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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 "platform/audio/AudioDestination.h"
#if ENABLE(WEB_AUDIO)
#include "platform/audio/AudioFIFO.h"
#include "platform/audio/AudioPullFIFO.h"
#include "public/platform/Platform.h"
namespace blink {
// 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 = 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(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>(Platform::current()->audioHardwareSampleRate());
}
unsigned long AudioDestination::maxChannelCount()
{
return static_cast<float>(Platform::current()->audioHardwareOutputChannels());
}
void AudioDestination::render(const WebVector<float*>& sourceData, const 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 = nullptr;
if (m_inputFifo->framesInFifo() >= framesToProcess) {
m_inputFifo->consume(m_inputBus.get(), framesToProcess);
sourceBus = m_inputBus.get();
}
m_callback.render(sourceBus, bus, framesToProcess);
}
} // namespace blink
#endif // ENABLE(WEB_AUDIO)
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