| /* | |
| * 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/corruption_detection/frame_instrumentation_data.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 { | |
| 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 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<FrameInstrumentationData> frame_instrumentation_data; | |
| }; | |
| } // namespace webrtc | |
| #endif // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_ |