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Issue 2133073002: Add periodic logging of video stats. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 4 months 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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194 std::map<std::string, rtc::NetworkRoute> network_routes_; 194 std::map<std::string, rtc::NetworkRoute> network_routes_;
195 195
196 VieRemb remb_; 196 VieRemb remb_;
197 const std::unique_ptr<CongestionController> congestion_controller_; 197 const std::unique_ptr<CongestionController> congestion_controller_;
198 const std::unique_ptr<SendDelayStats> video_send_delay_stats_; 198 const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
199 199
200 RTC_DISALLOW_COPY_AND_ASSIGN(Call); 200 RTC_DISALLOW_COPY_AND_ASSIGN(Call);
201 }; 201 };
202 } // namespace internal 202 } // namespace internal
203 203
204 std::string Call::Stats::ToString(int64_t time_ms) const {
205 std::stringstream ss;
206 ss << "Call stats: " << time_ms << ", {";
207 ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
208 ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
209 ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
210 ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
211 ss << "rtt_ms: " << rtt_ms;
212 ss << '}';
213 return ss.str();
214 }
215
204 Call* Call::Create(const Call::Config& config) { 216 Call* Call::Create(const Call::Config& config) {
205 return new internal::Call(config); 217 return new internal::Call(config);
206 } 218 }
207 219
208 namespace internal { 220 namespace internal {
209 221
210 Call::Call(const Call::Config& config) 222 Call::Call(const Call::Config& config)
211 : clock_(Clock::GetRealTimeClock()), 223 : clock_(Clock::GetRealTimeClock()),
212 num_cpu_cores_(CpuInfo::DetectNumberOfCores()), 224 num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
213 module_process_thread_(ProcessThread::Create("ModuleProcessThread")), 225 module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
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877 // thread. Then this check can be enabled. 889 // thread. Then this check can be enabled.
878 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); 890 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
879 if (RtpHeaderParser::IsRtcp(packet, length)) 891 if (RtpHeaderParser::IsRtcp(packet, length))
880 return DeliverRtcp(media_type, packet, length); 892 return DeliverRtcp(media_type, packet, length);
881 893
882 return DeliverRtp(media_type, packet, length, packet_time); 894 return DeliverRtp(media_type, packet, length, packet_time);
883 } 895 }
884 896
885 } // namespace internal 897 } // namespace internal
886 } // namespace webrtc 898 } // namespace webrtc
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