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 // TODO(https://crbug.com/1315574): Remove the ability to set it in Chromium 22 // and delete this flag. 23 bool enable_dscp = true; 24 25 // Video-specific config. 26 struct Video { 27 // Enable WebRTC CPU Overuse Detection. This flag comes from the 28 // PeerConnection constraint 'googCpuOveruseDetection'. 29 // TODO(https://crbug.com/1315569): Remove the ability to set it in Chromium 30 // and delete this flag. 31 bool enable_cpu_adaptation = true; 32 33 // Enable WebRTC suspension of video. No video frames will be sent 34 // when the bitrate is below the configured minimum bitrate. This 35 // flag comes from the PeerConnection constraint 36 // 'googSuspendBelowMinBitrate', and WebRtcVideoChannel copies it 37 // to VideoSendStream::Config::suspend_below_min_bitrate. 38 // TODO(https://crbug.com/1315564): Remove the ability to set it in Chromium 39 // and delete this flag. 40 bool suspend_below_min_bitrate = false; 41 42 // Enable buffering and playout timing smoothing of decoded frames. 43 // If set to true, then WebRTC will buffer and potentially drop decoded 44 // frames in order to keep a smooth rendering. 45 // If set to false, then WebRTC will hand over the frame from the decoder 46 // to the renderer as soon as possible, meaning that the renderer is 47 // responsible for smooth rendering. 48 // Note that even if this flag is set to false, dropping of frames can 49 // still happen pre-decode, e.g., dropping of higher temporal layers. 50 // This flag comes from the PeerConnection RtcConfiguration. 51 bool enable_prerenderer_smoothing = true; 52 53 // Enables periodic bandwidth probing in application-limited region. 54 bool periodic_alr_bandwidth_probing = false; 55 56 // Enables the new method to estimate the cpu load from encoding, used for 57 // cpu adaptation. This flag is intended to be controlled primarily by a 58 // Chrome origin-trial. 59 // TODO(bugs.webrtc.org/8504): If all goes well, the flag will be removed 60 // together with the old method of estimation. 61 bool experiment_cpu_load_estimator = false; 62 63 // Time interval between RTCP report for video 64 int rtcp_report_interval_ms = 1000; 65 } video; 66 67 // Audio-specific config. 68 struct Audio { 69 // Time interval between RTCP report for audio 70 int rtcp_report_interval_ms = 5000; 71 } audio; 72 73 bool operator==(const MediaConfig& o) const { 74 return enable_dscp == o.enable_dscp && 75 video.enable_cpu_adaptation == o.video.enable_cpu_adaptation && 76 video.suspend_below_min_bitrate == 77 o.video.suspend_below_min_bitrate && 78 video.enable_prerenderer_smoothing == 79 o.video.enable_prerenderer_smoothing && 80 video.periodic_alr_bandwidth_probing == 81 o.video.periodic_alr_bandwidth_probing && 82 video.experiment_cpu_load_estimator == 83 o.video.experiment_cpu_load_estimator && 84 video.rtcp_report_interval_ms == o.video.rtcp_report_interval_ms && 85 audio.rtcp_report_interval_ms == o.audio.rtcp_report_interval_ms; 86 } 87 88 bool operator!=(const MediaConfig& o) const { return !(*this == o); } 89 }; 90 91 } // namespace cricket 92 93 #endif // MEDIA_BASE_MEDIA_CONFIG_H_ 94