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

Issue 1713493003: Enabling rtcp-rsize negotiation and fixing some issues with it. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 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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471 return rtp_sender_.SetMaxPayloadLength(length, packet_overhead_); 471 return rtp_sender_.SetMaxPayloadLength(length, packet_overhead_);
472 } 472 }
473 473
474 int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) { 474 int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) {
475 RTC_DCHECK_LE(mtu, IP_PACKET_SIZE) << "Invalid mtu: " << mtu; 475 RTC_DCHECK_LE(mtu, IP_PACKET_SIZE) << "Invalid mtu: " << mtu;
476 return rtp_sender_.SetMaxPayloadLength(mtu - packet_overhead_, 476 return rtp_sender_.SetMaxPayloadLength(mtu - packet_overhead_,
477 packet_overhead_); 477 packet_overhead_);
478 } 478 }
479 479
480 RtcpMode ModuleRtpRtcpImpl::RTCP() const { 480 RtcpMode ModuleRtpRtcpImpl::RTCP() const {
481 if (rtcp_sender_.Status() != RtcpMode::kOff) { 481 return rtcp_sender_.Status();
482 return rtcp_receiver_.Status();
483 }
484 return RtcpMode::kOff;
485 } 482 }
486 483
487 // Configure RTCP status i.e on/off. 484 // Configure RTCP status i.e on/off.
488 void ModuleRtpRtcpImpl::SetRTCPStatus(const RtcpMode method) { 485 void ModuleRtpRtcpImpl::SetRTCPStatus(const RtcpMode method) {
489 rtcp_sender_.SetRTCPStatus(method); 486 rtcp_sender_.SetRTCPStatus(method);
490 rtcp_receiver_.SetRTCPStatus(method);
491 } 487 }
492 488
493 int32_t ModuleRtpRtcpImpl::SetCNAME(const char* c_name) { 489 int32_t ModuleRtpRtcpImpl::SetCNAME(const char* c_name) {
494 return rtcp_sender_.SetCNAME(c_name); 490 return rtcp_sender_.SetCNAME(c_name);
495 } 491 }
496 492
497 int32_t ModuleRtpRtcpImpl::AddMixedCNAME(uint32_t ssrc, const char* c_name) { 493 int32_t ModuleRtpRtcpImpl::AddMixedCNAME(uint32_t ssrc, const char* c_name) {
498 return rtcp_sender_.AddMixedCNAME(ssrc, c_name); 494 return rtcp_sender_.AddMixedCNAME(ssrc, c_name);
499 } 495 }
500 496
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984 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 980 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
985 StreamDataCountersCallback* callback) { 981 StreamDataCountersCallback* callback) {
986 rtp_sender_.RegisterRtpStatisticsCallback(callback); 982 rtp_sender_.RegisterRtpStatisticsCallback(callback);
987 } 983 }
988 984
989 StreamDataCountersCallback* 985 StreamDataCountersCallback*
990 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 986 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
991 return rtp_sender_.GetRtpStatisticsCallback(); 987 return rtp_sender_.GetRtpStatisticsCallback();
992 } 988 }
993 } // namespace webrtc 989 } // namespace webrtc
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