123456789101112131415161718192021222324252627282930313233343536373839404142434445464748495051525354555657585960616263646566676869707172737475767778798081828384858687 |
- /*
- * 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'.
- 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;
- // 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;
- } 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_MEDIA_CONFIG_H_
|