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| 1 /* | |
| 2 * Copyright (c) 2015 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 <algorithm> | |
| 12 #include <limits> | |
| 13 #include <list> | |
| 14 #include <memory> | |
| 15 #include <numeric> | |
| 16 #include <string> | |
| 17 #include <vector> | |
| 18 | |
| 19 #include "webrtc/base/arraysize.h" | |
| 20 #include "webrtc/base/criticalsection.h" | |
| 21 #include "webrtc/base/format_macros.h" | |
| 22 #include "webrtc/base/logging.h" | |
| 23 #include "webrtc/base/scoped_ref_ptr.h" | |
| 24 #include "webrtc/base/timeutils.h" | |
| 25 #include "webrtc/modules/audio_device/audio_device_impl.h" | |
| 26 #include "webrtc/modules/audio_device/include/audio_device.h" | |
| 27 #include "webrtc/modules/audio_device/include/mock_audio_transport.h" | |
| 28 #include "webrtc/modules/audio_device/ios/audio_device_ios.h" | |
| 29 #include "webrtc/system_wrappers/include/event_wrapper.h" | |
| 30 #include "webrtc/system_wrappers/include/sleep.h" | |
| 31 #include "webrtc/test/gmock.h" | |
| 32 #include "webrtc/test/gtest.h" | |
| 33 #include "webrtc/test/testsupport/fileutils.h" | |
| 34 | |
| 35 using std::cout; | |
| 36 using std::endl; | |
| 37 using ::testing::_; | |
| 38 using ::testing::AtLeast; | |
| 39 using ::testing::Gt; | |
| 40 using ::testing::Invoke; | |
| 41 using ::testing::NiceMock; | |
| 42 using ::testing::NotNull; | |
| 43 using ::testing::Return; | |
| 44 | |
| 45 // #define ENABLE_DEBUG_PRINTF | |
| 46 #ifdef ENABLE_DEBUG_PRINTF | |
| 47 #define PRINTD(...) fprintf(stderr, __VA_ARGS__); | |
| 48 #else | |
| 49 #define PRINTD(...) ((void)0) | |
| 50 #endif | |
| 51 #define PRINT(...) fprintf(stderr, __VA_ARGS__); | |
| 52 | |
| 53 namespace webrtc { | |
| 54 | |
| 55 // Number of callbacks (input or output) the tests waits for before we set | |
| 56 // an event indicating that the test was OK. | |
| 57 static const size_t kNumCallbacks = 10; | |
| 58 // Max amount of time we wait for an event to be set while counting callbacks. | |
| 59 static const int kTestTimeOutInMilliseconds = 10 * 1000; | |
| 60 // Number of bits per PCM audio sample. | |
| 61 static const size_t kBitsPerSample = 16; | |
| 62 // Number of bytes per PCM audio sample. | |
| 63 static const size_t kBytesPerSample = kBitsPerSample / 8; | |
| 64 // Average number of audio callbacks per second assuming 10ms packet size. | |
| 65 static const size_t kNumCallbacksPerSecond = 100; | |
| 66 // Play out a test file during this time (unit is in seconds). | |
| 67 static const int kFilePlayTimeInSec = 15; | |
| 68 // Run the full-duplex test during this time (unit is in seconds). | |
| 69 // Note that first |kNumIgnoreFirstCallbacks| are ignored. | |
| 70 static const int kFullDuplexTimeInSec = 10; | |
| 71 // Wait for the callback sequence to stabilize by ignoring this amount of the | |
| 72 // initial callbacks (avoids initial FIFO access). | |
| 73 // Only used in the RunPlayoutAndRecordingInFullDuplex test. | |
| 74 static const size_t kNumIgnoreFirstCallbacks = 50; | |
| 75 // Sets the number of impulses per second in the latency test. | |
| 76 // TODO(henrika): fine tune this setting for iOS. | |
| 77 static const int kImpulseFrequencyInHz = 1; | |
| 78 // Length of round-trip latency measurements. Number of transmitted impulses | |
| 79 // is kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1. | |
| 80 // TODO(henrika): fine tune this setting for iOS. | |
| 81 static const int kMeasureLatencyTimeInSec = 5; | |
| 82 // Utilized in round-trip latency measurements to avoid capturing noise samples. | |
| 83 // TODO(henrika): fine tune this setting for iOS. | |
| 84 static const int kImpulseThreshold = 50; | |
| 85 static const char kTag[] = "[..........] "; | |
| 86 | |
| 87 enum TransportType { | |
| 88 kPlayout = 0x1, | |
| 89 kRecording = 0x2, | |
| 90 }; | |
| 91 | |
| 92 // Interface for processing the audio stream. Real implementations can e.g. | |
| 93 // run audio in loopback, read audio from a file or perform latency | |
| 94 // measurements. | |
| 95 class AudioStreamInterface { | |
| 96 public: | |
| 97 virtual void Write(const void* source, size_t num_frames) = 0; | |
| 98 virtual void Read(void* destination, size_t num_frames) = 0; | |
| 99 | |
| 100 protected: | |
| 101 virtual ~AudioStreamInterface() {} | |
| 102 }; | |
| 103 | |
| 104 // Reads audio samples from a PCM file where the file is stored in memory at | |
| 105 // construction. | |
| 106 class FileAudioStream : public AudioStreamInterface { | |
| 107 public: | |
| 108 FileAudioStream(size_t num_callbacks, | |
| 109 const std::string& file_name, | |
| 110 int sample_rate) | |
| 111 : file_size_in_bytes_(0), sample_rate_(sample_rate), file_pos_(0) { | |
| 112 file_size_in_bytes_ = test::GetFileSize(file_name); | |
| 113 sample_rate_ = sample_rate; | |
| 114 EXPECT_GE(file_size_in_callbacks(), num_callbacks) | |
| 115 << "Size of test file is not large enough to last during the test."; | |
| 116 const size_t num_16bit_samples = | |
| 117 test::GetFileSize(file_name) / kBytesPerSample; | |
| 118 file_.reset(new int16_t[num_16bit_samples]); | |
| 119 FILE* audio_file = fopen(file_name.c_str(), "rb"); | |
| 120 EXPECT_NE(audio_file, nullptr); | |
| 121 size_t num_samples_read = | |
| 122 fread(file_.get(), sizeof(int16_t), num_16bit_samples, audio_file); | |
| 123 EXPECT_EQ(num_samples_read, num_16bit_samples); | |
| 124 fclose(audio_file); | |
| 125 } | |
| 126 | |
| 127 // AudioStreamInterface::Write() is not implemented. | |
| 128 void Write(const void* source, size_t num_frames) override {} | |
| 129 | |
| 130 // Read samples from file stored in memory (at construction) and copy | |
| 131 // |num_frames| (<=> 10ms) to the |destination| byte buffer. | |
| 132 void Read(void* destination, size_t num_frames) override { | |
| 133 memcpy(destination, static_cast<int16_t*>(&file_[file_pos_]), | |
| 134 num_frames * sizeof(int16_t)); | |
| 135 file_pos_ += num_frames; | |
| 136 } | |
| 137 | |
| 138 int file_size_in_seconds() const { | |
| 139 return static_cast<int>( | |
| 140 file_size_in_bytes_ / (kBytesPerSample * sample_rate_)); | |
| 141 } | |
| 142 size_t file_size_in_callbacks() const { | |
| 143 return file_size_in_seconds() * kNumCallbacksPerSecond; | |
| 144 } | |
| 145 | |
| 146 private: | |
| 147 size_t file_size_in_bytes_; | |
| 148 int sample_rate_; | |
| 149 std::unique_ptr<int16_t[]> file_; | |
| 150 size_t file_pos_; | |
| 151 }; | |
| 152 | |
| 153 // Simple first in first out (FIFO) class that wraps a list of 16-bit audio | |
| 154 // buffers of fixed size and allows Write and Read operations. The idea is to | |
| 155 // store recorded audio buffers (using Write) and then read (using Read) these | |
| 156 // stored buffers with as short delay as possible when the audio layer needs | |
| 157 // data to play out. The number of buffers in the FIFO will stabilize under | |
| 158 // normal conditions since there will be a balance between Write and Read calls. | |
| 159 // The container is a std::list container and access is protected with a lock | |
| 160 // since both sides (playout and recording) are driven by its own thread. | |
| 161 class FifoAudioStream : public AudioStreamInterface { | |
| 162 public: | |
| 163 explicit FifoAudioStream(size_t frames_per_buffer) | |
| 164 : frames_per_buffer_(frames_per_buffer), | |
| 165 bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)), | |
| 166 fifo_(new AudioBufferList), | |
| 167 largest_size_(0), | |
| 168 total_written_elements_(0), | |
| 169 write_count_(0) { | |
| 170 EXPECT_NE(fifo_.get(), nullptr); | |
| 171 } | |
| 172 | |
| 173 ~FifoAudioStream() { Flush(); } | |
| 174 | |
| 175 // Allocate new memory, copy |num_frames| samples from |source| into memory | |
| 176 // and add pointer to the memory location to end of the list. | |
| 177 // Increases the size of the FIFO by one element. | |
| 178 void Write(const void* source, size_t num_frames) override { | |
| 179 ASSERT_EQ(num_frames, frames_per_buffer_); | |
| 180 PRINTD("+"); | |
| 181 if (write_count_++ < kNumIgnoreFirstCallbacks) { | |
| 182 return; | |
| 183 } | |
| 184 int16_t* memory = new int16_t[frames_per_buffer_]; | |
| 185 memcpy(static_cast<int16_t*>(&memory[0]), source, bytes_per_buffer_); | |
| 186 rtc::CritScope lock(&lock_); | |
| 187 fifo_->push_back(memory); | |
| 188 const size_t size = fifo_->size(); | |
| 189 if (size > largest_size_) { | |
| 190 largest_size_ = size; | |
| 191 PRINTD("(%" PRIuS ")", largest_size_); | |
| 192 } | |
| 193 total_written_elements_ += size; | |
| 194 } | |
| 195 | |
| 196 // Read pointer to data buffer from front of list, copy |num_frames| of stored | |
| 197 // data into |destination| and delete the utilized memory allocation. | |
| 198 // Decreases the size of the FIFO by one element. | |
| 199 void Read(void* destination, size_t num_frames) override { | |
| 200 ASSERT_EQ(num_frames, frames_per_buffer_); | |
| 201 PRINTD("-"); | |
| 202 rtc::CritScope lock(&lock_); | |
| 203 if (fifo_->empty()) { | |
| 204 memset(destination, 0, bytes_per_buffer_); | |
| 205 } else { | |
| 206 int16_t* memory = fifo_->front(); | |
| 207 fifo_->pop_front(); | |
| 208 memcpy(destination, static_cast<int16_t*>(&memory[0]), bytes_per_buffer_); | |
| 209 delete memory; | |
| 210 } | |
| 211 } | |
| 212 | |
| 213 size_t size() const { return fifo_->size(); } | |
| 214 | |
| 215 size_t largest_size() const { return largest_size_; } | |
| 216 | |
| 217 size_t average_size() const { | |
| 218 return (total_written_elements_ == 0) | |
| 219 ? 0.0 | |
| 220 : 0.5 + | |
| 221 static_cast<float>(total_written_elements_) / | |
| 222 (write_count_ - kNumIgnoreFirstCallbacks); | |
| 223 } | |
| 224 | |
| 225 private: | |
| 226 void Flush() { | |
| 227 for (auto it = fifo_->begin(); it != fifo_->end(); ++it) { | |
| 228 delete *it; | |
| 229 } | |
| 230 fifo_->clear(); | |
| 231 } | |
| 232 | |
| 233 using AudioBufferList = std::list<int16_t*>; | |
| 234 rtc::CriticalSection lock_; | |
| 235 const size_t frames_per_buffer_; | |
| 236 const size_t bytes_per_buffer_; | |
| 237 std::unique_ptr<AudioBufferList> fifo_; | |
| 238 size_t largest_size_; | |
| 239 size_t total_written_elements_; | |
| 240 size_t write_count_; | |
| 241 }; | |
| 242 | |
| 243 // Inserts periodic impulses and measures the latency between the time of | |
| 244 // transmission and time of receiving the same impulse. | |
| 245 // Usage requires a special hardware called Audio Loopback Dongle. | |
| 246 // See http://source.android.com/devices/audio/loopback.html for details. | |
| 247 class LatencyMeasuringAudioStream : public AudioStreamInterface { | |
| 248 public: | |
| 249 explicit LatencyMeasuringAudioStream(size_t frames_per_buffer) | |
| 250 : frames_per_buffer_(frames_per_buffer), | |
| 251 bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)), | |
| 252 play_count_(0), | |
| 253 rec_count_(0), | |
| 254 pulse_time_(0) {} | |
| 255 | |
| 256 // Insert periodic impulses in first two samples of |destination|. | |
| 257 void Read(void* destination, size_t num_frames) override { | |
| 258 ASSERT_EQ(num_frames, frames_per_buffer_); | |
| 259 if (play_count_ == 0) { | |
| 260 PRINT("["); | |
| 261 } | |
| 262 play_count_++; | |
| 263 memset(destination, 0, bytes_per_buffer_); | |
| 264 if (play_count_ % (kNumCallbacksPerSecond / kImpulseFrequencyInHz) == 0) { | |
| 265 if (pulse_time_ == 0) { | |
| 266 pulse_time_ = rtc::TimeMillis(); | |
| 267 } | |
| 268 PRINT("."); | |
| 269 const int16_t impulse = std::numeric_limits<int16_t>::max(); | |
| 270 int16_t* ptr16 = static_cast<int16_t*>(destination); | |
| 271 for (size_t i = 0; i < 2; ++i) { | |
| 272 ptr16[i] = impulse; | |
| 273 } | |
| 274 } | |
| 275 } | |
| 276 | |
| 277 // Detect received impulses in |source|, derive time between transmission and | |
| 278 // detection and add the calculated delay to list of latencies. | |
| 279 void Write(const void* source, size_t num_frames) override { | |
| 280 ASSERT_EQ(num_frames, frames_per_buffer_); | |
| 281 rec_count_++; | |
| 282 if (pulse_time_ == 0) { | |
| 283 // Avoid detection of new impulse response until a new impulse has | |
| 284 // been transmitted (sets |pulse_time_| to value larger than zero). | |
| 285 return; | |
| 286 } | |
| 287 const int16_t* ptr16 = static_cast<const int16_t*>(source); | |
| 288 std::vector<int16_t> vec(ptr16, ptr16 + num_frames); | |
| 289 // Find max value in the audio buffer. | |
| 290 int max = *std::max_element(vec.begin(), vec.end()); | |
| 291 // Find index (element position in vector) of the max element. | |
| 292 int index_of_max = | |
| 293 std::distance(vec.begin(), std::find(vec.begin(), vec.end(), max)); | |
| 294 if (max > kImpulseThreshold) { | |
| 295 PRINTD("(%d,%d)", max, index_of_max); | |
| 296 int64_t now_time = rtc::TimeMillis(); | |
| 297 int extra_delay = IndexToMilliseconds(static_cast<double>(index_of_max)); | |
| 298 PRINTD("[%d]", static_cast<int>(now_time - pulse_time_)); | |
| 299 PRINTD("[%d]", extra_delay); | |
| 300 // Total latency is the difference between transmit time and detection | |
| 301 // tome plus the extra delay within the buffer in which we detected the | |
| 302 // received impulse. It is transmitted at sample 0 but can be received | |
| 303 // at sample N where N > 0. The term |extra_delay| accounts for N and it | |
| 304 // is a value between 0 and 10ms. | |
| 305 latencies_.push_back(now_time - pulse_time_ + extra_delay); | |
| 306 pulse_time_ = 0; | |
| 307 } else { | |
| 308 PRINTD("-"); | |
| 309 } | |
| 310 } | |
| 311 | |
| 312 size_t num_latency_values() const { return latencies_.size(); } | |
| 313 | |
| 314 int min_latency() const { | |
| 315 if (latencies_.empty()) | |
| 316 return 0; | |
| 317 return *std::min_element(latencies_.begin(), latencies_.end()); | |
| 318 } | |
| 319 | |
| 320 int max_latency() const { | |
| 321 if (latencies_.empty()) | |
| 322 return 0; | |
| 323 return *std::max_element(latencies_.begin(), latencies_.end()); | |
| 324 } | |
| 325 | |
| 326 int average_latency() const { | |
| 327 if (latencies_.empty()) | |
| 328 return 0; | |
| 329 return 0.5 + | |
| 330 static_cast<double>( | |
| 331 std::accumulate(latencies_.begin(), latencies_.end(), 0)) / | |
| 332 latencies_.size(); | |
| 333 } | |
| 334 | |
| 335 void PrintResults() const { | |
| 336 PRINT("] "); | |
| 337 for (auto it = latencies_.begin(); it != latencies_.end(); ++it) { | |
| 338 PRINT("%d ", *it); | |
| 339 } | |
| 340 PRINT("\n"); | |
| 341 PRINT("%s[min, max, avg]=[%d, %d, %d] ms\n", kTag, min_latency(), | |
| 342 max_latency(), average_latency()); | |
| 343 } | |
| 344 | |
| 345 int IndexToMilliseconds(double index) const { | |
| 346 return 10.0 * (index / frames_per_buffer_) + 0.5; | |
| 347 } | |
| 348 | |
| 349 private: | |
| 350 const size_t frames_per_buffer_; | |
| 351 const size_t bytes_per_buffer_; | |
| 352 size_t play_count_; | |
| 353 size_t rec_count_; | |
| 354 int64_t pulse_time_; | |
| 355 std::vector<int> latencies_; | |
| 356 }; | |
| 357 // Mocks the AudioTransport object and proxies actions for the two callbacks | |
| 358 // (RecordedDataIsAvailable and NeedMorePlayData) to different implementations | |
| 359 // of AudioStreamInterface. | |
| 360 class MockAudioTransportIOS : public test::MockAudioTransport { | |
| 361 public: | |
| 362 explicit MockAudioTransportIOS(int type) | |
| 363 : num_callbacks_(0), | |
| 364 type_(type), | |
| 365 play_count_(0), | |
| 366 rec_count_(0), | |
| 367 audio_stream_(nullptr) {} | |
| 368 | |
| 369 virtual ~MockAudioTransportIOS() {} | |
| 370 | |
| 371 // Set default actions of the mock object. We are delegating to fake | |
| 372 // implementations (of AudioStreamInterface) here. | |
| 373 void HandleCallbacks(EventWrapper* test_is_done, | |
| 374 AudioStreamInterface* audio_stream, | |
| 375 size_t num_callbacks) { | |
| 376 test_is_done_ = test_is_done; | |
| 377 audio_stream_ = audio_stream; | |
| 378 num_callbacks_ = num_callbacks; | |
| 379 if (play_mode()) { | |
| 380 ON_CALL(*this, NeedMorePlayData(_, _, _, _, _, _, _, _)) | |
| 381 .WillByDefault( | |
| 382 Invoke(this, &MockAudioTransportIOS::RealNeedMorePlayData)); | |
| 383 } | |
| 384 if (rec_mode()) { | |
| 385 ON_CALL(*this, RecordedDataIsAvailable(_, _, _, _, _, _, _, _, _, _)) | |
| 386 .WillByDefault(Invoke( | |
| 387 this, &MockAudioTransportIOS::RealRecordedDataIsAvailable)); | |
| 388 } | |
| 389 } | |
| 390 | |
| 391 int32_t RealRecordedDataIsAvailable(const void* audioSamples, | |
| 392 const size_t nSamples, | |
| 393 const size_t nBytesPerSample, | |
| 394 const size_t nChannels, | |
| 395 const uint32_t samplesPerSec, | |
| 396 const uint32_t totalDelayMS, | |
| 397 const int32_t clockDrift, | |
| 398 const uint32_t currentMicLevel, | |
| 399 const bool keyPressed, | |
| 400 uint32_t& newMicLevel) { | |
| 401 EXPECT_TRUE(rec_mode()) << "No test is expecting these callbacks."; | |
| 402 rec_count_++; | |
| 403 // Process the recorded audio stream if an AudioStreamInterface | |
| 404 // implementation exists. | |
| 405 if (audio_stream_) { | |
| 406 audio_stream_->Write(audioSamples, nSamples); | |
| 407 } | |
| 408 if (ReceivedEnoughCallbacks()) { | |
| 409 if (test_is_done_) { | |
| 410 test_is_done_->Set(); | |
| 411 } | |
| 412 } | |
| 413 return 0; | |
| 414 } | |
| 415 | |
| 416 int32_t RealNeedMorePlayData(const size_t nSamples, | |
| 417 const size_t nBytesPerSample, | |
| 418 const size_t nChannels, | |
| 419 const uint32_t samplesPerSec, | |
| 420 void* audioSamples, | |
| 421 size_t& nSamplesOut, | |
| 422 int64_t* elapsed_time_ms, | |
| 423 int64_t* ntp_time_ms) { | |
| 424 EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks."; | |
| 425 play_count_++; | |
| 426 nSamplesOut = nSamples; | |
| 427 // Read (possibly processed) audio stream samples to be played out if an | |
| 428 // AudioStreamInterface implementation exists. | |
| 429 if (audio_stream_) { | |
| 430 audio_stream_->Read(audioSamples, nSamples); | |
| 431 } | |
| 432 if (ReceivedEnoughCallbacks()) { | |
| 433 if (test_is_done_) { | |
| 434 test_is_done_->Set(); | |
| 435 } | |
| 436 } | |
| 437 return 0; | |
| 438 } | |
| 439 | |
| 440 bool ReceivedEnoughCallbacks() { | |
| 441 bool recording_done = false; | |
| 442 if (rec_mode()) | |
| 443 recording_done = rec_count_ >= num_callbacks_; | |
| 444 else | |
| 445 recording_done = true; | |
| 446 | |
| 447 bool playout_done = false; | |
| 448 if (play_mode()) | |
| 449 playout_done = play_count_ >= num_callbacks_; | |
| 450 else | |
| 451 playout_done = true; | |
| 452 | |
| 453 return recording_done && playout_done; | |
| 454 } | |
| 455 | |
| 456 bool play_mode() const { return type_ & kPlayout; } | |
| 457 bool rec_mode() const { return type_ & kRecording; } | |
| 458 | |
| 459 private: | |
| 460 EventWrapper* test_is_done_; | |
| 461 size_t num_callbacks_; | |
| 462 int type_; | |
| 463 size_t play_count_; | |
| 464 size_t rec_count_; | |
| 465 AudioStreamInterface* audio_stream_; | |
| 466 }; | |
| 467 | |
| 468 // AudioDeviceTest test fixture. | |
| 469 class AudioDeviceTest : public ::testing::Test { | |
| 470 protected: | |
| 471 AudioDeviceTest() : test_is_done_(EventWrapper::Create()) { | |
| 472 old_sev_ = rtc::LogMessage::GetLogToDebug(); | |
| 473 // Set suitable logging level here. Change to rtc::LS_INFO for more verbose | |
| 474 // output. See webrtc/base/logging.h for complete list of options. | |
| 475 rtc::LogMessage::LogToDebug(rtc::LS_INFO); | |
| 476 // Add extra logging fields here (timestamps and thread id). | |
| 477 // rtc::LogMessage::LogTimestamps(); | |
| 478 rtc::LogMessage::LogThreads(); | |
| 479 // Creates an audio device using a default audio layer. | |
| 480 audio_device_ = CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio); | |
| 481 EXPECT_NE(audio_device_.get(), nullptr); | |
| 482 EXPECT_EQ(0, audio_device_->Init()); | |
| 483 EXPECT_EQ(0, | |
| 484 audio_device()->GetPlayoutAudioParameters(&playout_parameters_)); | |
| 485 EXPECT_EQ(0, audio_device()->GetRecordAudioParameters(&record_parameters_)); | |
| 486 } | |
| 487 virtual ~AudioDeviceTest() { | |
| 488 EXPECT_EQ(0, audio_device_->Terminate()); | |
| 489 rtc::LogMessage::LogToDebug(old_sev_); | |
| 490 } | |
| 491 | |
| 492 int playout_sample_rate() const { return playout_parameters_.sample_rate(); } | |
| 493 int record_sample_rate() const { return record_parameters_.sample_rate(); } | |
| 494 int playout_channels() const { return playout_parameters_.channels(); } | |
| 495 int record_channels() const { return record_parameters_.channels(); } | |
| 496 size_t playout_frames_per_10ms_buffer() const { | |
| 497 return playout_parameters_.frames_per_10ms_buffer(); | |
| 498 } | |
| 499 size_t record_frames_per_10ms_buffer() const { | |
| 500 return record_parameters_.frames_per_10ms_buffer(); | |
| 501 } | |
| 502 | |
| 503 rtc::scoped_refptr<AudioDeviceModule> audio_device() const { | |
| 504 return audio_device_; | |
| 505 } | |
| 506 | |
| 507 AudioDeviceModuleImpl* audio_device_impl() const { | |
| 508 return static_cast<AudioDeviceModuleImpl*>(audio_device_.get()); | |
| 509 } | |
| 510 | |
| 511 AudioDeviceBuffer* audio_device_buffer() const { | |
| 512 return audio_device_impl()->GetAudioDeviceBuffer(); | |
| 513 } | |
| 514 | |
| 515 rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice( | |
| 516 AudioDeviceModule::AudioLayer audio_layer) { | |
| 517 rtc::scoped_refptr<AudioDeviceModule> module( | |
| 518 AudioDeviceModule::Create(0, audio_layer)); | |
| 519 return module; | |
| 520 } | |
| 521 | |
| 522 // Returns file name relative to the resource root given a sample rate. | |
| 523 std::string GetFileName(int sample_rate) { | |
| 524 EXPECT_TRUE(sample_rate == 48000 || sample_rate == 44100 || | |
| 525 sample_rate == 16000); | |
| 526 char fname[64]; | |
| 527 snprintf(fname, sizeof(fname), "audio_device/audio_short%d", | |
| 528 sample_rate / 1000); | |
| 529 std::string file_name(webrtc::test::ResourcePath(fname, "pcm")); | |
| 530 EXPECT_TRUE(test::FileExists(file_name)); | |
| 531 #ifdef ENABLE_DEBUG_PRINTF | |
| 532 PRINTD("file name: %s\n", file_name.c_str()); | |
| 533 const size_t bytes = test::GetFileSize(file_name); | |
| 534 PRINTD("file size: %" PRIuS " [bytes]\n", bytes); | |
| 535 PRINTD("file size: %" PRIuS " [samples]\n", bytes / kBytesPerSample); | |
| 536 const int seconds = | |
| 537 static_cast<int>(bytes / (sample_rate * kBytesPerSample)); | |
| 538 PRINTD("file size: %d [secs]\n", seconds); | |
| 539 PRINTD("file size: %" PRIuS " [callbacks]\n", | |
| 540 seconds * kNumCallbacksPerSecond); | |
| 541 #endif | |
| 542 return file_name; | |
| 543 } | |
| 544 | |
| 545 void StartPlayout() { | |
| 546 EXPECT_FALSE(audio_device()->Playing()); | |
| 547 EXPECT_EQ(0, audio_device()->InitPlayout()); | |
| 548 EXPECT_TRUE(audio_device()->PlayoutIsInitialized()); | |
| 549 EXPECT_EQ(0, audio_device()->StartPlayout()); | |
| 550 EXPECT_TRUE(audio_device()->Playing()); | |
| 551 } | |
| 552 | |
| 553 void StopPlayout() { | |
| 554 EXPECT_EQ(0, audio_device()->StopPlayout()); | |
| 555 EXPECT_FALSE(audio_device()->Playing()); | |
| 556 } | |
| 557 | |
| 558 void StartRecording() { | |
| 559 EXPECT_FALSE(audio_device()->Recording()); | |
| 560 EXPECT_EQ(0, audio_device()->InitRecording()); | |
| 561 EXPECT_TRUE(audio_device()->RecordingIsInitialized()); | |
| 562 EXPECT_EQ(0, audio_device()->StartRecording()); | |
| 563 EXPECT_TRUE(audio_device()->Recording()); | |
| 564 } | |
| 565 | |
| 566 void StopRecording() { | |
| 567 EXPECT_EQ(0, audio_device()->StopRecording()); | |
| 568 EXPECT_FALSE(audio_device()->Recording()); | |
| 569 } | |
| 570 | |
| 571 std::unique_ptr<EventWrapper> test_is_done_; | |
| 572 rtc::scoped_refptr<AudioDeviceModule> audio_device_; | |
| 573 AudioParameters playout_parameters_; | |
| 574 AudioParameters record_parameters_; | |
| 575 rtc::LoggingSeverity old_sev_; | |
| 576 }; | |
| 577 | |
| 578 TEST_F(AudioDeviceTest, ConstructDestruct) { | |
| 579 // Using the test fixture to create and destruct the audio device module. | |
| 580 } | |
| 581 | |
| 582 TEST_F(AudioDeviceTest, InitTerminate) { | |
| 583 // Initialization is part of the test fixture. | |
| 584 EXPECT_TRUE(audio_device()->Initialized()); | |
| 585 EXPECT_EQ(0, audio_device()->Terminate()); | |
| 586 EXPECT_FALSE(audio_device()->Initialized()); | |
| 587 } | |
| 588 | |
| 589 // Tests that playout can be initiated, started and stopped. No audio callback | |
| 590 // is registered in this test. | |
| 591 // Failing when running on real iOS devices: bugs.webrtc.org/6889. | |
| 592 TEST_F(AudioDeviceTest, DISABLED_StartStopPlayout) { | |
| 593 StartPlayout(); | |
| 594 StopPlayout(); | |
| 595 StartPlayout(); | |
| 596 StopPlayout(); | |
| 597 } | |
| 598 | |
| 599 // Tests that recording can be initiated, started and stopped. No audio callback | |
| 600 // is registered in this test. | |
| 601 TEST_F(AudioDeviceTest, StartStopRecording) { | |
| 602 StartRecording(); | |
| 603 StopRecording(); | |
| 604 StartRecording(); | |
| 605 StopRecording(); | |
| 606 } | |
| 607 | |
| 608 // Verify that calling StopPlayout() will leave us in an uninitialized state | |
| 609 // which will require a new call to InitPlayout(). This test does not call | |
| 610 // StartPlayout() while being uninitialized since doing so will hit a | |
| 611 // RTC_DCHECK. | |
| 612 TEST_F(AudioDeviceTest, StopPlayoutRequiresInitToRestart) { | |
| 613 EXPECT_EQ(0, audio_device()->InitPlayout()); | |
| 614 EXPECT_EQ(0, audio_device()->StartPlayout()); | |
| 615 EXPECT_EQ(0, audio_device()->StopPlayout()); | |
| 616 EXPECT_FALSE(audio_device()->PlayoutIsInitialized()); | |
| 617 } | |
| 618 | |
| 619 // Verify that we can create two ADMs and start playing on the second ADM. | |
| 620 // Only the first active instance shall activate an audio session and the | |
| 621 // last active instance shall deactivate the audio session. The test does not | |
| 622 // explicitly verify correct audio session calls but instead focuses on | |
| 623 // ensuring that audio starts for both ADMs. | |
| 624 | |
| 625 // Failing when running on real iOS devices: bugs.webrtc.org/6889. | |
| 626 TEST_F(AudioDeviceTest, DISABLED_StartPlayoutOnTwoInstances) { | |
| 627 // Create and initialize a second/extra ADM instance. The default ADM is | |
| 628 // created by the test harness. | |
| 629 rtc::scoped_refptr<AudioDeviceModule> second_audio_device = | |
| 630 CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio); | |
| 631 EXPECT_NE(second_audio_device.get(), nullptr); | |
| 632 EXPECT_EQ(0, second_audio_device->Init()); | |
| 633 | |
| 634 // Start playout for the default ADM but don't wait here. Instead use the | |
| 635 // upcoming second stream for that. We set the same expectation on number | |
| 636 // of callbacks as for the second stream. | |
| 637 NiceMock<MockAudioTransportIOS> mock(kPlayout); | |
| 638 mock.HandleCallbacks(nullptr, nullptr, 0); | |
| 639 EXPECT_CALL( | |
| 640 mock, NeedMorePlayData(playout_frames_per_10ms_buffer(), kBytesPerSample, | |
| 641 playout_channels(), playout_sample_rate(), | |
| 642 NotNull(), _, _, _)) | |
| 643 .Times(AtLeast(kNumCallbacks)); | |
| 644 EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); | |
| 645 StartPlayout(); | |
| 646 | |
| 647 // Initialize playout for the second ADM. If all is OK, the second ADM shall | |
| 648 // reuse the audio session activated when the first ADM started playing. | |
| 649 // This call will also ensure that we avoid a problem related to initializing | |
| 650 // two different audio unit instances back to back (see webrtc:5166 for | |
| 651 // details). | |
| 652 EXPECT_EQ(0, second_audio_device->InitPlayout()); | |
| 653 EXPECT_TRUE(second_audio_device->PlayoutIsInitialized()); | |
| 654 | |
| 655 // Start playout for the second ADM and verify that it starts as intended. | |
| 656 // Passing this test ensures that initialization of the second audio unit | |
| 657 // has been done successfully and that there is no conflict with the already | |
| 658 // playing first ADM. | |
| 659 MockAudioTransportIOS mock2(kPlayout); | |
| 660 mock2.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks); | |
| 661 EXPECT_CALL( | |
| 662 mock2, NeedMorePlayData(playout_frames_per_10ms_buffer(), kBytesPerSample, | |
| 663 playout_channels(), playout_sample_rate(), | |
| 664 NotNull(), _, _, _)) | |
| 665 .Times(AtLeast(kNumCallbacks)); | |
| 666 EXPECT_EQ(0, second_audio_device->RegisterAudioCallback(&mock2)); | |
| 667 EXPECT_EQ(0, second_audio_device->StartPlayout()); | |
| 668 EXPECT_TRUE(second_audio_device->Playing()); | |
| 669 test_is_done_->Wait(kTestTimeOutInMilliseconds); | |
| 670 EXPECT_EQ(0, second_audio_device->StopPlayout()); | |
| 671 EXPECT_FALSE(second_audio_device->Playing()); | |
| 672 EXPECT_FALSE(second_audio_device->PlayoutIsInitialized()); | |
| 673 | |
| 674 EXPECT_EQ(0, second_audio_device->Terminate()); | |
| 675 } | |
| 676 | |
| 677 // Start playout and verify that the native audio layer starts asking for real | |
| 678 // audio samples to play out using the NeedMorePlayData callback. | |
| 679 TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) { | |
| 680 MockAudioTransportIOS mock(kPlayout); | |
| 681 mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks); | |
| 682 EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(), | |
| 683 kBytesPerSample, playout_channels(), | |
| 684 playout_sample_rate(), NotNull(), _, _, _)) | |
| 685 .Times(AtLeast(kNumCallbacks)); | |
| 686 EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); | |
| 687 StartPlayout(); | |
| 688 test_is_done_->Wait(kTestTimeOutInMilliseconds); | |
| 689 StopPlayout(); | |
| 690 } | |
| 691 | |
| 692 // Start recording and verify that the native audio layer starts feeding real | |
| 693 // audio samples via the RecordedDataIsAvailable callback. | |
| 694 TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) { | |
| 695 MockAudioTransportIOS mock(kRecording); | |
| 696 mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks); | |
| 697 EXPECT_CALL(mock, | |
| 698 RecordedDataIsAvailable( | |
| 699 NotNull(), record_frames_per_10ms_buffer(), kBytesPerSample, | |
| 700 record_channels(), record_sample_rate(), | |
| 701 _, // TODO(henrika): fix delay | |
| 702 0, 0, false, _)).Times(AtLeast(kNumCallbacks)); | |
| 703 | |
| 704 EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); | |
| 705 StartRecording(); | |
| 706 test_is_done_->Wait(kTestTimeOutInMilliseconds); | |
| 707 StopRecording(); | |
| 708 } | |
| 709 | |
| 710 // Start playout and recording (full-duplex audio) and verify that audio is | |
| 711 // active in both directions. | |
| 712 TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) { | |
| 713 MockAudioTransportIOS mock(kPlayout | kRecording); | |
| 714 mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks); | |
| 715 EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(), | |
| 716 kBytesPerSample, playout_channels(), | |
| 717 playout_sample_rate(), NotNull(), _, _, _)) | |
| 718 .Times(AtLeast(kNumCallbacks)); | |
| 719 EXPECT_CALL(mock, | |
| 720 RecordedDataIsAvailable( | |
| 721 NotNull(), record_frames_per_10ms_buffer(), kBytesPerSample, | |
| 722 record_channels(), record_sample_rate(), | |
| 723 _, // TODO(henrika): fix delay | |
| 724 0, 0, false, _)).Times(AtLeast(kNumCallbacks)); | |
| 725 EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); | |
| 726 StartPlayout(); | |
| 727 StartRecording(); | |
| 728 test_is_done_->Wait(kTestTimeOutInMilliseconds); | |
| 729 StopRecording(); | |
| 730 StopPlayout(); | |
| 731 } | |
| 732 | |
| 733 // Start playout and read audio from an external PCM file when the audio layer | |
| 734 // asks for data to play out. Real audio is played out in this test but it does | |
| 735 // not contain any explicit verification that the audio quality is perfect. | |
| 736 TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) { | |
| 737 // TODO(henrika): extend test when mono output is supported. | |
| 738 EXPECT_EQ(1, playout_channels()); | |
| 739 NiceMock<MockAudioTransportIOS> mock(kPlayout); | |
| 740 const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond; | |
| 741 std::string file_name = GetFileName(playout_sample_rate()); | |
| 742 std::unique_ptr<FileAudioStream> file_audio_stream( | |
| 743 new FileAudioStream(num_callbacks, file_name, playout_sample_rate())); | |
| 744 mock.HandleCallbacks(test_is_done_.get(), file_audio_stream.get(), | |
| 745 num_callbacks); | |
| 746 // SetMaxPlayoutVolume(); | |
| 747 EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); | |
| 748 StartPlayout(); | |
| 749 test_is_done_->Wait(kTestTimeOutInMilliseconds); | |
| 750 StopPlayout(); | |
| 751 } | |
| 752 | |
| 753 TEST_F(AudioDeviceTest, Devices) { | |
| 754 // Device enumeration is not supported. Verify fixed values only. | |
| 755 EXPECT_EQ(1, audio_device()->PlayoutDevices()); | |
| 756 EXPECT_EQ(1, audio_device()->RecordingDevices()); | |
| 757 } | |
| 758 | |
| 759 // Start playout and recording and store recorded data in an intermediate FIFO | |
| 760 // buffer from which the playout side then reads its samples in the same order | |
| 761 // as they were stored. Under ideal circumstances, a callback sequence would | |
| 762 // look like: ...+-+-+-+-+-+-+-..., where '+' means 'packet recorded' and '-' | |
| 763 // means 'packet played'. Under such conditions, the FIFO would only contain | |
| 764 // one packet on average. However, under more realistic conditions, the size | |
| 765 // of the FIFO will vary more due to an unbalance between the two sides. | |
| 766 // This test tries to verify that the device maintains a balanced callback- | |
| 767 // sequence by running in loopback for ten seconds while measuring the size | |
| 768 // (max and average) of the FIFO. The size of the FIFO is increased by the | |
| 769 // recording side and decreased by the playout side. | |
| 770 // TODO(henrika): tune the final test parameters after running tests on several | |
| 771 // different devices. | |
| 772 TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) { | |
| 773 EXPECT_EQ(record_channels(), playout_channels()); | |
| 774 EXPECT_EQ(record_sample_rate(), playout_sample_rate()); | |
| 775 NiceMock<MockAudioTransportIOS> mock(kPlayout | kRecording); | |
| 776 std::unique_ptr<FifoAudioStream> fifo_audio_stream( | |
| 777 new FifoAudioStream(playout_frames_per_10ms_buffer())); | |
| 778 mock.HandleCallbacks(test_is_done_.get(), fifo_audio_stream.get(), | |
| 779 kFullDuplexTimeInSec * kNumCallbacksPerSecond); | |
| 780 // SetMaxPlayoutVolume(); | |
| 781 EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); | |
| 782 StartRecording(); | |
| 783 StartPlayout(); | |
| 784 test_is_done_->Wait( | |
| 785 std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec)); | |
| 786 StopPlayout(); | |
| 787 StopRecording(); | |
| 788 EXPECT_LE(fifo_audio_stream->average_size(), 10u); | |
| 789 EXPECT_LE(fifo_audio_stream->largest_size(), 20u); | |
| 790 } | |
| 791 | |
| 792 // Measures loopback latency and reports the min, max and average values for | |
| 793 // a full duplex audio session. | |
| 794 // The latency is measured like so: | |
| 795 // - Insert impulses periodically on the output side. | |
| 796 // - Detect the impulses on the input side. | |
| 797 // - Measure the time difference between the transmit time and receive time. | |
| 798 // - Store time differences in a vector and calculate min, max and average. | |
| 799 // This test requires a special hardware called Audio Loopback Dongle. | |
| 800 // See http://source.android.com/devices/audio/loopback.html for details. | |
| 801 TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) { | |
| 802 EXPECT_EQ(record_channels(), playout_channels()); | |
| 803 EXPECT_EQ(record_sample_rate(), playout_sample_rate()); | |
| 804 NiceMock<MockAudioTransportIOS> mock(kPlayout | kRecording); | |
| 805 std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream( | |
| 806 new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer())); | |
| 807 mock.HandleCallbacks(test_is_done_.get(), latency_audio_stream.get(), | |
| 808 kMeasureLatencyTimeInSec * kNumCallbacksPerSecond); | |
| 809 EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock)); | |
| 810 // SetMaxPlayoutVolume(); | |
| 811 // DisableBuiltInAECIfAvailable(); | |
| 812 StartRecording(); | |
| 813 StartPlayout(); | |
| 814 test_is_done_->Wait( | |
| 815 std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec)); | |
| 816 StopPlayout(); | |
| 817 StopRecording(); | |
| 818 // Verify that the correct number of transmitted impulses are detected. | |
| 819 EXPECT_EQ(latency_audio_stream->num_latency_values(), | |
| 820 static_cast<size_t>( | |
| 821 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1)); | |
| 822 latency_audio_stream->PrintResults(); | |
| 823 } | |
| 824 | |
| 825 } // namespace webrtc | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 2872953002:
iOS audio session isInterrupted flag does not get reset correctly: (Closed)
