Use separate rtp module lists for send and receive in PacketRouter.

webrtc/audio/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 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 99 matching lines @@
     } else if (extension.name == RtpExtension::kTransportSequenceNumber) {
       channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransportSequenceNumber, extension.id);
       RTC_DCHECK(registered);
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
   // Configure bandwidth estimation.
-  channel_proxy_->SetCongestionControlObjects(
-      nullptr, nullptr, congestion_controller->packet_router());
+  channel_proxy_->RegisterReceiverCongestionControlObjects(
+      congestion_controller->packet_router());
   if (config.combined_audio_video_bwe) {
     if (UseSendSideBwe(config)) {
       remote_bitrate_estimator_ =
           congestion_controller->GetRemoteBitrateEstimator(true);
     } else {
       remote_bitrate_estimator_ =
           congestion_controller->GetRemoteBitrateEstimator(false);
     }
     RTC_DCHECK(remote_bitrate_estimator_);
   }
 }
 
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
-  channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
+  channel_proxy_->ResetCongestionControlObjects();
   if (remote_bitrate_estimator_) {
     remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
   }
 }
 
 void AudioReceiveStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 void AudioReceiveStream::Stop() {
@@ skipping 100 matching lines @@
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc