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

Issue 2133073002: Add periodic logging of video stats. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: use rtc::TimeMillis Created 4 years, 5 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 << "],";
pbos-webrtc 2016/07/10 16:58:50 Pref. removing []s because they look like arrays o
åsapersson 2016/08/01 15:55:22 Done.
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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868 // thread. Then this check can be enabled. 880 // thread. Then this check can be enabled.
869 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); 881 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
870 if (RtpHeaderParser::IsRtcp(packet, length)) 882 if (RtpHeaderParser::IsRtcp(packet, length))
871 return DeliverRtcp(media_type, packet, length); 883 return DeliverRtcp(media_type, packet, length);
872 884
873 return DeliverRtp(media_type, packet, length, packet_time); 885 return DeliverRtp(media_type, packet, length, packet_time);
874 } 886 }
875 887
876 } // namespace internal 888 } // namespace internal
877 } // namespace webrtc 889 } // namespace webrtc
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