webrtc/modules/video_coding/codecs/h264/include/h264_globals.h
Tove Petersson 1fccaa4485 Reland "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
This reverts commit 8bf3210629.

Reason for revert: Initialized an uninitialized member in GofInfoVP9 (+ removed some redundant initialization of members already initialized by SetGofInfoVP9())

Original change's description:
> Revert "operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test"
>
> This reverts commit 437bf78ed9.
>
> Reason for revert: Breaks upstream project
>
> Original change's description:
> > operator== for VideoFrameMetadata + used in CloneSenderVideoFrame test
> >
> > Added equality and inequality operators for VideoFrameMetadata and used the equality operator to check that the cloned metadata property is equal to the original metadata in RtpSenderVideoFrameTransformerDelegateTest.CloneSenderVideoFrame.
> >
> > Also default-initialized VideoFrameMetadata::ssrc_ to 0.
> >
> > Bug: webrtc:14708
> > Change-Id: If1f5153069bc986061ff9f0a6abaa2a4a5a98dd1
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293560
> > Commit-Queue: Tove Petersson <tovep@google.com>
> > Reviewed-by: Tony Herre <herre@google.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39411}
>
> Bug: webrtc:14708
> Change-Id: Icbec1b65ed22b89766606cb9514dde6f4e9124be
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295500
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Auto-Submit: Andrey Logvin <landrey@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39413}

Bug: webrtc:14708
Change-Id: I843d29f7dd0da2c7f16968a7fc08dc02cd359fc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tove Petersson <tovep@google.com>
Cr-Commit-Position: refs/heads/main@{#39418}
2023-02-28 15:44:21 +00:00

109 lines
4 KiB
C++

/*
* Copyright (c) 2012 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.
*/
// This file contains codec dependent definitions that are needed in
// order to compile the WebRTC codebase, even if this codec is not used.
#ifndef MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
#define MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
#include <algorithm>
#include <string>
#include "modules/video_coding/codecs/interface/common_constants.h"
#include "rtc_base/checks.h"
namespace webrtc {
// The packetization types that we support: single, aggregated, and fragmented.
enum H264PacketizationTypes {
kH264SingleNalu, // This packet contains a single NAL unit.
kH264StapA, // This packet contains STAP-A (single time
// aggregation) packets. If this packet has an
// associated NAL unit type, it'll be for the
// first such aggregated packet.
kH264FuA, // This packet contains a FU-A (fragmentation
// unit) packet, meaning it is a part of a frame
// that was too large to fit into a single packet.
};
// Packetization modes are defined in RFC 6184 section 6
// Due to the structure containing this being initialized with zeroes
// in some places, and mode 1 being default, mode 1 needs to have the value
// zero. https://crbug.com/webrtc/6803
enum class H264PacketizationMode {
NonInterleaved = 0, // Mode 1 - STAP-A, FU-A is allowed
SingleNalUnit // Mode 0 - only single NALU allowed
};
// This function is declared inline because it is not clear which
// .cc file it should belong to.
// TODO(hta): Refactor. https://bugs.webrtc.org/6842
// TODO(jonasolsson): Use absl::string_view instead when that's available.
inline std::string ToString(H264PacketizationMode mode) {
if (mode == H264PacketizationMode::NonInterleaved) {
return "NonInterleaved";
} else if (mode == H264PacketizationMode::SingleNalUnit) {
return "SingleNalUnit";
}
RTC_DCHECK_NOTREACHED();
return "";
}
struct NaluInfo {
uint8_t type;
int sps_id;
int pps_id;
friend bool operator==(const NaluInfo& lhs, const NaluInfo& rhs) {
return lhs.type == rhs.type && lhs.sps_id == rhs.sps_id &&
lhs.pps_id == rhs.pps_id;
}
friend bool operator!=(const NaluInfo& lhs, const NaluInfo& rhs) {
return !(lhs == rhs);
}
};
const size_t kMaxNalusPerPacket = 10;
struct RTPVideoHeaderH264 {
// The NAL unit type. If this is a header for a
// fragmented packet, it's the NAL unit type of
// the original data. If this is the header for an
// aggregated packet, it's the NAL unit type of
// the first NAL unit in the packet.
uint8_t nalu_type;
// The packetization type of this buffer - single, aggregated or fragmented.
H264PacketizationTypes packetization_type;
NaluInfo nalus[kMaxNalusPerPacket];
size_t nalus_length;
// The packetization mode of this transport. Packetization mode
// determines which packetization types are allowed when packetizing.
H264PacketizationMode packetization_mode;
friend bool operator==(const RTPVideoHeaderH264& lhs,
const RTPVideoHeaderH264& rhs) {
return lhs.nalu_type == rhs.nalu_type &&
lhs.packetization_type == rhs.packetization_type &&
std::equal(lhs.nalus, lhs.nalus + lhs.nalus_length, rhs.nalus,
rhs.nalus + rhs.nalus_length) &&
lhs.packetization_mode == rhs.packetization_mode;
}
friend bool operator!=(const RTPVideoHeaderH264& lhs,
const RTPVideoHeaderH264& rhs) {
return !(lhs == rhs);
}
};
} // namespace webrtc
#endif // MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_