Control rtt_mult addition cap via experiment.
Bug: webrtc:10717 Change-Id: I68f7d8216e1a1611e692dd82ba96890cad98c7de Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140284 Commit-Queue: Michael Horowitz <mhoro@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28191}
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@ -62,7 +62,8 @@ FrameBuffer::FrameBuffer(Clock* clock,
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stats_callback_(stats_callback),
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last_log_non_decoded_ms_(-kLogNonDecodedIntervalMs),
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add_rtt_to_playout_delay_(
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webrtc::field_trial::IsEnabled("WebRTC-AddRttToPlayoutDelay")) {}
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webrtc::field_trial::IsEnabled("WebRTC-AddRttToPlayoutDelay")),
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rtt_mult_settings_(RttMultExperiment::GetRttMultValue()) {}
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FrameBuffer::~FrameBuffer() {}
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@ -299,11 +300,10 @@ EncodedFrame* FrameBuffer::GetNextFrame() {
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}
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float rtt_mult = protection_mode_ == kProtectionNackFEC ? 0.0 : 1.0;
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absl::optional<double> rtt_mult_add_cap_ms = absl::nullopt;
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if (RttMultExperiment::RttMultEnabled()) {
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rtt_mult = RttMultExperiment::GetRttMultValue();
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// TODO(mhoro): add RttMultExperiment::GetJitterEstCapValue();
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rtt_mult_add_cap_ms = 200.0;
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absl::optional<float> rtt_mult_add_cap_ms = absl::nullopt;
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if (rtt_mult_settings_.has_value()) {
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rtt_mult = rtt_mult_settings_->rtt_mult_setting;
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rtt_mult_add_cap_ms = rtt_mult_settings_->rtt_mult_add_cap_ms;
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}
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timing_->SetJitterDelay(
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jitter_estimator_.GetJitterEstimate(rtt_mult, rtt_mult_add_cap_ms));
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