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diff --git a/chromium/third_party/WebKit/Source/platform/audio/AudioDestination.cpp b/chromium/third_party/WebKit/Source/platform/audio/AudioDestination.cpp
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+++ b/chromium/third_party/WebKit/Source/platform/audio/AudioDestination.cpp
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+/*
+ * 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 "platform/audio/AudioDestination.h"
+
+#include "platform/audio/AudioFIFO.h"
+#include "platform/audio/AudioPullFIFO.h"
+#include "public/platform/Platform.h"
+
+namespace WebCore {
+
+// Buffer size at which the web audio engine will render.
+const unsigned renderBufferSize = 128;
+
+// Size of the FIFO
+const size_t fifoSize = 8192;
+
+// Factory method: Chromium-implementation
+PassOwnPtr<AudioDestination> AudioDestination::create(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate)
+{
+    return adoptPtr(new AudioDestination(callback, inputDeviceId, numberOfInputChannels, numberOfOutputChannels, sampleRate));
+}
+
+AudioDestination::AudioDestination(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate)
+    : m_callback(callback)
+    , m_numberOfOutputChannels(numberOfOutputChannels)
+    , m_inputBus(AudioBus::create(numberOfInputChannels, renderBufferSize))
+    , m_renderBus(AudioBus::create(numberOfOutputChannels, renderBufferSize, false))
+    , m_sampleRate(sampleRate)
+    , m_isPlaying(false)
+{
+    // Use the optimal buffer size recommended by the audio backend.
+    m_callbackBufferSize = blink::Platform::current()->audioHardwareBufferSize();
+
+#if OS(ANDROID)
+    // The optimum low-latency hardware buffer size is usually too small on Android for WebAudio to
+    // render without glitching. So, if it is small, use a larger size. If it was already large, use
+    // the requested size.
+    //
+    // Since WebAudio renders in 128-frame blocks, the small buffer sizes (144 for a Galaxy Nexus),
+    // cause significant processing jitter. Sometimes multiple blocks will processed, but other
+    // times will not be since the FIFO can satisfy the request. By using a larger
+    // callbackBufferSize, we smooth out the jitter.
+    const size_t kSmallBufferSize = 1024;
+    const size_t kDefaultCallbackBufferSize = 2048;
+
+    if (m_callbackBufferSize <= kSmallBufferSize)
+        m_callbackBufferSize = kDefaultCallbackBufferSize;
+#endif
+
+    // Quick exit if the requested size is too large.
+    ASSERT(m_callbackBufferSize + renderBufferSize <= fifoSize);
+    if (m_callbackBufferSize + renderBufferSize > fifoSize)
+        return;
+
+    m_audioDevice = adoptPtr(blink::Platform::current()->createAudioDevice(m_callbackBufferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, this, inputDeviceId));
+    ASSERT(m_audioDevice);
+
+    // Create a FIFO to handle the possibility of the callback size
+    // not being a multiple of the render size. If the FIFO already
+    // contains enough data, the data will be provided directly.
+    // Otherwise, the FIFO will call the provider enough times to
+    // satisfy the request for data.
+    m_fifo = adoptPtr(new AudioPullFIFO(*this, numberOfOutputChannels, fifoSize, renderBufferSize));
+
+    // Input buffering.
+    m_inputFifo = adoptPtr(new AudioFIFO(numberOfInputChannels, fifoSize));
+
+    // If the callback size does not match the render size, then we need to buffer some
+    // extra silence for the input. Otherwise, we can over-consume the input FIFO.
+    if (m_callbackBufferSize != renderBufferSize) {
+        // FIXME: handle multi-channel input and don't hard-code to stereo.
+        RefPtr<AudioBus> silence = AudioBus::create(2, renderBufferSize);
+        m_inputFifo->push(silence.get());
+    }
+}
+
+AudioDestination::~AudioDestination()
+{
+    stop();
+}
+
+void AudioDestination::start()
+{
+    if (!m_isPlaying && m_audioDevice) {
+        m_audioDevice->start();
+        m_isPlaying = true;
+    }
+}
+
+void AudioDestination::stop()
+{
+    if (m_isPlaying && m_audioDevice) {
+        m_audioDevice->stop();
+        m_isPlaying = false;
+    }
+}
+
+float AudioDestination::hardwareSampleRate()
+{
+    return static_cast<float>(blink::Platform::current()->audioHardwareSampleRate());
+}
+
+unsigned long AudioDestination::maxChannelCount()
+{
+    return static_cast<float>(blink::Platform::current()->audioHardwareOutputChannels());
+}
+
+void AudioDestination::render(const blink::WebVector<float*>& sourceData, const blink::WebVector<float*>& audioData, size_t numberOfFrames)
+{
+    bool isNumberOfChannelsGood = audioData.size() == m_numberOfOutputChannels;
+    if (!isNumberOfChannelsGood) {
+        ASSERT_NOT_REACHED();
+        return;
+    }
+
+    bool isBufferSizeGood = numberOfFrames == m_callbackBufferSize;
+    if (!isBufferSizeGood) {
+        ASSERT_NOT_REACHED();
+        return;
+    }
+
+    // Buffer optional live input.
+    if (sourceData.size() >= 2) {
+        // FIXME: handle multi-channel input and don't hard-code to stereo.
+        RefPtr<AudioBus> wrapperBus = AudioBus::create(2, numberOfFrames, false);
+        wrapperBus->setChannelMemory(0, sourceData[0], numberOfFrames);
+        wrapperBus->setChannelMemory(1, sourceData[1], numberOfFrames);
+        m_inputFifo->push(wrapperBus.get());
+    }
+
+    for (unsigned i = 0; i < m_numberOfOutputChannels; ++i)
+        m_renderBus->setChannelMemory(i, audioData[i], numberOfFrames);
+
+    m_fifo->consume(m_renderBus.get(), numberOfFrames);
+}
+
+void AudioDestination::provideInput(AudioBus* bus, size_t framesToProcess)
+{
+    AudioBus* sourceBus = 0;
+    if (m_inputFifo->framesInFifo() >= framesToProcess) {
+        m_inputFifo->consume(m_inputBus.get(), framesToProcess);
+        sourceBus = m_inputBus.get();
+    }
+
+    m_callback.render(sourceBus, bus, framesToProcess);
+}
+
+} // namespace WebCore
+
+#endif // ENABLE(WEB_AUDIO)