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Issue 11773017: Avoids crash in WebRTC audio clients for 96kHz render rate on Mac OSX. (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: nit Created 7 years, 11 months ago
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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #include "content/renderer/media/webrtc_audio_renderer.h" 5 #include "content/renderer/media/webrtc_audio_renderer.h"
6 6
7 #include "base/logging.h" 7 #include "base/logging.h"
8 #include "base/metrics/histogram.h" 8 #include "base/metrics/histogram.h"
9 #include "base/string_util.h" 9 #include "base/string_util.h"
10 #include "content/renderer/media/audio_device_factory.h" 10 #include "content/renderer/media/audio_device_factory.h"
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153 if (base::win::GetVersion() < base::win::VERSION_VISTA) { 153 if (base::win::GetVersion() < base::win::VERSION_VISTA) {
154 buffer_size = 3 * buffer_size; 154 buffer_size = 3 * buffer_size;
155 DLOG(WARNING) << "Extending the output buffer size by a factor of three " 155 DLOG(WARNING) << "Extending the output buffer size by a factor of three "
156 << "since Windows XP has been detected."; 156 << "since Windows XP has been detected.";
157 } 157 }
158 #elif defined(OS_MACOSX) 158 #elif defined(OS_MACOSX)
159 channel_layout = media::CHANNEL_LAYOUT_MONO; 159 channel_layout = media::CHANNEL_LAYOUT_MONO;
160 160
161 // Render side: AUDIO_PCM_LOW_LATENCY on Mac OS X is based on a callback- 161 // Render side: AUDIO_PCM_LOW_LATENCY on Mac OS X is based on a callback-
162 // driven Core Audio implementation. Tests have shown that 10ms is a suitable 162 // driven Core Audio implementation. Tests have shown that 10ms is a suitable
163 // frame size to use, both for 48kHz and 44.1kHz. 163 // frame size to use for 96kHz, 48kHz and 44.1kHz.
164 164
165 // Use different buffer sizes depending on the current hardware sample rate. 165 // Use different buffer sizes depending on the current hardware sample rate.
166 if (sample_rate == 48000) { 166 if (sample_rate == 96000 || sample_rate == 48000) {
167 buffer_size = 480; 167 buffer_size = (sample_rate / 100);
168 } else { 168 } else {
169 // We do run at 44.1kHz at the actual audio layer, but ask for frames 169 // We do run at 44.1kHz at the actual audio layer, but ask for frames
170 // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine. 170 // at 44.0kHz to ensure that we can feed them to the webrtc::VoiceEngine.
171 buffer_size = 440; 171 buffer_size = 440;
172 } 172 }
173 #elif defined(OS_LINUX) || defined(OS_OPENBSD) 173 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
174 channel_layout = media::CHANNEL_LAYOUT_MONO; 174 channel_layout = media::CHANNEL_LAYOUT_MONO;
175 175
176 // Based on tests using the current ALSA implementation in Chrome, we have 176 // Based on tests using the current ALSA implementation in Chrome, we have
177 // found that 10ms buffer size on the output side works fine. 177 // found that 10ms buffer size on the output side works fine.
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282 params_.bits_per_sample() / 8); 282 params_.bits_per_sample() / 8);
283 return audio_bus->frames(); 283 return audio_bus->frames();
284 } 284 }
285 285
286 void WebRtcAudioRenderer::OnRenderError() { 286 void WebRtcAudioRenderer::OnRenderError() {
287 NOTIMPLEMENTED(); 287 NOTIMPLEMENTED();
288 LOG(ERROR) << "OnRenderError()"; 288 LOG(ERROR) << "OnRenderError()";
289 } 289 }
290 290
291 } // namespace content 291 } // namespace content
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