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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: Fixing unit test and adding more TODOs. 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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490 return rtp_sender_.SetMaxPayloadLength(length, packet_overhead_); 490 return rtp_sender_.SetMaxPayloadLength(length, packet_overhead_);
491 } 491 }
492 492
493 int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) { 493 int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) {
494 RTC_DCHECK_LE(mtu, IP_PACKET_SIZE) << "Invalid mtu: " << mtu; 494 RTC_DCHECK_LE(mtu, IP_PACKET_SIZE) << "Invalid mtu: " << mtu;
495 return rtp_sender_.SetMaxPayloadLength(mtu - packet_overhead_, 495 return rtp_sender_.SetMaxPayloadLength(mtu - packet_overhead_,
496 packet_overhead_); 496 packet_overhead_);
497 } 497 }
498 498
499 RtcpMode ModuleRtpRtcpImpl::RTCP() const { 499 RtcpMode ModuleRtpRtcpImpl::RTCP() const {
500 if (rtcp_sender_.Status() != RtcpMode::kOff) { 500 return rtcp_sender_.Status();
501 return rtcp_receiver_.Status();
502 }
503 return RtcpMode::kOff;
504 } 501 }
505 502
506 // Configure RTCP status i.e on/off. 503 // Configure RTCP status i.e on/off.
507 void ModuleRtpRtcpImpl::SetRTCPStatus(const RtcpMode method) { 504 void ModuleRtpRtcpImpl::SetRTCPStatus(const RtcpMode method) {
508 rtcp_sender_.SetRTCPStatus(method); 505 rtcp_sender_.SetRTCPStatus(method);
509 rtcp_receiver_.SetRTCPStatus(method);
510 } 506 }
511 507
512 int32_t ModuleRtpRtcpImpl::SetCNAME(const char* c_name) { 508 int32_t ModuleRtpRtcpImpl::SetCNAME(const char* c_name) {
513 return rtcp_sender_.SetCNAME(c_name); 509 return rtcp_sender_.SetCNAME(c_name);
514 } 510 }
515 511
516 int32_t ModuleRtpRtcpImpl::AddMixedCNAME(uint32_t ssrc, const char* c_name) { 512 int32_t ModuleRtpRtcpImpl::AddMixedCNAME(uint32_t ssrc, const char* c_name) {
517 return rtcp_sender_.AddMixedCNAME(ssrc, c_name); 513 return rtcp_sender_.AddMixedCNAME(ssrc, c_name);
518 } 514 }
519 515
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996 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 992 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
997 StreamDataCountersCallback* callback) { 993 StreamDataCountersCallback* callback) {
998 rtp_sender_.RegisterRtpStatisticsCallback(callback); 994 rtp_sender_.RegisterRtpStatisticsCallback(callback);
999 } 995 }
1000 996
1001 StreamDataCountersCallback* 997 StreamDataCountersCallback*
1002 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 998 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
1003 return rtp_sender_.GetRtpStatisticsCallback(); 999 return rtp_sender_.GetRtpStatisticsCallback();
1004 } 1000 }
1005 } // namespace webrtc 1001 } // namespace webrtc
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