| OLD | NEW |
|---|
| (Empty) |
| 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 "webrtc/modules/audio_coding/main/test/opus_test.h" | |
| 12 | |
| 13 #include <assert.h> | |
| 14 | |
| 15 #include <string> | |
| 16 | |
| 17 #include "testing/gtest/include/gtest/gtest.h" | |
| 18 #include "webrtc/common_types.h" | |
| 19 #include "webrtc/engine_configurations.h" | |
| 20 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h" | |
| 21 #include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.
h" | |
| 22 #include "webrtc/modules/audio_coding/main/test/TestStereo.h" | |
| 23 #include "webrtc/modules/audio_coding/main/test/utility.h" | |
| 24 #include "webrtc/system_wrappers/include/trace.h" | |
| 25 #include "webrtc/test/testsupport/fileutils.h" | |
| 26 | |
| 27 namespace webrtc { | |
| 28 | |
| 29 OpusTest::OpusTest() | |
| 30 : acm_receiver_(AudioCodingModule::Create(0)), | |
| 31 channel_a2b_(NULL), | |
| 32 counter_(0), | |
| 33 payload_type_(255), | |
| 34 rtp_timestamp_(0) {} | |
| 35 | |
| 36 OpusTest::~OpusTest() { | |
| 37 if (channel_a2b_ != NULL) { | |
| 38 delete channel_a2b_; | |
| 39 channel_a2b_ = NULL; | |
| 40 } | |
| 41 if (opus_mono_encoder_ != NULL) { | |
| 42 WebRtcOpus_EncoderFree(opus_mono_encoder_); | |
| 43 opus_mono_encoder_ = NULL; | |
| 44 } | |
| 45 if (opus_stereo_encoder_ != NULL) { | |
| 46 WebRtcOpus_EncoderFree(opus_stereo_encoder_); | |
| 47 opus_stereo_encoder_ = NULL; | |
| 48 } | |
| 49 if (opus_mono_decoder_ != NULL) { | |
| 50 WebRtcOpus_DecoderFree(opus_mono_decoder_); | |
| 51 opus_mono_decoder_ = NULL; | |
| 52 } | |
| 53 if (opus_stereo_decoder_ != NULL) { | |
| 54 WebRtcOpus_DecoderFree(opus_stereo_decoder_); | |
| 55 opus_stereo_decoder_ = NULL; | |
| 56 } | |
| 57 } | |
| 58 | |
| 59 void OpusTest::Perform() { | |
| 60 #ifndef WEBRTC_CODEC_OPUS | |
| 61 // Opus isn't defined, exit. | |
| 62 return; | |
| 63 #else | |
| 64 uint16_t frequency_hz; | |
| 65 int audio_channels; | |
| 66 int16_t test_cntr = 0; | |
| 67 | |
| 68 // Open both mono and stereo test files in 32 kHz. | |
| 69 const std::string file_name_stereo = | |
| 70 webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"); | |
| 71 const std::string file_name_mono = | |
| 72 webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); | |
| 73 frequency_hz = 32000; | |
| 74 in_file_stereo_.Open(file_name_stereo, frequency_hz, "rb"); | |
| 75 in_file_stereo_.ReadStereo(true); | |
| 76 in_file_mono_.Open(file_name_mono, frequency_hz, "rb"); | |
| 77 in_file_mono_.ReadStereo(false); | |
| 78 | |
| 79 // Create Opus encoders for mono and stereo. | |
| 80 ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_mono_encoder_, 1, 0), -1); | |
| 81 ASSERT_GT(WebRtcOpus_EncoderCreate(&opus_stereo_encoder_, 2, 1), -1); | |
| 82 | |
| 83 // Create Opus decoders for mono and stereo for stand-alone testing of Opus. | |
| 84 ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_mono_decoder_, 1), -1); | |
| 85 ASSERT_GT(WebRtcOpus_DecoderCreate(&opus_stereo_decoder_, 2), -1); | |
| 86 WebRtcOpus_DecoderInit(opus_mono_decoder_); | |
| 87 WebRtcOpus_DecoderInit(opus_stereo_decoder_); | |
| 88 | |
| 89 ASSERT_TRUE(acm_receiver_.get() != NULL); | |
| 90 EXPECT_EQ(0, acm_receiver_->InitializeReceiver()); | |
| 91 | |
| 92 // Register Opus stereo as receiving codec. | |
| 93 CodecInst opus_codec_param; | |
| 94 int codec_id = acm_receiver_->Codec("opus", 48000, 2); | |
| 95 EXPECT_EQ(0, acm_receiver_->Codec(codec_id, &opus_codec_param)); | |
| 96 payload_type_ = opus_codec_param.pltype; | |
| 97 EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param)); | |
| 98 | |
| 99 // Create and connect the channel. | |
| 100 channel_a2b_ = new TestPackStereo; | |
| 101 channel_a2b_->RegisterReceiverACM(acm_receiver_.get()); | |
| 102 | |
| 103 // | |
| 104 // Test Stereo. | |
| 105 // | |
| 106 | |
| 107 channel_a2b_->set_codec_mode(kStereo); | |
| 108 audio_channels = 2; | |
| 109 test_cntr++; | |
| 110 OpenOutFile(test_cntr); | |
| 111 | |
| 112 // Run Opus with 2.5 ms frame size. | |
| 113 Run(channel_a2b_, audio_channels, 64000, 120); | |
| 114 | |
| 115 // Run Opus with 5 ms frame size. | |
| 116 Run(channel_a2b_, audio_channels, 64000, 240); | |
| 117 | |
| 118 // Run Opus with 10 ms frame size. | |
| 119 Run(channel_a2b_, audio_channels, 64000, 480); | |
| 120 | |
| 121 // Run Opus with 20 ms frame size. | |
| 122 Run(channel_a2b_, audio_channels, 64000, 960); | |
| 123 | |
| 124 // Run Opus with 40 ms frame size. | |
| 125 Run(channel_a2b_, audio_channels, 64000, 1920); | |
| 126 | |
| 127 // Run Opus with 60 ms frame size. | |
| 128 Run(channel_a2b_, audio_channels, 64000, 2880); | |
| 129 | |
| 130 out_file_.Close(); | |
| 131 out_file_standalone_.Close(); | |
| 132 | |
| 133 // | |
| 134 // Test Opus stereo with packet-losses. | |
| 135 // | |
| 136 | |
| 137 test_cntr++; | |
| 138 OpenOutFile(test_cntr); | |
| 139 | |
| 140 // Run Opus with 20 ms frame size, 1% packet loss. | |
| 141 Run(channel_a2b_, audio_channels, 64000, 960, 1); | |
| 142 | |
| 143 // Run Opus with 20 ms frame size, 5% packet loss. | |
| 144 Run(channel_a2b_, audio_channels, 64000, 960, 5); | |
| 145 | |
| 146 // Run Opus with 20 ms frame size, 10% packet loss. | |
| 147 Run(channel_a2b_, audio_channels, 64000, 960, 10); | |
| 148 | |
| 149 out_file_.Close(); | |
| 150 out_file_standalone_.Close(); | |
| 151 | |
| 152 // | |
| 153 // Test Mono. | |
| 154 // | |
| 155 channel_a2b_->set_codec_mode(kMono); | |
| 156 audio_channels = 1; | |
| 157 test_cntr++; | |
| 158 OpenOutFile(test_cntr); | |
| 159 | |
| 160 // Register Opus mono as receiving codec. | |
| 161 opus_codec_param.channels = 1; | |
| 162 EXPECT_EQ(0, acm_receiver_->RegisterReceiveCodec(opus_codec_param)); | |
| 163 | |
| 164 // Run Opus with 2.5 ms frame size. | |
| 165 Run(channel_a2b_, audio_channels, 32000, 120); | |
| 166 | |
| 167 // Run Opus with 5 ms frame size. | |
| 168 Run(channel_a2b_, audio_channels, 32000, 240); | |
| 169 | |
| 170 // Run Opus with 10 ms frame size. | |
| 171 Run(channel_a2b_, audio_channels, 32000, 480); | |
| 172 | |
| 173 // Run Opus with 20 ms frame size. | |
| 174 Run(channel_a2b_, audio_channels, 32000, 960); | |
| 175 | |
| 176 // Run Opus with 40 ms frame size. | |
| 177 Run(channel_a2b_, audio_channels, 32000, 1920); | |
| 178 | |
| 179 // Run Opus with 60 ms frame size. | |
| 180 Run(channel_a2b_, audio_channels, 32000, 2880); | |
| 181 | |
| 182 out_file_.Close(); | |
| 183 out_file_standalone_.Close(); | |
| 184 | |
| 185 // | |
| 186 // Test Opus mono with packet-losses. | |
| 187 // | |
| 188 test_cntr++; | |
| 189 OpenOutFile(test_cntr); | |
| 190 | |
| 191 // Run Opus with 20 ms frame size, 1% packet loss. | |
| 192 Run(channel_a2b_, audio_channels, 64000, 960, 1); | |
| 193 | |
| 194 // Run Opus with 20 ms frame size, 5% packet loss. | |
| 195 Run(channel_a2b_, audio_channels, 64000, 960, 5); | |
| 196 | |
| 197 // Run Opus with 20 ms frame size, 10% packet loss. | |
| 198 Run(channel_a2b_, audio_channels, 64000, 960, 10); | |
| 199 | |
| 200 // Close the files. | |
| 201 in_file_stereo_.Close(); | |
| 202 in_file_mono_.Close(); | |
| 203 out_file_.Close(); | |
| 204 out_file_standalone_.Close(); | |
| 205 #endif | |
| 206 } | |
| 207 | |
| 208 void OpusTest::Run(TestPackStereo* channel, int channels, int bitrate, | |
| 209 int frame_length, int percent_loss) { | |
| 210 AudioFrame audio_frame; | |
| 211 int32_t out_freq_hz_b = out_file_.SamplingFrequency(); | |
| 212 const int kBufferSizeSamples = 480 * 12 * 2; // Can hold 120 ms stereo audio. | |
| 213 int16_t audio[kBufferSizeSamples]; | |
| 214 int16_t out_audio[kBufferSizeSamples]; | |
| 215 int16_t audio_type; | |
| 216 int written_samples = 0; | |
| 217 int read_samples = 0; | |
| 218 int decoded_samples = 0; | |
| 219 bool first_packet = true; | |
| 220 uint32_t start_time_stamp = 0; | |
| 221 | |
| 222 channel->reset_payload_size(); | |
| 223 counter_ = 0; | |
| 224 | |
| 225 // Set encoder rate. | |
| 226 EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_mono_encoder_, bitrate)); | |
| 227 EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_stereo_encoder_, bitrate)); | |
| 228 | |
| 229 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM) | |
| 230 // If we are on Android, iOS and/or ARM, use a lower complexity setting as | |
| 231 // default. | |
| 232 const int kOpusComplexity5 = 5; | |
| 233 EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, kOpusComplexity5)); | |
| 234 EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_stereo_encoder_, | |
| 235 kOpusComplexity5)); | |
| 236 #endif | |
| 237 | |
| 238 // Make sure the runtime is less than 60 seconds to pass Android test. | |
| 239 for (size_t audio_length = 0; audio_length < 10000; audio_length += 10) { | |
| 240 bool lost_packet = false; | |
| 241 | |
| 242 // Get 10 msec of audio. | |
| 243 if (channels == 1) { | |
| 244 if (in_file_mono_.EndOfFile()) { | |
| 245 break; | |
| 246 } | |
| 247 in_file_mono_.Read10MsData(audio_frame); | |
| 248 } else { | |
| 249 if (in_file_stereo_.EndOfFile()) { | |
| 250 break; | |
| 251 } | |
| 252 in_file_stereo_.Read10MsData(audio_frame); | |
| 253 } | |
| 254 | |
| 255 // If input audio is sampled at 32 kHz, resampling to 48 kHz is required. | |
| 256 EXPECT_EQ(480, | |
| 257 resampler_.Resample10Msec(audio_frame.data_, | |
| 258 audio_frame.sample_rate_hz_, | |
| 259 48000, | |
| 260 channels, | |
| 261 kBufferSizeSamples - written_samples, | |
| 262 &audio[written_samples])); | |
| 263 written_samples += 480 * channels; | |
| 264 | |
| 265 // Sometimes we need to loop over the audio vector to produce the right | |
| 266 // number of packets. | |
| 267 int loop_encode = (written_samples - read_samples) / | |
| 268 (channels * frame_length); | |
| 269 | |
| 270 if (loop_encode > 0) { | |
| 271 const int kMaxBytes = 1000; // Maximum number of bytes for one packet. | |
| 272 size_t bitstream_len_byte; | |
| 273 uint8_t bitstream[kMaxBytes]; | |
| 274 for (int i = 0; i < loop_encode; i++) { | |
| 275 int bitstream_len_byte_int = WebRtcOpus_Encode( | |
| 276 (channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_, | |
| 277 &audio[read_samples], frame_length, kMaxBytes, bitstream); | |
| 278 ASSERT_GE(bitstream_len_byte_int, 0); | |
| 279 bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int); | |
| 280 | |
| 281 // Simulate packet loss by setting |packet_loss_| to "true" in | |
| 282 // |percent_loss| percent of the loops. | |
| 283 // TODO(tlegrand): Move handling of loss simulation to TestPackStereo. | |
| 284 if (percent_loss > 0) { | |
| 285 if (counter_ == floor((100 / percent_loss) + 0.5)) { | |
| 286 counter_ = 0; | |
| 287 lost_packet = true; | |
| 288 channel->set_lost_packet(true); | |
| 289 } else { | |
| 290 lost_packet = false; | |
| 291 channel->set_lost_packet(false); | |
| 292 } | |
| 293 counter_++; | |
| 294 } | |
| 295 | |
| 296 // Run stand-alone Opus decoder, or decode PLC. | |
| 297 if (channels == 1) { | |
| 298 if (!lost_packet) { | |
| 299 decoded_samples += WebRtcOpus_Decode( | |
| 300 opus_mono_decoder_, bitstream, bitstream_len_byte, | |
| 301 &out_audio[decoded_samples * channels], &audio_type); | |
| 302 } else { | |
| 303 decoded_samples += WebRtcOpus_DecodePlc( | |
| 304 opus_mono_decoder_, &out_audio[decoded_samples * channels], 1); | |
| 305 } | |
| 306 } else { | |
| 307 if (!lost_packet) { | |
| 308 decoded_samples += WebRtcOpus_Decode( | |
| 309 opus_stereo_decoder_, bitstream, bitstream_len_byte, | |
| 310 &out_audio[decoded_samples * channels], &audio_type); | |
| 311 } else { | |
| 312 decoded_samples += WebRtcOpus_DecodePlc( | |
| 313 opus_stereo_decoder_, &out_audio[decoded_samples * channels], | |
| 314 1); | |
| 315 } | |
| 316 } | |
| 317 | |
| 318 // Send data to the channel. "channel" will handle the loss simulation. | |
| 319 channel->SendData(kAudioFrameSpeech, payload_type_, rtp_timestamp_, | |
| 320 bitstream, bitstream_len_byte, NULL); | |
| 321 if (first_packet) { | |
| 322 first_packet = false; | |
| 323 start_time_stamp = rtp_timestamp_; | |
| 324 } | |
| 325 rtp_timestamp_ += frame_length; | |
| 326 read_samples += frame_length * channels; | |
| 327 } | |
| 328 if (read_samples == written_samples) { | |
| 329 read_samples = 0; | |
| 330 written_samples = 0; | |
| 331 } | |
| 332 } | |
| 333 | |
| 334 // Run received side of ACM. | |
| 335 ASSERT_EQ(0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame)); | |
| 336 | |
| 337 // Write output speech to file. | |
| 338 out_file_.Write10MsData( | |
| 339 audio_frame.data_, | |
| 340 audio_frame.samples_per_channel_ * audio_frame.num_channels_); | |
| 341 | |
| 342 // Write stand-alone speech to file. | |
| 343 out_file_standalone_.Write10MsData( | |
| 344 out_audio, static_cast<size_t>(decoded_samples) * channels); | |
| 345 | |
| 346 if (audio_frame.timestamp_ > start_time_stamp) { | |
| 347 // Number of channels should be the same for both stand-alone and | |
| 348 // ACM-decoding. | |
| 349 EXPECT_EQ(audio_frame.num_channels_, channels); | |
| 350 } | |
| 351 | |
| 352 decoded_samples = 0; | |
| 353 } | |
| 354 | |
| 355 if (in_file_mono_.EndOfFile()) { | |
| 356 in_file_mono_.Rewind(); | |
| 357 } | |
| 358 if (in_file_stereo_.EndOfFile()) { | |
| 359 in_file_stereo_.Rewind(); | |
| 360 } | |
| 361 // Reset in case we ended with a lost packet. | |
| 362 channel->set_lost_packet(false); | |
| 363 } | |
| 364 | |
| 365 void OpusTest::OpenOutFile(int test_number) { | |
| 366 std::string file_name; | |
| 367 std::stringstream file_stream; | |
| 368 file_stream << webrtc::test::OutputPath() << "opustest_out_" | |
| 369 << test_number << ".pcm"; | |
| 370 file_name = file_stream.str(); | |
| 371 out_file_.Open(file_name, 48000, "wb"); | |
| 372 file_stream.str(""); | |
| 373 file_name = file_stream.str(); | |
| 374 file_stream << webrtc::test::OutputPath() << "opusstandalone_out_" | |
| 375 << test_number << ".pcm"; | |
| 376 file_name = file_stream.str(); | |
| 377 out_file_standalone_.Open(file_name, 48000, "wb"); | |
| 378 } | |
| 379 | |
| 380 } // namespace webrtc | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 1481493004:
audio_coding: remove "main" directory (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master