Index: webrtc/call/call.cc |
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc |
index 2cc296dd36e9de80417f817d38e66afae6ccee18..3b6b8a0911dcd2070a603b05e390faa84c6efebb 100644 |
--- a/webrtc/call/call.cc |
+++ b/webrtc/call/call.cc |
@@ -380,7 +380,7 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream( |
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread()); |
AudioSendStream* send_stream = new AudioSendStream( |
config, config_.audio_state, &worker_queue_, congestion_controller_.get(), |
- bitrate_allocator_.get()); |
+ bitrate_allocator_.get(), event_log_.get()); |
{ |
WriteLockScoped write_lock(*send_crit_); |
RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) == |