| /* |
| * Copyright (c) 2015 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/extended_jitter_report.h" |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/logging.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
| #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
| |
| using webrtc::RTCPUtility::RtcpCommonHeader; |
| |
| namespace webrtc { |
| namespace rtcp { |
| |
| // Transmission Time Offsets in RTP Streams (RFC 5450). |
| // |
| // 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 |
| // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| // hdr |V=2|P| RC | PT=IJ=195 | length | |
| // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| // | inter-arrival jitter | |
| // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| // . . |
| // . . |
| // . . |
| // | inter-arrival jitter | |
| // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| // |
| // If present, this RTCP packet must be placed after a receiver report |
| // (inside a compound RTCP packet), and MUST have the same value for RC |
| // (reception report count) as the receiver report. |
| |
| bool ExtendedJitterReport::Parse(const RtcpCommonHeader& header, |
| const uint8_t* payload) { |
| RTC_DCHECK(header.packet_type == kPacketType); |
| |
| const uint8_t jitters_count = header.count_or_format; |
| const size_t kJitterSizeBytes = 4u; |
| |
| if (header.payload_size_bytes < jitters_count * kJitterSizeBytes) { |
| LOG(LS_WARNING) << "Packet is too small to contain all the jitter."; |
| return false; |
| } |
| |
| inter_arrival_jitters_.resize(jitters_count); |
| for (size_t index = 0; index < jitters_count; ++index) { |
| inter_arrival_jitters_[index] = |
| ByteReader<uint32_t>::ReadBigEndian(&payload[index * kJitterSizeBytes]); |
| } |
| |
| return true; |
| } |
| |
| bool ExtendedJitterReport::WithJitter(uint32_t jitter) { |
| if (inter_arrival_jitters_.size() >= kMaxNumberOfJitters) { |
| LOG(LS_WARNING) << "Max inter-arrival jitter items reached."; |
| return false; |
| } |
| inter_arrival_jitters_.push_back(jitter); |
| return true; |
| } |
| |
| bool ExtendedJitterReport::Create( |
| uint8_t* packet, |
| size_t* index, |
| size_t max_length, |
| RtcpPacket::PacketReadyCallback* callback) const { |
| while (*index + BlockLength() > max_length) { |
| if (!OnBufferFull(packet, index, callback)) |
| return false; |
| } |
| const size_t index_end = *index + BlockLength(); |
| size_t length = inter_arrival_jitters_.size(); |
| CreateHeader(length, kPacketType, length, packet, index); |
| |
| for (uint32_t jitter : inter_arrival_jitters_) { |
| ByteWriter<uint32_t>::WriteBigEndian(packet + *index, jitter); |
| *index += sizeof(uint32_t); |
| } |
| // Sanity check. |
| RTC_DCHECK_EQ(index_end, *index); |
| return true; |
| } |
| |
| } // namespace rtcp |
| } // namespace webrtc |