chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.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_video_generic.h"

#include <string.h>

#include "absl/types/optional.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"

namespace webrtc {

static const size_t kGenericHeaderLength = …;
static const size_t kExtendedHeaderLength = …;

RtpPacketizerGeneric::RtpPacketizerGeneric(
    rtc::ArrayView<const uint8_t> payload,
    PayloadSizeLimits limits,
    const RTPVideoHeader& rtp_video_header)
    : … { … }

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

RtpPacketizerGeneric::~RtpPacketizerGeneric() = default;

size_t RtpPacketizerGeneric::NumPackets() const { … }

bool RtpPacketizerGeneric::NextPacket(RtpPacketToSend* packet) { … }

void RtpPacketizerGeneric::BuildHeader(const RTPVideoHeader& rtp_video_header) { … }
}  // namespace webrtc