1 /*
2 * Copyright (c) 2013 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 <assert.h>
12 #include <math.h>
13
14 #include <iostream>
15
16 #include "gflags/gflags.h"
17 #include "testing/gtest/include/gtest/gtest.h"
18 #include "webrtc/base/scoped_ptr.h"
19 #include "webrtc/common.h"
20 #include "webrtc/common_types.h"
21 #include "webrtc/engine_configurations.h"
22 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
23 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
24 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
25 #include "webrtc/modules/audio_coding/test/Channel.h"
26 #include "webrtc/modules/audio_coding/test/PCMFile.h"
27 #include "webrtc/modules/audio_coding/test/utility.h"
28 #include "webrtc/system_wrappers/include/event_wrapper.h"
29 #include "webrtc/test/testsupport/fileutils.h"
30
31 DEFINE_string(codec, "isac", "Codec Name");
32 DEFINE_int32(sample_rate_hz, 16000, "Sampling rate in Hertz.");
33 DEFINE_int32(num_channels, 1, "Number of Channels.");
34 DEFINE_string(input_file, "", "Input file, PCM16 32 kHz, optional.");
35 DEFINE_int32(delay, 0, "Delay in millisecond.");
36 DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
37 DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
38 DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
39
40 namespace webrtc {
41
42 namespace {
43
44 struct CodecSettings {
45 char name[50];
46 int sample_rate_hz;
47 int num_channels;
48 };
49
50 struct AcmSettings {
51 bool dtx;
52 bool fec;
53 };
54
55 struct TestSettings {
56 CodecSettings codec;
57 AcmSettings acm;
58 bool packet_loss;
59 };
60
61 } // namespace
62
63 class DelayTest {
64 public:
DelayTest()65 DelayTest()
66 : acm_a_(AudioCodingModule::Create(0)),
67 acm_b_(AudioCodingModule::Create(1)),
68 channel_a2b_(new Channel),
69 test_cntr_(0),
70 encoding_sample_rate_hz_(8000) {}
71
~DelayTest()72 ~DelayTest() {
73 if (channel_a2b_ != NULL) {
74 delete channel_a2b_;
75 channel_a2b_ = NULL;
76 }
77 in_file_a_.Close();
78 }
79
Initialize()80 void Initialize() {
81 test_cntr_ = 0;
82 std::string file_name = webrtc::test::ResourcePath(
83 "audio_coding/testfile32kHz", "pcm");
84 if (FLAGS_input_file.size() > 0)
85 file_name = FLAGS_input_file;
86 in_file_a_.Open(file_name, 32000, "rb");
87 ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
88 "Couldn't initialize receiver.\n";
89 ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
90 "Couldn't initialize receiver.\n";
91
92 if (FLAGS_delay > 0) {
93 ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
94 "Failed to set minimum delay.\n";
95 }
96
97 int num_encoders = acm_a_->NumberOfCodecs();
98 CodecInst my_codec_param;
99 for (int n = 0; n < num_encoders; n++) {
100 EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
101 "Failed to get codec.";
102 if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
103 my_codec_param.channels = 1;
104 else if (my_codec_param.channels > 1)
105 continue;
106 if (STR_CASE_CMP(my_codec_param.plname, "CN") == 0 &&
107 my_codec_param.plfreq == 48000)
108 continue;
109 if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
110 continue;
111 ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
112 "Couldn't register receive codec.\n";
113 }
114
115 // Create and connect the channel
116 ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
117 "Couldn't register Transport callback.\n";
118 channel_a2b_->RegisterReceiverACM(acm_b_.get());
119 }
120
Perform(const TestSettings * config,size_t num_tests,int duration_sec,const char * output_prefix)121 void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
122 const char* output_prefix) {
123 for (size_t n = 0; n < num_tests; ++n) {
124 ApplyConfig(config[n]);
125 Run(duration_sec, output_prefix);
126 }
127 }
128
129 private:
ApplyConfig(const TestSettings & config)130 void ApplyConfig(const TestSettings& config) {
131 printf("====================================\n");
132 printf("Test %d \n"
133 "Codec: %s, %d kHz, %d channel(s)\n"
134 "ACM: DTX %s, FEC %s\n"
135 "Channel: %s\n",
136 ++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
137 config.codec.num_channels, config.acm.dtx ? "on" : "off",
138 config.acm.fec ? "on" : "off",
139 config.packet_loss ? "with packet-loss" : "no packet-loss");
140 SendCodec(config.codec);
141 ConfigAcm(config.acm);
142 ConfigChannel(config.packet_loss);
143 }
144
SendCodec(const CodecSettings & config)145 void SendCodec(const CodecSettings& config) {
146 CodecInst my_codec_param;
147 ASSERT_EQ(0, AudioCodingModule::Codec(
148 config.name, &my_codec_param, config.sample_rate_hz,
149 config.num_channels)) << "Specified codec is not supported.\n";
150
151 encoding_sample_rate_hz_ = my_codec_param.plfreq;
152 ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) <<
153 "Failed to register send-codec.\n";
154 }
155
ConfigAcm(const AcmSettings & config)156 void ConfigAcm(const AcmSettings& config) {
157 ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) <<
158 "Failed to set VAD.\n";
159 ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) <<
160 "Failed to set RED.\n";
161 }
162
ConfigChannel(bool packet_loss)163 void ConfigChannel(bool packet_loss) {
164 channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
165 }
166
OpenOutFile(const char * output_id)167 void OpenOutFile(const char* output_id) {
168 std::stringstream file_stream;
169 file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
170 << "Hz" << "_" << FLAGS_delay << "ms.pcm";
171 std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
172 std::string file_name = webrtc::test::OutputPath() + file_stream.str();
173 out_file_b_.Open(file_name.c_str(), 32000, "wb");
174 }
175
Run(int duration_sec,const char * output_prefix)176 void Run(int duration_sec, const char* output_prefix) {
177 OpenOutFile(output_prefix);
178 AudioFrame audio_frame;
179 uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
180
181 int num_frames = 0;
182 int in_file_frames = 0;
183 uint32_t playout_ts;
184 uint32_t received_ts;
185 double average_delay = 0;
186 double inst_delay_sec = 0;
187 while (num_frames < (duration_sec * 100)) {
188 if (in_file_a_.EndOfFile()) {
189 in_file_a_.Rewind();
190 }
191
192 // Print delay information every 16 frame
193 if ((num_frames & 0x3F) == 0x3F) {
194 NetworkStatistics statistics;
195 acm_b_->GetNetworkStatistics(&statistics);
196 fprintf(stdout, "delay: min=%3d max=%3d mean=%3d median=%3d"
197 " ts-based average = %6.3f, "
198 "curr buff-lev = %4u opt buff-lev = %4u \n",
199 statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs,
200 statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs,
201 average_delay, statistics.currentBufferSize,
202 statistics.preferredBufferSize);
203 fflush (stdout);
204 }
205
206 in_file_a_.Read10MsData(audio_frame);
207 ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0);
208 ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
209 out_file_b_.Write10MsData(
210 audio_frame.data_,
211 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
212 acm_b_->PlayoutTimestamp(&playout_ts);
213 received_ts = channel_a2b_->LastInTimestamp();
214 inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts)
215 / static_cast<double>(encoding_sample_rate_hz_);
216
217 if (num_frames > 10)
218 average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
219
220 ++num_frames;
221 ++in_file_frames;
222 }
223 out_file_b_.Close();
224 }
225
226 rtc::scoped_ptr<AudioCodingModule> acm_a_;
227 rtc::scoped_ptr<AudioCodingModule> acm_b_;
228
229 Channel* channel_a2b_;
230
231 PCMFile in_file_a_;
232 PCMFile out_file_b_;
233 int test_cntr_;
234 int encoding_sample_rate_hz_;
235 };
236
237 } // namespace webrtc
238
main(int argc,char * argv[])239 int main(int argc, char* argv[]) {
240 google::ParseCommandLineFlags(&argc, &argv, true);
241 webrtc::TestSettings test_setting;
242 strcpy(test_setting.codec.name, FLAGS_codec.c_str());
243
244 if (FLAGS_sample_rate_hz != 8000 &&
245 FLAGS_sample_rate_hz != 16000 &&
246 FLAGS_sample_rate_hz != 32000 &&
247 FLAGS_sample_rate_hz != 48000) {
248 std::cout << "Invalid sampling rate.\n";
249 return 1;
250 }
251 test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
252 if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) {
253 std::cout << "Only mono and stereo are supported.\n";
254 return 1;
255 }
256 test_setting.codec.num_channels = FLAGS_num_channels;
257 test_setting.acm.dtx = FLAGS_dtx;
258 test_setting.acm.fec = FLAGS_fec;
259 test_setting.packet_loss = FLAGS_packet_loss;
260
261 webrtc::DelayTest delay_test;
262 delay_test.Initialize();
263 delay_test.Perform(&test_setting, 1, 240, "delay_test");
264 return 0;
265 }
266