webrtc/modules/rtp_rtcp/source/rtp_video_header.h
philipel ef615ea7a3 Added is_last_packet_in_frame to match is_first_packet_in_frame.
Today we use |is_first_packet_in_frame| to know when a frame begins and the
|markerBit| to know when it ends, but the markerbit does not actually mark the
end of a frame, it marks the end of a picture.

Bug: webrtc:9361
Change-Id: Icc70e6075590cdc31e875a4eb9d489868adbb67c
Reviewed-on: https://webrtc-review.googlesource.com/100160
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24722}
2018-09-13 11:07:10 +00:00

67 lines
2.3 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 "absl/container/inlined_vector.h"
#include "absl/types/variant.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame_marking.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;
};
RTPVideoHeader();
RTPVideoHeader(const RTPVideoHeader& other);
~RTPVideoHeader();
absl::optional<GenericDescriptorInfo> generic;
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;
VideoSendTiming video_timing;
FrameMarking frame_marking;
RTPVideoTypeHeader video_type_header;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_