Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index e76df360bcbf747d713ed21e3041bb64b2e9296a..3d940b619a073b29221e341977f82a371b762ce8 100644 |
--- a/webrtc/call/call.cc |
+++ b/webrtc/call/call.cc |
@@ -426,7 +426,8 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream( |
event_log_->LogAudioSendStreamConfig(config); |
AudioSendStream* send_stream = new AudioSendStream( |
config, config_.audio_state, &worker_queue_, &packet_router_, |
- congestion_controller_.get(), bitrate_allocator_.get(), event_log_); |
+ congestion_controller_.get(), bitrate_allocator_.get(), event_log_, |
+ call_stats_->rtcp_rtt_stats()); |
{ |
WriteLockScoped write_lock(*send_crit_); |
RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) == |