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Cleaup WebRTC-AddRttToPlayoutDelay field trial

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Field trial is not used in any rollouts and should be removed.

R=mhoro@webrtc.org

Bug: webrtc:13264
Change-Id: Ib896dcdec81db7c3f4e68a8dda266d96dfdc6aed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234865
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35195}
This commit is contained in:
Evan Shrubsole
2021-10-13 11:19:48 +02:00
committed by WebRTC LUCI CQ
parent af1038d97c
commit 9146d765e2
3 changed files with 2 additions and 31 deletions
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4
modules/video_coding/frame_buffer2.cc
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@ -65,8 +65,6 @@ FrameBuffer::FrameBuffer(Clock* clock,
protection_mode_(kProtectionNack),
stats_callback_(stats_callback),
last_log_non_decoded_ms_(-kLogNonDecodedIntervalMs),
add_rtt_to_playout_delay_(
webrtc::field_trial::IsEnabled("WebRTC-AddRttToPlayoutDelay")),
rtt_mult_settings_(RttMultExperiment::GetRttMultValue()),
zero_playout_delay_max_decode_queue_size_(
"max_decode_queue_size",
@ -309,7 +307,7 @@ EncodedFrame* FrameBuffer::GetNextFrame() {
jitter_estimator_.GetJitterEstimate(rtt_mult, rtt_mult_add_cap_ms));
timing_->UpdateCurrentDelay(render_time_ms, now_ms);
} else {
if (RttMultExperiment::RttMultEnabled() || add_rtt_to_playout_delay_)
if (RttMultExperiment::RttMultEnabled())
jitter_estimator_.FrameNacked();
}