| Index: webrtc/call/call.cc
|
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
|
| index f8611398c3a5dfe95c4e376b0caa802a911a05a2..5c46a48f14e30760e702369dd118f434997b0073 100644
|
| --- a/webrtc/call/call.cc
|
| +++ b/webrtc/call/call.cc
|
| @@ -336,8 +336,7 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
|
| TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
|
| RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
|
| AudioReceiveStream* receive_stream = new AudioReceiveStream(
|
| - congestion_controller_->GetRemoteBitrateEstimator(false), config,
|
| - config_.audio_state);
|
| + congestion_controller_.get(), config, config_.audio_state);
|
| {
|
| WriteLockScoped write_lock(*receive_crit_);
|
| RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
|
|
|