1 /* 2 * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MEDIA_BASE_MEDIA_CONFIG_H_ 12 #define MEDIA_BASE_MEDIA_CONFIG_H_ 13 14 namespace cricket { 15 16 // Construction-time settings, passed on when creating 17 // MediaChannels. 18 struct MediaConfig { 19 // Set DSCP value on packets. This flag comes from the 20 // PeerConnection constraint 'googDscp'. 21 bool enable_dscp = false; 22 23 // Video-specific config. 24 struct Video { 25 // Enable WebRTC CPU Overuse Detection. This flag comes from the 26 // PeerConnection constraint 'googCpuOveruseDetection'. 27 bool enable_cpu_adaptation = true; 28 29 // Enable WebRTC suspension of video. No video frames will be sent 30 // when the bitrate is below the configured minimum bitrate. This 31 // flag comes from the PeerConnection constraint 32 // 'googSuspendBelowMinBitrate', and WebRtcVideoChannel copies it 33 // to VideoSendStream::Config::suspend_below_min_bitrate. 34 bool suspend_below_min_bitrate = false; 35 36 // Enable buffering and playout timing smoothing of decoded frames. 37 // If set to true, then WebRTC will buffer and potentially drop decoded 38 // frames in order to keep a smooth rendering. 39 // If set to false, then WebRTC will hand over the frame from the decoder 40 // to the renderer as soon as possible, meaning that the renderer is 41 // responsible for smooth rendering. 42 // Note that even if this flag is set to false, dropping of frames can 43 // still happen pre-decode, e.g., dropping of higher temporal layers. 44 // This flag comes from the PeerConnection RtcConfiguration. 45 bool enable_prerenderer_smoothing = true; 46 47 // Enables periodic bandwidth probing in application-limited region. 48 bool periodic_alr_bandwidth_probing = false; 49 50 // Enables the new method to estimate the cpu load from encoding, used for 51 // cpu adaptation. This flag is intended to be controlled primarily by a 52 // Chrome origin-trial. 53 // TODO(bugs.webrtc.org/8504): If all goes well, the flag will be removed 54 // together with the old method of estimation. 55 bool experiment_cpu_load_estimator = false; 56 57 // Time interval between RTCP report for video 58 int rtcp_report_interval_ms = 1000; 59 } video; 60 61 // Audio-specific config. 62 struct Audio { 63 // Time interval between RTCP report for audio 64 int rtcp_report_interval_ms = 5000; 65 } audio; 66 67 bool operator==(const MediaConfig& o) const { 68 return enable_dscp == o.enable_dscp && 69 video.enable_cpu_adaptation == o.video.enable_cpu_adaptation && 70 video.suspend_below_min_bitrate == 71 o.video.suspend_below_min_bitrate && 72 video.enable_prerenderer_smoothing == 73 o.video.enable_prerenderer_smoothing && 74 video.periodic_alr_bandwidth_probing == 75 o.video.periodic_alr_bandwidth_probing && 76 video.experiment_cpu_load_estimator == 77 o.video.experiment_cpu_load_estimator && 78 video.rtcp_report_interval_ms == o.video.rtcp_report_interval_ms && 79 audio.rtcp_report_interval_ms == o.audio.rtcp_report_interval_ms; 80 } 81 82 bool operator!=(const MediaConfig& o) const { return !(*this == o); } 83 }; 84 85 } // namespace cricket 86 87 #endif // MEDIA_BASE_MEDIA_CONFIG_H_ 88