Break out pacer thread from CongestionController to increase testability.

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 103 matching lines @@
       return nullptr;
   }
 
   void UpdateSendHistograms() EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
   void UpdateReceiveHistograms();
 
   Clock* const clock_;
 
   const int num_cpu_cores_;
   const rtc::scoped_ptr<ProcessThread> module_process_thread_;
+  const rtc::scoped_ptr<ProcessThread> pacer_thread_;
   const rtc::scoped_ptr<CallStats> call_stats_;
   const rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_;
   Call::Config config_;
   rtc::ThreadChecker configuration_thread_checker_;
 
   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_
@@ skipping 42 matching lines @@
 Call* Call::Create(const Call::Config& config) {
   return new internal::Call(config);
 }
 
 namespace internal {
 
 Call::Call(const Call::Config& config)
     : clock_(Clock::GetRealTimeClock()),
       num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
       module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
+      pacer_thread_(ProcessThread::Create("PacerThread")),
       call_stats_(new CallStats(clock_)),
       bitrate_allocator_(new BitrateAllocator()),
       config_(config),
       network_enabled_(true),
       receive_crit_(RWLockWrapper::CreateRWLock()),
       send_crit_(RWLockWrapper::CreateRWLock()),
       received_video_bytes_(0),
       received_audio_bytes_(0),
       received_rtcp_bytes_(0),
       first_rtp_packet_received_ms_(-1),
@@ skipping 14 matching lines @@
   }
   if (config.audio_state.get()) {
     ScopedVoEInterface<VoECodec> voe_codec(voice_engine());
     event_log_ = voe_codec->GetEventLog();
   }
 
   Trace::CreateTrace();
   module_process_thread_->Start();
   module_process_thread_->RegisterModule(call_stats_.get());
   module_process_thread_->RegisterModule(congestion_controller_.get());
+  pacer_thread_->RegisterModule(congestion_controller_->pacer());
+  pacer_thread_->RegisterModule(
+      congestion_controller_->GetRemoteBitrateEstimator(true));
+  pacer_thread_->Start();

[pbos-webrtc, 2016/02/24 14:52:13]

Move to end of block so the controller settings are configured before starting.

   call_stats_->RegisterStatsObserver(congestion_controller_.get());
 
   congestion_controller_->SetBweBitrates(
       config_.bitrate_config.min_bitrate_bps,
       config_.bitrate_config.start_bitrate_bps,
       config_.bitrate_config.max_bitrate_bps);
 
   congestion_controller_->GetBitrateController()->SetEventLog(event_log_);
 }
 
 Call::~Call() {
   RTC_DCHECK(!remb_.InUse());
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   UpdateSendHistograms();
   UpdateReceiveHistograms();
   RTC_CHECK(audio_send_ssrcs_.empty());
   RTC_CHECK(video_send_ssrcs_.empty());
   RTC_CHECK(video_send_streams_.empty());
   RTC_CHECK(audio_receive_ssrcs_.empty());
   RTC_CHECK(video_receive_ssrcs_.empty());
   RTC_CHECK(video_receive_streams_.empty());
 
   call_stats_->DeregisterStatsObserver(congestion_controller_.get());
+  pacer_thread_->Stop();
+  pacer_thread_->DeRegisterModule(congestion_controller_->pacer());
+  pacer_thread_->DeRegisterModule(
+      congestion_controller_->GetRemoteBitrateEstimator(true));
   module_process_thread_->DeRegisterModule(congestion_controller_.get());
   module_process_thread_->DeRegisterModule(call_stats_.get());
   module_process_thread_->Stop();
   Trace::ReturnTrace();
 }
 
 void Call::UpdateSendHistograms() {
   if (num_bitrate_updates_ == 0 || first_packet_sent_ms_ == -1)
     return;
   int64_t elapsed_sec =
@@ skipping 481 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