1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "webrtc/modules/audio_coding/main/test/PacketLossTest.h"
12
13 #include "testing/gtest/include/gtest/gtest.h"
14 #include "webrtc/common.h"
15 #include "webrtc/test/testsupport/fileutils.h"
16
17 namespace webrtc {
18
ReceiverWithPacketLoss()19 ReceiverWithPacketLoss::ReceiverWithPacketLoss()
20 : loss_rate_(0),
21 burst_length_(1),
22 packet_counter_(0),
23 lost_packet_counter_(0),
24 burst_lost_counter_(burst_length_) {
25 }
26
Setup(AudioCodingModule * acm,RTPStream * rtpStream,std::string out_file_name,int channels,int loss_rate,int burst_length)27 void ReceiverWithPacketLoss::Setup(AudioCodingModule *acm,
28 RTPStream *rtpStream,
29 std::string out_file_name,
30 int channels,
31 int loss_rate,
32 int burst_length) {
33 loss_rate_ = loss_rate;
34 burst_length_ = burst_length;
35 burst_lost_counter_ = burst_length_; // To prevent first packet gets lost.
36 std::stringstream ss;
37 ss << out_file_name << "_" << loss_rate_ << "_" << burst_length_ << "_";
38 Receiver::Setup(acm, rtpStream, ss.str(), channels);
39 }
40
IncomingPacket()41 bool ReceiverWithPacketLoss::IncomingPacket() {
42 if (!_rtpStream->EndOfFile()) {
43 if (packet_counter_ == 0) {
44 _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
45 _payloadSizeBytes, &_nextTime);
46 if (_realPayloadSizeBytes == 0) {
47 if (_rtpStream->EndOfFile()) {
48 packet_counter_ = 0;
49 return true;
50 } else {
51 return false;
52 }
53 }
54 }
55
56 if (!PacketLost()) {
57 _acm->IncomingPacket(_incomingPayload, _realPayloadSizeBytes, _rtpInfo);
58 }
59 packet_counter_++;
60 _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
61 _payloadSizeBytes, &_nextTime);
62 if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
63 packet_counter_ = 0;
64 lost_packet_counter_ = 0;
65 }
66 }
67 return true;
68 }
69
PacketLost()70 bool ReceiverWithPacketLoss::PacketLost() {
71 if (burst_lost_counter_ < burst_length_) {
72 lost_packet_counter_++;
73 burst_lost_counter_++;
74 return true;
75 }
76
77 if (lost_packet_counter_ * 100 < loss_rate_ * packet_counter_) {
78 lost_packet_counter_++;
79 burst_lost_counter_ = 1;
80 return true;
81 }
82 return false;
83 }
84
SenderWithFEC()85 SenderWithFEC::SenderWithFEC()
86 : expected_loss_rate_(0) {
87 }
88
Setup(AudioCodingModule * acm,RTPStream * rtpStream,std::string in_file_name,int sample_rate,int channels,int expected_loss_rate)89 void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
90 std::string in_file_name, int sample_rate,
91 int channels, int expected_loss_rate) {
92 Sender::Setup(acm, rtpStream, in_file_name, sample_rate, channels);
93 EXPECT_TRUE(SetFEC(true));
94 EXPECT_TRUE(SetPacketLossRate(expected_loss_rate));
95 }
96
SetFEC(bool enable_fec)97 bool SenderWithFEC::SetFEC(bool enable_fec) {
98 if (_acm->SetCodecFEC(enable_fec) == 0) {
99 return true;
100 }
101 return false;
102 }
103
SetPacketLossRate(int expected_loss_rate)104 bool SenderWithFEC::SetPacketLossRate(int expected_loss_rate) {
105 if (_acm->SetPacketLossRate(expected_loss_rate) == 0) {
106 expected_loss_rate_ = expected_loss_rate;
107 return true;
108 }
109 return false;
110 }
111
PacketLossTest(int channels,int expected_loss_rate,int actual_loss_rate,int burst_length)112 PacketLossTest::PacketLossTest(int channels, int expected_loss_rate,
113 int actual_loss_rate, int burst_length)
114 : channels_(channels),
115 in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" :
116 "audio_coding/teststereo32kHz"),
117 sample_rate_hz_(32000),
118 sender_(new SenderWithFEC),
119 receiver_(new ReceiverWithPacketLoss),
120 expected_loss_rate_(expected_loss_rate),
121 actual_loss_rate_(actual_loss_rate),
122 burst_length_(burst_length) {
123 }
124
Perform()125 void PacketLossTest::Perform() {
126 #ifndef WEBRTC_CODEC_OPUS
127 return;
128 #else
129 scoped_ptr<AudioCodingModule> acm(AudioCodingModule::Create(0));
130
131 int codec_id = acm->Codec("opus", 48000, channels_);
132
133 RTPFile rtpFile;
134 std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
135
136 // Encode to file
137 rtpFile.Open(fileName.c_str(), "wb+");
138 rtpFile.WriteHeader();
139
140 sender_->testMode = 0;
141 sender_->codeId = codec_id;
142
143 sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
144 expected_loss_rate_);
145 struct CodecInst sendCodecInst;
146 if (acm->SendCodec(&sendCodecInst) >= 0) {
147 sender_->Run();
148 }
149 sender_->Teardown();
150 rtpFile.Close();
151
152 // Decode to file
153 rtpFile.Open(fileName.c_str(), "rb");
154 rtpFile.ReadHeader();
155
156 receiver_->testMode = 0;
157 receiver_->codeId = codec_id;
158
159 receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
160 actual_loss_rate_, burst_length_);
161 receiver_->Run();
162 receiver_->Teardown();
163 rtpFile.Close();
164 #endif
165 }
166
167 } // namespace webrtc
168