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/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MEDIA_BASE_MEDIACONFIG_H_
#define MEDIA_BASE_MEDIACONFIG_H_
namespace cricket {
// Construction-time settings, passed on when creating
// MediaChannels.
struct MediaConfig {
// Set DSCP value on packets. This flag comes from the
// PeerConnection constraint 'googDscp'.
bool enable_dscp = false;
// Video-specific config.
struct Video {
// Enable WebRTC CPU Overuse Detection. This flag comes from the
// PeerConnection constraint 'googCpuOveruseDetection'.
bool enable_cpu_adaptation = true;
// Enable WebRTC suspension of video. No video frames will be sent
// when the bitrate is below the configured minimum bitrate. This
// flag comes from the PeerConnection constraint
// 'googSuspendBelowMinBitrate', and WebRtcVideoChannel copies it
// to VideoSendStream::Config::suspend_below_min_bitrate.
bool suspend_below_min_bitrate = false;
// Set to true if the renderer has an algorithm of frame selection.
// If the value is true, then WebRTC will hand over a frame as soon as
// possible without delay, and rendering smoothness is completely the duty
// of the renderer;
// If the value is false, then WebRTC is responsible to delay frame release
// in order to increase rendering smoothness.
//
// This flag comes from PeerConnection's RtcConfiguration, but is
// currently only set by the command line flag
// 'disable-rtc-smoothness-algorithm'.
// WebRtcVideoChannel::AddRecvStream copies it to the created
// WebRtcVideoReceiveStream, where it is returned by the
// SmoothsRenderedFrames method. This method is used by the
// VideoReceiveStream, where the value is passed on to the
// IncomingVideoStream constructor.
bool enable_prerenderer_smoothing = true;
// Enables periodic bandwidth probing in application-limited region.
bool periodic_alr_bandwidth_probing = false;
// Enables the new method to estimate the cpu load from encoding, used for
// cpu adaptation. This flag is intended to be controlled primarily by a
// Chrome origin-trial.
// TODO(bugs.webrtc.org/8504): If all goes well, the flag will be removed
// together with the old method of estimation.
bool experiment_cpu_load_estimator = false;
// Time interval between RTCP report for video
int rtcp_report_interval_ms = 1000;
} video;
// Audio-specific config.
struct Audio {
// Time interval between RTCP report for audio
int rtcp_report_interval_ms = 5000;
} audio;
bool operator==(const MediaConfig& o) const {
return enable_dscp == o.enable_dscp &&
video.enable_cpu_adaptation == o.video.enable_cpu_adaptation &&
video.suspend_below_min_bitrate ==
o.video.suspend_below_min_bitrate &&
video.enable_prerenderer_smoothing ==
o.video.enable_prerenderer_smoothing &&
video.periodic_alr_bandwidth_probing ==
o.video.periodic_alr_bandwidth_probing &&
video.experiment_cpu_load_estimator ==
o.video.experiment_cpu_load_estimator &&
video.rtcp_report_interval_ms == o.video.rtcp_report_interval_ms &&
audio.rtcp_report_interval_ms == o.audio.rtcp_report_interval_ms;
}
bool operator!=(const MediaConfig& o) const { return !(*this == o); }
};
} // namespace cricket
#endif // MEDIA_BASE_MEDIACONFIG_H_