Index: Source/platform/audio/chromium/AudioDestinationChromium.cpp |
diff --git a/Source/platform/audio/chromium/AudioDestinationChromium.cpp b/Source/platform/audio/chromium/AudioDestinationChromium.cpp |
deleted file mode 100644 |
index 22b68305ee19837617988ed6f9333c06afa550c1..0000000000000000000000000000000000000000 |
--- a/Source/platform/audio/chromium/AudioDestinationChromium.cpp |
+++ /dev/null |
@@ -1,180 +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 "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) |