Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(265)

Side by Side Diff: webrtc/call/call.cc

Issue 2712683002: Add |protected_by_flexfec| flag to VideoReceiveStream::Config. (Closed)
Patch Set: Rebase. 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
« no previous file with comments | « no previous file | webrtc/media/engine/webrtcvideoengine2.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 664 matching lines...) Expand 10 before | Expand all | Expand 10 after
675 675
676 UpdateAggregateNetworkState(); 676 UpdateAggregateNetworkState();
677 delete send_stream_impl; 677 delete send_stream_impl;
678 } 678 }
679 679
680 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream( 680 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
681 webrtc::VideoReceiveStream::Config configuration) { 681 webrtc::VideoReceiveStream::Config configuration) {
682 TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream"); 682 TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
683 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); 683 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
684 684
685 bool protected_by_flexfec = false;
686 {
687 ReadLockScoped read_lock(*receive_crit_);
688 protected_by_flexfec =
689 flexfec_receive_ssrcs_media_.find(configuration.rtp.remote_ssrc) !=
690 flexfec_receive_ssrcs_media_.end();
691 }
692 VideoReceiveStream* receive_stream = new VideoReceiveStream( 685 VideoReceiveStream* receive_stream = new VideoReceiveStream(
693 num_cpu_cores_, protected_by_flexfec, 686 num_cpu_cores_, &packet_router_, std::move(configuration),
694 &packet_router_, std::move(configuration), module_process_thread_.get(), 687 module_process_thread_.get(), call_stats_.get(), &remb_);
695 call_stats_.get(), &remb_);
696 688
697 const webrtc::VideoReceiveStream::Config& config = receive_stream->config(); 689 const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
698 ReceiveRtpConfig receive_config(config.rtp.extensions, 690 ReceiveRtpConfig receive_config(config.rtp.extensions,
699 UseSendSideBwe(config)); 691 UseSendSideBwe(config));
700 { 692 {
701 WriteLockScoped write_lock(*receive_crit_); 693 WriteLockScoped write_lock(*receive_crit_);
702 RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) == 694 RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
703 video_receive_ssrcs_.end()); 695 video_receive_ssrcs_.end());
704 video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream; 696 video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
705 if (config.rtp.rtx_ssrc) { 697 if (config.rtp.rtx_ssrc) {
(...skipping 581 matching lines...) Expand 10 before | Expand all | Expand 10 after
1287 if (media_type != MediaType::AUDIO || 1279 if (media_type != MediaType::AUDIO ||
1288 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { 1280 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
1289 congestion_controller_->OnReceivedPacket( 1281 congestion_controller_->OnReceivedPacket(
1290 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), 1282 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
1291 header); 1283 header);
1292 } 1284 }
1293 } 1285 }
1294 1286
1295 } // namespace internal 1287 } // namespace internal
1296 } // namespace webrtc 1288 } // namespace webrtc
OLDNEW
« no previous file with comments | « no previous file | webrtc/media/engine/webrtcvideoengine2.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698