Index: webrtc/audio/audio_send_stream.cc |
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc |
index b031762b4cecd173e21766e4ab3f91c070a37fd9..179edf2d3d55393b302ee5008a4296a61d186777 100644 |
--- a/webrtc/audio/audio_send_stream.cc |
+++ b/webrtc/audio/audio_send_stream.cc |
@@ -45,6 +45,7 @@ AudioSendStream::AudioSendStream( |
const webrtc::AudioSendStream::Config& config, |
const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
rtc::TaskQueue* worker_queue, |
+ PacketRouter* packet_router, |
CongestionController* congestion_controller, |
BitrateAllocator* bitrate_allocator, |
RtcEventLog* event_log) |
@@ -62,8 +63,7 @@ AudioSendStream::AudioSendStream( |
channel_proxy_->SetRtcEventLog(event_log); |
channel_proxy_->RegisterSenderCongestionControlObjects( |
congestion_controller->pacer(), |
- congestion_controller->GetTransportFeedbackObserver(), |
- congestion_controller->packet_router()); |
+ congestion_controller->GetTransportFeedbackObserver(), packet_router); |
channel_proxy_->SetRTCPStatus(true); |
channel_proxy_->SetLocalSSRC(config.rtp.ssrc); |
channel_proxy_->SetRTCP_CNAME(config.rtp.c_name); |