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Issue 1213603002: Remove ResetStatistics from RTP feedback. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: comment + rebase 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
(...skipping 522 matching lines...)
533 int64_t* min_rtt, 533 int64_t* min_rtt,
534 int64_t* max_rtt) const { 534 int64_t* max_rtt) const {
535 int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt); 535 int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt);
536 if (rtt && *rtt == 0) { 536 if (rtt && *rtt == 0) {
537 // Try to get RTT from RtcpRttStats class. 537 // Try to get RTT from RtcpRttStats class.
538 *rtt = rtt_ms(); 538 *rtt = rtt_ms();
539 } 539 }
540 return ret; 540 return ret;
541 } 541 }
542 542
543 // Reset RTP data counters for the sending side.
544 int32_t ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
545 rtp_sender_.ResetDataCounters();
546 return 0; // TODO(pwestin): change to void.
547 }
548
549 // Force a send of an RTCP packet. 543 // Force a send of an RTCP packet.
550 // Normal SR and RR are triggered via the process function. 544 // Normal SR and RR are triggered via the process function.
551 int32_t ModuleRtpRtcpImpl::SendRTCP(RTCPPacketType packet_type) { 545 int32_t ModuleRtpRtcpImpl::SendRTCP(RTCPPacketType packet_type) {
552 return rtcp_sender_.SendRTCP(GetFeedbackState(), packet_type); 546 return rtcp_sender_.SendRTCP(GetFeedbackState(), packet_type);
553 } 547 }
554 548
555 // Force a send of an RTCP packet. 549 // Force a send of an RTCP packet.
556 // Normal SR and RR are triggered via the process function. 550 // Normal SR and RR are triggered via the process function.
557 int32_t ModuleRtpRtcpImpl::SendCompoundRTCP( 551 int32_t ModuleRtpRtcpImpl::SendCompoundRTCP(
558 const std::set<RTCPPacketType>& packet_types) { 552 const std::set<RTCPPacketType>& packet_types) {
(...skipping 416 matching lines...)
975 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 969 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
976 StreamDataCountersCallback* callback) { 970 StreamDataCountersCallback* callback) {
977 rtp_sender_.RegisterRtpStatisticsCallback(callback); 971 rtp_sender_.RegisterRtpStatisticsCallback(callback);
978 } 972 }
979 973
980 StreamDataCountersCallback* 974 StreamDataCountersCallback*
981 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 975 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
982 return rtp_sender_.GetRtpStatisticsCallback(); 976 return rtp_sender_.GetRtpStatisticsCallback();
983 } 977 }
984 } // namespace webrtc 978 } // namespace webrtc
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