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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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641 num_cpu_cores_, congestion_controller_.get(), &packet_router_, | 641 num_cpu_cores_, congestion_controller_.get(), &packet_router_, |
642 std::move(configuration), voice_engine(), module_process_thread_.get(), | 642 std::move(configuration), voice_engine(), module_process_thread_.get(), |
643 call_stats_.get(), &remb_); | 643 call_stats_.get(), &remb_); |
644 | 644 |
645 const webrtc::VideoReceiveStream::Config& config = receive_stream->config(); | 645 const webrtc::VideoReceiveStream::Config& config = receive_stream->config(); |
646 { | 646 { |
647 WriteLockScoped write_lock(*receive_crit_); | 647 WriteLockScoped write_lock(*receive_crit_); |
648 RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) == | 648 RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) == |
649 video_receive_ssrcs_.end()); | 649 video_receive_ssrcs_.end()); |
650 video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream; | 650 video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream; |
651 // TODO(pbos): Configure different RTX payloads per receive payload. | 651 if (!config.rtp.rtx_payload_types.empty()) |
652 VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it = | 652 video_receive_ssrcs_[config.rtp.rtx_ssrc] = receive_stream; |
653 config.rtp.rtx.begin(); | |
654 if (it != config.rtp.rtx.end()) | |
655 video_receive_ssrcs_[it->second.ssrc] = receive_stream; | |
656 video_receive_streams_.insert(receive_stream); | 653 video_receive_streams_.insert(receive_stream); |
657 ConfigureSync(config.sync_group); | 654 ConfigureSync(config.sync_group); |
658 } | 655 } |
659 receive_stream->SignalNetworkState(video_network_state_); | 656 receive_stream->SignalNetworkState(video_network_state_); |
660 UpdateAggregateNetworkState(); | 657 UpdateAggregateNetworkState(); |
661 event_log_->LogVideoReceiveStreamConfig(config); | 658 event_log_->LogVideoReceiveStreamConfig(config); |
662 return receive_stream; | 659 return receive_stream; |
663 } | 660 } |
664 | 661 |
665 void Call::DestroyVideoReceiveStream( | 662 void Call::DestroyVideoReceiveStream( |
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1201 | 1198 |
1202 void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet) { | 1199 void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet) { |
1203 RTPHeader header; | 1200 RTPHeader header; |
1204 packet.GetHeader(&header); | 1201 packet.GetHeader(&header); |
1205 congestion_controller_->OnReceivedPacket(packet.arrival_time_ms(), | 1202 congestion_controller_->OnReceivedPacket(packet.arrival_time_ms(), |
1206 packet.payload_size(), header); | 1203 packet.payload_size(), header); |
1207 } | 1204 } |
1208 | 1205 |
1209 } // namespace internal | 1206 } // namespace internal |
1210 } // namespace webrtc | 1207 } // namespace webrtc |
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