chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc

/*
 *  Copyright (c) 2014 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/rtp_format_h264.h"

#include <string.h>

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

#include "absl/algorithm/container.h"
#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_packet_to_send.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"

namespace webrtc {
namespace {

static const size_t kNalHeaderSize = …;
static const size_t kFuAHeaderSize = …;
static const size_t kLengthFieldSize = …;

}  // namespace

RtpPacketizerH264::RtpPacketizerH264(rtc::ArrayView<const uint8_t> payload,
                                     PayloadSizeLimits limits,
                                     H264PacketizationMode packetization_mode)
    : … { … }

RtpPacketizerH264::~RtpPacketizerH264() = default;

size_t RtpPacketizerH264::NumPackets() const { … }

bool RtpPacketizerH264::GeneratePackets(
    H264PacketizationMode packetization_mode) { … }

bool RtpPacketizerH264::PacketizeFuA(size_t fragment_index) { … }

size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) { … }

bool RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) { … }

bool RtpPacketizerH264::NextPacket(RtpPacketToSend* rtp_packet) { … }

void RtpPacketizerH264::NextAggregatePacket(RtpPacketToSend* rtp_packet) { … }

void RtpPacketizerH264::NextFragmentPacket(RtpPacketToSend* rtp_packet) { … }

}  // namespace webrtc