a0b22af9e1
This reverts commit 04dd95fcac549fbdc330cee1de65074961db5934. Reason for revert: code has been updated Original change's description: > Temporary add 'RTPVideoHeaderH264::nalus_length'. > > This is a forward fix for https://webrtc-review.googlesource.com/c/src/+/354622 that breaks client code using nalus_length. > > No-Try: true > Change-Id: Ic0fc41696e408adefe4eb8792150a64b1eab49da > Bug: None > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354840 > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Commit-Queue: Jeremy Leconte <jleconte@google.com> > Owners-Override: Jeremy Leconte <jleconte@google.com> > Cr-Commit-Position: refs/heads/main@{#42493} Bug: None Change-Id: I1b65fe94ca07efdb8c7643e2ac46517050095018 No-Presubmit: true No-Tree-Checks: true No-Try: true Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354860 Commit-Queue: Jeremy Leconte <jleconte@google.com> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42496}
107 lines
3.8 KiB
C++
107 lines
3.8 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 <vector>
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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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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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std::vector<NaluInfo> nalus;
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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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lhs.nalus == rhs.nalus &&
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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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