2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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.
16 #include "gflags/gflags.h"
17 #include "testing/gtest/include/gtest/gtest.h"
18 #include "webrtc/common.h"
19 #include "webrtc/common_types.h"
20 #include "webrtc/engine_configurations.h"
21 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
22 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
23 #include "webrtc/modules/audio_coding/main/acm2/acm_common_defs.h"
24 #include "webrtc/modules/audio_coding/main/test/Channel.h"
25 #include "webrtc/modules/audio_coding/main/test/PCMFile.h"
26 #include "webrtc/modules/audio_coding/main/test/utility.h"
27 #include "webrtc/system_wrappers/interface/event_wrapper.h"
28 #include "webrtc/system_wrappers/interface/scoped_ptr.h"
29 #include "webrtc/test/testsupport/fileutils.h"
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).");
45 struct CodecSettings {
67 : acm_a_(AudioCodingModule::Create(0)),
68 acm_b_(AudioCodingModule::Create(1)),
69 channel_a2b_(new Channel),
71 encoding_sample_rate_hz_(8000) {}
74 if (channel_a2b_ != NULL) {
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";
97 if (FLAGS_delay > 0) {
98 ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAGS_delay)) <<
99 "Failed to set minimum delay.\n";
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)
111 if (STR_CASE_CMP(my_codec_param.plname, "CN") == 0 &&
112 my_codec_param.plfreq == 48000)
114 if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
116 ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
117 "Couldn't register receive codec.\n";
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());
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);
135 void ApplyConfig(const TestSettings& config) {
136 printf("====================================\n");
138 "Codec: %s, %d kHz, %d channel(s)\n"
139 "ACM: DTX %s, FEC %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);
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";
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";
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_->SetFECStatus(config.fec)) <<
165 "Failed to set FEC.\n";
168 void ConfigChannel(bool packet_loss) {
169 channel_a2b_->SetFECTestWithPacketLoss(packet_loss);
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");
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();
187 int in_file_frames = 0;
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()) {
197 // Print delay information every 16 frame
198 if ((num_frames & 0x3F) == 0x3F) {
199 ACMNetworkStatistics statistics;
200 acm_b_->NetworkStatistics(&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);
211 in_file_a_.Read10MsData(audio_frame);
212 ASSERT_EQ(0, acm_a_->Add10MsData(audio_frame));
213 ASSERT_LE(0, acm_a_->Process());
214 ASSERT_EQ(0, acm_b_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
215 out_file_b_.Write10MsData(
217 audio_frame.samples_per_channel_ * audio_frame.num_channels_);
218 acm_b_->PlayoutTimestamp(&playout_ts);
219 received_ts = channel_a2b_->LastInTimestamp();
220 inst_delay_sec = static_cast<uint32_t>(received_ts - playout_ts)
221 / static_cast<double>(encoding_sample_rate_hz_);
224 average_delay = 0.95 * average_delay + 0.05 * inst_delay_sec;
232 scoped_ptr<AudioCodingModule> acm_a_;
233 scoped_ptr<AudioCodingModule> acm_b_;
235 Channel* channel_a2b_;
240 int encoding_sample_rate_hz_;
243 } // namespace webrtc
245 int main(int argc, char* argv[]) {
246 google::ParseCommandLineFlags(&argc, &argv, true);
247 webrtc::TestSettings test_setting;
248 strcpy(test_setting.codec.name, FLAGS_codec.c_str());
250 if (FLAGS_sample_rate_hz != 8000 &&
251 FLAGS_sample_rate_hz != 16000 &&
252 FLAGS_sample_rate_hz != 32000 &&
253 FLAGS_sample_rate_hz != 48000) {
254 std::cout << "Invalid sampling rate.\n";
257 test_setting.codec.sample_rate_hz = FLAGS_sample_rate_hz;
258 if (FLAGS_num_channels < 1 || FLAGS_num_channels > 2) {
259 std::cout << "Only mono and stereo are supported.\n";
262 test_setting.codec.num_channels = FLAGS_num_channels;
263 test_setting.acm.dtx = FLAGS_dtx;
264 test_setting.acm.fec = FLAGS_fec;
265 test_setting.packet_loss = FLAGS_packet_loss;
267 webrtc::DelayTest delay_test;
268 delay_test.Initialize();
269 delay_test.Perform(&test_setting, 1, 240, "delay_test");