/* * 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 "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" #include "webrtc/modules/rtp_rtcp/source/byte_io.h" namespace webrtc { namespace rtcp { namespace { void AssignUWord8(uint8_t* buffer, size_t* offset, uint8_t value) { buffer[(*offset)++] = value; } void AssignUWord16(uint8_t* buffer, size_t* offset, uint16_t value) { ByteWriter::WriteBigEndian(buffer + *offset, value); *offset += 2; } } // namespace rtc::Buffer RtcpPacket::Build() const { size_t length = 0; rtc::Buffer packet(IP_PACKET_SIZE); class PacketVerifier : public PacketReadyCallback { public: explicit PacketVerifier(rtc::Buffer* packet) : called_(false), packet_(packet) {} virtual ~PacketVerifier() {} void OnPacketReady(uint8_t* data, size_t length) override { RTC_CHECK(!called_) << "Fragmentation not supported."; called_ = true; packet_->SetSize(length); } private: bool called_; rtc::Buffer* const packet_; } verifier(&packet); Create(packet.data(), &length, packet.capacity(), &verifier); OnBufferFull(packet.data(), &length, &verifier); return packet; } bool RtcpPacket::Build(PacketReadyCallback* callback) const { uint8_t buffer[IP_PACKET_SIZE]; return BuildExternalBuffer(buffer, IP_PACKET_SIZE, callback); } bool RtcpPacket::BuildExternalBuffer(uint8_t* buffer, size_t max_length, PacketReadyCallback* callback) const { size_t index = 0; if (!Create(buffer, &index, max_length, callback)) return false; return OnBufferFull(buffer, &index, callback); } bool RtcpPacket::OnBufferFull(uint8_t* packet, size_t* index, RtcpPacket::PacketReadyCallback* callback) const { if (*index == 0) return false; callback->OnPacketReady(packet, *index); *index = 0; return true; } size_t RtcpPacket::HeaderLength() const { size_t length_in_bytes = BlockLength(); // Length in 32-bit words minus 1. assert(length_in_bytes > 0); return ((length_in_bytes + 3) / 4) - 1; } // 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( uint8_t count_or_format, // Depends on packet type. uint8_t packet_type, size_t length, uint8_t* buffer, size_t* pos) { assert(length <= 0xffff); const uint8_t kVersion = 2; AssignUWord8(buffer, pos, (kVersion << 6) + count_or_format); AssignUWord8(buffer, pos, packet_type); AssignUWord16(buffer, pos, length); } } // namespace rtcp } // namespace webrtc