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| 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/include/audio_coding_module.h" | |
| 23 #include "webrtc/modules/audio_coding/main/include/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/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 } | |
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Chromium Code Reviews
Issue 1481493004:
audio_coding: remove "main" directory (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master