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) == |