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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_MEDIA_CONFIG_H_
#define MEDIA_BASE_MEDIA_CONFIG_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'.
// TODO(https://crbug.com/1315574): Remove the ability to set it in Chromium
// and delete this flag.
bool enable_dscp = true;
// Video-specific config.
struct Video {
// Enable WebRTC CPU Overuse Detection. This flag comes from the
// PeerConnection constraint 'googCpuOveruseDetection'.
// TODO(https://crbug.com/1315569): Remove the ability to set it in Chromium
// and delete this flag.
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.
// TODO(https://crbug.com/1315564): Remove the ability to set it in Chromium
// and delete this flag.
bool suspend_below_min_bitrate = false;
// Enable buffering and playout timing smoothing of decoded frames.
// If set to true, then WebRTC will buffer and potentially drop decoded
// frames in order to keep a smooth rendering.
// If set to false, then WebRTC will hand over the frame from the decoder
// to the renderer as soon as possible, meaning that the renderer is
// responsible for smooth rendering.
// Note that even if this flag is set to false, dropping of frames can
// still happen pre-decode, e.g., dropping of higher temporal layers.
// This flag comes from the PeerConnection RtcConfiguration.
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;
// Enables send packet batching from the egress RTP sender.
bool enable_send_packet_batching = false;
} 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 &&
video.enable_send_packet_batching ==
o.video.enable_send_packet_batching &&
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_MEDIA_CONFIG_H_
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