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

Issue 1452733002: rtcp::VoipMetric block moved into own file and got Parse function (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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785 } 785 }
786 786
787 // TODO(sprang): Add a unit test for this, or remove if the code isn't used. 787 // TODO(sprang): Add a unit test for this, or remove if the code isn't used.
788 rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildVoIPMetric( 788 rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildVoIPMetric(
789 const RtcpContext& context) { 789 const RtcpContext& context) {
790 rtcp::Xr* xr = new rtcp::Xr(); 790 rtcp::Xr* xr = new rtcp::Xr();
791 xr->From(ssrc_); 791 xr->From(ssrc_);
792 792
793 rtcp::VoipMetric voip; 793 rtcp::VoipMetric voip;
794 voip.To(remote_ssrc_); 794 voip.To(remote_ssrc_);
795 voip.LossRate(xr_voip_metric_.lossRate); 795 voip.WithVoipMetric(xr_voip_metric_);
796 voip.DiscardRate(xr_voip_metric_.discardRate);
797 voip.BurstDensity(xr_voip_metric_.burstDensity);
798 voip.GapDensity(xr_voip_metric_.gapDensity);
799 voip.BurstDuration(xr_voip_metric_.burstDuration);
800 voip.GapDuration(xr_voip_metric_.gapDuration);
801 voip.RoundTripDelay(xr_voip_metric_.roundTripDelay);
802 voip.EndSystemDelay(xr_voip_metric_.endSystemDelay);
803 voip.SignalLevel(xr_voip_metric_.signalLevel);
804 voip.NoiseLevel(xr_voip_metric_.noiseLevel);
805 voip.Rerl(xr_voip_metric_.RERL);
806 voip.Gmin(xr_voip_metric_.Gmin);
807 voip.Rfactor(xr_voip_metric_.Rfactor);
808 voip.ExtRfactor(xr_voip_metric_.extRfactor);
809 voip.MosLq(xr_voip_metric_.MOSLQ);
810 voip.MosCq(xr_voip_metric_.MOSCQ);
811 voip.RxConfig(xr_voip_metric_.RXconfig);
812 voip.JbNominal(xr_voip_metric_.JBnominal);
813 voip.JbMax(xr_voip_metric_.JBmax);
814 voip.JbAbsMax(xr_voip_metric_.JBabsMax);
815 796
816 xr->WithVoipMetric(&voip); 797 xr->WithVoipMetric(&voip);
817 798
818 return rtc::scoped_ptr<rtcp::Xr>(xr); 799 return rtc::scoped_ptr<rtcp::Xr>(xr);
819 } 800 }
820 801
821 int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state, 802 int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
822 RTCPPacketType packetType, 803 RTCPPacketType packetType,
823 int32_t nack_size, 804 int32_t nack_size,
824 const uint16_t* nack_list, 805 const uint16_t* nack_list,
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1088 Transport* const transport_; 1069 Transport* const transport_;
1089 bool send_failure_; 1070 bool send_failure_;
1090 } sender(transport_); 1071 } sender(transport_);
1091 1072
1092 uint8_t buffer[IP_PACKET_SIZE]; 1073 uint8_t buffer[IP_PACKET_SIZE];
1093 return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) && 1074 return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
1094 !sender.send_failure_; 1075 !sender.send_failure_;
1095 } 1076 }
1096 1077
1097 } // namespace webrtc 1078 } // namespace webrtc
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