKernel_h
-
-#include "AudioArray.h"
-#include "AudioDSPKernel.h"
-#include "DelayProcessor.h"
-
-namespace WebCore {
-
-class DelayProcessor;
-    
-class DelayDSPKernel : public AudioDSPKernel {
-public:  
-    DelayDSPKernel(DelayProcessor*);
-    DelayDSPKernel(double maxDelayTime, double sampleRate);
-    
-    virtual void process(const float* source, float* destination, size_t framesToProcess);
-    virtual void reset();
-    
-    double maxDelayTime() const { return m_maxDelayTime; }
-    
-    void setDelayFrames(double numberOfFrames) { m_desiredDelayFrames = numberOfFrames; }
-    
-private:
-    AudioFloatArray m_buffer;
-    double m_maxDelayTime;
-    int m_writeIndex;
-    double m_currentDelayTime;
-    double m_smoothingRate;
-    bool m_firstTime;
-    double m_desiredDelayFrames;
-
-    DelayProcessor* delayProcessor() { return static_cast<DelayProcessor*>(processor()); }
-};
-
-} // namespace WebCore
-
-#endif // DelayDSPKernel_h
diff --git a/WebCore/webaudio/DelayNode.cpp b/WebCore/webaudio/DelayNode.cpp
deleted file mode 100644
index 29fceae..0000000
--- a/WebCore/webaudio/DelayNode.cpp
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * 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 "DelayNode.h"
-
-namespace WebCore {
-
-DelayNode::DelayNode(AudioContext* context, double sampleRate)
-    : AudioBasicProcessorNode(context, sampleRate)
-{
-    m_processor = adoptPtr(new DelayProcessor(sampleRate, 1));    
-    setType(NodeTypeDelay);
-}
-
-AudioParam* DelayNode::delayTime()
-{
-    return delayProcessor()->delayTime();
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/DelayNode.h b/WebCore/webaudio/DelayNode.h
deleted file mode 100644
index 93ad227..0000000
--- a/WebCore/webaudio/DelayNode.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef DelayNode_h
-#define DelayNode_h
-
-#include "AudioBasicProcessorNode.h"
-#include "DelayProcessor.h"
-#include <wtf/PassRefPtr.h>
-
-namespace WebCore {
-
-class AudioParam;
-
-class DelayNode : public AudioBasicProcessorNode {
-public:
-    static PassRefPtr<DelayNode> create(AudioContext* context, double sampleRate)
-    {
-        return adoptRef(new DelayNode(context, sampleRate));      
-    }
-
-    AudioParam* delayTime();
-
-private:
-    DelayNode(AudioContext*, double sampleRate);
-
-    DelayProcessor* delayProcessor() { return static_cast<DelayProcessor*>(processor()); }
-};
-
-} // namespace WebCore
-
-#endif // DelayNode_h
diff --git a/WebCore/webaudio/DelayNode.idl b/WebCore/webaudio/DelayNode.idl
deleted file mode 100644
index 7756627..0000000
--- a/WebCore/webaudio/DelayNode.idl
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO,
-        GenerateToJS
-    ] DelayNode : AudioNode {
-        readonly attribute AudioParam delayTime;
-    };
-}
diff --git a/WebCore/webaudio/DelayProcessor.cpp b/WebCore/webaudio/DelayProcessor.cpp
deleted file mode 100644
index 5fdc8df..0000000
--- a/WebCore/webaudio/DelayProcessor.cpp
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * 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 "DelayProcessor.h"
- 
-#include "DelayDSPKernel.h"
-
-namespace WebCore {
-
-DelayProcessor::DelayProcessor(double sampleRate, unsigned numberOfChannels)
-    : AudioDSPKernelProcessor(sampleRate, numberOfChannels)
-{
-    m_delayTime = AudioParam::create("delayTime", 0.0, 0.0, 1.0);
-}
-
-DelayProcessor::~DelayProcessor()
-{
-    if (isInitialized())
-        uninitialize();
-}
-
-PassOwnPtr<AudioDSPKernel> DelayProcessor::createKernel()
-{
-    return adoptPtr(new DelayDSPKernel(this));
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/DelayProcessor.h b/WebCore/webaudio/DelayProcessor.h
deleted file mode 100644
index 4844c4b..0000000
--- a/WebCore/webaudio/DelayProcessor.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef DelayProcessor_h
-#define DelayProcessor_h
-
-#include "AudioDSPKernelProcessor.h"
-#include "AudioParam.h"
-
-#include <wtf/PassOwnPtr.h>
-#include <wtf/RefPtr.h>
-
-namespace WebCore {
-
-class AudioDSPKernel;
-    
-class DelayProcessor : public AudioDSPKernelProcessor {
-public:
-    DelayProcessor(double sampleRate, unsigned numberOfChannels);
-    virtual ~DelayProcessor();
-    
-    virtual PassOwnPtr<AudioDSPKernel> createKernel();
-        
-    AudioParam* delayTime() const { return m_delayTime.get(); }
-
-private:
-    RefPtr<AudioParam> m_delayTime;
-};
-
-} // namespace WebCore
-
-#endif // DelayProcessor_h
diff --git a/WebCore/webaudio/HighPass2FilterNode.cpp b/WebCore/webaudio/HighPass2FilterNode.cpp
deleted file mode 100644
index ca33784..0000000
--- a/WebCore/webaudio/HighPass2FilterNode.cpp
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * 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 "HighPass2FilterNode.h"
-
-namespace WebCore {
-
-HighPass2FilterNode::HighPass2FilterNode(AudioContext* context, double sampleRate)
-    : AudioBasicProcessorNode(context, sampleRate)
-{
-    m_processor = adoptPtr(new BiquadProcessor(BiquadProcessor::HighPass2, sampleRate, 1, false));
-    setType(NodeTypeHighPass2Filter);
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/HighPass2FilterNode.h b/WebCore/webaudio/HighPass2FilterNode.h
deleted file mode 100644
index be0beb6..0000000
--- a/WebCore/webaudio/HighPass2FilterNode.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef HighPass2FilterNode_h
-#define HighPass2FilterNode_h
-
-#include "AudioBasicProcessorNode.h"
-#include "BiquadProcessor.h"
-
-namespace WebCore {
-
-class AudioParam;
-    
-class HighPass2FilterNode : public AudioBasicProcessorNode {
-public:
-    static PassRefPtr<HighPass2FilterNode> create(AudioContext* context, double sampleRate)
-    {
-        return adoptRef(new HighPass2FilterNode(context, sampleRate));      
-    }
-
-    AudioParam* cutoff() { return biquadProcessor()->parameter1(); }
-    AudioParam* resonance() { return biquadProcessor()->parameter2(); }
-    
-private:
-    HighPass2FilterNode(AudioContext*, double sampleRate);
-
-    BiquadProcessor* biquadProcessor() { return static_cast<BiquadProcessor*>(processor()); }
-};
-
-} // namespace WebCore
-
-#endif // HighPass2FilterNode_h
diff --git a/WebCore/webaudio/HighPass2FilterNode.idl b/WebCore/webaudio/HighPass2FilterNode.idl
deleted file mode 100644
index 399f9b5..0000000
--- a/WebCore/webaudio/HighPass2FilterNode.idl
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    // Two-pole highpass filter
-    // FIXME: design BiquadNode and use instead of this
-    interface [
-        Conditional=WEB_AUDIO,
-        GenerateToJS
-    ] HighPass2FilterNode : AudioNode {
-        readonly attribute AudioParam cutoff;
-        readonly attribute AudioParam resonance;
-    };
-}
diff --git a/WebCore/webaudio/JavaScriptAudioNode.cpp b/WebCore/webaudio/JavaScriptAudioNode.cpp
deleted file mode 100644
index 15a8cf7..0000000
--- a/WebCore/webaudio/JavaScriptAudioNode.cpp
+++ /dev/null
@@ -1,272 +0,0 @@
-/*
- * 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 "JavaScriptAudioNode.h"
-
-#include "AudioBuffer.h"
-#include "AudioBus.h"
-#include "AudioContext.h"
-#include "AudioNodeInput.h"
-#include "AudioNodeOutput.h"
-#include "AudioProcessingEvent.h"
-#include "Document.h"
-#include "Float32Array.h"
-#include <wtf/MainThread.h>
-
-namespace WebCore {
-
-const size_t DefaultBufferSize = 4096;
-
-PassRefPtr<JavaScriptAudioNode> JavaScriptAudioNode::create(AudioContext* context, double sampleRate, size_t bufferSize, unsigned numberOfInputs, unsigned numberOfOutputs)
-{
-    return adoptRef(new JavaScriptAudioNode(context, sampleRate, bufferSize, numberOfInputs, numberOfOutputs));
-}
-
-JavaScriptAudioNode::JavaScriptAudioNode(AudioContext* context, double sampleRate, size_t bufferSize, unsigned numberOfInputs, unsigned numberOfOutputs)
-    : AudioNode(context, sampleRate)
-    , m_doubleBufferIndex(0)
-    , m_doubleBufferIndexForEvent(0)
-    , m_bufferSize(bufferSize)
-    , m_bufferReadWriteIndex(0)
-    , m_isRequestOutstanding(false)
-{
-    // Check for valid buffer size.
-    switch (bufferSize) {
-    case 256:
-    case 512:
-    case 1024:
-    case 2048:
-    case 4096:
-    case 8192:
-    case 16384:
-        m_bufferSize = bufferSize;
-        break;
-    default:
-        m_bufferSize = DefaultBufferSize;
-    }
-        
-    // Regardless of the allowed buffer sizes above, we still need to process at the granularity of the AudioNode.
-    if (m_bufferSize < AudioNode::ProcessingSizeInFrames)
-        m_bufferSize = AudioNode::ProcessingSizeInFrames;
-
-    // FIXME: Right now we're hardcoded to single input and single output.
-    // Although the specification says this is OK for a simple implementation, multiple inputs and outputs would be good.
-    ASSERT_UNUSED(numberOfInputs, numberOfInputs == 1);
-    ASSERT_UNUSED(numberOfOutputs, numberOfOutputs == 1);
-    addInput(adoptPtr(new AudioNodeInput(this)));
-    addOutput(adoptPtr(new AudioNodeOutput(this, 2)));
-
-    setType(NodeTypeJavaScript);
-
-    initialize();
-}
-
-JavaScriptAudioNode::~JavaScriptAudioNode()
-{
-    uninitialize();
-}
-
-void JavaScriptAudioNode::initialize()
-{
-    if (isInitialized())
-        return;
-
-    double sampleRate = context()->sampleRate();
-
-    // Create double buffers on both the input and output sides.
-    // These AudioBuffers will be directly accessed in the main thread by JavaScript.
-    for (unsigned i = 0; i < 2; ++i) {
-        m_inputBuffers.append(AudioBuffer::create(2, bufferSize(), sampleRate));
-        m_outputBuffers.append(AudioBuffer::create(2, bufferSize(), sampleRate));
-    }
-
-    AudioNode::initialize();
-}
-
-void JavaScriptAudioNode::uninitialize()
-{
-    if (!isInitialized())
-        return;
-
-    m_inputBuffers.clear();
-    m_outputBuffers.clear();
-
-    AudioNode::uninitialize();
-}
-
-JavaScriptAudioNode* JavaScriptAudioNode::toJavaScriptAudioNode()
-{
-    return this;
-}
-
-void JavaScriptAudioNode::process(size_t framesToProcess)
-{
-    // Discussion about inputs and outputs:
-    // As in other AudioNodes, JavaScriptAudioNode uses an AudioBus for its input and output (see inputBus and outputBus below).
-    // Additionally, there is a double-buffering for input and output which is exposed directly to JavaScript (see inputBuffer and outputBuffer below).
-    // This node is the producer for inputBuffer and the consumer for outputBuffer.
-    // The JavaScript code is the consumer of inputBuffer and the producer for outputBuffer.
-    
-    // Get input and output busses.
-    AudioBus* inputBus = this->input(0)->bus();
-    AudioBus* outputBus = this->output(0)->bus();
-
-    // Get input and output buffers.  We double-buffer both the input and output sides.
-    unsigned doubleBufferIndex = this->doubleBufferIndex();
-    bool isDoubleBufferIndexGood = doubleBufferIndex < 2 && doubleBufferIndex < m_inputBuffers.size() && doubleBufferIndex < m_outputBuffers.size();
-    ASSERT(isDoubleBufferIndexGood);
-    if (!isDoubleBufferIndexGood)
-        return;
-    
-    AudioBuffer* inputBuffer = m_inputBuffers[doubleBufferIndex].get();
-    AudioBuffer* outputBuffer = m_outputBuffers[doubleBufferIndex].get();
-
-    // Check the consistency of input and output buffers.
-    bool buffersAreGood = inputBuffer && outputBuffer && bufferSize() == inputBuffer->length() && bufferSize() == outputBuffer->length()
-        && m_bufferReadWriteIndex + framesToProcess <= bufferSize();
-    ASSERT(buffersAreGood);
-    if (!buffersAreGood)
-        return;
-
-    // We assume that bufferSize() is evenly divisible by framesToProcess - should always be true, but we should still check.
-    bool isFramesToProcessGood = framesToProcess && bufferSize() >= framesToProcess && !(bufferSize() % framesToProcess);
-    ASSERT(isFramesToProcessGood);
-    if (!isFramesToProcessGood)
-        return;
-        
-    unsigned numberOfInputChannels = inputBus->numberOfChannels();
-    
-    bool channelsAreGood = (numberOfInputChannels == 1 || numberOfInputChannels == 2) && outputBus->numberOfChannels() == 2;
-    ASSERT(channelsAreGood);
-    if (!channelsAreGood)
-        return;
-
-    float* sourceL = inputBus->channel(0)->data();
-    float* sourceR = numberOfInputChannels > 1 ? inputBus->channel(1)->data() : 0;
-    float* destinationL = outputBus->channel(0)->data();
-    float* destinationR = outputBus->channel(1)->data();
-
-    // Copy from the input to the input buffer.  See "buffersAreGood" check above for safety.
-    size_t bytesToCopy = sizeof(float) * framesToProcess;
-    memcpy(inputBuffer->getChannelData(0)->data() + m_bufferReadWriteIndex, sourceL, bytesToCopy);
-    
-    if (numberOfInputChannels == 2)
-        memcpy(inputBuffer->getChannelData(1)->data() + m_bufferReadWriteIndex, sourceR, bytesToCopy);
-    else if (numberOfInputChannels == 1) {
-        // If the input is mono, then also copy the mono input to the right channel of the AudioBuffer which the AudioProcessingEvent uses.
-        // FIXME: it is likely the audio API will evolve to present an AudioBuffer with the same number of channels as our input.
-        memcpy(inputBuffer->getChannelData(1)->data() + m_bufferReadWriteIndex, sourceL, bytesToCopy);
-    }
-    
-    // Copy from the output buffer to the output.  See "buffersAreGood" check above for safety.
-    memcpy(destinationL, outputBuffer->getChannelData(0)->data() + m_bufferReadWriteIndex, bytesToCopy);
-    memcpy(destinationR, outputBuffer->getChannelData(1)->data() + m_bufferReadWriteIndex, bytesToCopy);
-
-    // Update the buffering index.
-    m_bufferReadWriteIndex = (m_bufferReadWriteIndex + framesToProcess) % bufferSize();
-
-    // m_bufferReadWriteIndex will wrap back around to 0 when the current input and output buffers are full.
-    // When this happens, fire an event and swap buffers.
-    if (!m_bufferReadWriteIndex) {
-        // Avoid building up requests on the main thread to fire process events when they're not being handled.
-        // This could be a problem if the main thread is very busy doing other things and is being held up handling previous requests.
-        if (m_isRequestOutstanding) {
-            // We're late in handling the previous request.  The main thread must be very busy.
-            // The best we can do is clear out the buffer ourself here.
-            outputBuffer->zero();            
-        } else {
-            // Reference ourself so we don't accidentally get deleted before fireProcessEvent() gets called.
-            ref();
-            
-            // Fire the event on the main thread, not this one (which is the realtime audio thread).
-            m_doubleBufferIndexForEvent = m_doubleBufferIndex;
-            callOnMainThread(fireProcessEventDispatch, this);
-            m_isRequestOutstanding = true;
-        }
-
-        swapBuffers();
-    }
-}
-
-void JavaScriptAudioNode::fireProcessEventDispatch(void* userData)
-{
-    JavaScriptAudioNode* jsAudioNode = static_cast<JavaScriptAudioNode*>(userData);
-    ASSERT(jsAudioNode);
-    if (!jsAudioNode)
-        return;
-
-    jsAudioNode->fireProcessEvent();
-
-    // De-reference to match the ref() call in process().
-    jsAudioNode->deref();
-}
-
-void JavaScriptAudioNode::fireProcessEvent()
-{
-    ASSERT(isMainThread() && m_isRequestOutstanding);
-    
-    bool isIndexGood = m_doubleBufferIndexForEvent < 2;
-    ASSERT(isIndexGood);
-    if (!isIndexGood)
-        return;
-        
-    AudioBuffer* inputBuffer = m_inputBuffers[m_doubleBufferIndexForEvent].get();
-    AudioBuffer* outputBuffer = m_outputBuffers[m_doubleBufferIndexForEvent].get();
-    ASSERT(inputBuffer && outputBuffer);
-    if (!inputBuffer || !outputBuffer)
-        return;
-
-    // Avoid firing the event if the document has already gone away.
-    if (context()->hasDocument()) {
-        // Let the audio thread know we've gotten to the point where it's OK for it to make another request.
-        m_isRequestOutstanding = false;
-        
-        // Call the JavaScript event handler which will do the audio processing.
-        dispatchEvent(AudioProcessingEvent::create(inputBuffer, outputBuffer));
-    }
-}
-
-void JavaScriptAudioNode::reset()
-{
-    m_bufferReadWriteIndex = 0;
-    m_doubleBufferIndex = 0;
-
-    for (unsigned i = 0; i < 2; ++i) {
-        m_inputBuffers[i]->zero();
-        m_outputBuffers[i]->zero();
-    }
-}
-
-ScriptExecutionContext* JavaScriptAudioNode::scriptExecutionContext() const
-{
-    return const_cast<JavaScriptAudioNode*>(this)->context()->document();
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/JavaScriptAudioNode.h b/WebCore/webaudio/JavaScriptAudioNode.h
deleted file mode 100644
index e99a25d..0000000
--- a/WebCore/webaudio/JavaScriptAudioNode.h
+++ /dev/null
@@ -1,104 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef JavaScriptAudioNode_h
-#define JavaScriptAudioNode_h
-
-#include "ActiveDOMObject.h"
-#include "AudioNode.h"
-#include "EventListener.h"
-#include "EventTarget.h"
-#include <wtf/PassRefPtr.h>
-#include <wtf/RefPtr.h>
-#include <wtf/Vector.h>
-
-namespace WebCore {
-
-class AudioBuffer;
-class AudioContext;
-class AudioProcessingEvent;
-class Float32Array;
-
-// JavaScriptAudioNode is an AudioNode which allows for arbitrary synthesis or processing directly using JavaScript.
-// The API allows for a variable number of inputs and outputs, although it must have at least one input or output.
-// This basic implementation supports no more than one input and output.
-// The "onaudioprocess" attribute is an event listener which will get called periodically with an AudioProcessingEvent which has
-// AudioBuffers for each input and output.
-
-class JavaScriptAudioNode : public AudioNode, public EventTarget {
-public:
-    // bufferSize must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384.
-    // This value controls how frequently the onaudioprocess event handler is called and how many sample-frames need to be processed each call.
-    // Lower numbers for bufferSize will result in a lower (better) latency. Higher numbers will be necessary to avoid audio breakup and glitches.
-    // The value chosen must carefully balance between latency and audio quality.
-    static PassRefPtr<JavaScriptAudioNode> create(AudioContext*, double sampleRate, size_t bufferSize, unsigned numberOfInputs = 1, unsigned numberOfOutputs = 1);
-
-    virtual ~JavaScriptAudioNode();
-
-    // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void reset();
-    virtual void initialize();
-    virtual void uninitialize();
-
-    // EventTarget
-    virtual ScriptExecutionContext* scriptExecutionContext() const;
-    virtual JavaScriptAudioNode* toJavaScriptAudioNode();
-    virtual EventTargetData* eventTargetData() { return &m_eventTargetData; }
-    virtual EventTargetData* ensureEventTargetData()  { return &m_eventTargetData; }
-
-    size_t bufferSize() const { return m_bufferSize; }
-
-    DEFINE_ATTRIBUTE_EVENT_LISTENER(audioprocess);
-
-    // Reconcile ref/deref which are defined both in AudioNode and EventTarget.
-    using AudioNode::ref;
-    using AudioNode::deref;
-    
-private:
-    JavaScriptAudioNode(AudioContext*, double sampleRate, size_t bufferSize, unsigned numberOfInputs, unsigned numberOfOutputs);
-
-    static void fireProcessEventDispatch(void* userData);
-    void fireProcessEvent();
-
-    // Double buffering
-    unsigned doubleBufferIndex() const { return m_doubleBufferIndex; }
-    void swapBuffers() { m_doubleBufferIndex = 1 - m_doubleBufferIndex; }
-    unsigned m_doubleBufferIndex;
-    unsigned m_doubleBufferIndexForEvent;
-    Vector<RefPtr<AudioBuffer> > m_inputBuffers;
-    Vector<RefPtr<AudioBuffer> > m_outputBuffers;
-
-    virtual void refEventTarget() { ref(); }
-    virtual void derefEventTarget() { deref(); }
-    EventTargetData m_eventTargetData;
-
-    size_t m_bufferSize;
-    unsigned m_bufferReadWriteIndex;
-    volatile bool m_isRequestOutstanding;
-};
-
-} // namespace WebCore
-
-#endif // JavaScriptAudioNode_h
diff --git a/WebCore/webaudio/JavaScriptAudioNode.idl b/WebCore/webaudio/JavaScriptAudioNode.idl
deleted file mode 100644
index ef5359b..0000000
--- a/WebCore/webaudio/JavaScriptAudioNode.idl
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    // For real-time audio stream synthesis/processing in JavaScript 
-    interface [
-        Conditional=WEB_AUDIO,
-        GenerateToJS,
-        CustomMarkFunction,
-#if defined(V8_BINDING) && V8_BINDING
-        EventTarget
-#endif
-    ] JavaScriptAudioNode : AudioNode {
-        // Rendering callback
-        attribute EventListener onaudioprocess;
-        
-        readonly attribute long bufferSize;
-    };
-}
diff --git a/WebCore/webaudio/LowPass2FilterNode.cpp b/WebCore/webaudio/LowPass2FilterNode.cpp
deleted file mode 100644
index 691f4ed..0000000
--- a/WebCore/webaudio/LowPass2FilterNode.cpp
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * 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 "LowPass2FilterNode.h"
-
-namespace WebCore {
-
-LowPass2FilterNode::LowPass2FilterNode(AudioContext* context, double sampleRate)
-    : AudioBasicProcessorNode(context, sampleRate)
-{
-    m_processor = adoptPtr(new BiquadProcessor(BiquadProcessor::LowPass2, sampleRate, 1, false));
-    setType(NodeTypeLowPass2Filter);
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/LowPass2FilterNode.h b/WebCore/webaudio/LowPass2FilterNode.h
deleted file mode 100644
index 43d7051..0000000
--- a/WebCore/webaudio/LowPass2FilterNode.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef LowPass2FilterNode_h
-#define LowPass2FilterNode_h
-
-#include "AudioBasicProcessorNode.h"
-#include "BiquadProcessor.h"
-
-namespace WebCore {
-
-class AudioParam;
-    
-class LowPass2FilterNode : public AudioBasicProcessorNode {
-public:
-    static PassRefPtr<LowPass2FilterNode> create(AudioContext* context, double sampleRate)
-    {
-        return adoptRef(new LowPass2FilterNode(context, sampleRate));      
-    }
-
-    AudioParam* cutoff() { return biquadProcessor()->parameter1(); }
-    AudioParam* resonance() { return biquadProcessor()->parameter2(); }
-    
-private:
-    LowPass2FilterNode(AudioContext*, double sampleRate);
-
-    BiquadProcessor* biquadProcessor() { return static_cast<BiquadProcessor*>(processor()); }
-};
-
-} // namespace WebCore
-
-#endif // LowPass2FilterNode_h
diff --git a/WebCore/webaudio/LowPass2FilterNode.idl b/WebCore/webaudio/LowPass2FilterNode.idl
deleted file mode 100644
index 310c21e..0000000
--- a/WebCore/webaudio/LowPass2FilterNode.idl
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    // Two-pole lowpass filter
-    // FIXME: design BiquadNode and use instead of this
-    interface [
-        Conditional=WEB_AUDIO,
-        GenerateToJS
-    ] LowPass2FilterNode : AudioNode {
-        readonly attribute AudioParam cutoff;
-        readonly attribute AudioParam resonance;
-    };
-}
diff --git a/WebCore/webaudio/RealtimeAnalyser.cpp b/WebCore/webaudio/RealtimeAnalyser.cpp
deleted file mode 100644
index 30a7de1..0000000
--- a/WebCore/webaudio/RealtimeAnalyser.cpp
+++ /dev/null
@@ -1,301 +0,0 @@
-/*
- * 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 "RealtimeAnalyser.h"
-
-#include "AudioBus.h"
-#include "AudioUtilities.h"
-#include "FFTFrame.h"
-
-#if ENABLE(3D_CANVAS)
-#include "Float32Array.h"
-#include "Uint8Array.h"
-#endif
-
-#include <algorithm>
-#include <limits.h>
-#include <wtf/Complex.h>
-#include <wtf/MathExtras.h>
-#include <wtf/Threading.h>
-
-using namespace std;
-
-namespace WebCore {
-
-const double RealtimeAnalyser::DefaultSmoothingTimeConstant  = 0.8;
-const double RealtimeAnalyser::DefaultMinDecibels = -100.0;
-const double RealtimeAnalyser::DefaultMaxDecibels = -30.0;
-
-const unsigned RealtimeAnalyser::DefaultFFTSize = 2048;
-const unsigned RealtimeAnalyser::MaxFFTSize = 2048;
-const unsigned RealtimeAnalyser::InputBufferSize = RealtimeAnalyser::MaxFFTSize * 2;
-
-RealtimeAnalyser::RealtimeAnalyser()
-    : m_inputBuffer(InputBufferSize)
-    , m_writeIndex(0)
-    , m_fftSize(DefaultFFTSize)
-    , m_magnitudeBuffer(DefaultFFTSize / 2)
-    , m_smoothingTimeConstant(DefaultSmoothingTimeConstant)
-    , m_minDecibels(DefaultMinDecibels)
-    , m_maxDecibels(DefaultMaxDecibels)
-{
-    m_analysisFrame = adoptPtr(new FFTFrame(DefaultFFTSize));
-}
-
-RealtimeAnalyser::~RealtimeAnalyser()
-{
-}
-
-void RealtimeAnalyser::reset()
-{
-    m_writeIndex = 0;
-    m_inputBuffer.zero();
-    m_magnitudeBuffer.zero();
-}
-
-void RealtimeAnalyser::setFftSize(size_t size)
-{
-    ASSERT(isMainThread());
-
-    // Only allow powers of two.
-    unsigned log2size = static_cast<unsigned>(log2(size));
-    bool isPOT(1UL << log2size == size);
-    
-    if (!isPOT || size > MaxFFTSize) {
-        // FIXME: It would be good to also set an exception.
-        return;
-    }
-
-    if (m_fftSize != size) {
-        m_analysisFrame = adoptPtr(new FFTFrame(m_fftSize));
-        m_magnitudeBuffer.resize(size);
-        m_fftSize = size;
-    }
-}
-
-void RealtimeAnalyser::writeInput(AudioBus* bus, size_t framesToProcess)
-{
-    bool isBusGood = bus && bus->numberOfChannels() > 0 && bus->channel(0)->length() >= framesToProcess;
-    ASSERT(isBusGood);
-    if (!isBusGood)
-        return;
-        
-    // FIXME : allow to work with non-FFTSize divisible chunking
-    bool isDestinationGood = m_writeIndex < m_inputBuffer.size() && m_writeIndex + framesToProcess <= m_inputBuffer.size();
-    ASSERT(isDestinationGood);
-    if (!isDestinationGood)
-        return;    
-    
-    // Perform real-time analysis
-    // FIXME : for now just use left channel (must mix if stereo source)
-    float* source = bus->channel(0)->data();
-
-    // The source has already been sanity checked with isBusGood above.
-    
-    memcpy(m_inputBuffer.data() + m_writeIndex, source, sizeof(float) * framesToProcess);
-
-    m_writeIndex += framesToProcess;
-    if (m_writeIndex >= InputBufferSize)
-        m_writeIndex = 0;
-}
-
-namespace {
-
-void applyWindow(float* p, size_t n)
-{
-    ASSERT(isMainThread());
-    
-    // Blackman window
-    double alpha = 0.16;
-    double a0 = 0.5 * (1.0 - alpha);
-    double a1 = 0.5;
-    double a2 = 0.5 * alpha;
-    
-    for (unsigned i = 0; i < n; ++i) {
-        double x = static_cast<double>(i) / static_cast<double>(n);
-        double window = a0 - a1 * cos(2.0 * piDouble * x) + a2 * cos(4.0 * piDouble * x);
-        p[i] *= float(window);
-    }
-}
-
-} // namespace
-
-void RealtimeAnalyser::doFFTAnalysis()
-{    
-    ASSERT(isMainThread());
-
-    // Unroll the input buffer into a temporary buffer, where we'll apply an analysis window followed by an FFT.
-    size_t fftSize = this->fftSize();
-    
-    AudioFloatArray temporaryBuffer(fftSize);
-    float* inputBuffer = m_inputBuffer.data();
-    float* tempP = temporaryBuffer.data();
-
-    // Take the previous fftSize values from the input buffer and copy into the temporary buffer.
-    // FIXME : optimize with memcpy().
-    unsigned writeIndex = m_writeIndex;
-    for (unsigned i = 0; i < fftSize; ++i)
-        tempP[i] = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
-    
-    // Window the input samples.
-    applyWindow(tempP, fftSize);
-    
-    // Do the analysis.
-    m_analysisFrame->doFFT(tempP);
-
-    size_t n = DefaultFFTSize / 2;
-
-    float* realP = m_analysisFrame->realData();
-    float* imagP = m_analysisFrame->imagData();
-
-    // Blow away the packed nyquist component.
-    imagP[0] = 0.0f;
-    
-    // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT scaling factor).
-    const double MagnitudeScale = 1.0 / DefaultFFTSize;
-
-    // A value of 0 does no averaging with the previous result.  Larger values produce slower, but smoother changes.
-    double k = m_smoothingTimeConstant;
-    k = max(0.0, k);
-    k = min(1.0, k);    
-    
-    // Convert the analysis data from complex to magnitude and average with the previous result.
-    float* destination = magnitudeBuffer().data();
-    for (unsigned i = 0; i < n; ++i) {
-        Complex c(realP[i], imagP[i]);
-        double scalarMagnitude = abs(c) * MagnitudeScale;        
-        destination[i] = float(k * destination[i] + (1.0 - k) * scalarMagnitude);
-    }
-}
-
-#if ENABLE(3D_CANVAS)
-
-void RealtimeAnalyser::getFloatFrequencyData(Float32Array* destinationArray)
-{
-    ASSERT(isMainThread());
-
-    if (!destinationArray)
-        return;
-        
-    doFFTAnalysis();
-    
-    // Convert from linear magnitude to floating-point decibels.
-    const double MinDecibels = m_minDecibels;
-    unsigned sourceLength = magnitudeBuffer().size();
-    size_t len = min(sourceLength, destinationArray->length());
-    if (len > 0) {
-        const float* source = magnitudeBuffer().data();
-        float* destination = destinationArray->data();
-        
-        for (unsigned i = 0; i < len; ++i) {
-            float linearValue = source[i];
-            double dbMag = !linearValue ? MinDecibels : AudioUtilities::linearToDecibels(linearValue);
-            destination[i] = float(dbMag);
-        }
-    }
-}
-
-void RealtimeAnalyser::getByteFrequencyData(Uint8Array* destinationArray)
-{
-    ASSERT(isMainThread());
-
-    if (!destinationArray)
-        return;
-        
-    doFFTAnalysis();
-    
-    // Convert from linear magnitude to unsigned-byte decibels.
-    unsigned sourceLength = magnitudeBuffer().size();
-    size_t len = min(sourceLength, destinationArray->length());
-    if (len > 0) {
-        const double RangeScaleFactor = m_maxDecibels == m_minDecibels ? 1.0 : 1.0 / (m_maxDecibels - m_minDecibels);
-
-        const float* source = magnitudeBuffer().data();
-        unsigned char* destination = destinationArray->data();        
-        
-        for (unsigned i = 0; i < len; ++i) {
-            float linearValue = source[i];
-            double dbMag = !linearValue ? m_minDecibels : AudioUtilities::linearToDecibels(linearValue);
-            
-            // The range m_minDecibels to m_maxDecibels will be scaled to byte values from 0 to UCHAR_MAX.
-            double scaledValue = UCHAR_MAX * (dbMag - m_minDecibels) * RangeScaleFactor;
-
-            // Clip to valid range.
-            if (scaledValue < 0.0)
-                scaledValue = 0.0;
-            if (scaledValue > UCHAR_MAX)
-                scaledValue = UCHAR_MAX;
-            
-            destination[i] = static_cast<unsigned char>(scaledValue);
-        }
-    }
-}
-
-void RealtimeAnalyser::getByteTimeDomainData(Uint8Array* destinationArray)
-{
-    ASSERT(isMainThread());
-
-    if (!destinationArray)
-        return;
-        
-    unsigned fftSize = this->fftSize();
-    size_t len = min(fftSize, destinationArray->length());
-    if (len > 0) {
-        bool isInputBufferGood = m_inputBuffer.size() == InputBufferSize && m_inputBuffer.size() > fftSize;
-        ASSERT(isInputBufferGood);
-        if (!isInputBufferGood)
-            return;
-
-        float* inputBuffer = m_inputBuffer.data();        
-        unsigned char* destination = destinationArray->data();
-        
-        unsigned writeIndex = m_writeIndex;
-
-        for (unsigned i = 0; i < len; ++i) {
-            // Buffer access is protected due to modulo operation.
-            float value = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
-
-            // Scale from nominal -1.0 -> +1.0 to unsigned byte.
-            double scaledValue = 128.0 * (value + 1.0);
-
-            // Clip to valid range.
-            if (scaledValue < 0.0)
-                scaledValue = 0.0;
-            if (scaledValue > UCHAR_MAX)
-                scaledValue = UCHAR_MAX;
-            
-            destination[i] = static_cast<unsigned char>(scaledValue);
-        }
-    }
-}
-
-#endif // 3D_CANVAS
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/RealtimeAnalyser.h b/WebCore/webaudio/RealtimeAnalyser.h
deleted file mode 100644
index 686c17c..0000000
--- a/WebCore/webaudio/RealtimeAnalyser.h
+++ /dev/null
@@ -1,103 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef RealtimeAnalyser_h
-#define RealtimeAnalyser_h
-
-#include "AudioArray.h"
-#include <wtf/NonCopyable.h>
-#include <wtf/OwnPtr.h>
-
-namespace WebCore {
-
-class AudioBus;
-class FFTFrame;
-
-#if ENABLE(3D_CANVAS)
-class Float32Array;
-class Uint8Array;
-#endif
-
-class RealtimeAnalyser : public Noncopyable {
-public:
-    RealtimeAnalyser();
-    virtual ~RealtimeAnalyser();
-    
-    void reset();
-
-    size_t fftSize() const { return m_fftSize; }
-    void setFftSize(size_t size);
-
-    unsigned frequencyBinCount() const { return m_fftSize / 2; }
-
-    void setMinDecibels(float k) { m_minDecibels = k; }
-    float minDecibels() const { return static_cast<float>(m_minDecibels); }
-
-    void setMaxDecibels(float k) { m_maxDecibels = k; }
-    float maxDecibels() const { return static_cast<float>(m_maxDecibels); }
-
-    void setSmoothingTimeConstant(float k) { m_smoothingTimeConstant = k; }
-    float smoothingTimeConstant() const { return static_cast<float>(m_smoothingTimeConstant); }
-
-#if ENABLE(3D_CANVAS)
-    void getFloatFrequencyData(Float32Array*);
-    void getByteFrequencyData(Uint8Array*);
-    void getByteTimeDomainData(Uint8Array*);
-#endif
-
-    // The audio thread writes input data here.
-    void writeInput(AudioBus*, size_t framesToProcess);
-
-    static const double DefaultSmoothingTimeConstant;
-    static const double DefaultMinDecibels;
-    static const double DefaultMaxDecibels;
-
-    static const unsigned DefaultFFTSize;
-    static const unsigned MaxFFTSize;
-    static const unsigned InputBufferSize;
-
-private:
-    // The audio thread writes the input audio here.
-    AudioFloatArray m_inputBuffer;
-    unsigned m_writeIndex;
-    
-    size_t m_fftSize;
-    OwnPtr<FFTFrame> m_analysisFrame;
-    void doFFTAnalysis();
-    
-    // doFFTAnalysis() stores the floating-point magnitude analysis data here.
-    AudioFloatArray m_magnitudeBuffer;
-    AudioFloatArray& magnitudeBuffer() { return m_magnitudeBuffer; }
-
-    // A value between 0 and 1 which averages the previous version of m_magnitudeBuffer with the current analysis magnitude data.
-    double m_smoothingTimeConstant;    
-
-    // The range used when converting when using getByteFrequencyData(). 
-    double m_minDecibels;
-    double m_maxDecibels;
-};
-
-} // namespace WebCore
-
-#endif // RealtimeAnalyser_h
diff --git a/WebCore/webaudio/RealtimeAnalyserNode.cpp b/WebCore/webaudio/RealtimeAnalyserNode.cpp
deleted file mode 100644
index 2ba751a..0000000
--- a/WebCore/webaudio/RealtimeAnalyserNode.cpp
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- * 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 "RealtimeAnalyserNode.h"
-
-#include "AudioNodeInput.h"
-#include "AudioNodeOutput.h"
-
-namespace WebCore {
-
-RealtimeAnalyserNode::RealtimeAnalyserNode(AudioContext* context, double sampleRate)
-    : AudioNode(context, sampleRate)
-{
-    addInput(adoptPtr(new AudioNodeInput(this)));
-    addOutput(adoptPtr(new AudioNodeOutput(this, 2)));
-    
-    setType(NodeTypeAnalyser);
-    
-    initialize();
-}
-
-RealtimeAnalyserNode::~RealtimeAnalyserNode()
-{
-    uninitialize();
-}
-
-void RealtimeAnalyserNode::process(size_t framesToProcess)
-{
-    AudioBus* outputBus = output(0)->bus();
-
-    if (!isInitialized() || !input(0)->isConnected()) {
-        outputBus->zero();
-        return;
-    }
-
-    AudioBus* inputBus = input(0)->bus();
-    
-    // Give the analyser the audio which is passing through this AudioNode.
-    m_analyser.writeInput(inputBus, framesToProcess);
-
-    // For in-place processing, our override of pullInputs() will just pass the audio data through unchanged if the channel count matches from input to output
-    // (resulting in inputBus == outputBus).  Otherwise, do an up-mix to stereo.
-    if (inputBus != outputBus)
-        outputBus->copyFrom(*inputBus);
-}
-
-// We override pullInputs() as an optimization allowing this node to take advantage of in-place processing,
-// where the input is simply passed through unprocessed to the output.
-// Note: this only applies if the input and output channel counts match.
-void RealtimeAnalyserNode::pullInputs(size_t framesToProcess)
-{
-    // Render input stream - try to render directly into output bus for pass-through processing where process() doesn't need to do anything...
-    input(0)->pull(output(0)->bus(), framesToProcess);
-}
-
-void RealtimeAnalyserNode::reset()
-{
-    m_analyser.reset();
-}
-
-} // namespace WebCore
-
-#endif // ENABLE(WEB_AUDIO)
diff --git a/WebCore/webaudio/RealtimeAnalyserNode.h b/WebCore/webaudio/RealtimeAnalyserNode.h
deleted file mode 100644
index 9f62464..0000000
--- a/WebCore/webaudio/RealtimeAnalyserNode.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef RealtimeAnalyserNode_h
-#define RealtimeAnalyserNode_h
-
-#include "AudioNode.h"
-#include "RealtimeAnalyser.h"
-
-namespace WebCore {
-
-class RealtimeAnalyserNode : public AudioNode {
-public:
-    static PassRefPtr<RealtimeAnalyserNode> create(AudioContext* context, double sampleRate)
-    {
-        return adoptRef(new RealtimeAnalyserNode(context, sampleRate));      
-    }
-
-    virtual ~RealtimeAnalyserNode();
-    
-    // AudioNode
-    virtual void process(size_t framesToProcess);
-    virtual void pullInputs(size_t framesToProcess);
-    virtual void reset();
-
-    // Javascript bindings
-    unsigned int fftSize() const { return m_analyser.fftSize(); }
-    void setFftSize(unsigned int size) { m_analyser.setFftSize(size); }
-
-    unsigned frequencyBinCount() const { return m_analyser.frequencyBinCount(); }
-
-    void setMinDecibels(float k) { m_analyser.setMinDecibels(k); }
-    float minDecibels() const { return m_analyser.minDecibels(); }
-
-    void setMaxDecibels(float k) { m_analyser.setMaxDecibels(k); }
-    float maxDecibels() const { return m_analyser.maxDecibels(); }
-
-    void setSmoothingTimeConstant(float k) { m_analyser.setSmoothingTimeConstant(k); }
-    float smoothingTimeConstant() const { return m_analyser.smoothingTimeConstant(); }
-
-#if ENABLE(3D_CANVAS)
-    void getFloatFrequencyData(Float32Array* array) { m_analyser.getFloatFrequencyData(array); }
-    void getByteFrequencyData(Uint8Array* array) { m_analyser.getByteFrequencyData(array); }
-    void getByteTimeDomainData(Uint8Array* array) { m_analyser.getByteTimeDomainData(array); }
-#endif
-
-private:
-    RealtimeAnalyserNode(AudioContext*, double sampleRate);
-
-    RealtimeAnalyser m_analyser;
-};
-
-} // namespace WebCore
-
-#endif // RealtimeAnalyserNode_h
diff --git a/WebCore/webaudio/RealtimeAnalyserNode.idl b/WebCore/webaudio/RealtimeAnalyserNode.idl
deleted file mode 100644
index 5b2b223..0000000
--- a/WebCore/webaudio/RealtimeAnalyserNode.idl
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * 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.
- */
-
-module audio {
-    interface [
-        Conditional=WEB_AUDIO,
-        GenerateToJS
-    ] RealtimeAnalyserNode : AudioNode {
-        attribute unsigned long fftSize;
-        readonly attribute unsigned long frequencyBinCount;
-
-        // minDecibels / maxDecibels represent the range to scale the FFT analysis data for conversion to unsigned byte values.
-        attribute float minDecibels;
-        attribute float maxDecibels;
-        
-        // A value from 0.0 -> 1.0 where 0.0 represents no time averaging with the last analysis frame.
-        attribute float smoothingTimeConstant;
-        
-        // Copies the current frequency data into the passed array.
-        // If the array has fewer elements than the frequencyBinCount, the excess elements will be dropped.
-        [Conditional=3D_CANVAS] void getFloatFrequencyData(in Float32Array array);
-        [Conditional=3D_CANVAS] void getByteFrequencyData(in Uint8Array array);
-
-        // Real-time waveform data
-        [Conditional=3D_CANVAS] void getByteTimeDomainData(in Uint8Array array);
-    };
-}