Add RtcpRttStats to AudioStream

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index ee84b96aaa3700a8e7d937e0cf3d4ec1dfd0990d..377002f929023d645eaccc0fd2fb0bf09a837db5 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -64,7 +64,8 @@ AudioSendStream::AudioSendStream(
     rtc::TaskQueue* worker_queue,
     CongestionController* congestion_controller,
     BitrateAllocator* bitrate_allocator,
-    RtcEventLog* event_log)
+    RtcEventLog* event_log,
+    RtcpRttStats* rtcp_rtt_stats)
     : worker_queue_(worker_queue),
       config_(config),
       audio_state_(audio_state),
@@ -77,6 +78,7 @@ AudioSendStream::AudioSendStream(
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetRtcEventLog(event_log);
+  channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
   channel_proxy_->RegisterSenderCongestionControlObjects(
       congestion_controller->pacer(),
       congestion_controller->GetTransportFeedbackObserver(),