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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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| 314 int64_t capture_time_ms, | 314 int64_t capture_time_ms, |
| 315 size_t bytes, | 315 size_t bytes, |
| 316 bool retransmission) override { | 316 bool retransmission) override { |
| 317 rtc::CritScope lock(&crit_); | 317 rtc::CritScope lock(&crit_); |
| 318 if (rtp_packet_sender_) { | 318 if (rtp_packet_sender_) { |
| 319 rtp_packet_sender_->InsertPacket(priority, ssrc, sequence_number, | 319 rtp_packet_sender_->InsertPacket(priority, ssrc, sequence_number, |
| 320 capture_time_ms, bytes, retransmission); | 320 capture_time_ms, bytes, retransmission); |
| 321 } | 321 } |
| 322 } | 322 } |
| 323 | 323 |
| 324 void AccountForAudioPackets(bool account_for_audio) override { | |
|
stefan-webrtc
2017/08/14 07:46:03
I don't think this should ever be called as it's a
alexnarest
2017/08/15 06:02:37
Done.
| |
| 325 rtc::CritScope lock(&crit_); | |
| 326 if (rtp_packet_sender_) { | |
| 327 rtp_packet_sender_->AccountForAudioPackets(account_for_audio); | |
| 328 } | |
| 329 } | |
| 330 | |
| 324 private: | 331 private: |
| 325 rtc::ThreadChecker thread_checker_; | 332 rtc::ThreadChecker thread_checker_; |
| 326 rtc::CriticalSection crit_; | 333 rtc::CriticalSection crit_; |
| 327 RtpPacketSender* rtp_packet_sender_ GUARDED_BY(&crit_); | 334 RtpPacketSender* rtp_packet_sender_ GUARDED_BY(&crit_); |
| 328 }; | 335 }; |
| 329 | 336 |
| 330 class VoERtcpObserver : public RtcpBandwidthObserver { | 337 class VoERtcpObserver : public RtcpBandwidthObserver { |
| 331 public: | 338 public: |
| 332 explicit VoERtcpObserver(Channel* owner) | 339 explicit VoERtcpObserver(Channel* owner) |
| 333 : owner_(owner), bandwidth_observer_(nullptr) {} | 340 : owner_(owner), bandwidth_observer_(nullptr) {} |
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| 3140 int64_t min_rtt = 0; | 3147 int64_t min_rtt = 0; |
| 3141 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != | 3148 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != |
| 3142 0) { | 3149 0) { |
| 3143 return 0; | 3150 return 0; |
| 3144 } | 3151 } |
| 3145 return rtt; | 3152 return rtt; |
| 3146 } | 3153 } |
| 3147 | 3154 |
| 3148 } // namespace voe | 3155 } // namespace voe |
| 3149 } // namespace webrtc | 3156 } // namespace webrtc |
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