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Side by Side Diff: webrtc/video/video_receive_stream.cc

Issue 1452883002: Require negotiation to send transport cc feedback over RTCP. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Comments addressed Created 5 years, 1 month ago
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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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74 ss << "{remote_ssrc: " << remote_ssrc; 74 ss << "{remote_ssrc: " << remote_ssrc;
75 ss << ", local_ssrc: " << local_ssrc; 75 ss << ", local_ssrc: " << local_ssrc;
76 ss << ", rtcp_mode: " 76 ss << ", rtcp_mode: "
77 << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound" 77 << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
78 : "RtcpMode::kReducedSize"); 78 : "RtcpMode::kReducedSize");
79 ss << ", rtcp_xr: "; 79 ss << ", rtcp_xr: ";
80 ss << "{receiver_reference_time_report: " 80 ss << "{receiver_reference_time_report: "
81 << (rtcp_xr.receiver_reference_time_report ? "on" : "off"); 81 << (rtcp_xr.receiver_reference_time_report ? "on" : "off");
82 ss << '}'; 82 ss << '}';
83 ss << ", remb: " << (remb ? "on" : "off"); 83 ss << ", remb: " << (remb ? "on" : "off");
84 ss << ", transport_cc: " << (transport_cc ? "on" : "off");
84 ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}'; 85 ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
85 ss << ", fec: " << fec.ToString(); 86 ss << ", fec: " << fec.ToString();
86 ss << ", rtx: {"; 87 ss << ", rtx: {";
87 for (auto& kv : rtx) { 88 for (auto& kv : rtx) {
88 ss << kv.first << " -> "; 89 ss << kv.first << " -> ";
89 ss << "{ssrc: " << kv.second.ssrc; 90 ss << "{ssrc: " << kv.second.ssrc;
90 ss << ", payload_type: " << kv.second.payload_type; 91 ss << ", payload_type: " << kv.second.payload_type;
91 ss << '}'; 92 ss << '}';
92 } 93 }
93 ss << '}'; 94 ss << '}';
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146 ProcessThread* process_thread, 147 ProcessThread* process_thread,
147 CallStats* call_stats) 148 CallStats* call_stats)
148 : transport_adapter_(config.rtcp_send_transport), 149 : transport_adapter_(config.rtcp_send_transport),
149 encoded_frame_proxy_(config.pre_decode_callback), 150 encoded_frame_proxy_(config.pre_decode_callback),
150 config_(config), 151 config_(config),
151 clock_(Clock::GetRealTimeClock()), 152 clock_(Clock::GetRealTimeClock()),
152 congestion_controller_(congestion_controller), 153 congestion_controller_(congestion_controller),
153 call_stats_(call_stats) { 154 call_stats_(call_stats) {
154 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString(); 155 LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
155 156
156 bool send_side_bwe = UseSendSideBwe(config_.rtp.extensions); 157 bool send_side_bwe =
158 config.rtp.transport_cc && UseSendSideBwe(config_.rtp.extensions);
157 159
158 RemoteBitrateEstimator* bitrate_estimator = 160 RemoteBitrateEstimator* bitrate_estimator =
159 congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe); 161 congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe);
160 162
161 vie_channel_.reset(new ViEChannel( 163 vie_channel_.reset(new ViEChannel(
162 num_cpu_cores, &transport_adapter_, process_thread, nullptr, 164 num_cpu_cores, &transport_adapter_, process_thread, nullptr,
163 nullptr, nullptr, bitrate_estimator, call_stats_->rtcp_rtt_stats(), 165 nullptr, nullptr, bitrate_estimator, call_stats_->rtcp_rtt_stats(),
164 congestion_controller_->pacer(), congestion_controller_->packet_router(), 166 congestion_controller_->pacer(), congestion_controller_->packet_router(),
165 1, false)); 167 1, false));
166 168
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376 return 0; 378 return 0;
377 } 379 }
378 380
379 void VideoReceiveStream::SignalNetworkState(NetworkState state) { 381 void VideoReceiveStream::SignalNetworkState(NetworkState state) {
380 vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode 382 vie_channel_->SetRTCPMode(state == kNetworkUp ? config_.rtp.rtcp_mode
381 : RtcpMode::kOff); 383 : RtcpMode::kOff);
382 } 384 }
383 385
384 } // namespace internal 386 } // namespace internal
385 } // namespace webrtc 387 } // namespace webrtc
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