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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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