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The frame time deltas are now capped based on the current noise. This has been tested in various conditions using both screen content and typical mobile video settings, to produce delays that are not overly high screen content, and simultaneously not negatively affect mobile calls on really bad network that may have high natural jitter. Bug: webrtc:9898 Change-Id: I51ad279af156aba1b5cc75ae203334a34bce9d48 Reviewed-on: https://webrtc-review.googlesource.com/c/107349 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25469}
31 lines
1.1 KiB
C++
31 lines
1.1 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef RTC_BASE_EXPERIMENTS_JITTER_UPPER_BOUND_EXPERIMENT_H_
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#define RTC_BASE_EXPERIMENTS_JITTER_UPPER_BOUND_EXPERIMENT_H_
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#include "absl/types/optional.h"
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namespace webrtc {
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class JitterUpperBoundExperiment {
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public:
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// Returns nullopt if experiment is not on, otherwise returns the configured
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// upper bound for frame delay delta used in jitter estimation, expressed as
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// number of standard deviations of the current deviation from the expected
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// delay.
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static absl::optional<double> GetUpperBoundSigmas();
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static const char kJitterUpperBoundExperimentName[];
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};
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} // namespace webrtc
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#endif // RTC_BASE_EXPERIMENTS_JITTER_UPPER_BOUND_EXPERIMENT_H_
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