| /* |
| * 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 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 = 0; |
| 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) |