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

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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}
This commit is contained in:
philipel
2017-10-02 14:10:28 +02:00
committed by Commit Bot
parent 2c72fe8a2a
commit 707f278299
1 changed files with 3 additions and 1 deletions
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4
modules/video_coding/frame_buffer2.cc
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@ -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.