| 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) ==
|
|
|