From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/media/base/media_config.h | 93 +++++++++++++++++++++++++ 1 file changed, 93 insertions(+) create mode 100644 third_party/libwebrtc/media/base/media_config.h (limited to 'third_party/libwebrtc/media/base/media_config.h') diff --git a/third_party/libwebrtc/media/base/media_config.h b/third_party/libwebrtc/media/base/media_config.h new file mode 100644 index 0000000000..b383c9aa3d --- /dev/null +++ b/third_party/libwebrtc/media/base/media_config.h @@ -0,0 +1,93 @@ +/* + * 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_ -- cgit v1.2.3