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Issue 1588693002: Revert of Storing raw audio sink for default audio track. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 11 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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1178 } 1178 }
1179 1179
1180 int32_t 1180 int32_t
1181 Channel::UpdateLocalTimeStamp() 1181 Channel::UpdateLocalTimeStamp()
1182 { 1182 {
1183 1183
1184 _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_); 1184 _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
1185 return 0; 1185 return 0;
1186 } 1186 }
1187 1187
1188 void Channel::SetSink(const rtc::scoped_refptr<AudioSinkInterface>& sink) { 1188 void Channel::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
1189 CriticalSectionScoped cs(&_callbackCritSect); 1189 CriticalSectionScoped cs(&_callbackCritSect);
1190 audio_sink_ = sink; 1190 audio_sink_ = std::move(sink);
1191 } 1191 }
1192 1192
1193 int32_t 1193 int32_t
1194 Channel::StartPlayout() 1194 Channel::StartPlayout()
1195 { 1195 {
1196 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId), 1196 WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
1197 "Channel::StartPlayout()"); 1197 "Channel::StartPlayout()");
1198 if (channel_state_.Get().playing) 1198 if (channel_state_.Get().playing)
1199 { 1199 {
1200 return 0; 1200 return 0;
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4103 int64_t min_rtt = 0; 4103 int64_t min_rtt = 0;
4104 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) 4104 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
4105 != 0) { 4105 != 0) {
4106 return 0; 4106 return 0;
4107 } 4107 }
4108 return rtt; 4108 return rtt;
4109 } 4109 }
4110 4110
4111 } // namespace voe 4111 } // namespace voe
4112 } // namespace webrtc 4112 } // namespace webrtc
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