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: 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 72 ~DelayTest() { 73 if (channel_a2b_ != NULL) { 74 delete channel_a2b_; 75 channel_a2b_ = NULL; 76 } 77 in_file_a_.Close(); 78 } 79 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 } 120 121 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: 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 } 144 145 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 } 155 156 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 } 162 163 void ConfigChannel(bool packet_loss) { 164 channel_a2b_->SetFECTestWithPacketLoss(packet_loss); 165 } 166 167 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 } 175 176 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 238 239 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
