Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(240)

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

Issue 2234783002: Cleaned out candidateSet member from TMMBRHelp class (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 4 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 1215 matching lines...) Expand 10 before | Expand all | Expand 10 after
1226 RTCPHelp::RTCPPacketInformation* rtcp_packet_information) { 1226 RTCPHelp::RTCPPacketInformation* rtcp_packet_information) {
1227 rtcp::RtcpPacket* packet = rtcp_parser->ReleaseRtcpPacket(); 1227 rtcp::RtcpPacket* packet = rtcp_parser->ReleaseRtcpPacket();
1228 RTC_DCHECK(packet != nullptr); 1228 RTC_DCHECK(packet != nullptr);
1229 rtcp_packet_information->rtcpPacketTypeFlags |= kRtcpTransportFeedback; 1229 rtcp_packet_information->rtcpPacketTypeFlags |= kRtcpTransportFeedback;
1230 rtcp_packet_information->transport_feedback_.reset( 1230 rtcp_packet_information->transport_feedback_.reset(
1231 static_cast<rtcp::TransportFeedback*>(packet)); 1231 static_cast<rtcp::TransportFeedback*>(packet));
1232 1232
1233 rtcp_parser->Iterate(); 1233 rtcp_parser->Iterate();
1234 } 1234 }
1235 int32_t RTCPReceiver::UpdateTMMBR() { 1235 int32_t RTCPReceiver::UpdateTMMBR() {
1236 TMMBRHelp tmmbr_help;
1237 uint32_t bitrate = 0;
1238 uint32_t accNumCandidates = 0;
1239
1240 int32_t size = TMMBRReceived(0, 0, NULL); 1236 int32_t size = TMMBRReceived(0, 0, NULL);
1237 TMMBRSet candidates;
1241 if (size > 0) { 1238 if (size > 0) {
1242 TMMBRSet* candidateSet = tmmbr_help.VerifyAndAllocateCandidateSet(size); 1239 candidates.reserve(size);
1243 // Get candidate set from receiver. 1240 // Get candidate set from receiver.
1244 accNumCandidates = TMMBRReceived(size, accNumCandidates, candidateSet); 1241 TMMBRReceived(size, 0, &candidates);
1245 } 1242 }
1246 // Find bounding set 1243 // Find bounding set
1247 std::vector<rtcp::TmmbItem> bounding = tmmbr_help.FindTMMBRBoundingSet(); 1244 std::vector<rtcp::TmmbItem> bounding =
1245 TMMBRHelp::FindBoundingSet(std::move(candidates));
1248 // Set bounding set 1246 // Set bounding set
1249 // Inform remote clients about the new bandwidth 1247 // Inform remote clients about the new bandwidth
1250 // inform the remote client 1248 // inform the remote client
1251 _rtpRtcp.SetTMMBN(&bounding); 1249 _rtpRtcp.SetTMMBN(&bounding);
1252 1250
1253 // might trigger a TMMBN 1251 // might trigger a TMMBN
1254 if (bounding.empty()) { 1252 if (bounding.empty()) {
1255 // owner of max bitrate request has timed out 1253 // owner of max bitrate request has timed out
1256 // empty bounding set has been sent 1254 // empty bounding set has been sent
1257 return 0; 1255 return 0;
1258 } 1256 }
1259 // Get net bitrate from bounding set depending on sent packet rate 1257 // We have a new bandwidth estimate on this channel.
danilchap 2016/08/10 11:18:48 comment say bounding set is used, but code used ca
1260 if (tmmbr_help.CalcMinBitRate(&bitrate)) { 1258 if (_cbRtcpBandwidthObserver) {
1261 // we have a new bandwidth estimate on this channel 1259 uint64_t bitrate_bps = TMMBRHelp::CalcMinBitrateBps(bounding);
1262 if (_cbRtcpBandwidthObserver) { 1260 _cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(bitrate_bps);
1263 _cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(bitrate * 1000);
1264 }
1265 } 1261 }
1266 return 0; 1262 return 0;
1267 } 1263 }
1268 1264
1269 void RTCPReceiver::RegisterRtcpStatisticsCallback( 1265 void RTCPReceiver::RegisterRtcpStatisticsCallback(
1270 RtcpStatisticsCallback* callback) { 1266 RtcpStatisticsCallback* callback) {
1271 rtc::CritScope cs(&_criticalSectionFeedbacks); 1267 rtc::CritScope cs(&_criticalSectionFeedbacks);
1272 stats_callback_ = callback; 1268 stats_callback_ = callback;
1273 } 1269 }
1274 1270
(...skipping 159 matching lines...) Expand 10 before | Expand all | Expand 10 after
1434 return -1; 1430 return -1;
1435 } 1431 }
1436 num += receiveInfo->TmmbrSet.lengthOfSet(); 1432 num += receiveInfo->TmmbrSet.lengthOfSet();
1437 receiveInfoIt++; 1433 receiveInfoIt++;
1438 } 1434 }
1439 } 1435 }
1440 return num; 1436 return num;
1441 } 1437 }
1442 1438
1443 } // namespace webrtc 1439 } // namespace webrtc
OLDNEW
« no previous file with comments | « no previous file | webrtc/modules/rtp_rtcp/source/rtcp_sender.cc » ('j') | webrtc/modules/rtp_rtcp/source/tmmbr_help.cc » ('J')

Powered by Google App Engine
This is Rietveld 408576698