1 /* 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h" 12 13 #include "webrtc/base/checks.h" 14 #include "webrtc/base/logging.h" 15 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" 16 17 namespace webrtc { 18 namespace rtcp { 19 20 // From RFC 3550, RTP: A Transport Protocol for Real-Time Applications. 21 // 22 // RTCP report block (RFC 3550). 23 // 24 // 0 1 2 3 25 // 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 26 // +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ 27 // 0 | SSRC_1 (SSRC of first source) | 28 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 29 // 4 | fraction lost | cumulative number of packets lost | 30 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 31 // 8 | extended highest sequence number received | 32 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 33 // 12 | interarrival jitter | 34 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 35 // 16 | last SR (LSR) | 36 // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 37 // 20 | delay since last SR (DLSR) | 38 // 24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ 39 ReportBlock::ReportBlock() 40 : source_ssrc_(0), 41 fraction_lost_(0), 42 cumulative_lost_(0), 43 extended_high_seq_num_(0), 44 jitter_(0), 45 last_sr_(0), 46 delay_since_last_sr_(0) {} 47 48 bool ReportBlock::Parse(const uint8_t* buffer, size_t length) { 49 RTC_DCHECK(buffer != nullptr); 50 if (length < ReportBlock::kLength) { 51 LOG(LS_ERROR) << "Report Block should be 24 bytes long"; 52 return false; 53 } 54 55 source_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[0]); 56 fraction_lost_ = buffer[4]; 57 cumulative_lost_ = ByteReader<uint32_t, 3>::ReadBigEndian(&buffer[5]); 58 extended_high_seq_num_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]); 59 jitter_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[12]); 60 last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[16]); 61 delay_since_last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[20]); 62 63 return true; 64 } 65 66 void ReportBlock::Create(uint8_t* buffer) const { 67 // Runtime check should be done while setting cumulative_lost. 68 RTC_DCHECK_LT(cumulative_lost(), (1u << 24)); // Have only 3 bytes for it. 69 70 ByteWriter<uint32_t>::WriteBigEndian(&buffer[0], source_ssrc()); 71 ByteWriter<uint8_t>::WriteBigEndian(&buffer[4], fraction_lost()); 72 ByteWriter<uint32_t, 3>::WriteBigEndian(&buffer[5], cumulative_lost()); 73 ByteWriter<uint32_t>::WriteBigEndian(&buffer[8], extended_high_seq_num()); 74 ByteWriter<uint32_t>::WriteBigEndian(&buffer[12], jitter()); 75 ByteWriter<uint32_t>::WriteBigEndian(&buffer[16], last_sr()); 76 ByteWriter<uint32_t>::WriteBigEndian(&buffer[20], delay_since_last_sr()); 77 } 78 79 bool ReportBlock::WithCumulativeLost(uint32_t cumulative_lost) { 80 if (cumulative_lost >= (1u << 24)) { // Have only 3 bytes to store it. 81 LOG(LS_WARNING) << "Cumulative lost is too big to fit into Report Block"; 82 return false; 83 } 84 cumulative_lost_ = cumulative_lost; 85 return true; 86 } 87 88 } // namespace rtcp 89 } // namespace webrtc 90
