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This reverts commit8bf3210629
. 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 commit437bf78ed9
. > > 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}
109 lines
4 KiB
C++
109 lines
4 KiB
C++
/*
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* Copyright (c) 2012 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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// This file contains codec dependent definitions that are needed in
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// order to compile the WebRTC codebase, even if this codec is not used.
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#ifndef MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
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#define MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
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#include <algorithm>
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#include <string>
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#include "modules/video_coding/codecs/interface/common_constants.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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// The packetization types that we support: single, aggregated, and fragmented.
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enum H264PacketizationTypes {
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kH264SingleNalu, // This packet contains a single NAL unit.
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kH264StapA, // This packet contains STAP-A (single time
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// aggregation) packets. If this packet has an
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// associated NAL unit type, it'll be for the
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// first such aggregated packet.
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kH264FuA, // This packet contains a FU-A (fragmentation
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// unit) packet, meaning it is a part of a frame
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// that was too large to fit into a single packet.
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};
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// Packetization modes are defined in RFC 6184 section 6
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// Due to the structure containing this being initialized with zeroes
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// in some places, and mode 1 being default, mode 1 needs to have the value
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// zero. https://crbug.com/webrtc/6803
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enum class H264PacketizationMode {
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NonInterleaved = 0, // Mode 1 - STAP-A, FU-A is allowed
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SingleNalUnit // Mode 0 - only single NALU allowed
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};
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// This function is declared inline because it is not clear which
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// .cc file it should belong to.
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// TODO(hta): Refactor. https://bugs.webrtc.org/6842
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// TODO(jonasolsson): Use absl::string_view instead when that's available.
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inline std::string ToString(H264PacketizationMode mode) {
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if (mode == H264PacketizationMode::NonInterleaved) {
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return "NonInterleaved";
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} else if (mode == H264PacketizationMode::SingleNalUnit) {
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return "SingleNalUnit";
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}
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RTC_DCHECK_NOTREACHED();
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return "";
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}
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struct NaluInfo {
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uint8_t type;
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int sps_id;
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int pps_id;
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friend bool operator==(const NaluInfo& lhs, const NaluInfo& rhs) {
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return lhs.type == rhs.type && lhs.sps_id == rhs.sps_id &&
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lhs.pps_id == rhs.pps_id;
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}
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friend bool operator!=(const NaluInfo& lhs, const NaluInfo& rhs) {
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return !(lhs == rhs);
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}
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};
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const size_t kMaxNalusPerPacket = 10;
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struct RTPVideoHeaderH264 {
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// The NAL unit type. If this is a header for a
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// fragmented packet, it's the NAL unit type of
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// the original data. If this is the header for an
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// aggregated packet, it's the NAL unit type of
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// the first NAL unit in the packet.
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uint8_t nalu_type;
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// The packetization type of this buffer - single, aggregated or fragmented.
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H264PacketizationTypes packetization_type;
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NaluInfo nalus[kMaxNalusPerPacket];
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size_t nalus_length;
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// The packetization mode of this transport. Packetization mode
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// determines which packetization types are allowed when packetizing.
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H264PacketizationMode packetization_mode;
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friend bool operator==(const RTPVideoHeaderH264& lhs,
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const RTPVideoHeaderH264& rhs) {
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return lhs.nalu_type == rhs.nalu_type &&
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lhs.packetization_type == rhs.packetization_type &&
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std::equal(lhs.nalus, lhs.nalus + lhs.nalus_length, rhs.nalus,
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rhs.nalus + rhs.nalus_length) &&
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lhs.packetization_mode == rhs.packetization_mode;
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}
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friend bool operator!=(const RTPVideoHeaderH264& lhs,
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const RTPVideoHeaderH264& rhs) {
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return !(lhs == rhs);
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}
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};
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} // namespace webrtc
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#endif // MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
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