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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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363 // TODO(solenberg): Some test cases in EndToEndTest use this from a different | 363 // TODO(solenberg): Some test cases in EndToEndTest use this from a different |
364 // thread. Re-enable once that is fixed. | 364 // thread. Re-enable once that is fixed. |
365 // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); | 365 // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
366 return this; | 366 return this; |
367 } | 367 } |
368 | 368 |
369 webrtc::AudioSendStream* Call::CreateAudioSendStream( | 369 webrtc::AudioSendStream* Call::CreateAudioSendStream( |
370 const webrtc::AudioSendStream::Config& config) { | 370 const webrtc::AudioSendStream::Config& config) { |
371 TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream"); | 371 TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream"); |
372 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); | 372 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
| 373 event_log_->LogAudioSendStreamConfig(config); |
373 AudioSendStream* send_stream = new AudioSendStream( | 374 AudioSendStream* send_stream = new AudioSendStream( |
374 config, config_.audio_state, &worker_queue_, congestion_controller_.get(), | 375 config, config_.audio_state, &worker_queue_, congestion_controller_.get(), |
375 bitrate_allocator_.get(), event_log_); | 376 bitrate_allocator_.get(), event_log_); |
376 { | 377 { |
377 WriteLockScoped write_lock(*send_crit_); | 378 WriteLockScoped write_lock(*send_crit_); |
378 RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) == | 379 RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) == |
379 audio_send_ssrcs_.end()); | 380 audio_send_ssrcs_.end()); |
380 audio_send_ssrcs_[config.rtp.ssrc] = send_stream; | 381 audio_send_ssrcs_[config.rtp.ssrc] = send_stream; |
381 } | 382 } |
382 send_stream->SignalNetworkState(audio_network_state_); | 383 send_stream->SignalNetworkState(audio_network_state_); |
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400 RTC_DCHECK(num_deleted == 1); | 401 RTC_DCHECK(num_deleted == 1); |
401 } | 402 } |
402 UpdateAggregateNetworkState(); | 403 UpdateAggregateNetworkState(); |
403 delete audio_send_stream; | 404 delete audio_send_stream; |
404 } | 405 } |
405 | 406 |
406 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( | 407 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( |
407 const webrtc::AudioReceiveStream::Config& config) { | 408 const webrtc::AudioReceiveStream::Config& config) { |
408 TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream"); | 409 TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream"); |
409 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); | 410 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
| 411 event_log_->LogAudioReceiveStreamConfig(config); |
410 AudioReceiveStream* receive_stream = new AudioReceiveStream( | 412 AudioReceiveStream* receive_stream = new AudioReceiveStream( |
411 congestion_controller_.get(), config, config_.audio_state, event_log_); | 413 congestion_controller_.get(), config, config_.audio_state, event_log_); |
412 { | 414 { |
413 WriteLockScoped write_lock(*receive_crit_); | 415 WriteLockScoped write_lock(*receive_crit_); |
414 RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) == | 416 RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) == |
415 audio_receive_ssrcs_.end()); | 417 audio_receive_ssrcs_.end()); |
416 audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream; | 418 audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream; |
417 ConfigureSync(config.sync_group); | 419 ConfigureSync(config.sync_group); |
418 } | 420 } |
419 receive_stream->SignalNetworkState(audio_network_state_); | 421 receive_stream->SignalNetworkState(audio_network_state_); |
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933 // thread. Then this check can be enabled. | 935 // thread. Then this check can be enabled. |
934 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); | 936 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); |
935 if (RtpHeaderParser::IsRtcp(packet, length)) | 937 if (RtpHeaderParser::IsRtcp(packet, length)) |
936 return DeliverRtcp(media_type, packet, length); | 938 return DeliverRtcp(media_type, packet, length); |
937 | 939 |
938 return DeliverRtp(media_type, packet, length, packet_time); | 940 return DeliverRtp(media_type, packet, length, packet_time); |
939 } | 941 } |
940 | 942 |
941 } // namespace internal | 943 } // namespace internal |
942 } // namespace webrtc | 944 } // namespace webrtc |
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