chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_packet.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/rtcp_packet.h"

#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/checks.h"

namespace webrtc {
namespace rtcp {
constexpr size_t RtcpPacket::kHeaderLength;

rtc::Buffer RtcpPacket::Build() const { … }

bool RtcpPacket::Build(size_t max_length, PacketReadyCallback callback) const { … }

bool RtcpPacket::OnBufferFull(uint8_t* packet,
                              size_t* index,
                              PacketReadyCallback callback) const { … }

size_t RtcpPacket::HeaderLength() const { … }

// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
//
// RTP header format.
//   0                   1                   2                   3
//   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//  |V=2|P| RC/FMT  |      PT       |             length            |
//  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
void RtcpPacket::CreateHeader(
    size_t count_or_format,  // Depends on packet type.
    uint8_t packet_type,
    size_t length,
    uint8_t* buffer,
    size_t* pos) { … }

void RtcpPacket::CreateHeader(
    size_t count_or_format,  // Depends on packet type.
    uint8_t packet_type,
    size_t length,
    bool padding,
    uint8_t* buffer,
    size_t* pos) { … }

}  // namespace rtcp
}  // namespace webrtc