109 lines
4.7 KiB
C++
109 lines
4.7 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_MEDIA_CONFIG_H_
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#define MEDIA_BASE_MEDIA_CONFIG_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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// TODO(https://crbug.com/1315574): Remove the ability to set it in Chromium
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// and delete this flag.
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bool enable_dscp = true;
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// If true, RTCStats timestamps are sourced from the monotonically increasing
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// environment Clock, where the epoch is unspecified (i.e. up to the Clock
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// implementation). If false, RTCStats timestamps are either sourced from
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// system clock via rtc::TimeUTCMicros() which is relative to 1970 but not
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// necessarily monotonically increasing, or from a monotonic clock that is
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// set to rtc::TimeUTCMicros() at first call, and then procceeds to increase
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// monotonically.
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// TODO: bugs.webrtc.org/370535296 - Change default value to true and delete
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// this flag once downstream projects have migrated.
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bool stats_timestamp_with_environment_clock = 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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// TODO(https://crbug.com/1315569): Remove the ability to set it in Chromium
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// and delete this flag.
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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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// TODO(https://crbug.com/1315564): Remove the ability to set it in Chromium
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// and delete this flag.
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bool suspend_below_min_bitrate = false;
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// Enable buffering and playout timing smoothing of decoded frames.
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// If set to true, then WebRTC will buffer and potentially drop decoded
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// frames in order to keep a smooth rendering.
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// If set to false, then WebRTC will hand over the frame from the decoder
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// to the renderer as soon as possible, meaning that the renderer is
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// responsible for smooth rendering.
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// Note that even if this flag is set to false, dropping of frames can
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// still happen pre-decode, e.g., dropping of higher temporal layers.
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// This flag comes from the PeerConnection RtcConfiguration.
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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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// Time interval between RTCP report for video
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int rtcp_report_interval_ms = 1000;
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// Enables send packet batching from the egress RTP sender.
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bool enable_send_packet_batching = false;
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} video;
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// Audio-specific config.
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struct Audio {
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// Time interval between RTCP report for audio
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int rtcp_report_interval_ms = 5000;
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} audio;
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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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video.rtcp_report_interval_ms == o.video.rtcp_report_interval_ms &&
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video.enable_send_packet_batching ==
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o.video.enable_send_packet_batching &&
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audio.rtcp_report_interval_ms == o.audio.rtcp_report_interval_ms;
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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_MEDIA_CONFIG_H_
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