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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Issue 1763823003: rtt calculation handles time go backwards (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 9 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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579 return rtcp_sender_.SetApplicationSpecificData(sub_type, name, data, length); 579 return rtcp_sender_.SetApplicationSpecificData(sub_type, name, data, length);
580 } 580 }
581 581
582 // (XR) VOIP metric. 582 // (XR) VOIP metric.
583 int32_t ModuleRtpRtcpImpl::SetRTCPVoIPMetrics( 583 int32_t ModuleRtpRtcpImpl::SetRTCPVoIPMetrics(
584 const RTCPVoIPMetric* voip_metric) { 584 const RTCPVoIPMetric* voip_metric) {
585 return rtcp_sender_.SetRTCPVoIPMetrics(voip_metric); 585 return rtcp_sender_.SetRTCPVoIPMetrics(voip_metric);
586 } 586 }
587 587
588 void ModuleRtpRtcpImpl::SetRtcpXrRrtrStatus(bool enable) { 588 void ModuleRtpRtcpImpl::SetRtcpXrRrtrStatus(bool enable) {
589 return rtcp_sender_.SendRtcpXrReceiverReferenceTime(enable); 589 rtcp_receiver_.SetRtcpXrRrtrStatus(enable);
590 rtcp_sender_.SendRtcpXrReceiverReferenceTime(enable);
590 } 591 }
591 592
592 bool ModuleRtpRtcpImpl::RtcpXrRrtrStatus() const { 593 bool ModuleRtpRtcpImpl::RtcpXrRrtrStatus() const {
593 return rtcp_sender_.RtcpXrReceiverReferenceTime(); 594 return rtcp_sender_.RtcpXrReceiverReferenceTime();
594 } 595 }
595 596
596 // TODO(asapersson): Replace this method with the one below. 597 // TODO(asapersson): Replace this method with the one below.
597 int32_t ModuleRtpRtcpImpl::DataCountersRTP( 598 int32_t ModuleRtpRtcpImpl::DataCountersRTP(
598 size_t* bytes_sent, 599 size_t* bytes_sent,
599 uint32_t* packets_sent) const { 600 uint32_t* packets_sent) const {
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991 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 992 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
992 StreamDataCountersCallback* callback) { 993 StreamDataCountersCallback* callback) {
993 rtp_sender_.RegisterRtpStatisticsCallback(callback); 994 rtp_sender_.RegisterRtpStatisticsCallback(callback);
994 } 995 }
995 996
996 StreamDataCountersCallback* 997 StreamDataCountersCallback*
997 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 998 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
998 return rtp_sender_.GetRtpStatisticsCallback(); 999 return rtp_sender_.GetRtpStatisticsCallback();
999 } 1000 }
1000 } // namespace webrtc 1001 } // namespace webrtc
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