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Unified Diff: webrtc/call/call.cc

Issue 2402333002: Add RtcpRttStats to AudioStream (Closed)
Patch Set: Created 4 years, 2 months ago
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Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 9515ac10f14efe7c87ae2708a42cf49d55ba5452..a68a60e120c709f25e1bbd5f2c62d0794b2e5ce6 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -373,7 +373,7 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
event_log_->LogAudioSendStreamConfig(config);
AudioSendStream* send_stream = new AudioSendStream(
config, config_.audio_state, &worker_queue_, congestion_controller_.get(),
- bitrate_allocator_.get(), event_log_);
+ bitrate_allocator_.get(), event_log_, call_stats_->rtcp_rtt_stats());
{
WriteLockScoped write_lock(*send_crit_);
RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
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