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[fix] fix qos module
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87
src/qos/report_block.cc
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87
src/qos/report_block.cc
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@@ -0,0 +1,87 @@
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/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "report_block.h"
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#include "byte_io.h"
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#include "log.h"
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namespace webrtc {
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namespace rtcp {
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// From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.
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//
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// RTCP report block (RFC 3550).
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//
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// 0 1 2 3
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// 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
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// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
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// 0 | SSRC_1 (SSRC of first source) |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// 4 | fraction lost | cumulative number of packets lost |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// 8 | extended highest sequence number received |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// 12 | interarrival jitter |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// 16 | last SR (LSR) |
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// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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// 20 | delay since last SR (DLSR) |
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// 24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
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ReportBlock::ReportBlock()
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: source_ssrc_(0),
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fraction_lost_(0),
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cumulative_lost_(0),
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extended_high_seq_num_(0),
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jitter_(0),
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last_sr_(0),
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delay_since_last_sr_(0) {}
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bool ReportBlock::Parse(const uint8_t* buffer, size_t length) {
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if (length < ReportBlock::kLength) {
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LOG_ERROR("Report Block should be 24 bytes long");
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return false;
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}
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source_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[0]);
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fraction_lost_ = buffer[4];
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cumulative_lost_ = ByteReader<int32_t, 3>::ReadBigEndian(&buffer[5]);
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extended_high_seq_num_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
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jitter_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[12]);
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last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[16]);
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delay_since_last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[20]);
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return true;
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}
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void ReportBlock::Create(uint8_t* buffer) const {
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// Runtime check should be done while setting cumulative_lost.
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ByteWriter<uint32_t>::WriteBigEndian(&buffer[0], source_ssrc());
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ByteWriter<uint8_t>::WriteBigEndian(&buffer[4], fraction_lost());
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ByteWriter<int32_t, 3>::WriteBigEndian(&buffer[5], cumulative_lost());
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ByteWriter<uint32_t>::WriteBigEndian(&buffer[8], extended_high_seq_num());
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ByteWriter<uint32_t>::WriteBigEndian(&buffer[12], jitter());
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ByteWriter<uint32_t>::WriteBigEndian(&buffer[16], last_sr());
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ByteWriter<uint32_t>::WriteBigEndian(&buffer[20], delay_since_last_sr());
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}
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bool ReportBlock::SetCumulativeLost(int32_t cumulative_lost) {
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// We have only 3 bytes to store it, and it's a signed value.
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if (cumulative_lost >= (1 << 23) || cumulative_lost < -(1 << 23)) {
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LOG_WARN("Cumulative lost is too big to fit into Report Block");
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return false;
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}
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cumulative_lost_ = cumulative_lost;
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return true;
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}
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} // namespace rtcp
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} // namespace webrtc
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