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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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947 return rtcp_receiver_.LastReceivedXrReferenceTimeInfo(info); | 947 return rtcp_receiver_.LastReceivedXrReferenceTimeInfo(info); |
948 } | 948 } |
949 | 949 |
950 bool ModuleRtpRtcpImpl::UpdateRTCPReceiveInformationTimers() { | 950 bool ModuleRtpRtcpImpl::UpdateRTCPReceiveInformationTimers() { |
951 // If this returns true this channel has timed out. | 951 // If this returns true this channel has timed out. |
952 // Periodically check if this is true and if so call UpdateTMMBR. | 952 // Periodically check if this is true and if so call UpdateTMMBR. |
953 return rtcp_receiver_.UpdateRTCPReceiveInformationTimers(); | 953 return rtcp_receiver_.UpdateRTCPReceiveInformationTimers(); |
954 } | 954 } |
955 | 955 |
956 // Called from RTCPsender. | 956 // Called from RTCPsender. |
957 int32_t ModuleRtpRtcpImpl::BoundingSet(bool* tmmbr_owner, | 957 int32_t ModuleRtpRtcpImpl::BoundingSet( |
958 TMMBRSet* bounding_set) { | 958 bool* tmmbr_owner, |
| 959 std::vector<rtcp::TmmbItem>* bounding_set) { |
959 return rtcp_receiver_.BoundingSet(tmmbr_owner, bounding_set); | 960 return rtcp_receiver_.BoundingSet(tmmbr_owner, bounding_set); |
960 } | 961 } |
961 | 962 |
962 int64_t ModuleRtpRtcpImpl::RtcpReportInterval() { | 963 int64_t ModuleRtpRtcpImpl::RtcpReportInterval() { |
963 if (audio_) | 964 if (audio_) |
964 return RTCP_INTERVAL_AUDIO_MS; | 965 return RTCP_INTERVAL_AUDIO_MS; |
965 else | 966 else |
966 return RTCP_INTERVAL_VIDEO_MS; | 967 return RTCP_INTERVAL_VIDEO_MS; |
967 } | 968 } |
968 | 969 |
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987 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( | 988 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( |
988 StreamDataCountersCallback* callback) { | 989 StreamDataCountersCallback* callback) { |
989 rtp_sender_.RegisterRtpStatisticsCallback(callback); | 990 rtp_sender_.RegisterRtpStatisticsCallback(callback); |
990 } | 991 } |
991 | 992 |
992 StreamDataCountersCallback* | 993 StreamDataCountersCallback* |
993 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { | 994 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { |
994 return rtp_sender_.GetRtpStatisticsCallback(); | 995 return rtp_sender_.GetRtpStatisticsCallback(); |
995 } | 996 } |
996 } // namespace webrtc | 997 } // namespace webrtc |
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