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| 1 /* | 1 /* |
| 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2015 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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| 253 return rtc::Optional<Syncable::Info>(); | 253 return rtc::Optional<Syncable::Info>(); |
| 254 } | 254 } |
| 255 if (rtp_rtcp->RemoteNTP(&info.capture_time_ntp_secs, | 255 if (rtp_rtcp->RemoteNTP(&info.capture_time_ntp_secs, |
| 256 &info.capture_time_ntp_frac, | 256 &info.capture_time_ntp_frac, |
| 257 nullptr, | 257 nullptr, |
| 258 nullptr, | 258 nullptr, |
| 259 &info.capture_time_source_clock) != 0) { | 259 &info.capture_time_source_clock) != 0) { |
| 260 return rtc::Optional<Syncable::Info>(); | 260 return rtc::Optional<Syncable::Info>(); |
| 261 } | 261 } |
| 262 | 262 |
| 263 int jitter_buffer_delay_ms = 0; | 263 info.current_delay_ms = channel_proxy_->GetDelayEstimate(); |
| 264 int playout_buffer_delay_ms = 0; | |
| 265 channel_proxy_->GetDelayEstimate(&jitter_buffer_delay_ms, | |
| 266 &playout_buffer_delay_ms); | |
| 267 info.current_delay_ms = jitter_buffer_delay_ms + playout_buffer_delay_ms; | |
| 268 return rtc::Optional<Syncable::Info>(info); | 264 return rtc::Optional<Syncable::Info>(info); |
| 269 } | 265 } |
| 270 | 266 |
| 271 uint32_t AudioReceiveStream::GetPlayoutTimestamp() const { | 267 uint32_t AudioReceiveStream::GetPlayoutTimestamp() const { |
| 272 // Called on video capture thread. | 268 // Called on video capture thread. |
| 273 return channel_proxy_->GetPlayoutTimestamp(); | 269 return channel_proxy_->GetPlayoutTimestamp(); |
| 274 } | 270 } |
| 275 | 271 |
| 276 void AudioReceiveStream::SetMinimumPlayoutDelay(int delay_ms) { | 272 void AudioReceiveStream::SetMinimumPlayoutDelay(int delay_ms) { |
| 277 RTC_DCHECK_RUN_ON(&module_process_thread_checker_); | 273 RTC_DCHECK_RUN_ON(&module_process_thread_checker_); |
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| 332 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) { | 328 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) { |
| 333 ScopedVoEInterface<VoEBase> base(voice_engine()); | 329 ScopedVoEInterface<VoEBase> base(voice_engine()); |
| 334 if (playout) { | 330 if (playout) { |
| 335 return base->StartPlayout(config_.voe_channel_id); | 331 return base->StartPlayout(config_.voe_channel_id); |
| 336 } else { | 332 } else { |
| 337 return base->StopPlayout(config_.voe_channel_id); | 333 return base->StopPlayout(config_.voe_channel_id); |
| 338 } | 334 } |
| 339 } | 335 } |
| 340 } // namespace internal | 336 } // namespace internal |
| 341 } // namespace webrtc | 337 } // namespace webrtc |
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