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Delete inter arrival jitter rtcp packet as unused

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WebRTC doesn't produces such packet and ignores it when receive.

Bug: None
Change-Id: I4af8cb3308cb2422808bdfc420a85fa175085bfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269181
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37627}
This commit is contained in:
Danil Chapovalov
2022-07-21 13:01:08 +02:00
committed by WebRTC LUCI CQ
parent 23370f22c1
commit 300a230f16
9 changed files with 4 additions and 261 deletions
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101
modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc
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@ -1,101 +0,0 @@
/*
* 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

54
modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h
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@ -1,54 +0,0 @@
/*
* 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.
*/
#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
#include <vector>
#include "modules/rtp_rtcp/source/rtcp_packet.h"
namespace webrtc {
namespace rtcp {
class CommonHeader;
class ExtendedJitterReport : public RtcpPacket {
public:
static constexpr uint8_t kPacketType = 195;
static constexpr size_t kMaxNumberOfJitterValues = 0x1f;
ExtendedJitterReport();
~ExtendedJitterReport() override;
// Parse assumes header is already parsed and validated.
bool Parse(const CommonHeader& packet);
bool SetJitterValues(std::vector<uint32_t> jitter_values);
const std::vector<uint32_t>& jitter_values() {
return inter_arrival_jitters_;
}
size_t BlockLength() const override;
bool Create(uint8_t* packet,
size_t* index,
size_t max_length,
PacketReadyCallback callback) const override;
private:
static constexpr size_t kJitterSizeBytes = 4;
std::vector<uint32_t> inter_arrival_jitters_;
};
} // namespace rtcp
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_

76
modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc
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@ -1,76 +0,0 @@
/*
* 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 "test/gmock.h"
#include "test/gtest.h"
#include "test/rtcp_packet_parser.h"
using ::testing::ElementsAre;
using ::testing::IsEmpty;
using webrtc::rtcp::ExtendedJitterReport;
namespace webrtc {
namespace {
constexpr uint32_t kJitter1 = 0x11121314;
constexpr uint32_t kJitter2 = 0x22242628;
} // namespace
TEST(RtcpPacketExtendedJitterReportTest, CreateAndParseWithoutItems) {
ExtendedJitterReport ij;
rtc::Buffer raw = ij.Build();
ExtendedJitterReport parsed;
EXPECT_TRUE(test::ParseSinglePacket(raw, &parsed));
EXPECT_THAT(parsed.jitter_values(), IsEmpty());
}
TEST(RtcpPacketExtendedJitterReportTest, CreateAndParseWithOneItem) {
ExtendedJitterReport ij;
EXPECT_TRUE(ij.SetJitterValues({kJitter1}));
rtc::Buffer raw = ij.Build();
ExtendedJitterReport parsed;
EXPECT_TRUE(test::ParseSinglePacket(raw, &parsed));
EXPECT_THAT(parsed.jitter_values(), ElementsAre(kJitter1));
}
TEST(RtcpPacketExtendedJitterReportTest, CreateAndParseWithTwoItems) {
ExtendedJitterReport ij;
EXPECT_TRUE(ij.SetJitterValues({kJitter1, kJitter2}));
rtc::Buffer raw = ij.Build();
ExtendedJitterReport parsed;
EXPECT_TRUE(test::ParseSinglePacket(raw, &parsed));
EXPECT_THAT(parsed.jitter_values(), ElementsAre(kJitter1, kJitter2));
}
TEST(RtcpPacketExtendedJitterReportTest, CreateWithTooManyItems) {
ExtendedJitterReport ij;
const int kMaxItems = ExtendedJitterReport::kMaxNumberOfJitterValues;
EXPECT_FALSE(
ij.SetJitterValues(std::vector<uint32_t>(kMaxItems + 1, kJitter1)));
EXPECT_TRUE(ij.SetJitterValues(std::vector<uint32_t>(kMaxItems, kJitter1)));
}
TEST(RtcpPacketExtendedJitterReportTest, ParseFailsWithTooManyItems) {
ExtendedJitterReport ij;
ij.SetJitterValues({kJitter1});
rtc::Buffer raw = ij.Build();
raw[0]++; // Damage packet: increase jitter count by 1.
ExtendedJitterReport parsed;
EXPECT_FALSE(test::ParseSinglePacket(raw, &parsed));
}
} // namespace webrtc