Remove network_enabled_crit_ in call.cc.

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 83 matching lines @@
   void ConfigureSync(const std::string& sync_group)
       EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
 
   const int num_cpu_cores_;
   const rtc::scoped_ptr<ProcessThread> module_process_thread_;
   const rtc::scoped_ptr<CallStats> call_stats_;
   const rtc::scoped_ptr<CongestionController> congestion_controller_;
   Call::Config config_;
   rtc::ThreadChecker configuration_thread_checker_;
 
-  // Needs to be held while write-locking |receive_crit_| or |send_crit_|. This
-  // ensures that we have a consistent network state signalled to all senders
-  // and receivers.
-  rtc::CriticalSection network_enabled_crit_;
-  bool network_enabled_ GUARDED_BY(network_enabled_crit_);
+ bool network_enabled_;
 
   rtc::scoped_ptr<RWLockWrapper> receive_crit_;
   // Audio and Video receive streams are owned by the client that creates them.
   std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_
       GUARDED_BY(receive_crit_);
   std::map<uint32_t, VideoReceiveStream*> video_receive_ssrcs_
       GUARDED_BY(receive_crit_);
   std::set<VideoReceiveStream*> video_receive_streams_
       GUARDED_BY(receive_crit_);
   std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
@@ skipping 76 matching lines @@
   // RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   return this;
 }
 
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   AudioSendStream* send_stream = new AudioSendStream(config);
   {
-    rtc::CritScope lock(&network_enabled_crit_);
     WriteLockScoped write_lock(*send_crit_);
     RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
                audio_send_ssrcs_.end());
     audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
 
     if (!network_enabled_)

[pbos-webrtc, 2015/10/22 15:19:11]

Do this outside write_lock.

[mflodman, 2015/10/22 15:26:15]

Done.

       send_stream->SignalNetworkState(kNetworkDown);
   }
   return send_stream;
 }
 
 void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(send_stream != nullptr);
 
@@ skipping 55 matching lines @@
   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   // 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(), call_stats_.get(),
       congestion_controller_.get(), 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.
-  rtc::CritScope lock(&network_enabled_crit_);
   WriteLockScoped write_lock(*send_crit_);
   for (uint32_t ssrc : config.rtp.ssrcs) {
     RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
     video_send_ssrcs_[ssrc] = send_stream;
   }
   video_send_streams_.insert(send_stream);
 
   if (event_log_)
     event_log_->LogVideoSendStreamConfig(config);
 
   if (!network_enabled_)
     send_stream->SignalNetworkState(kNetworkDown);

[pbos-webrtc, 2015/10/22 15:19:10]

Do this before taking write_lock.

[mflodman, 2015/10/22 15:26:15]

Done.

   return send_stream;
 }
 
 void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
   TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
   RTC_DCHECK(send_stream != nullptr);
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
 
   send_stream->Stop();
 
@@ skipping 25 matching lines @@
 }
 
 webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
     const webrtc::VideoReceiveStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   VideoReceiveStream* receive_stream = new VideoReceiveStream(
       num_cpu_cores_, congestion_controller_.get(), config,
       config_.voice_engine, module_process_thread_.get(), call_stats_.get());
 
-  // 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) ==
              video_receive_ssrcs_.end());
   video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
   // TODO(pbos): Configure different RTX payloads per receive payload.
   VideoReceiveStream::Config::Rtp::RtxMap::const_iterator it =
       config.rtp.rtx.begin();
   if (it != config.rtp.rtx.end())
     video_receive_ssrcs_[it->second.ssrc] = receive_stream;
   video_receive_streams_.insert(receive_stream);
@@ skipping 83 matching lines @@
     return;
   }
   config_.bitrate_config = bitrate_config;
   congestion_controller_->SetBweBitrates(bitrate_config.min_bitrate_bps,
                                          bitrate_config.start_bitrate_bps,
                                          bitrate_config.max_bitrate_bps);
 }
 
 void Call::SignalNetworkState(NetworkState state) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
-  // Take crit for entire function, it needs to be held while updating streams
-  // to guarantee a consistent state across streams.
-  rtc::CritScope lock(&network_enabled_crit_);
   network_enabled_ = state == kNetworkUp;
   congestion_controller_->SignalNetworkState(state);
   {
     ReadLockScoped write_lock(*send_crit_);
     for (auto& kv : audio_send_ssrcs_) {
       kv.second->SignalNetworkState(state);
     }
     for (auto& kv : video_send_ssrcs_) {
       kv.second->SignalNetworkState(state);
     }
@@ skipping 136 matching lines @@
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!configuration_thread_checker_.CalledOnValidThread());
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 }  // namespace internal
 }  // namespace webrtc