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 #include <memory>
16
17 #include "gflags/gflags.h"
18 #include "webrtc/common_types.h"
19 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
20 #include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
21 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
22 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
23 #include "webrtc/modules/audio_coding/test/Channel.h"
24 #include "webrtc/modules/audio_coding/test/PCMFile.h"
25 #include "webrtc/modules/audio_coding/test/utility.h"
26 #include "webrtc/system_wrappers/include/event_wrapper.h"
27 #include "webrtc/test/gtest.h"
28 #include "webrtc/test/testsupport/fileutils.h"
29 #include "webrtc/typedefs.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(true,
112 acm_b_->RegisterReceiveCodec(my_codec_param.pltype,
113 CodecInstToSdp(my_codec_param)));
114 }
115
116 // Create and connect the channel
117 ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
118 "Couldn't register Transport callback.\n";
119 channel_a2b_->RegisterReceiverACM(acm_b_.get());
120 }
121
Perform(const TestSettings * config,size_t num_tests,int duration_sec,const char * output_prefix)122 void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
123 const char* output_prefix) {
124 for (size_t n = 0; n < num_tests; ++n) {
125 ApplyConfig(config[n]);
126 Run(duration_sec, output_prefix);
127 }
128 }
129
130 private:
ApplyConfig(const TestSettings & config)131 void ApplyConfig(const TestSettings& config) {
132 printf("====================================\n");
133 printf("Test %d \n"
134 "Codec: %s, %d kHz, %d channel(s)\n"
135 "ACM: DTX %s, FEC %s\n"
136 "Channel: %s\n",
137 ++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
138 config.codec.num_channels, config.acm.dtx ? "on" : "off",
139 config.acm.fec ? "on" : "off",
140 config.packet_loss ? "with packet-loss" : "no packet-loss");
141 SendCodec(config.codec);
142 ConfigAcm(config.acm);
143 ConfigChannel(config.packet_loss);
144 }
145
SendCodec(const CodecSettings & config)146 void SendCodec(const CodecSettings& config) {
147 CodecInst my_codec_param;
148 ASSERT_EQ(0, AudioCodingModule::Codec(
149 config.name, &my_codec_param, config.sample_rate_hz,
150 config.num_channels)) << "Specified codec is not supported.\n";
151
152 encoding_sample_rate_hz_ = my_codec_param.plfreq;
153 ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) <<
154 "Failed to register send-codec.\n";
155 }
156
ConfigAcm(const AcmSettings & config)157 void ConfigAcm(const AcmSettings& config) {
158 ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) <<
159 "Failed to set VAD.\n";
160 ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) <<
161 "Failed to set RED.\n";
162 }
163
ConfigChannel(bool packet_loss)164 void ConfigChannel(bool packet_loss) {
165 channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
166 }
167
OpenOutFile(const char * output_id)168 void OpenOutFile(const char* output_id) {
169 std::stringstream file_stream;
170 file_stream << "delay_test_" << FLAGS_codec << "_" << FLAGS_sample_rate_hz
171 << "Hz" << "_" << FLAGS_delay << "ms.pcm";
172 std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
173 std::string file_name = webrtc::test::OutputPath() + file_stream.str();
174 out_file_b_.Open(file_name.c_str(), 32000, "wb");
175 }
176
Run(int duration_sec,const char * output_prefix)177 void Run(int duration_sec, const char* output_prefix) {
178 OpenOutFile(output_prefix);
179 AudioFrame audio_frame;
180 uint32_t out_freq_hz_b = out_file_b_.SamplingFrequency();
181
182 int num_frames = 0;
183 int in_file_frames = 0;
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 bool muted;
209 ASSERT_EQ(0,
210 acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame, &muted));
211 RTC_DCHECK(!muted);
212 out_file_b_.Write10MsData(
213 audio_frame.data_,
214 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
215 received_ts = channel_a2b_->LastInTimestamp();
216 rtc::Optional<uint32_t> playout_timestamp = acm_b_->PlayoutTimestamp();
217 ASSERT_TRUE(playout_timestamp);
218 inst_delay_sec = static_cast<uint32_t>(received_ts - *playout_timestamp) /
219 static_cast<double>(encoding_sample_rate_hz_);
220
221 if (num_frames > 10)
222 average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
223
224 ++num_frames;
225 ++in_file_frames;
226 }
227 out_file_b_.Close();
228 }
229
230 std::unique_ptr<AudioCodingModule> acm_a_;
231 std::unique_ptr<AudioCodingModule> acm_b_;
232
233 Channel* channel_a2b_;
234
235 PCMFile in_file_a_;
236 PCMFile out_file_b_;
237 int test_cntr_;
238 int encoding_sample_rate_hz_;
239 };
240
241 } // namespace webrtc
242
main(int argc,char * argv[])243 int main(int argc, char* argv[]) {
244 google::ParseCommandLineFlags(&argc, &argv, true);
245 webrtc::TestSettings test_setting;
246 strcpy(test_setting.codec.name, FLAGS_codec.c_str());
247
248 if (FLAGS_sample_rate_hz != 8000 &&
249 FLAGS_sample_rate_hz != 16000 &&
250 FLAGS_sample_rate_hz != 32000 &&
251 FLAGS_sample_rate_hz != 48000) {
252 std::cout << "Invalid sampling rate.\n";
253 return 1;
254 }
255 test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
256 if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) {
257 std::cout << "Only mono and stereo are supported.\n";
258 return 1;
259 }
260 test_setting.codec.num_channels = FLAGS_num_channels;
261 test_setting.acm.dtx = FLAGS_dtx;
262 test_setting.acm.fec = FLAGS_fec;
263 test_setting.packet_loss = FLAGS_packet_loss;
264
265 webrtc::DelayTest delay_test;
266 delay_test.Initialize();
267 delay_test.Perform(&test_setting, 1, 240, "delay_test");
268 return 0;
269 }
270