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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 <bitset>
#include <cstdint>
#include <optional>
#include <variant>
#include "absl/container/inlined_vector.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 "common_video/frame_instrumentation_data.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 {
uint16_t picture_id;
};
using RTPVideoTypeHeader = std::variant<std::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;
absl::InlinedVector<int, 4> chain_diffs;
std::bitset<32> active_decode_targets = ~uint32_t{0};
};
static RTPVideoHeader FromMetadata(const VideoFrameMetadata& metadata);
RTPVideoHeader();
RTPVideoHeader(const RTPVideoHeader& other);
~RTPVideoHeader();
// The subset of RTPVideoHeader that is exposed in the Insertable Streams API.
VideoFrameMetadata GetAsMetadata() const;
void SetFromMetadata(const VideoFrameMetadata& metadata);
std::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;
bool is_last_frame_in_picture = true;
uint8_t simulcastIdx = 0;
VideoCodecType codec = VideoCodecType::kVideoCodecGeneric;
std::optional<VideoPlayoutDelay> playout_delay;
VideoSendTiming video_timing;
std::optional<ColorSpace> color_space;
// This field is meant for media quality testing purpose only. When enabled it
// carries the webrtc::VideoFrame id field from the sender to the receiver.
std::optional<uint16_t> video_frame_tracking_id;
RTPVideoTypeHeader video_type_header;
// When provided, is sent as is as an RTP header extension according to
// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time.
// Otherwise, it is derived from other relevant information.
std::optional<AbsoluteCaptureTime> absolute_capture_time;
// Required for automatic corruption detection.
std::optional<
std::variant<FrameInstrumentationSyncData, FrameInstrumentationData>>
frame_instrumentation_data;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
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