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rtcp::Ij renamed to rtcp::ExtendedJitterReport

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to match name given in the RFC5450
  private member renamed to inter_arrival_jitters_ for the same reason.
rtcp::ExtendedJitterReport moved into own file
accessors and Parse function added
  to make class usable for parsing packet

Review URL: https://codereview.webrtc.org/1434213004

Cr-Commit-Position: refs/heads/master@{#10636}
This commit is contained in:
danilchap 2015-11-13 07:33:20 -08:00 committed by Commit bot
parent cbe9f51cf8
commit f8506cbdd8
10 changed files with 266 additions and 139 deletions
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1
webrtc/modules/modules.gyp
View File

@ -247,6 +247,7 @@
'rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc',
'rtp_rtcp/source/rtcp_format_remb_unittest.cc',
'rtp_rtcp/source/rtcp_packet_unittest.cc',
'rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc',
'rtp_rtcp/source/rtcp_packet/report_block_unittest.cc',
'rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc',
'rtp_rtcp/source/rtcp_receiver_unittest.cc',

2
webrtc/modules/rtp_rtcp/BUILD.gn
View File

@ -46,6 +46,8 @@ source_set("rtp_rtcp") {
"source/remote_ntp_time_estimator.cc",
"source/rtcp_packet.cc",
"source/rtcp_packet.h",
"source/rtcp_packet/extended_jitter_report.cc",
"source/rtcp_packet/extended_jitter_report.h",
"source/rtcp_packet/report_block.cc",
"source/rtcp_packet/report_block.h",
"source/rtcp_packet/transport_feedback.cc",

2
webrtc/modules/rtp_rtcp/rtp_rtcp.gypi
View File

@ -41,6 +41,8 @@
'source/rtp_rtcp_impl.h',
'source/rtcp_packet.cc',
'source/rtcp_packet.h',
'source/rtcp_packet/extended_jitter_report.cc',
'source/rtcp_packet/extended_jitter_report.h',
'source/rtcp_packet/report_block.cc',
'source/rtcp_packet/report_block.h',
'source/rtcp_packet/transport_feedback.cc',

45
webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
View File

@ -163,28 +163,6 @@ void CreateReportBlocks(const std::vector<ReportBlock>& blocks,
}
}
// 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 |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
void CreateIj(const std::vector<uint32_t>& ij_items,
uint8_t* buffer,
size_t* pos) {
for (uint32_t item : ij_items)
AssignUWord32(buffer, pos, item);
}
// Source Description (SDES) (RFC 3550).
//
// 0 1 2 3
@ -804,29 +782,6 @@ bool ReceiverReport::WithReportBlock(const ReportBlock& block) {
return true;
}
bool Ij::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;
}
size_t length = ij_items_.size();
CreateHeader(length, PT_IJ, length, packet, index);
CreateIj(ij_items_, packet, index);
return true;
}
bool Ij::WithJitterItem(uint32_t jitter) {
if (ij_items_.size() >= kMaxNumberOfIjItems) {
LOG(LS_WARNING) << "Max inter-arrival jitter items reached.";
return false;
}
ij_items_.push_back(jitter);
return true;
}
bool Sdes::Create(uint8_t* packet,
size_t* index,
size_t max_length,

45
webrtc/modules/rtp_rtcp/source/rtcp_packet.h
View File

@ -263,51 +263,6 @@ class ReceiverReport : public RtcpPacket {
RTC_DISALLOW_COPY_AND_ASSIGN(ReceiverReport);
};
// 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.
class Ij : public RtcpPacket {
public:
Ij() : RtcpPacket() {}
virtual ~Ij() {}
bool WithJitterItem(uint32_t jitter);
protected:
bool Create(uint8_t* packet,
size_t* index,
size_t max_length,
RtcpPacket::PacketReadyCallback* callback) const override;
private:
static const int kMaxNumberOfIjItems = 0x1f;
size_t BlockLength() const {
return kHeaderLength + 4 * ij_items_.size();
}
std::vector<uint32_t> ij_items_;
RTC_DISALLOW_COPY_AND_ASSIGN(Ij);
};
// Source Description (SDES) (RFC 3550).
//
// 0 1 2 3

95
webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc Normal file
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@ -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/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

63
webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h Normal file
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@ -0,0 +1,63 @@
/*
* 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_
#include <vector>
#include "webrtc/base/checks.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
namespace webrtc {
namespace rtcp {
class ExtendedJitterReport : public RtcpPacket {
public:
static const uint8_t kPacketType = 195;
ExtendedJitterReport() : RtcpPacket() {}
virtual ~ExtendedJitterReport() {}
// Parse assumes header is already parsed and validated.
bool Parse(const RTCPUtility::RtcpCommonHeader& header,
const uint8_t* payload); // Size of the payload is in the header.
bool WithJitter(uint32_t jitter);
size_t jitters_count() const { return inter_arrival_jitters_.size(); }
uint32_t jitter(size_t index) const {
RTC_DCHECK_LT(index, jitters_count());
return inter_arrival_jitters_[index];
}
protected:
bool Create(uint8_t* packet,
size_t* index,
size_t max_length,
RtcpPacket::PacketReadyCallback* callback) const override;
private:
static const int kMaxNumberOfJitters = 0x1f;
size_t BlockLength() const override {
return kHeaderLength + 4 * inter_arrival_jitters_.size();
}
std::vector<uint32_t> inter_arrival_jitters_;
RTC_DISALLOW_COPY_AND_ASSIGN(ExtendedJitterReport);
};
} // namespace rtcp
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_JITTER_REPORT_H_

98
webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc Normal file
View File

@ -0,0 +1,98 @@
/*
* 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 <limits>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
using webrtc::rtcp::RawPacket;
using webrtc::rtcp::ExtendedJitterReport;
using webrtc::RTCPUtility::RtcpCommonHeader;
using webrtc::RTCPUtility::RtcpParseCommonHeader;
namespace webrtc {
namespace {
class RtcpPacketExtendedJitterReportTest : public ::testing::Test {
protected:
void BuildPacket() { packet = ij.Build().Pass(); }
void ParsePacket() {
RtcpCommonHeader header;
EXPECT_TRUE(
RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header));
EXPECT_EQ(header.BlockSize(), packet->Length());
EXPECT_TRUE(parsed_.Parse(
header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
}
ExtendedJitterReport ij;
rtc::scoped_ptr<RawPacket> packet;
const ExtendedJitterReport& parsed() { return parsed_; }
private:
ExtendedJitterReport parsed_;
};
TEST_F(RtcpPacketExtendedJitterReportTest, NoItem) {
// No initialization because packet is empty.
BuildPacket();
ParsePacket();
EXPECT_EQ(0u, parsed().jitters_count());
}
TEST_F(RtcpPacketExtendedJitterReportTest, OneItem) {
EXPECT_TRUE(ij.WithJitter(0x11121314));
BuildPacket();
ParsePacket();
EXPECT_EQ(1u, parsed().jitters_count());
EXPECT_EQ(0x11121314U, parsed().jitter(0));
}
TEST_F(RtcpPacketExtendedJitterReportTest, TwoItems) {
EXPECT_TRUE(ij.WithJitter(0x11121418));
EXPECT_TRUE(ij.WithJitter(0x22242628));
BuildPacket();
ParsePacket();
EXPECT_EQ(2u, parsed().jitters_count());
EXPECT_EQ(0x11121418U, parsed().jitter(0));
EXPECT_EQ(0x22242628U, parsed().jitter(1));
}
TEST_F(RtcpPacketExtendedJitterReportTest, TooManyItems) {
const int kMaxIjItems = (1 << 5) - 1;
for (int i = 0; i < kMaxIjItems; ++i) {
EXPECT_TRUE(ij.WithJitter(i));
}
EXPECT_FALSE(ij.WithJitter(kMaxIjItems));
}
TEST_F(RtcpPacketExtendedJitterReportTest, ParseFailWithTooManyItems) {
ij.WithJitter(0x11121418);
BuildPacket();
RtcpCommonHeader header;
RtcpParseCommonHeader(packet->Buffer(), packet->Length(), &header);
header.count_or_format++; // Damage package.
ExtendedJitterReport parsed;
EXPECT_FALSE(parsed.Parse(
header, packet->Buffer() + RtcpCommonHeader::kHeaderSizeBytes));
}
} // namespace
} // namespace webrtc

45
webrtc/modules/rtp_rtcp/source/rtcp_packet_unittest.cc
View File

@ -23,7 +23,6 @@ using webrtc::rtcp::Bye;
using webrtc::rtcp::Dlrr;
using webrtc::rtcp::Empty;
using webrtc::rtcp::Fir;
using webrtc::rtcp::Ij;
using webrtc::rtcp::Nack;
using webrtc::rtcp::Pli;
using webrtc::rtcp::Sdes;
@ -196,50 +195,6 @@ TEST(RtcpPacketTest, SrWithTooManyReportBlocks) {
EXPECT_FALSE(sr.WithReportBlock(rb));
}
TEST(RtcpPacketTest, IjNoItem) {
Ij ij;
rtc::scoped_ptr<RawPacket> packet(ij.Build());
RtcpPacketParser parser;
parser.Parse(packet->Buffer(), packet->Length());
EXPECT_EQ(1, parser.ij()->num_packets());
EXPECT_EQ(0, parser.ij_item()->num_packets());
}
TEST(RtcpPacketTest, IjOneItem) {
Ij ij;
EXPECT_TRUE(ij.WithJitterItem(0x11111111));
rtc::scoped_ptr<RawPacket> packet(ij.Build());
RtcpPacketParser parser;
parser.Parse(packet->Buffer(), packet->Length());
EXPECT_EQ(1, parser.ij()->num_packets());
EXPECT_EQ(1, parser.ij_item()->num_packets());
EXPECT_EQ(0x11111111U, parser.ij_item()->Jitter());
}
TEST(RtcpPacketTest, IjTwoItems) {
Ij ij;
EXPECT_TRUE(ij.WithJitterItem(0x11111111));
EXPECT_TRUE(ij.WithJitterItem(0x22222222));
rtc::scoped_ptr<RawPacket> packet(ij.Build());
RtcpPacketParser parser;
parser.Parse(packet->Buffer(), packet->Length());
EXPECT_EQ(1, parser.ij()->num_packets());
EXPECT_EQ(2, parser.ij_item()->num_packets());
EXPECT_EQ(0x22222222U, parser.ij_item()->Jitter());
}
TEST(RtcpPacketTest, IjTooManyItems) {
Ij ij;
const int kMaxIjItems = (1 << 5) - 1;
for (int i = 0; i < kMaxIjItems; ++i) {
EXPECT_TRUE(ij.WithJitterItem(i));
}
EXPECT_FALSE(ij.WithJitterItem(kMaxIjItems));
}
TEST(RtcpPacketTest, AppWithNoData) {
App app;
app.WithSubType(30);

9
webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
View File

@ -25,6 +25,7 @@
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
namespace webrtc {
@ -384,20 +385,20 @@ TEST_F(RtcpReceiverTest, GetRtt) {
}
TEST_F(RtcpReceiverTest, InjectIjWithNoItem) {
rtcp::Ij ij;
rtcp::ExtendedJitterReport ij;
rtc::scoped_ptr<rtcp::RawPacket> packet(ij.Build());
EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
}
TEST_F(RtcpReceiverTest, InjectIjWithOneItem) {
rtcp::Ij ij;
ij.WithJitterItem(0x11111111);
rtcp::ExtendedJitterReport ij;
ij.WithJitter(0x11213141);
rtc::scoped_ptr<rtcp::RawPacket> packet(ij.Build());
EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
EXPECT_EQ(kRtcpTransmissionTimeOffset, rtcp_packet_info_.rtcpPacketTypeFlags);
EXPECT_EQ(0x11111111U, rtcp_packet_info_.interArrivalJitter);
EXPECT_EQ(0x11213141U, rtcp_packet_info_.interArrivalJitter);
}
TEST_F(RtcpReceiverTest, InjectAppWithNoData) {