chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_video_header.h

/*
 *  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 MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_

#include <bitset>
#include <cstdint>

#include "absl/container/inlined_vector.h"
#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/rtp_headers.h"
#include "api/transport/rtp/dependency_descriptor.h"
#include "api/video/color_space.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame_metadata.h"
#include "api/video/video_frame_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
#include "modules/video_coding/codecs/h264/include/h264_globals.h"
#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"

namespace webrtc {
// Details passed in the rtp payload for legacy generic rtp packetizer.
// TODO(bugs.webrtc.org/9772): Deprecate in favor of passing generic video
// details in an rtp header extension.
struct RTPVideoHeaderLegacyGeneric { … };

RTPVideoTypeHeader;

struct RTPVideoHeader { … };

}  // namespace webrtc

#endif  // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_