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Side by Side Diff: third_party/WebKit/Source/platform/audio/AudioDestination.cpp

Issue 2060833002: Implementation of 'AudioContext.getOutputTimestamp' method (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Added implementation for ALSA. Created 4 years, 5 months ago
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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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141 float AudioDestination::hardwareSampleRate() 141 float AudioDestination::hardwareSampleRate()
142 { 142 {
143 return static_cast<float>(Platform::current()->audioHardwareSampleRate()); 143 return static_cast<float>(Platform::current()->audioHardwareSampleRate());
144 } 144 }
145 145
146 unsigned long AudioDestination::maxChannelCount() 146 unsigned long AudioDestination::maxChannelCount()
147 { 147 {
148 return static_cast<float>(Platform::current()->audioHardwareOutputChannels() ); 148 return static_cast<float>(Platform::current()->audioHardwareOutputChannels() );
149 } 149 }
150 150
151 void AudioDestination::render(const WebVector<float*>& sourceData, const WebVect or<float*>& audioData, size_t numberOfFrames) 151 void AudioDestination::render(const WebVector<float*>& sourceData, const WebVect or<float*>& audioData, size_t numberOfFrames, const WebAudioTimestamp& timestamp )
152 { 152 {
153 m_outputTimestamp = timestamp;
153 bool isNumberOfChannelsGood = audioData.size() == m_numberOfOutputChannels; 154 bool isNumberOfChannelsGood = audioData.size() == m_numberOfOutputChannels;
154 if (!isNumberOfChannelsGood) { 155 if (!isNumberOfChannelsGood) {
155 ASSERT_NOT_REACHED(); 156 ASSERT_NOT_REACHED();
156 return; 157 return;
157 } 158 }
158 159
159 bool isBufferSizeGood = numberOfFrames == m_callbackBufferSize; 160 bool isBufferSizeGood = numberOfFrames == m_callbackBufferSize;
160 if (!isBufferSizeGood) { 161 if (!isBufferSizeGood) {
161 ASSERT_NOT_REACHED(); 162 ASSERT_NOT_REACHED();
162 return; 163 return;
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178 } 179 }
179 180
180 void AudioDestination::provideInput(AudioBus* bus, size_t framesToProcess) 181 void AudioDestination::provideInput(AudioBus* bus, size_t framesToProcess)
181 { 182 {
182 AudioBus* sourceBus = nullptr; 183 AudioBus* sourceBus = nullptr;
183 if (m_inputFifo->framesInFifo() >= framesToProcess) { 184 if (m_inputFifo->framesInFifo() >= framesToProcess) {
184 m_inputFifo->consume(m_inputBus.get(), framesToProcess); 185 m_inputFifo->consume(m_inputBus.get(), framesToProcess);
185 sourceBus = m_inputBus.get(); 186 sourceBus = m_inputBus.get();
186 } 187 }
187 188
188 m_callback.render(sourceBus, bus, framesToProcess); 189 m_callback.render(sourceBus, bus, framesToProcess, m_outputTimestamp);
189 } 190 }
190 191
191 } // namespace blink 192 } // namespace blink
192 193
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