mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00

The latter is also a member of the former. This cleanup is also a preparation for dropping WebRtcRTPHeader::frameType (or deleting WebRtcRTPHeader right away), now that it's a video-specific member. Tbr: kwiberg@webrtc.org # Comment change in modules/include/ Bug: None Change-Id: I5c1f3f981f0d750713fc9b9b145278150fe32b5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133024 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27740}
76 lines
2.6 KiB
C++
76 lines
2.6 KiB
C++
/*
|
|
* 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/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 {
|
|
using RTPVideoTypeHeader = absl::variant<absl::monostate,
|
|
RTPVideoHeaderVP8,
|
|
RTPVideoHeaderVP9,
|
|
RTPVideoHeaderH264>;
|
|
|
|
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<int64_t, 5> dependencies;
|
|
absl::InlinedVector<int, 5> higher_spatial_layers;
|
|
bool discardable = false;
|
|
};
|
|
|
|
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_
|