| 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) ==
|
|
|