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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 |
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673 | 673 |
674 UpdateAggregateNetworkState(); | 674 UpdateAggregateNetworkState(); |
675 delete send_stream_impl; | 675 delete send_stream_impl; |
676 } | 676 } |
677 | 677 |
678 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream( | 678 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream( |
679 webrtc::VideoReceiveStream::Config configuration) { | 679 webrtc::VideoReceiveStream::Config configuration) { |
680 TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream"); | 680 TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream"); |
681 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); | 681 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
682 | 682 |
683 bool protected_by_flexfec = false; | |
684 { | |
685 ReadLockScoped read_lock(*receive_crit_); | |
686 protected_by_flexfec = | |
687 flexfec_receive_ssrcs_media_.find(configuration.rtp.remote_ssrc) != | |
688 flexfec_receive_ssrcs_media_.end(); | |
689 } | |
690 VideoReceiveStream* receive_stream = new VideoReceiveStream( | 683 VideoReceiveStream* receive_stream = new VideoReceiveStream( |
691 num_cpu_cores_, protected_by_flexfec, | 684 num_cpu_cores_, &packet_router_, std::move(configuration), |
692 &packet_router_, std::move(configuration), module_process_thread_.get(), | 685 module_process_thread_.get(), call_stats_.get(), &remb_); |
693 call_stats_.get(), &remb_); | |
694 | 686 |
695 const webrtc::VideoReceiveStream::Config& config = receive_stream->config(); | 687 const webrtc::VideoReceiveStream::Config& config = receive_stream->config(); |
696 ReceiveRtpConfig receive_config(config.rtp.extensions, | 688 ReceiveRtpConfig receive_config(config.rtp.extensions, |
697 UseSendSideBwe(config)); | 689 UseSendSideBwe(config)); |
698 { | 690 { |
699 WriteLockScoped write_lock(*receive_crit_); | 691 WriteLockScoped write_lock(*receive_crit_); |
700 RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) == | 692 RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) == |
701 video_receive_ssrcs_.end()); | 693 video_receive_ssrcs_.end()); |
702 video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream; | 694 video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream; |
703 if (config.rtp.rtx_ssrc) { | 695 if (config.rtp.rtx_ssrc) { |
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1285 if (media_type != MediaType::AUDIO || | 1277 if (media_type != MediaType::AUDIO || |
1286 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { | 1278 (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) { |
1287 congestion_controller_->OnReceivedPacket( | 1279 congestion_controller_->OnReceivedPacket( |
1288 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), | 1280 packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(), |
1289 header); | 1281 header); |
1290 } | 1282 } |
1291 } | 1283 } |
1292 | 1284 |
1293 } // namespace internal | 1285 } // namespace internal |
1294 } // namespace webrtc | 1286 } // namespace webrtc |
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