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Unified Diff: webrtc/call/call.cc

Issue 2226823003: Set the event log in Channel from AudioSendStream (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 4 years, 4 months ago
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« webrtc/audio/audio_send_stream.cc ('K') | « webrtc/audio/audio_send_stream_unittest.cc ('k') | no next file » | no next file with comments »
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Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index f6354ade28e89e17037e9b03a3750058efc4b3fc..e5936fb10f712542d13e10470b8bf5f14155bf39 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -370,7 +370,7 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
AudioSendStream* send_stream = new AudioSendStream(
config, config_.audio_state, 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) ==
« webrtc/audio/audio_send_stream.cc ('K') | « webrtc/audio/audio_send_stream_unittest.cc ('k') | no next file » | no next file with comments »

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