Move chrome specific code from platform/audio/chromium to platform/audio

Source/platform/audio/chromium/AudioDestinationChromium.cpp

-/*
- * 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/chromium/AudioDestinationChromium.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 AudioDestinationChromium(callback, inputDeviceId, numberOfInputChannels, numberOfOutputChannels, sampleRate));
-}
-
-AudioDestinationChromium::AudioDestinationChromium(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());
-    }
-}
-
-AudioDestinationChromium::~AudioDestinationChromium()
-{
-    stop();
-}
-
-void AudioDestinationChromium::start()
-{
-    if (!m_isPlaying && m_audioDevice) {
-        m_audioDevice->start();
-        m_isPlaying = true;
-    }
-}
-
-void AudioDestinationChromium::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 AudioDestinationChromium::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 AudioDestinationChromium::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)