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Issue 2353543003: Added logging for audio send/receive stream configs. (Closed)
Patch Set: Refactored setting of header extensions into separate function. Created 4 years, 3 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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371 // TODO(solenberg): Some test cases in EndToEndTest use this from a different 371 // TODO(solenberg): Some test cases in EndToEndTest use this from a different
372 // thread. Re-enable once that is fixed. 372 // thread. Re-enable once that is fixed.
373 // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); 373 // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
374 return this; 374 return this;
375 } 375 }
376 376
377 webrtc::AudioSendStream* Call::CreateAudioSendStream( 377 webrtc::AudioSendStream* Call::CreateAudioSendStream(
378 const webrtc::AudioSendStream::Config& config) { 378 const webrtc::AudioSendStream::Config& config) {
379 TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream"); 379 TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
380 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); 380 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
381 event_log_->LogAudioSendStreamConfig(config);
381 AudioSendStream* send_stream = new AudioSendStream( 382 AudioSendStream* send_stream = new AudioSendStream(
382 config, config_.audio_state, &worker_queue_, congestion_controller_.get(), 383 config, config_.audio_state, &worker_queue_, congestion_controller_.get(),
383 bitrate_allocator_.get()); 384 bitrate_allocator_.get());
384 { 385 {
385 WriteLockScoped write_lock(*send_crit_); 386 WriteLockScoped write_lock(*send_crit_);
386 RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) == 387 RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
387 audio_send_ssrcs_.end()); 388 audio_send_ssrcs_.end());
388 audio_send_ssrcs_[config.rtp.ssrc] = send_stream; 389 audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
389 } 390 }
390 send_stream->SignalNetworkState(audio_network_state_); 391 send_stream->SignalNetworkState(audio_network_state_);
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408 RTC_DCHECK(num_deleted == 1); 409 RTC_DCHECK(num_deleted == 1);
409 } 410 }
410 UpdateAggregateNetworkState(); 411 UpdateAggregateNetworkState();
411 delete audio_send_stream; 412 delete audio_send_stream;
412 } 413 }
413 414
414 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( 415 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
415 const webrtc::AudioReceiveStream::Config& config) { 416 const webrtc::AudioReceiveStream::Config& config) {
416 TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream"); 417 TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
417 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); 418 RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
419 event_log_->LogAudioReceiveStreamConfig(config);
418 AudioReceiveStream* receive_stream = 420 AudioReceiveStream* receive_stream =
419 new AudioReceiveStream(congestion_controller_.get(), config, 421 new AudioReceiveStream(congestion_controller_.get(), config,
420 config_.audio_state, event_log_.get()); 422 config_.audio_state, event_log_.get());
421 { 423 {
422 WriteLockScoped write_lock(*receive_crit_); 424 WriteLockScoped write_lock(*receive_crit_);
423 RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) == 425 RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
424 audio_receive_ssrcs_.end()); 426 audio_receive_ssrcs_.end());
425 audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream; 427 audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
426 ConfigureSync(config.sync_group); 428 ConfigureSync(config.sync_group);
427 } 429 }
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943 // thread. Then this check can be enabled. 945 // thread. Then this check can be enabled.
944 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread()); 946 // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
945 if (RtpHeaderParser::IsRtcp(packet, length)) 947 if (RtpHeaderParser::IsRtcp(packet, length))
946 return DeliverRtcp(media_type, packet, length); 948 return DeliverRtcp(media_type, packet, length);
947 949
948 return DeliverRtp(media_type, packet, length, packet_time); 950 return DeliverRtp(media_type, packet, length, packet_time);
949 } 951 }
950 952
951 } // namespace internal 953 } // namespace internal
952 } // namespace webrtc 954 } // namespace webrtc
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