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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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504 uint32_t send_bandwidth = 0; | 504 uint32_t send_bandwidth = 0; |
505 congestion_controller_->GetBitrateController()->AvailableBandwidth( | 505 congestion_controller_->GetBitrateController()->AvailableBandwidth( |
506 &send_bandwidth); | 506 &send_bandwidth); |
507 std::vector<unsigned int> ssrcs; | 507 std::vector<unsigned int> ssrcs; |
508 uint32_t recv_bandwidth = 0; | 508 uint32_t recv_bandwidth = 0; |
509 congestion_controller_->GetRemoteBitrateEstimator(false)->LatestEstimate( | 509 congestion_controller_->GetRemoteBitrateEstimator(false)->LatestEstimate( |
510 &ssrcs, &recv_bandwidth); | 510 &ssrcs, &recv_bandwidth); |
511 stats.send_bandwidth_bps = send_bandwidth; | 511 stats.send_bandwidth_bps = send_bandwidth; |
512 stats.recv_bandwidth_bps = recv_bandwidth; | 512 stats.recv_bandwidth_bps = recv_bandwidth; |
513 stats.pacer_delay_ms = congestion_controller_->GetPacerQueuingDelayMs(); | 513 stats.pacer_delay_ms = congestion_controller_->GetPacerQueuingDelayMs(); |
514 { | 514 stats.rtt_ms = call_stats_->rtcp_rtt_stats()->LastProcessedRtt(); |
515 ReadLockScoped read_lock(*send_crit_); | |
516 // TODO(solenberg): Add audio send streams. | |
517 for (const auto& kv : video_send_ssrcs_) { | |
518 int rtt_ms = kv.second->GetRtt(); | |
519 if (rtt_ms > 0) | |
520 stats.rtt_ms = rtt_ms; | |
521 } | |
522 } | |
523 return stats; | 515 return stats; |
524 } | 516 } |
525 | 517 |
526 void Call::SetBitrateConfig( | 518 void Call::SetBitrateConfig( |
527 const webrtc::Call::Config::BitrateConfig& bitrate_config) { | 519 const webrtc::Call::Config::BitrateConfig& bitrate_config) { |
528 TRACE_EVENT0("webrtc", "Call::SetBitrateConfig"); | 520 TRACE_EVENT0("webrtc", "Call::SetBitrateConfig"); |
529 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); | 521 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
530 RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0); | 522 RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0); |
531 if (bitrate_config.max_bitrate_bps != -1) | 523 if (bitrate_config.max_bitrate_bps != -1) |
532 RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0); | 524 RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0); |
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742 // thread. Then this check can be enabled. | 734 // thread. Then this check can be enabled. |
743 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); | 735 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); |
744 if (RtpHeaderParser::IsRtcp(packet, length)) | 736 if (RtpHeaderParser::IsRtcp(packet, length)) |
745 return DeliverRtcp(media_type, packet, length); | 737 return DeliverRtcp(media_type, packet, length); |
746 | 738 |
747 return DeliverRtp(media_type, packet, length, packet_time); | 739 return DeliverRtp(media_type, packet, length, packet_time); |
748 } | 740 } |
749 | 741 |
750 } // namespace internal | 742 } // namespace internal |
751 } // namespace webrtc | 743 } // namespace webrtc |
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