/* * 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; } 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_