| /* |
| * 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_ |