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af1038d97c
Bug: b/150859541 Change-Id: Iffb6ee84f49ffa64fdb0633248864d2dfd6e9ff3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234868 Commit-Queue: Paul Hallak <phallak@google.com> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35194}
94 lines
3.5 KiB
C++
94 lines
3.5 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
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#define MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
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#include <bitset>
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#include <cstdint>
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#include "absl/container/inlined_vector.h"
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#include "absl/types/optional.h"
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#include "absl/types/variant.h"
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#include "api/rtp_headers.h"
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#include "api/transport/rtp/dependency_descriptor.h"
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#include "api/video/color_space.h"
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#include "api/video/video_codec_type.h"
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#include "api/video/video_content_type.h"
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#include "api/video/video_frame_type.h"
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#include "api/video/video_rotation.h"
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#include "api/video/video_timing.h"
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#include "modules/video_coding/codecs/h264/include/h264_globals.h"
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#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
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#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
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namespace webrtc {
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// Details passed in the rtp payload for legacy generic rtp packetizer.
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// TODO(bugs.webrtc.org/9772): Deprecate in favor of passing generic video
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// details in an rtp header extension.
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struct RTPVideoHeaderLegacyGeneric {
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uint16_t picture_id;
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};
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using RTPVideoTypeHeader = absl::variant<absl::monostate,
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RTPVideoHeaderVP8,
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RTPVideoHeaderVP9,
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RTPVideoHeaderH264,
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RTPVideoHeaderLegacyGeneric>;
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struct RTPVideoHeader {
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struct GenericDescriptorInfo {
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GenericDescriptorInfo();
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GenericDescriptorInfo(const GenericDescriptorInfo& other);
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~GenericDescriptorInfo();
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int64_t frame_id = 0;
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int spatial_index = 0;
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int temporal_index = 0;
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absl::InlinedVector<DecodeTargetIndication, 10> decode_target_indications;
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absl::InlinedVector<int64_t, 5> dependencies;
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absl::InlinedVector<int, 4> chain_diffs;
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std::bitset<32> active_decode_targets = ~uint32_t{0};
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};
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RTPVideoHeader();
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RTPVideoHeader(const RTPVideoHeader& other);
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~RTPVideoHeader();
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absl::optional<GenericDescriptorInfo> generic;
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VideoFrameType frame_type = VideoFrameType::kEmptyFrame;
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uint16_t width = 0;
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uint16_t height = 0;
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VideoRotation rotation = VideoRotation::kVideoRotation_0;
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VideoContentType content_type = VideoContentType::UNSPECIFIED;
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bool is_first_packet_in_frame = false;
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bool is_last_packet_in_frame = false;
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bool is_last_frame_in_picture = true;
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uint8_t simulcastIdx = 0;
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VideoCodecType codec = VideoCodecType::kVideoCodecGeneric;
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VideoPlayoutDelay playout_delay;
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VideoSendTiming video_timing;
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absl::optional<ColorSpace> color_space;
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// This field is meant for media quality testing purpose only. When enabled it
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// carries the webrtc::VideoFrame id field from the sender to the receiver.
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absl::optional<uint16_t> video_frame_tracking_id;
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RTPVideoTypeHeader video_type_header;
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// When provided, is sent as is as an RTP header extension according to
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// http://www.webrtc.org/experiments/rtp-hdrext/abs-capture-time.
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// Otherwise, it is derived from other relevant information.
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absl::optional<AbsoluteCaptureTime> absolute_capture_time;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
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