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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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419 return this; | 419 return this; |
420 } | 420 } |
421 | 421 |
422 webrtc::AudioSendStream* Call::CreateAudioSendStream( | 422 webrtc::AudioSendStream* Call::CreateAudioSendStream( |
423 const webrtc::AudioSendStream::Config& config) { | 423 const webrtc::AudioSendStream::Config& config) { |
424 TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream"); | 424 TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream"); |
425 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); | 425 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
426 event_log_->LogAudioSendStreamConfig(config); | 426 event_log_->LogAudioSendStreamConfig(config); |
427 AudioSendStream* send_stream = new AudioSendStream( | 427 AudioSendStream* send_stream = new AudioSendStream( |
428 config, config_.audio_state, &worker_queue_, &packet_router_, | 428 config, config_.audio_state, &worker_queue_, &packet_router_, |
429 congestion_controller_.get(), bitrate_allocator_.get(), event_log_); | 429 congestion_controller_.get(), bitrate_allocator_.get(), event_log_, |
| 430 call_stats_->rtcp_rtt_stats()); |
430 { | 431 { |
431 WriteLockScoped write_lock(*send_crit_); | 432 WriteLockScoped write_lock(*send_crit_); |
432 RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) == | 433 RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) == |
433 audio_send_ssrcs_.end()); | 434 audio_send_ssrcs_.end()); |
434 audio_send_ssrcs_[config.rtp.ssrc] = send_stream; | 435 audio_send_ssrcs_[config.rtp.ssrc] = send_stream; |
435 } | 436 } |
436 { | 437 { |
437 ReadLockScoped read_lock(*receive_crit_); | 438 ReadLockScoped read_lock(*receive_crit_); |
438 for (const auto& kv : audio_receive_ssrcs_) { | 439 for (const auto& kv : audio_receive_ssrcs_) { |
439 if (kv.second->config().rtp.local_ssrc == config.rtp.ssrc) { | 440 if (kv.second->config().rtp.local_ssrc == config.rtp.ssrc) { |
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1122 uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]); | 1123 uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]); |
1123 ReadLockScoped read_lock(*receive_crit_); | 1124 ReadLockScoped read_lock(*receive_crit_); |
1124 auto it = video_receive_ssrcs_.find(ssrc); | 1125 auto it = video_receive_ssrcs_.find(ssrc); |
1125 if (it == video_receive_ssrcs_.end()) | 1126 if (it == video_receive_ssrcs_.end()) |
1126 return false; | 1127 return false; |
1127 return it->second->OnRecoveredPacket(packet, length); | 1128 return it->second->OnRecoveredPacket(packet, length); |
1128 } | 1129 } |
1129 | 1130 |
1130 } // namespace internal | 1131 } // namespace internal |
1131 } // namespace webrtc | 1132 } // namespace webrtc |
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