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/*
* 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 "modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
#include <cstdint>
#include <utility>
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace rtcp {
constexpr uint8_t ExtendedJitterReport::kPacketType;
// 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.
ExtendedJitterReport::ExtendedJitterReport() = default;
ExtendedJitterReport::~ExtendedJitterReport() = default;
bool ExtendedJitterReport::Parse(const CommonHeader& packet) {
RTC_DCHECK_EQ(packet.type(), kPacketType);
const uint8_t number_of_jitters = packet.count();
if (packet.payload_size_bytes() < number_of_jitters * kJitterSizeBytes) {
RTC_LOG(LS_WARNING) << "Packet is too small to contain all the jitter.";
return false;
}
inter_arrival_jitters_.resize(number_of_jitters);
for (size_t index = 0; index < number_of_jitters; ++index) {
inter_arrival_jitters_[index] = ByteReader<uint32_t>::ReadBigEndian(
&packet.payload()[index * kJitterSizeBytes]);
}
return true;
}
bool ExtendedJitterReport::SetJitterValues(std::vector<uint32_t> values) {
if (values.size() > kMaxNumberOfJitterValues) {
RTC_LOG(LS_WARNING) << "Too many inter-arrival jitter items.";
return false;
}
inter_arrival_jitters_ = std::move(values);
return true;
}
size_t ExtendedJitterReport::BlockLength() const {
return kHeaderLength + kJitterSizeBytes * inter_arrival_jitters_.size();
}
bool ExtendedJitterReport::Create(uint8_t* packet,
size_t* index,
size_t max_length,
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 += kJitterSizeBytes;
}
// Sanity check.
RTC_DCHECK_EQ(index_end, *index);
return true;
}
} // namespace rtcp
} // namespace webrtc