fc9dcb6a00
This experiment is now wired up inside of chrome using field trial and this passthrough is now obsolete. Bug: chromium:794608 Change-Id: I1407e391d39c7e8696add9f656f059e7d8a27a08 Reviewed-on: https://webrtc-review.googlesource.com/82780 Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23625}
83 lines
3.3 KiB
C++
83 lines
3.3 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 MEDIA_BASE_MEDIACONFIG_H_
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#define MEDIA_BASE_MEDIACONFIG_H_
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namespace cricket {
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// Construction-time settings, passed on when creating
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// MediaChannels.
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struct MediaConfig {
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// Set DSCP value on packets. This flag comes from the
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// PeerConnection constraint 'googDscp'.
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bool enable_dscp = false;
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// Video-specific config.
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struct Video {
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// Enable WebRTC CPU Overuse Detection. This flag comes from the
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// PeerConnection constraint 'googCpuOveruseDetection'.
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bool enable_cpu_adaptation = true;
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// Enable WebRTC suspension of video. No video frames will be sent
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// when the bitrate is below the configured minimum bitrate. This
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// flag comes from the PeerConnection constraint
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// 'googSuspendBelowMinBitrate', and WebRtcVideoChannel copies it
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// to VideoSendStream::Config::suspend_below_min_bitrate.
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bool suspend_below_min_bitrate = false;
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// Set to true if the renderer has an algorithm of frame selection.
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// If the value is true, then WebRTC will hand over a frame as soon as
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// possible without delay, and rendering smoothness is completely the duty
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// of the renderer;
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// If the value is false, then WebRTC is responsible to delay frame release
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// in order to increase rendering smoothness.
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//
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// This flag comes from PeerConnection's RtcConfiguration, but is
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// currently only set by the command line flag
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// 'disable-rtc-smoothness-algorithm'.
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// WebRtcVideoChannel::AddRecvStream copies it to the created
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// WebRtcVideoReceiveStream, where it is returned by the
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// SmoothsRenderedFrames method. This method is used by the
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// VideoReceiveStream, where the value is passed on to the
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// IncomingVideoStream constructor.
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bool enable_prerenderer_smoothing = true;
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// Enables periodic bandwidth probing in application-limited region.
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bool periodic_alr_bandwidth_probing = false;
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// Enables the new method to estimate the cpu load from encoding, used for
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// cpu adaptation. This flag is intended to be controlled primarily by a
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// Chrome origin-trial.
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// TODO(bugs.webrtc.org/8504): If all goes well, the flag will be removed
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// together with the old method of estimation.
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bool experiment_cpu_load_estimator = false;
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} video;
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bool operator==(const MediaConfig& o) const {
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return enable_dscp == o.enable_dscp &&
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video.enable_cpu_adaptation == o.video.enable_cpu_adaptation &&
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video.suspend_below_min_bitrate ==
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o.video.suspend_below_min_bitrate &&
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video.enable_prerenderer_smoothing ==
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o.video.enable_prerenderer_smoothing &&
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video.periodic_alr_bandwidth_probing ==
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o.video.periodic_alr_bandwidth_probing &&
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video.experiment_cpu_load_estimator ==
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o.video.experiment_cpu_load_estimator;
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}
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bool operator!=(const MediaConfig& o) const { return !(*this == o); }
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};
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} // namespace cricket
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#endif // MEDIA_BASE_MEDIACONFIG_H_
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