Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(83)

Side by Side Diff: webrtc/voice_engine/channel.cc

Issue 2661043003: Allow ANA to receive RPLR (recoverable packet loss rate) indications (Closed)
Patch Set: CR respone Created 3 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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 1295 matching lines...) Expand 10 before | Expand all | Expand 10 after
1306 1306
1307 void Channel::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) { 1307 void Channel::OnTwccBasedUplinkPacketLossRate(float packet_loss_rate) {
1308 if (!use_twcc_plr_for_ana_) 1308 if (!use_twcc_plr_for_ana_)
1309 return; 1309 return;
1310 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { 1310 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
1311 if (*encoder) 1311 if (*encoder)
1312 (*encoder)->OnReceivedUplinkPacketLossFraction(packet_loss_rate); 1312 (*encoder)->OnReceivedUplinkPacketLossFraction(packet_loss_rate);
1313 }); 1313 });
1314 } 1314 }
1315 1315
1316 void Channel::OnRecoverableUplinkPacketLossRate(
1317 float recoverable_packet_loss_rate) {
1318 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
1319 if (*encoder)
stefan-webrtc 2017/03/22 12:15:46 {}
elad.alon_webrtc.org 2017/03/22 15:15:55 Done.
1320 (*encoder)->OnReceivedUplinkRecoverablePacketLossFraction(
1321 recoverable_packet_loss_rate);
1322 });
1323 }
1324
1316 void Channel::OnUplinkPacketLossRate(float packet_loss_rate) { 1325 void Channel::OnUplinkPacketLossRate(float packet_loss_rate) {
1317 if (use_twcc_plr_for_ana_) 1326 if (use_twcc_plr_for_ana_)
1318 return; 1327 return;
1319 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) { 1328 audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
1320 if (*encoder) { 1329 if (*encoder) {
1321 (*encoder)->OnReceivedUplinkPacketLossFraction(packet_loss_rate); 1330 (*encoder)->OnReceivedUplinkPacketLossFraction(packet_loss_rate);
1322 } 1331 }
1323 }); 1332 });
1324 } 1333 }
1325 1334
(...skipping 1692 matching lines...) Expand 10 before | Expand all | Expand 10 after
3018 int64_t min_rtt = 0; 3027 int64_t min_rtt = 0;
3019 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) != 3028 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
3020 0) { 3029 0) {
3021 return 0; 3030 return 0;
3022 } 3031 }
3023 return rtt; 3032 return rtt;
3024 } 3033 }
3025 3034
3026 } // namespace voe 3035 } // namespace voe
3027 } // namespace webrtc 3036 } // namespace webrtc
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698