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

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 35a65521ddd79a6644e72d608b08f262cd0d4cec..62121c63a5f06a92499086f134e895507678f1da 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -68,7 +68,7 @@ AudioSendStream::AudioSendStream(
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
-  channel_proxy_->SetCongestionControlObjects(
+  channel_proxy_->RegisterSenderCongestionControlObjects(
       congestion_controller->pacer(),
       congestion_controller->GetTransportFeedbackObserver(),
       congestion_controller->packet_router());
@@ -92,7 +92,7 @@ AudioSendStream::AudioSendStream(
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
-  channel_proxy_->SetCongestionControlObjects(nullptr, nullptr, nullptr);
+  channel_proxy_->ResetCongestionControlObjects();
 }
 
 void AudioSendStream::Start() {