Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index d2bc153d32765f13d89d1a239b5a17c5e7c15c88..cb3c029070458ea735e9c74cb31a6588a56da1e8 100644 |
--- a/webrtc/call/call.cc |
+++ b/webrtc/call/call.cc |
@@ -413,7 +413,7 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream( |
event_log_->LogAudioSendStreamConfig(config); |
AudioSendStream* send_stream = new AudioSendStream( |
config, config_.audio_state, &worker_queue_, congestion_controller_.get(), |
- bitrate_allocator_.get(), event_log_); |
+ bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats()); |
{ |
WriteLockScoped write_lock(*send_crit_); |
RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) == |