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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 | |
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 | |
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 | |
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 | |
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 | |
85 SenderWithFEC::SenderWithFEC() | |
86 : expected_loss_rate_(0) { | |
87 } | |
88 | |
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 | |
97 bool SenderWithFEC::SetFEC(bool enable_fec) { | |
98 if (_acm->SetCodecFEC(enable_fec) == 0) { | |
99 return true; | |
100 } | |
101 return false; | |
102 } | |
103 | |
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 | |
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 | |
125 void PacketLossTest::Perform() { | |
126 #ifndef WEBRTC_CODEC_OPUS | |
127 return; | |
128 #else | |
129 rtc::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::TempFilename(webrtc::test::OutputPath(), | |
135 "packet_loss_test"); | |
136 | |
137 // Encode to file | |
138 rtpFile.Open(fileName.c_str(), "wb+"); | |
139 rtpFile.WriteHeader(); | |
140 | |
141 sender_->testMode = 0; | |
142 sender_->codeId = codec_id; | |
143 | |
144 sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_, | |
145 expected_loss_rate_); | |
146 if (acm->SendCodec()) { | |
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 | |
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