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

Issue 1208843003: Removed extended jitter report from RtcpSender (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed comments Created 5 years, 5 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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627 627
628 void ModuleRtpRtcpImpl::SetREMBStatus(const bool enable) { 628 void ModuleRtpRtcpImpl::SetREMBStatus(const bool enable) {
629 rtcp_sender_.SetREMBStatus(enable); 629 rtcp_sender_.SetREMBStatus(enable);
630 } 630 }
631 631
632 void ModuleRtpRtcpImpl::SetREMBData(const uint32_t bitrate, 632 void ModuleRtpRtcpImpl::SetREMBData(const uint32_t bitrate,
633 const std::vector<uint32_t>& ssrcs) { 633 const std::vector<uint32_t>& ssrcs) {
634 rtcp_sender_.SetREMBData(bitrate, ssrcs); 634 rtcp_sender_.SetREMBData(bitrate, ssrcs);
635 } 635 }
636 636
637 // (IJ) Extended jitter report.
638 bool ModuleRtpRtcpImpl::IJ() const {
639 return rtcp_sender_.IJ();
640 }
641
642 void ModuleRtpRtcpImpl::SetIJStatus(const bool enable) {
643 rtcp_sender_.SetIJStatus(enable);
644 }
645
646 int32_t ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension( 637 int32_t ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension(
647 const RTPExtensionType type, 638 const RTPExtensionType type,
648 const uint8_t id) { 639 const uint8_t id) {
649 return rtp_sender_.RegisterRtpHeaderExtension(type, id); 640 return rtp_sender_.RegisterRtpHeaderExtension(type, id);
650 } 641 }
651 642
652 int32_t ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension( 643 int32_t ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
653 const RTPExtensionType type) { 644 const RTPExtensionType type) {
654 return rtp_sender_.DeregisterRtpHeaderExtension(type); 645 return rtp_sender_.DeregisterRtpHeaderExtension(type);
655 } 646 }
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975 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 966 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
976 StreamDataCountersCallback* callback) { 967 StreamDataCountersCallback* callback) {
977 rtp_sender_.RegisterRtpStatisticsCallback(callback); 968 rtp_sender_.RegisterRtpStatisticsCallback(callback);
978 } 969 }
979 970
980 StreamDataCountersCallback* 971 StreamDataCountersCallback*
981 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 972 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
982 return rtp_sender_.GetRtpStatisticsCallback(); 973 return rtp_sender_.GetRtpStatisticsCallback();
983 } 974 }
984 } // namespace webrtc 975 } // namespace webrtc
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