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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/common_video/h264/h264_bitstream_parser.h | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/common_video/h264/h264_bitstream_parser.h')
-rw-r--r-- | third_party/libwebrtc/common_video/h264/h264_bitstream_parser.h | 58 |
1 files changed, 58 insertions, 0 deletions
diff --git a/third_party/libwebrtc/common_video/h264/h264_bitstream_parser.h b/third_party/libwebrtc/common_video/h264/h264_bitstream_parser.h new file mode 100644 index 0000000000..05427825ac --- /dev/null +++ b/third_party/libwebrtc/common_video/h264/h264_bitstream_parser.h @@ -0,0 +1,58 @@ +/* + * Copyright (c) 2015 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 COMMON_VIDEO_H264_H264_BITSTREAM_PARSER_H_ +#define COMMON_VIDEO_H264_H264_BITSTREAM_PARSER_H_ +#include <stddef.h> +#include <stdint.h> + +#include "absl/types/optional.h" +#include "api/video_codecs/bitstream_parser.h" +#include "common_video/h264/pps_parser.h" +#include "common_video/h264/sps_parser.h" + +namespace webrtc { + +// Stateful H264 bitstream parser (due to SPS/PPS). Used to parse out QP values +// from the bitstream. +// TODO(pbos): Unify with RTP SPS parsing and only use one H264 parser. +// TODO(pbos): If/when this gets used on the receiver side CHECKs must be +// removed and gracefully abort as we have no control over receive-side +// bitstreams. +class H264BitstreamParser : public BitstreamParser { + public: + H264BitstreamParser(); + ~H264BitstreamParser() override; + + void ParseBitstream(rtc::ArrayView<const uint8_t> bitstream) override; + absl::optional<int> GetLastSliceQp() const override; + + protected: + enum Result { + kOk, + kInvalidStream, + kUnsupportedStream, + }; + void ParseSlice(const uint8_t* slice, size_t length); + Result ParseNonParameterSetNalu(const uint8_t* source, + size_t source_length, + uint8_t nalu_type); + + // SPS/PPS state, updated when parsing new SPS/PPS, used to parse slices. + absl::optional<SpsParser::SpsState> sps_; + absl::optional<PpsParser::PpsState> pps_; + + // Last parsed slice QP. + absl::optional<int32_t> last_slice_qp_delta_; +}; + +} // namespace webrtc + +#endif // COMMON_VIDEO_H264_H264_BITSTREAM_PARSER_H_ |