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/main/interface/audio_coding_module.h"
23 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
24 #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
25 #include "webrtc/modules/audio_coding/main/test/Channel.h"
26 #include "webrtc/modules/audio_coding/main/test/PCMFile.h"
27 #include "webrtc/modules/audio_coding/main/test/utility.h"
28 #include "webrtc/system_wrappers/interface/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_int32(init_delay, 0, "Initial delay in millisecond.");
37 DEFINE_bool(dtx, false, "Enable DTX at the sender side.");
38 DEFINE_bool(packet_loss, false, "Apply packet loss, c.f. Channel{.cc, .h}.");
39 DEFINE_bool(fec, false, "Use Forward Error Correction (FEC).");
40
41 namespace webrtc {
42
43 namespace {
44
45 struct CodecSettings {
46 char name[50];
47 int sample_rate_hz;
48 int num_channels;
49 };
50
51 struct AcmSettings {
52 bool dtx;
53 bool fec;
54 };
55
56 struct TestSettings {
57 CodecSettings codec;
58 AcmSettings acm;
59 bool packet_loss;
60 };
61
62 } // namespace
63
64 class DelayTest {
65 public:
DelayTest()66 DelayTest()
67 : acm_a_(AudioCodingModule::Create(0)),
68 acm_b_(AudioCodingModule::Create(1)),
69 channel_a2b_(new Channel),
70 test_cntr_(0),
71 encoding_sample_rate_hz_(8000) {}
72
~DelayTest()73 ~DelayTest() {
74 if (channel_a2b_ != NULL) {
75 delete channel_a2b_;
76 channel_a2b_ = NULL;
77 }
78 in_file_a_.Close();
79 }
80
Initialize()81 void Initialize() {
82 test_cntr_ = 0;
83 std::string file_name = webrtc::test::ResourcePath(
84 "audio_coding/testfile32kHz", "pcm");
85 if (FLAGS_input_file.size() > 0)
86 file_name = FLAGS_input_file;
87 in_file_a_.Open(file_name, 32000, "rb");
88 ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
89 "Couldn't initialize receiver.\n";
90 ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
91 "Couldn't initialize receiver.\n";
92 if (FLAGS_init_delay > 0) {
93 ASSERT_EQ(0, acm_b_->SetInitialPlayoutDelay(FLAGS_init_delay)) <<
94 "Failed to set initial delay.\n";
95 }
96
97 if (FLAGS_delay > 0) {
98 ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
99 "Failed to set minimum delay.\n";
100 }
101
102 int num_encoders = acm_a_->NumberOfCodecs();
103 CodecInst my_codec_param;
104 for (int n = 0; n < num_encoders; n++) {
105 EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
106 "Failed to get codec.";
107 if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
108 my_codec_param.channels = 1;
109 else if (my_codec_param.channels > 1)
110 continue;
111 if (STR_CASE_CMP(my_codec_param.plname, "CN") == 0 &&
112 my_codec_param.plfreq == 48000)
113 continue;
114 if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
115 continue;
116 ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
117 "Couldn't register receive codec.\n";
118 }
119
120 // Create and connect the channel
121 ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
122 "Couldn't register Transport callback.\n";
123 channel_a2b_->RegisterReceiverACM(acm_b_.get());
124 }
125
Perform(const TestSettings * config,size_t num_tests,int duration_sec,const char * output_prefix)126 void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
127 const char* output_prefix) {
128 for (size_t n = 0; n < num_tests; ++n) {
129 ApplyConfig(config[n]);
130 Run(duration_sec, output_prefix);
131 }
132 }
133
134 private:
ApplyConfig(const TestSettings & config)135 void ApplyConfig(const TestSettings& config) {
136 printf("====================================\n");
137 printf("Test %d \n"
138 "Codec: %s, %d kHz, %d channel(s)\n"
139 "ACM: DTX %s, FEC %s\n"
140 "Channel: %s\n",
141 ++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
142 config.codec.num_channels, config.acm.dtx ? "on" : "off",
143 config.acm.fec ? "on" : "off",
144 config.packet_loss ? "with packet-loss" : "no packet-loss");
145 SendCodec(config.codec);
146 ConfigAcm(config.acm);
147 ConfigChannel(config.packet_loss);
148 }
149
SendCodec(const CodecSettings & config)150 void SendCodec(const CodecSettings& config) {
151 CodecInst my_codec_param;
152 ASSERT_EQ(0, AudioCodingModule::Codec(
153 config.name, &my_codec_param, config.sample_rate_hz,
154 config.num_channels)) << "Specified codec is not supported.\n";
155
156 encoding_sample_rate_hz_ = my_codec_param.plfreq;
157 ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) <<
158 "Failed to register send-codec.\n";
159 }
160
ConfigAcm(const AcmSettings & config)161 void ConfigAcm(const AcmSettings& config) {
162 ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) <<
163 "Failed to set VAD.\n";
164 ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) <<
165 "Failed to set RED.\n";
166 }
167
ConfigChannel(bool packet_loss)168 void ConfigChannel(bool packet_loss) {
169 channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
170 }
171
OpenOutFile(const char * output_id)172 void OpenOutFile(const char* output_id) {
173 std::stringstream file_stream;
174 file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
175 << "Hz" << "_" << FLAGS_init_delay << "ms_" << FLAGS_delay << "ms.pcm";
176 std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
177 std::string file_name = webrtc::test::OutputPath() + file_stream.str();
178 out_file_b_.Open(file_name.c_str(), 32000, "wb");
179 }
180
Run(int duration_sec,const char * output_prefix)181 void Run(int duration_sec, const char* output_prefix) {
182 OpenOutFile(output_prefix);
183 AudioFrame audio_frame;
184 uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
185
186 int num_frames = 0;
187 int in_file_frames = 0;
188 uint32_t playout_ts;
189 uint32_t received_ts;
190 double average_delay = 0;
191 double inst_delay_sec = 0;
192 while (num_frames < (duration_sec * 100)) {
193 if (in_file_a_.EndOfFile()) {
194 in_file_a_.Rewind();
195 }
196
197 // Print delay information every 16 frame
198 if ((num_frames & 0x3F) == 0x3F) {
199 NetworkStatistics statistics;
200 acm_b_->GetNetworkStatistics(&statistics);
201 fprintf(stdout, "delay: min=%3d max=%3d mean=%3d median=%3d"
202 " ts-based average = %6.3f, "
203 "curr buff-lev = %4u opt buff-lev = %4u \n",
204 statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs,
205 statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs,
206 average_delay, statistics.currentBufferSize,
207 statistics.preferredBufferSize);
208 fflush (stdout);
209 }
210
211 in_file_a_.Read10MsData(audio_frame);
212 ASSERT_GE(acm_a_->Add10MsData(audio_frame), 0);
213 ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
214 out_file_b_.Write10MsData(
215 audio_frame.data_,
216 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
217 acm_b_->PlayoutTimestamp(&playout_ts);
218 received_ts = channel_a2b_->LastInTimestamp();
219 inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts)
220 / static_cast<double>(encoding_sample_rate_hz_);
221
222 if (num_frames > 10)
223 average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
224
225 ++num_frames;
226 ++in_file_frames;
227 }
228 out_file_b_.Close();
229 }
230
231 rtc::scoped_ptr<AudioCodingModule> acm_a_;
232 rtc::scoped_ptr<AudioCodingModule> acm_b_;
233
234 Channel* channel_a2b_;
235
236 PCMFile in_file_a_;
237 PCMFile out_file_b_;
238 int test_cntr_;
239 int encoding_sample_rate_hz_;
240 };
241
242 } // namespace webrtc
243
main(int argc,char * argv[])244 int main(int argc, char* argv[]) {
245 google::ParseCommandLineFlags(&argc, &argv, true);
246 webrtc::TestSettings test_setting;
247 strcpy(test_setting.codec.name, FLAGS_codec.c_str());
248
249 if (FLAGS_sample_rate_hz != 8000 &&
250 FLAGS_sample_rate_hz != 16000 &&
251 FLAGS_sample_rate_hz != 32000 &&
252 FLAGS_sample_rate_hz != 48000) {
253 std::cout << "Invalid sampling rate.\n";
254 return 1;
255 }
256 test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
257 if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) {
258 std::cout << "Only mono and stereo are supported.\n";
259 return 1;
260 }
261 test_setting.codec.num_channels = FLAGS_num_channels;
262 test_setting.acm.dtx = FLAGS_dtx;
263 test_setting.acm.fec = FLAGS_fec;
264 test_setting.packet_loss = FLAGS_packet_loss;
265
266 webrtc::DelayTest delay_test;
267 delay_test.Initialize();
268 delay_test.Perform(&test_setting, 1, 240, "delay_test");
269 return 0;
270 }
271