Simplify CongestionController.

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 181 matching lines @@
       received_video_bytes_(0),
       received_audio_bytes_(0),
       received_rtcp_bytes_(0),
       first_rtp_packet_received_ms_(-1),
       last_rtp_packet_received_ms_(-1),
       first_packet_sent_ms_(-1),
       estimated_send_bitrate_sum_kbits_(0),
       pacer_bitrate_sum_kbits_(0),
       num_bitrate_updates_(0),
       remb_(clock_),
-      congestion_controller_(
-          new CongestionController(clock_,
-                                   module_process_thread_.get(),
-                                   call_stats_.get(),
-                                   this,
-                                   &remb_)) {
+      congestion_controller_(new CongestionController(clock_, this, &remb_)) {
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
   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());
+  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());
+  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 =
       (clock_->TimeInMilliseconds() - first_packet_sent_ms_) / 1000;
@@ skipping 488 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