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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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67 : m_callback(callback), | 67 : m_callback(callback), |
68 m_numberOfOutputChannels(numberOfOutputChannels), | 68 m_numberOfOutputChannels(numberOfOutputChannels), |
69 m_inputBus(AudioBus::create(numberOfInputChannels, renderBufferSize)), | 69 m_inputBus(AudioBus::create(numberOfInputChannels, renderBufferSize)), |
70 m_renderBus( | 70 m_renderBus( |
71 AudioBus::create(numberOfOutputChannels, renderBufferSize, false)), | 71 AudioBus::create(numberOfOutputChannels, renderBufferSize, false)), |
72 m_sampleRate(sampleRate), | 72 m_sampleRate(sampleRate), |
73 m_isPlaying(false) { | 73 m_isPlaying(false) { |
74 // Histogram for audioHardwareBufferSize | 74 // Histogram for audioHardwareBufferSize |
75 DEFINE_STATIC_LOCAL(SparseHistogram, hardwareBufferSizeHistogram, | 75 DEFINE_STATIC_LOCAL(SparseHistogram, hardwareBufferSizeHistogram, |
76 ("WebAudio.AudioDestination.HardwareBufferSize")); | 76 ("WebAudio.AudioDestination.HardwareBufferSize")); |
77 // Histogram for the actual callback size used. Typically, this is the same a
s | 77 // Histogram for the actual callback size used. Typically, this is the same |
78 // audioHardwareBufferSize, but can be adjusted depending on some heuristics b
elow. | 78 // as audioHardwareBufferSize, but can be adjusted depending on some |
| 79 // heuristics below. |
79 DEFINE_STATIC_LOCAL(SparseHistogram, callbackBufferSizeHistogram, | 80 DEFINE_STATIC_LOCAL(SparseHistogram, callbackBufferSizeHistogram, |
80 ("WebAudio.AudioDestination.CallbackBufferSize")); | 81 ("WebAudio.AudioDestination.CallbackBufferSize")); |
81 | 82 |
82 // Use the optimal buffer size recommended by the audio backend. | 83 // Use the optimal buffer size recommended by the audio backend. |
83 size_t recommendedHardwareBufferSize = | 84 size_t recommendedHardwareBufferSize = |
84 Platform::current()->audioHardwareBufferSize(); | 85 Platform::current()->audioHardwareBufferSize(); |
85 m_callbackBufferSize = recommendedHardwareBufferSize; | 86 m_callbackBufferSize = recommendedHardwareBufferSize; |
86 | 87 |
87 #if OS(ANDROID) | 88 #if OS(ANDROID) |
88 // The optimum low-latency hardware buffer size is usually too small on Androi
d for WebAudio to | 89 // The optimum low-latency hardware buffer size is usually too small on |
89 // render without glitching. So, if it is small, use a larger size. If it was
already large, use | 90 // Android for WebAudio to render without glitching. So, if it is small, use |
90 // the requested size. | 91 // a larger size. If it was already large, use the requested size. |
91 // | 92 // |
92 // Since WebAudio renders in 128-frame blocks, the small buffer sizes (144 for
a Galaxy Nexus), | 93 // Since WebAudio renders in 128-frame blocks, the small buffer sizes (144 |
93 // cause significant processing jitter. Sometimes multiple blocks will process
ed, but other | 94 // for a Galaxy Nexus), cause significant processing jitter. Sometimes |
94 // times will not be since the FIFO can satisfy the request. By using a larger | 95 // multiple blocks will processed, but other times will not be since the FIFO |
95 // callbackBufferSize, we smooth out the jitter. | 96 // can satisfy the request. By using a larger callbackBufferSize, we smooth |
| 97 // out the jitter. |
96 const size_t kSmallBufferSize = 1024; | 98 const size_t kSmallBufferSize = 1024; |
97 const size_t kDefaultCallbackBufferSize = 2048; | 99 const size_t kDefaultCallbackBufferSize = 2048; |
98 | 100 |
99 if (m_callbackBufferSize <= kSmallBufferSize) | 101 if (m_callbackBufferSize <= kSmallBufferSize) |
100 m_callbackBufferSize = kDefaultCallbackBufferSize; | 102 m_callbackBufferSize = kDefaultCallbackBufferSize; |
101 #endif | 103 #endif |
102 | 104 |
103 // Quick exit if the requested size is too large. | 105 // Quick exit if the requested size is too large. |
104 ASSERT(m_callbackBufferSize + renderBufferSize <= fifoSize); | 106 ASSERT(m_callbackBufferSize + renderBufferSize <= fifoSize); |
105 if (m_callbackBufferSize + renderBufferSize > fifoSize) | 107 if (m_callbackBufferSize + renderBufferSize > fifoSize) |
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118 // not being a multiple of the render size. If the FIFO already | 120 // not being a multiple of the render size. If the FIFO already |
119 // contains enough data, the data will be provided directly. | 121 // contains enough data, the data will be provided directly. |
120 // Otherwise, the FIFO will call the provider enough times to | 122 // Otherwise, the FIFO will call the provider enough times to |
121 // satisfy the request for data. | 123 // satisfy the request for data. |
122 m_fifo = wrapUnique(new AudioPullFIFO(*this, numberOfOutputChannels, fifoSize, | 124 m_fifo = wrapUnique(new AudioPullFIFO(*this, numberOfOutputChannels, fifoSize, |
123 renderBufferSize)); | 125 renderBufferSize)); |
124 | 126 |
125 // Input buffering. | 127 // Input buffering. |
126 m_inputFifo = wrapUnique(new AudioFIFO(numberOfInputChannels, fifoSize)); | 128 m_inputFifo = wrapUnique(new AudioFIFO(numberOfInputChannels, fifoSize)); |
127 | 129 |
128 // If the callback size does not match the render size, then we need to buffer
some | 130 // If the callback size does not match the render size, then we need to |
129 // extra silence for the input. Otherwise, we can over-consume the input FIFO. | 131 // buffer some extra silence for the input. Otherwise, we can over-consume |
| 132 // the input FIFO. |
130 if (m_callbackBufferSize != renderBufferSize) { | 133 if (m_callbackBufferSize != renderBufferSize) { |
131 // FIXME: handle multi-channel input and don't hard-code to stereo. | 134 // FIXME: handle multi-channel input and don't hard-code to stereo. |
132 RefPtr<AudioBus> silence = AudioBus::create(2, renderBufferSize); | 135 RefPtr<AudioBus> silence = AudioBus::create(2, renderBufferSize); |
133 m_inputFifo->push(silence.get()); | 136 m_inputFifo->push(silence.get()); |
134 } | 137 } |
135 } | 138 } |
136 | 139 |
137 AudioDestination::~AudioDestination() { | 140 AudioDestination::~AudioDestination() { |
138 stop(); | 141 stop(); |
139 } | 142 } |
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194 AudioBus* sourceBus = nullptr; | 197 AudioBus* sourceBus = nullptr; |
195 if (m_inputFifo->framesInFifo() >= framesToProcess) { | 198 if (m_inputFifo->framesInFifo() >= framesToProcess) { |
196 m_inputFifo->consume(m_inputBus.get(), framesToProcess); | 199 m_inputFifo->consume(m_inputBus.get(), framesToProcess); |
197 sourceBus = m_inputBus.get(); | 200 sourceBus = m_inputBus.get(); |
198 } | 201 } |
199 | 202 |
200 m_callback.render(sourceBus, bus, framesToProcess); | 203 m_callback.render(sourceBus, bus, framesToProcess); |
201 } | 204 } |
202 | 205 |
203 } // namespace blink | 206 } // namespace blink |
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