chromium/third_party/webrtc/modules/rtp_rtcp/source/video_rtp_depacketizer_h264.cc

/*
 *  Copyright (c) 2020 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.
 */

#include "modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h"

#include <cstddef>
#include <cstdint>
#include <utility>
#include <vector>

#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "common_video/h264/h264_common.h"
#include "common_video/h264/pps_parser.h"
#include "common_video/h264/sps_parser.h"
#include "common_video/h264/sps_vui_rewriter.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtp_format_h264.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/logging.h"

namespace webrtc {
namespace {

constexpr size_t kNalHeaderSize = …;
constexpr size_t kFuAHeaderSize = …;
constexpr size_t kLengthFieldSize = …;
constexpr size_t kStapAHeaderSize = …;

// TODO(pbos): Avoid parsing this here as well as inside the jitter buffer.
bool ParseStapAStartOffsets(const uint8_t* nalu_ptr,
                            size_t length_remaining,
                            std::vector<size_t>* offsets) { … }

absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ProcessStapAOrSingleNalu(
    rtc::CopyOnWriteBuffer rtp_payload) { … }

absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> ParseFuaNalu(
    rtc::CopyOnWriteBuffer rtp_payload) { … }

}  // namespace

absl::optional<VideoRtpDepacketizer::ParsedRtpPayload>
VideoRtpDepacketizerH264::Parse(rtc::CopyOnWriteBuffer rtp_payload) { … }

}  // namespace webrtc