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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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