| /* |
| * 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 "absl/container/inlined_vector.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 "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 = 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; |
| 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< |
| absl::variant<FrameInstrumentationSyncData, FrameInstrumentationData>> |
| frame_instrumentation_data; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_ |