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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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52 | 52 |
53 AudioDestination::AudioDestination(AudioIOCallback& callback, const String& inpu
tDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, floa
t sampleRate) | 53 AudioDestination::AudioDestination(AudioIOCallback& callback, const String& inpu
tDeviceId, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, floa
t sampleRate) |
54 : m_callback(callback) | 54 : m_callback(callback) |
55 , m_numberOfOutputChannels(numberOfOutputChannels) | 55 , m_numberOfOutputChannels(numberOfOutputChannels) |
56 , m_inputBus(AudioBus::create(numberOfInputChannels, renderBufferSize)) | 56 , m_inputBus(AudioBus::create(numberOfInputChannels, renderBufferSize)) |
57 , m_renderBus(AudioBus::create(numberOfOutputChannels, renderBufferSize, fal
se)) | 57 , m_renderBus(AudioBus::create(numberOfOutputChannels, renderBufferSize, fal
se)) |
58 , m_sampleRate(sampleRate) | 58 , m_sampleRate(sampleRate) |
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 = blink::Platform::current()->audioHardwareBufferSize()
; | 62 m_callbackBufferSize = Platform::current()->audioHardwareBufferSize(); |
63 | 63 |
64 #if OS(ANDROID) | 64 #if OS(ANDROID) |
65 // The optimum low-latency hardware buffer size is usually too small on Andr
oid for WebAudio to | 65 // The optimum low-latency hardware buffer size is usually too small on Andr
oid for WebAudio to |
66 // render without glitching. So, if it is small, use a larger size. If it wa
s already large, use | 66 // render without glitching. So, if it is small, use a larger size. If it wa
s already large, use |
67 // the requested size. | 67 // the requested size. |
68 // | 68 // |
69 // Since WebAudio renders in 128-frame blocks, the small buffer sizes (144 f
or a Galaxy Nexus), | 69 // Since WebAudio renders in 128-frame blocks, the small buffer sizes (144 f
or a Galaxy Nexus), |
70 // cause significant processing jitter. Sometimes multiple blocks will proce
ssed, but other | 70 // cause significant processing jitter. Sometimes multiple blocks will proce
ssed, but other |
71 // times will not be since the FIFO can satisfy the request. By using a larg
er | 71 // times will not be since the FIFO can satisfy the request. By using a larg
er |
72 // callbackBufferSize, we smooth out the jitter. | 72 // callbackBufferSize, we smooth out the jitter. |
73 const size_t kSmallBufferSize = 1024; | 73 const size_t kSmallBufferSize = 1024; |
74 const size_t kDefaultCallbackBufferSize = 2048; | 74 const size_t kDefaultCallbackBufferSize = 2048; |
75 | 75 |
76 if (m_callbackBufferSize <= kSmallBufferSize) | 76 if (m_callbackBufferSize <= kSmallBufferSize) |
77 m_callbackBufferSize = kDefaultCallbackBufferSize; | 77 m_callbackBufferSize = kDefaultCallbackBufferSize; |
78 #endif | 78 #endif |
79 | 79 |
80 // Quick exit if the requested size is too large. | 80 // Quick exit if the requested size is too large. |
81 ASSERT(m_callbackBufferSize + renderBufferSize <= fifoSize); | 81 ASSERT(m_callbackBufferSize + renderBufferSize <= fifoSize); |
82 if (m_callbackBufferSize + renderBufferSize > fifoSize) | 82 if (m_callbackBufferSize + renderBufferSize > fifoSize) |
83 return; | 83 return; |
84 | 84 |
85 m_audioDevice = adoptPtr(blink::Platform::current()->createAudioDevice(m_cal
lbackBufferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, this
, inputDeviceId)); | 85 m_audioDevice = adoptPtr(Platform::current()->createAudioDevice(m_callbackBu
fferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, this, input
DeviceId)); |
86 ASSERT(m_audioDevice); | 86 ASSERT(m_audioDevice); |
87 | 87 |
88 // Create a FIFO to handle the possibility of the callback size | 88 // Create a FIFO to handle the possibility of the callback size |
89 // not being a multiple of the render size. If the FIFO already | 89 // not being a multiple of the render size. If the FIFO already |
90 // contains enough data, the data will be provided directly. | 90 // contains enough data, the data will be provided directly. |
91 // Otherwise, the FIFO will call the provider enough times to | 91 // Otherwise, the FIFO will call the provider enough times to |
92 // satisfy the request for data. | 92 // satisfy the request for data. |
93 m_fifo = adoptPtr(new AudioPullFIFO(*this, numberOfOutputChannels, fifoSize,
renderBufferSize)); | 93 m_fifo = adoptPtr(new AudioPullFIFO(*this, numberOfOutputChannels, fifoSize,
renderBufferSize)); |
94 | 94 |
95 // Input buffering. | 95 // Input buffering. |
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120 void AudioDestination::stop() | 120 void AudioDestination::stop() |
121 { | 121 { |
122 if (m_isPlaying && m_audioDevice) { | 122 if (m_isPlaying && m_audioDevice) { |
123 m_audioDevice->stop(); | 123 m_audioDevice->stop(); |
124 m_isPlaying = false; | 124 m_isPlaying = false; |
125 } | 125 } |
126 } | 126 } |
127 | 127 |
128 float AudioDestination::hardwareSampleRate() | 128 float AudioDestination::hardwareSampleRate() |
129 { | 129 { |
130 return static_cast<float>(blink::Platform::current()->audioHardwareSampleRat
e()); | 130 return static_cast<float>(Platform::current()->audioHardwareSampleRate()); |
131 } | 131 } |
132 | 132 |
133 unsigned long AudioDestination::maxChannelCount() | 133 unsigned long AudioDestination::maxChannelCount() |
134 { | 134 { |
135 return static_cast<float>(blink::Platform::current()->audioHardwareOutputCha
nnels()); | 135 return static_cast<float>(Platform::current()->audioHardwareOutputChannels()
); |
136 } | 136 } |
137 | 137 |
138 void AudioDestination::render(const blink::WebVector<float*>& sourceData, const
blink::WebVector<float*>& audioData, size_t numberOfFrames) | 138 void AudioDestination::render(const WebVector<float*>& sourceData, const WebVect
or<float*>& audioData, size_t numberOfFrames) |
139 { | 139 { |
140 bool isNumberOfChannelsGood = audioData.size() == m_numberOfOutputChannels; | 140 bool isNumberOfChannelsGood = audioData.size() == m_numberOfOutputChannels; |
141 if (!isNumberOfChannelsGood) { | 141 if (!isNumberOfChannelsGood) { |
142 ASSERT_NOT_REACHED(); | 142 ASSERT_NOT_REACHED(); |
143 return; | 143 return; |
144 } | 144 } |
145 | 145 |
146 bool isBufferSizeGood = numberOfFrames == m_callbackBufferSize; | 146 bool isBufferSizeGood = numberOfFrames == m_callbackBufferSize; |
147 if (!isBufferSizeGood) { | 147 if (!isBufferSizeGood) { |
148 ASSERT_NOT_REACHED(); | 148 ASSERT_NOT_REACHED(); |
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171 m_inputFifo->consume(m_inputBus.get(), framesToProcess); | 171 m_inputFifo->consume(m_inputBus.get(), framesToProcess); |
172 sourceBus = m_inputBus.get(); | 172 sourceBus = m_inputBus.get(); |
173 } | 173 } |
174 | 174 |
175 m_callback.render(sourceBus, bus, framesToProcess); | 175 m_callback.render(sourceBus, bus, framesToProcess); |
176 } | 176 } |
177 | 177 |
178 } // namespace blink | 178 } // namespace blink |
179 | 179 |
180 #endif // ENABLE(WEB_AUDIO) | 180 #endif // ENABLE(WEB_AUDIO) |
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