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1 /* | 1 /* |
2 * Copyright (C) 2010 Google Inc. All rights reserved. | 2 * Copyright (C) 2010 Google Inc. All rights reserved. |
3 * | 3 * |
4 * Redistribution and use in source and binary forms, with or without | 4 * Redistribution and use in source and binary forms, with or without |
5 * modification, are permitted provided that the following conditions | 5 * modification, are permitted provided that the following conditions |
6 * are met: | 6 * are met: |
7 * | 7 * |
8 * 1. Redistributions of source code must retain the above copyright | 8 * 1. Redistributions of source code must retain the above copyright |
9 * notice, this list of conditions and the following disclaimer. | 9 * notice, this list of conditions and the following disclaimer. |
10 * 2. Redistributions in binary form must reproduce the above copyright | 10 * 2. Redistributions in binary form must reproduce the above copyright |
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46 | 46 |
47 // Factory method: Chromium-implementation | 47 // Factory method: Chromium-implementation |
48 PassOwnPtr<AudioDestination> AudioDestination::create(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfO utputChannels, float sampleRate) | 48 PassOwnPtr<AudioDestination> AudioDestination::create(AudioIOCallback& callback, const String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfO utputChannels, float sampleRate) |
49 { | 49 { |
50 return adoptPtr(new AudioDestinationChromium(callback, inputDeviceId, number OfInputChannels, numberOfOutputChannels, sampleRate)); | 50 return adoptPtr(new AudioDestinationChromium(callback, inputDeviceId, number OfInputChannels, numberOfOutputChannels, sampleRate)); |
51 } | 51 } |
52 | 52 |
53 AudioDestinationChromium::AudioDestinationChromium(AudioIOCallback& callback, co nst String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutp utChannels, float sampleRate) | 53 AudioDestinationChromium::AudioDestinationChromium(AudioIOCallback& callback, co nst String& inputDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutp utChannels, float sampleRate) |
54 : m_callback(callback) | 54 : m_callback(callback) |
55 , m_numberOfOutputChannels(numberOfOutputChannels) | 55 , m_numberOfOutputChannels(numberOfOutputChannels) |
56 , m_inputBus(numberOfInputChannels, renderBufferSize) | 56 , m_inputBus(new AudioBus(numberOfInputChannels, renderBufferSize)) |
57 , m_renderBus(numberOfOutputChannels, renderBufferSize, false) | 57 , m_renderBus(new AudioBus(numberOfOutputChannels, renderBufferSize, false)) |
58 , m_sampleRate(sampleRate) | 58 , m_sampleRate(sampleRate) |
Chris Rogers
2013/05/02 17:48:46
adoptRef()
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59 , m_isPlaying(false) | 59 , m_isPlaying(false) |
60 { | 60 { |
61 // Use the optimal buffer size recommended by the audio backend. | 61 // Use the optimal buffer size recommended by the audio backend. |
62 m_callbackBufferSize = WebKit::Platform::current()->audioHardwareBufferSize( ); | 62 m_callbackBufferSize = WebKit::Platform::current()->audioHardwareBufferSize( ); |
63 | 63 |
64 // Quick exit if the requested size is too large. | 64 // Quick exit if the requested size is too large. |
65 ASSERT(m_callbackBufferSize + renderBufferSize <= fifoSize); | 65 ASSERT(m_callbackBufferSize + renderBufferSize <= fifoSize); |
66 if (m_callbackBufferSize + renderBufferSize > fifoSize) | 66 if (m_callbackBufferSize + renderBufferSize > fifoSize) |
67 return; | 67 return; |
68 | 68 |
69 m_audioDevice = adoptPtr(WebKit::Platform::current()->createAudioDevice(m_ca llbackBufferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, thi s, inputDeviceId)); | 69 m_audioDevice = adoptPtr(WebKit::Platform::current()->createAudioDevice(m_ca llbackBufferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, thi s, inputDeviceId)); |
70 ASSERT(m_audioDevice); | 70 ASSERT(m_audioDevice); |
71 | 71 |
72 // Create a FIFO to handle the possibility of the callback size | 72 // Create a FIFO to handle the possibility of the callback size |
73 // not being a multiple of the render size. If the FIFO already | 73 // not being a multiple of the render size. If the FIFO already |
74 // contains enough data, the data will be provided directly. | 74 // contains enough data, the data will be provided directly. |
75 // Otherwise, the FIFO will call the provider enough times to | 75 // Otherwise, the FIFO will call the provider enough times to |
76 // satisfy the request for data. | 76 // satisfy the request for data. |
77 m_fifo = adoptPtr(new AudioPullFIFO(*this, numberOfOutputChannels, fifoSize, renderBufferSize)); | 77 m_fifo = adoptPtr(new AudioPullFIFO(*this, numberOfOutputChannels, fifoSize, renderBufferSize)); |
78 | 78 |
79 // Input buffering. | 79 // Input buffering. |
80 m_inputFifo = adoptPtr(new AudioFIFO(numberOfInputChannels, fifoSize)); | 80 m_inputFifo = adoptPtr(new AudioFIFO(numberOfInputChannels, fifoSize)); |
81 | 81 |
82 // If the callback size does not match the render size, then we need to buff er some | 82 // If the callback size does not match the render size, then we need to buff er some |
83 // extra silence for the input. Otherwise, we can over-consume the input FIF O. | 83 // extra silence for the input. Otherwise, we can over-consume the input FIF O. |
84 if (m_callbackBufferSize != renderBufferSize) { | 84 if (m_callbackBufferSize != renderBufferSize) { |
85 // FIXME: handle multi-channel input and don't hard-code to stereo. | 85 // FIXME: handle multi-channel input and don't hard-code to stereo. |
86 AudioBus silence(2, renderBufferSize); | 86 RefPtr<AudioBus> silence = adoptRef<AudioBus>(new AudioBus(2, renderBuff erSize)); |
Chris Rogers
2013/05/02 17:48:46
Does this need to be adoptRef<AudioBus> or can it
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87 m_inputFifo->push(&silence); | 87 m_inputFifo->push(silence.get()); |
88 } | 88 } |
89 } | 89 } |
90 | 90 |
91 AudioDestinationChromium::~AudioDestinationChromium() | 91 AudioDestinationChromium::~AudioDestinationChromium() |
92 { | 92 { |
93 stop(); | 93 stop(); |
94 } | 94 } |
95 | 95 |
96 void AudioDestinationChromium::start() | 96 void AudioDestinationChromium::start() |
97 { | 97 { |
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136 // Buffer optional live input. | 136 // Buffer optional live input. |
137 if (sourceData.size() >= 2) { | 137 if (sourceData.size() >= 2) { |
138 // FIXME: handle multi-channel input and don't hard-code to stereo. | 138 // FIXME: handle multi-channel input and don't hard-code to stereo. |
139 AudioBus wrapperBus(2, numberOfFrames, false); | 139 AudioBus wrapperBus(2, numberOfFrames, false); |
140 wrapperBus.setChannelMemory(0, sourceData[0], numberOfFrames); | 140 wrapperBus.setChannelMemory(0, sourceData[0], numberOfFrames); |
141 wrapperBus.setChannelMemory(1, sourceData[1], numberOfFrames); | 141 wrapperBus.setChannelMemory(1, sourceData[1], numberOfFrames); |
142 m_inputFifo->push(&wrapperBus); | 142 m_inputFifo->push(&wrapperBus); |
143 } | 143 } |
144 | 144 |
145 for (unsigned i = 0; i < m_numberOfOutputChannels; ++i) | 145 for (unsigned i = 0; i < m_numberOfOutputChannels; ++i) |
146 m_renderBus.setChannelMemory(i, audioData[i], numberOfFrames); | 146 m_renderBus->setChannelMemory(i, audioData[i], numberOfFrames); |
147 | 147 |
148 m_fifo->consume(&m_renderBus, numberOfFrames); | 148 m_fifo->consume(m_renderBus.get(), numberOfFrames); |
149 } | 149 } |
150 | 150 |
151 void AudioDestinationChromium::provideInput(AudioBus* bus, size_t framesToProces s) | 151 void AudioDestinationChromium::provideInput(AudioBus* bus, size_t framesToProces s) |
152 { | 152 { |
153 AudioBus* sourceBus = 0; | 153 AudioBus* sourceBus = 0; |
154 if (m_inputFifo->framesInFifo() >= framesToProcess) { | 154 if (m_inputFifo->framesInFifo() >= framesToProcess) { |
155 m_inputFifo->consume(&m_inputBus, framesToProcess); | 155 m_inputFifo->consume(m_inputBus.get(), framesToProcess); |
156 sourceBus = &m_inputBus; | 156 sourceBus = m_inputBus.get(); |
157 } | 157 } |
158 | 158 |
159 m_callback.render(sourceBus, bus, framesToProcess); | 159 m_callback.render(sourceBus, bus, framesToProcess); |
160 } | 160 } |
161 | 161 |
162 } // namespace WebCore | 162 } // namespace WebCore |
163 | 163 |
164 #endif // ENABLE(WEB_AUDIO) | 164 #endif // ENABLE(WEB_AUDIO) |
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