From 707f278299ab9226107e8df911b98e6f6a7cd590 Mon Sep 17 00:00:00 2001 From: philipel Date: Mon, 2 Oct 2017 14:10:28 +0200 Subject: [PATCH] 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 Reviewed-by: Stefan Holmer Cr-Commit-Position: refs/heads/master@{#20079} --- modules/video_coding/frame_buffer2.cc | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/modules/video_coding/frame_buffer2.cc b/modules/video_coding/frame_buffer2.cc index b4bb002802..e289a03bb2 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 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame( 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.