| Index: Source/platform/audio/chromium/AudioDestinationChromium.cpp
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| diff --git a/Source/platform/audio/chromium/AudioDestinationChromium.cpp b/Source/platform/audio/chromium/AudioDestinationChromium.cpp
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| deleted file mode 100644
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| index 22b68305ee19837617988ed6f9333c06afa550c1..0000000000000000000000000000000000000000
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| --- a/Source/platform/audio/chromium/AudioDestinationChromium.cpp
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| +++ /dev/null
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| @@ -1,180 +0,0 @@
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| -/*
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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
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| - * 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.
|
| - *
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| - * 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
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| - * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
|
| - * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
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| - * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
|
| - * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
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| - * ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
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| - * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
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| - * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
|
| - */
|
| -
|
| -#include "config.h"
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| -
|
| -#if ENABLE(WEB_AUDIO)
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| -
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| -#include "platform/audio/chromium/AudioDestinationChromium.h"
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| -
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| -#include "platform/audio/AudioFIFO.h"
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| -#include "platform/audio/AudioPullFIFO.h"
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| -#include "public/platform/Platform.h"
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| -
|
| -namespace WebCore {
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| -
|
| -// Buffer size at which the web audio engine will render.
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| -const unsigned renderBufferSize = 128;
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| -
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| -// Size of the FIFO
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| -const size_t fifoSize = 8192;
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| -
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| -// Factory method: Chromium-implementation
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| -PassOwnPtr<AudioDestination> AudioDestination::create(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate)
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| -{
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| - return adoptPtr(new AudioDestinationChromium(callback, inputDeviceId, numberOfInputChannels, numberOfOutputChannels, sampleRate));
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| -}
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| -
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| -AudioDestinationChromium::AudioDestinationChromium(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate)
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| - : m_callback(callback)
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| - , m_numberOfOutputChannels(numberOfOutputChannels)
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| - , m_inputBus(AudioBus::create(numberOfInputChannels, renderBufferSize))
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| - , m_renderBus(AudioBus::create(numberOfOutputChannels, renderBufferSize, false))
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| - , m_sampleRate(sampleRate)
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| - , m_isPlaying(false)
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| -{
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| - // Use the optimal buffer size recommended by the audio backend.
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| - m_callbackBufferSize = blink::Platform::current()->audioHardwareBufferSize();
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| -
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| -#if OS(ANDROID)
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| - // The optimum low-latency hardware buffer size is usually too small on Android for WebAudio to
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| - // render without glitching. So, if it is small, use a larger size. If it was already large, use
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| - // the requested size.
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| - //
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| - // Since WebAudio renders in 128-frame blocks, the small buffer sizes (144 for a Galaxy Nexus),
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| - // cause significant processing jitter. Sometimes multiple blocks will processed, but other
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| - // times will not be since the FIFO can satisfy the request. By using a larger
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| - // callbackBufferSize, we smooth out the jitter.
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| - const size_t kSmallBufferSize = 1024;
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| - const size_t kDefaultCallbackBufferSize = 2048;
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| -
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| - if (m_callbackBufferSize <= kSmallBufferSize)
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| - m_callbackBufferSize = kDefaultCallbackBufferSize;
|
| -#endif
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| -
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| - // Quick exit if the requested size is too large.
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| - ASSERT(m_callbackBufferSize + renderBufferSize <= fifoSize);
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| - if (m_callbackBufferSize + renderBufferSize > fifoSize)
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| - return;
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| -
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| - m_audioDevice = adoptPtr(blink::Platform::current()->createAudioDevice(m_callbackBufferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, this, inputDeviceId));
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| - ASSERT(m_audioDevice);
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| -
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| - // Create a FIFO to handle the possibility of the callback size
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| - // not being a multiple of the render size. If the FIFO already
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| - // contains enough data, the data will be provided directly.
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| - // Otherwise, the FIFO will call the provider enough times to
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| - // satisfy the request for data.
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| - m_fifo = adoptPtr(new AudioPullFIFO(*this, numberOfOutputChannels, fifoSize, renderBufferSize));
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| -
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| - // Input buffering.
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| - m_inputFifo = adoptPtr(new AudioFIFO(numberOfInputChannels, fifoSize));
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| -
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| - // If the callback size does not match the render size, then we need to buffer some
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| - // extra silence for the input. Otherwise, we can over-consume the input FIFO.
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| - if (m_callbackBufferSize != renderBufferSize) {
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| - // FIXME: handle multi-channel input and don't hard-code to stereo.
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| - RefPtr<AudioBus> silence = AudioBus::create(2, renderBufferSize);
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| - m_inputFifo->push(silence.get());
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| - }
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| -}
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| -
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| -AudioDestinationChromium::~AudioDestinationChromium()
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| -{
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| - stop();
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| -}
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| -
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| -void AudioDestinationChromium::start()
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| -{
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| - if (!m_isPlaying && m_audioDevice) {
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| - m_audioDevice->start();
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| - m_isPlaying = true;
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| - }
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| -}
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| -
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| -void AudioDestinationChromium::stop()
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| -{
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| - if (m_isPlaying && m_audioDevice) {
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| - m_audioDevice->stop();
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| - m_isPlaying = false;
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| - }
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| -}
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| -
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| -float AudioDestination::hardwareSampleRate()
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| -{
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| - return static_cast<float>(blink::Platform::current()->audioHardwareSampleRate());
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| -}
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| -
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| -unsigned long AudioDestination::maxChannelCount()
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| -{
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| - return static_cast<float>(blink::Platform::current()->audioHardwareOutputChannels());
|
| -}
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| -
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| -void AudioDestinationChromium::render(const blink::WebVector<float*>& sourceData, const blink::WebVector<float*>& audioData, size_t numberOfFrames)
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| -{
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| - bool isNumberOfChannelsGood = audioData.size() == m_numberOfOutputChannels;
|
| - if (!isNumberOfChannelsGood) {
|
| - ASSERT_NOT_REACHED();
|
| - return;
|
| - }
|
| -
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| - bool isBufferSizeGood = numberOfFrames == m_callbackBufferSize;
|
| - if (!isBufferSizeGood) {
|
| - ASSERT_NOT_REACHED();
|
| - return;
|
| - }
|
| -
|
| - // Buffer optional live input.
|
| - if (sourceData.size() >= 2) {
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| - // FIXME: handle multi-channel input and don't hard-code to stereo.
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| - RefPtr<AudioBus> wrapperBus = AudioBus::create(2, numberOfFrames, false);
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| - wrapperBus->setChannelMemory(0, sourceData[0], numberOfFrames);
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| - wrapperBus->setChannelMemory(1, sourceData[1], numberOfFrames);
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| - m_inputFifo->push(wrapperBus.get());
|
| - }
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| -
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| - for (unsigned i = 0; i < m_numberOfOutputChannels; ++i)
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| - m_renderBus->setChannelMemory(i, audioData[i], numberOfFrames);
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| -
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| - m_fifo->consume(m_renderBus.get(), numberOfFrames);
|
| -}
|
| -
|
| -void AudioDestinationChromium::provideInput(AudioBus* bus, size_t framesToProcess)
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| -{
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| - AudioBus* sourceBus = 0;
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| - if (m_inputFifo->framesInFifo() >= framesToProcess) {
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| - m_inputFifo->consume(m_inputBus.get(), framesToProcess);
|
| - sourceBus = m_inputBus.get();
|
| - }
|
| -
|
| - m_callback.render(sourceBus, bus, framesToProcess);
|
| -}
|
| -
|
| -} // namespace WebCore
|
| -
|
| -#endif // ENABLE(WEB_AUDIO)
|
|
|