Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 109 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/include/h264_globals.h b/third_party/libwebrtc/modules/video_coding/codecs/h264/include/h264_globals.h
new file mode 100644
index 0000000000..6a1de382dc
--- /dev/null
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/include/h264_globals.h
@@ -0,0 +1,109 @@
+/*
+ *  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_