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/*
* 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 <cstdint>
#include "absl/container/inlined_vector.h"
#include "absl/types/optional.h"
#include "absl/types/variant.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_marking.h"
#include "api/video/video_frame_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
#include "common_types.h" // NOLINT(build/include)
#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 {
uint16_t picture_id;
};
using RTPVideoTypeHeader = absl::variant<absl::monostate,
RTPVideoHeaderVP8,
RTPVideoHeaderVP9,
RTPVideoHeaderH264,
RTPVideoHeaderLegacyGeneric>;
struct RTPVideoHeader {
struct GenericDescriptorInfo {
GenericDescriptorInfo();
GenericDescriptorInfo(const GenericDescriptorInfo& other);
~GenericDescriptorInfo();
int64_t frame_id = 0;
int spatial_index = 0;
int temporal_index = 0;
absl::InlinedVector<DecodeTargetIndication, 10> decode_target_indications;
absl::InlinedVector<int64_t, 5> dependencies;
};
RTPVideoHeader();
RTPVideoHeader(const RTPVideoHeader& other);
~RTPVideoHeader();
absl::optional<GenericDescriptorInfo> generic;
VideoFrameType frame_type = VideoFrameType::kEmptyFrame;
uint16_t width = 0;
uint16_t height = 0;
VideoRotation rotation = VideoRotation::kVideoRotation_0;
VideoContentType content_type = VideoContentType::UNSPECIFIED;
bool is_first_packet_in_frame = false;
bool is_last_packet_in_frame = false;
uint8_t simulcastIdx = 0;
VideoCodecType codec = VideoCodecType::kVideoCodecGeneric;
PlayoutDelay playout_delay = {-1, -1};
VideoSendTiming video_timing;
FrameMarking frame_marking = {false, false, false, false, false, 0xFF, 0, 0};
absl::optional<ColorSpace> color_space;
RTPVideoTypeHeader video_type_header;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_