commit 707f278299ab9226107e8df911b98e6f6a7cd590
author philipel <philipel@webrtc.org> Mon Oct 02 14:10:28 2017 +0200
committer Commit Bot <commit-bot@chromium.org> Mon Oct 02 13:28:30 2017 +0000
tree 432f3d72a68c06ff6a0166dd0ca4b2217d8cc90a
parent 2c72fe8a2a0e540a66028254b26e2409f9360cd8

Add RTT to playout delay behind WebRTC-AddRttToPlayoutDelay field trial.

Bug: webrtc:8010
Change-Id: I78d2b5053521186b9bcc27eba264325b6f934a78
Reviewed-on: https://webrtc-review.googlesource.com/4666
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20079}

modules/video_coding/frame_buffer2.cc

diff --git a/modules/video_coding/frame_buffer2.cc b/modules/video_coding/frame_buffer2.cc
index b4bb002..e289a03 100644
--- a/modules/video_coding/frame_buffer2.cc
+++ b/modules/video_coding/frame_buffer2.cc
@@ -21,6 +21,7 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/trace_event.h"
 #include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/field_trial.h"
 #include "system_wrappers/include/metrics.h"
 
 namespace webrtc {
@@ -148,7 +149,8 @@
         timing_->SetJitterDelay(jitter_estimator_->GetJitterEstimate(rtt_mult));
         timing_->UpdateCurrentDelay(frame->RenderTime(), now_ms);
       } else {
-        jitter_estimator_->FrameNacked();
+        if (webrtc::field_trial::IsEnabled("WebRTC-AddRttToPlayoutDelay"))
+          jitter_estimator_->FrameNacked();
       }
 
       // Gracefully handle bad RTP timestamps and render time issues.