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