| 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);
|
|
|