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

Issue 1513303003: [rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years 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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949 return rtcp_receiver_.LastReceivedXrReferenceTimeInfo(info); 949 return rtcp_receiver_.LastReceivedXrReferenceTimeInfo(info);
950 } 950 }
951 951
952 bool ModuleRtpRtcpImpl::UpdateRTCPReceiveInformationTimers() { 952 bool ModuleRtpRtcpImpl::UpdateRTCPReceiveInformationTimers() {
953 // If this returns true this channel has timed out. 953 // If this returns true this channel has timed out.
954 // Periodically check if this is true and if so call UpdateTMMBR. 954 // Periodically check if this is true and if so call UpdateTMMBR.
955 return rtcp_receiver_.UpdateRTCPReceiveInformationTimers(); 955 return rtcp_receiver_.UpdateRTCPReceiveInformationTimers();
956 } 956 }
957 957
958 // Called from RTCPsender. 958 // Called from RTCPsender.
959 int32_t ModuleRtpRtcpImpl::BoundingSet(bool& tmmbr_owner, 959 int32_t ModuleRtpRtcpImpl::BoundingSet(bool* tmmbr_owner,
960 TMMBRSet*& bounding_set) { 960 TMMBRSet* bounding_set) {
961 return rtcp_receiver_.BoundingSet(tmmbr_owner, bounding_set); 961 return rtcp_receiver_.BoundingSet(tmmbr_owner, bounding_set);
962 } 962 }
963 963
964 int64_t ModuleRtpRtcpImpl::RtcpReportInterval() { 964 int64_t ModuleRtpRtcpImpl::RtcpReportInterval() {
965 if (audio_) 965 if (audio_)
966 return RTCP_INTERVAL_AUDIO_MS; 966 return RTCP_INTERVAL_AUDIO_MS;
967 else 967 else
968 return RTCP_INTERVAL_VIDEO_MS; 968 return RTCP_INTERVAL_VIDEO_MS;
969 } 969 }
970 970
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989 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 989 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
990 StreamDataCountersCallback* callback) { 990 StreamDataCountersCallback* callback) {
991 rtp_sender_.RegisterRtpStatisticsCallback(callback); 991 rtp_sender_.RegisterRtpStatisticsCallback(callback);
992 } 992 }
993 993
994 StreamDataCountersCallback* 994 StreamDataCountersCallback*
995 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 995 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
996 return rtp_sender_.GetRtpStatisticsCallback(); 996 return rtp_sender_.GetRtpStatisticsCallback();
997 } 997 }
998 } // namespace webrtc 998 } // namespace webrtc
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