Log Call {audio, video} stream deletions.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 158 matching lines @@
   RTC_CHECK_EQ(0u, video_receive_streams_.size());
 
   module_process_thread_->Stop();
   Trace::ReturnTrace();
 }
 
 PacketReceiver* Call::Receiver() { return this; }
 
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
+  // TODO(pbos): When adding AudioSendStream, add both TRACE_EVENT0 and config
+  // logging to AudioSendStream constructor.
   return nullptr;
 }
 
 void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
+  // TODO(pbos): When adding AudioSendStream, add both TRACE_EVENT0 and config
+  // logging to AudioSendStream destructor.
 }
 
 webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
-  LOG(LS_INFO) << "CreateAudioReceiveStream: " << config.ToString();
   AudioReceiveStream* receive_stream = new AudioReceiveStream(
       channel_group_->GetRemoteBitrateEstimator(), config);
   {
     WriteLockScoped write_lock(*receive_crit_);
     RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
                audio_receive_ssrcs_.end());
     audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
     ConfigureSync(config.sync_group);
   }
   return receive_stream;
@@ skipping 18 matching lines @@
       ConfigureSync(sync_group);
     }
   }
   delete audio_receive_stream;
 }
 
 webrtc::VideoSendStream* Call::CreateVideoSendStream(
     const webrtc::VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
-  LOG(LS_INFO) << "CreateVideoSendStream: " << config.ToString();
-  RTC_DCHECK(!config.rtp.ssrcs.empty());
 
   // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
   // the call has already started.
   VideoSendStream* send_stream = new VideoSendStream(num_cpu_cores_,
       module_process_thread_.get(), channel_group_.get(),
       rtc::AtomicOps::Increment(&next_channel_id_), config, encoder_config,
       suspended_video_send_ssrcs_);
 
   // This needs to be taken before send_crit_ as both locks need to be held
   // while changing network state.
@@ skipping 42 matching lines @@
        ++it) {
     suspended_video_send_ssrcs_[it->first] = it->second;
   }
 
   delete send_stream_impl;
 }
 
 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
     const webrtc::VideoReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
-  LOG(LS_INFO) << "CreateVideoReceiveStream: " << config.ToString();
   VideoReceiveStream* receive_stream = new VideoReceiveStream(
       num_cpu_cores_, channel_group_.get(),
       rtc::AtomicOps::Increment(&next_channel_id_), config,
       config_.voice_engine);
 
   // This needs to be taken before receive_crit_ as both locks need to be held
   // while changing network state.
   rtc::CritScope lock(&network_enabled_crit_);
   WriteLockScoped write_lock(*receive_crit_);
   RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
@@ skipping 230 matching lines @@
     size_t length,
     const PacketTime& packet_time) {
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc