diff options
Diffstat (limited to 'webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc')
-rw-r--r-- | webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc | 95 |
1 files changed, 95 insertions, 0 deletions
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc new file mode 100644 index 0000000000..030f9f81fa --- /dev/null +++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc @@ -0,0 +1,95 @@ +/* + * 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/byte_io.h" +#include "webrtc/modules/rtp_rtcp/source/rtp_utility.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 |