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| 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
| 2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
| 4 | 4 |
| 5 #include <windows.h> | 5 #include <windows.h> |
| 6 #include <mmsystem.h> | 6 #include <mmsystem.h> |
| 7 | 7 |
| 8 #include "base/basictypes.h" | 8 #include "base/basictypes.h" |
| 9 #include "base/environment.h" | 9 #include "base/environment.h" |
| 10 #include "base/file_util.h" | 10 #include "base/file_util.h" |
| (...skipping 20 matching lines...) Expand all Loading... |
| 31 namespace media { | 31 namespace media { |
| 32 | 32 |
| 33 ACTION_P3(CheckCountAndPostQuitTask, count, limit, loop) { | 33 ACTION_P3(CheckCountAndPostQuitTask, count, limit, loop) { |
| 34 if (++*count >= limit) { | 34 if (++*count >= limit) { |
| 35 loop->PostTask(FROM_HERE, base::MessageLoop::QuitClosure()); | 35 loop->PostTask(FROM_HERE, base::MessageLoop::QuitClosure()); |
| 36 } | 36 } |
| 37 } | 37 } |
| 38 | 38 |
| 39 class MockAudioInputCallback : public AudioInputStream::AudioInputCallback { | 39 class MockAudioInputCallback : public AudioInputStream::AudioInputCallback { |
| 40 public: | 40 public: |
| 41 MOCK_METHOD4(OnData, | 41 MOCK_METHOD5(OnData, void(AudioInputStream* stream, |
| 42 void(AudioInputStream* stream, | 42 const uint8* src, uint32 size, |
| 43 const AudioBus* src, | 43 uint32 hardware_delay_bytes, double volume)); |
| 44 uint32 hardware_delay_bytes, | |
| 45 double volume)); | |
| 46 MOCK_METHOD1(OnError, void(AudioInputStream* stream)); | 44 MOCK_METHOD1(OnError, void(AudioInputStream* stream)); |
| 47 }; | 45 }; |
| 48 | 46 |
| 49 class FakeAudioInputCallback : public AudioInputStream::AudioInputCallback { | 47 class FakeAudioInputCallback : public AudioInputStream::AudioInputCallback { |
| 50 public: | 48 public: |
| 51 FakeAudioInputCallback() | 49 FakeAudioInputCallback() |
| 52 : error_(false), | 50 : error_(false), |
| 53 data_event_(false, false), | 51 data_event_(false, false) { |
| 54 num_received_audio_frames_(0) {} | 52 } |
| 55 | 53 |
| 54 const std::vector<uint8>& received_data() const { return received_data_; } |
| 56 bool error() const { return error_; } | 55 bool error() const { return error_; } |
| 57 int num_received_audio_frames() const { return num_received_audio_frames_; } | |
| 58 | 56 |
| 59 // Waits until OnData() is called on another thread. | 57 // Waits until OnData() is called on another thread. |
| 60 void WaitForData() { | 58 void WaitForData() { |
| 61 data_event_.Wait(); | 59 data_event_.Wait(); |
| 62 } | 60 } |
| 63 | 61 |
| 64 virtual void OnData(AudioInputStream* stream, | 62 virtual void OnData(AudioInputStream* stream, |
| 65 const AudioBus* src, | 63 const uint8* src, uint32 size, |
| 66 uint32 hardware_delay_bytes, | 64 uint32 hardware_delay_bytes, double volume) OVERRIDE { |
| 67 double volume) OVERRIDE { | 65 received_data_.insert(received_data_.end(), src, src + size); |
| 68 EXPECT_NE(hardware_delay_bytes, 0u); | |
| 69 num_received_audio_frames_ += src->frames(); | |
| 70 data_event_.Signal(); | 66 data_event_.Signal(); |
| 71 } | 67 } |
| 72 | 68 |
| 73 virtual void OnError(AudioInputStream* stream) OVERRIDE { | 69 virtual void OnError(AudioInputStream* stream) OVERRIDE { |
| 74 error_ = true; | 70 error_ = true; |
| 75 } | 71 } |
| 76 | 72 |
| 77 private: | 73 private: |
| 78 int num_received_audio_frames_; | 74 std::vector<uint8> received_data_; |
| 79 base::WaitableEvent data_event_; | 75 base::WaitableEvent data_event_; |
| 80 bool error_; | 76 bool error_; |
| 81 | 77 |
| 82 DISALLOW_COPY_AND_ASSIGN(FakeAudioInputCallback); | 78 DISALLOW_COPY_AND_ASSIGN(FakeAudioInputCallback); |
| 83 }; | 79 }; |
| 84 | 80 |
| 85 // This audio sink implementation should be used for manual tests only since | 81 // This audio sink implementation should be used for manual tests only since |
| 86 // the recorded data is stored on a raw binary data file. | 82 // the recorded data is stored on a raw binary data file. |
| 87 class WriteToFileAudioSink : public AudioInputStream::AudioInputCallback { | 83 class WriteToFileAudioSink : public AudioInputStream::AudioInputCallback { |
| 88 public: | 84 public: |
| 89 // Allocate space for ~10 seconds of data @ 48kHz in stereo: | 85 // Allocate space for ~10 seconds of data @ 48kHz in stereo: |
| 90 // 2 bytes per sample, 2 channels, 10ms @ 48kHz, 10 seconds <=> 1920000 bytes. | 86 // 2 bytes per sample, 2 channels, 10ms @ 48kHz, 10 seconds <=> 1920000 bytes. |
| 91 static const size_t kMaxBufferSize = 2 * 2 * 480 * 100 * 10; | 87 static const size_t kMaxBufferSize = 2 * 2 * 480 * 100 * 10; |
| 92 | 88 |
| 93 explicit WriteToFileAudioSink(const char* file_name, int bits_per_sample) | 89 explicit WriteToFileAudioSink(const char* file_name) |
| 94 : bits_per_sample_(bits_per_sample), | 90 : buffer_(0, kMaxBufferSize), |
| 95 buffer_(0, kMaxBufferSize), | |
| 96 bytes_to_write_(0) { | 91 bytes_to_write_(0) { |
| 97 base::FilePath file_path; | 92 base::FilePath file_path; |
| 98 EXPECT_TRUE(PathService::Get(base::DIR_EXE, &file_path)); | 93 EXPECT_TRUE(PathService::Get(base::DIR_EXE, &file_path)); |
| 99 file_path = file_path.AppendASCII(file_name); | 94 file_path = file_path.AppendASCII(file_name); |
| 100 binary_file_ = base::OpenFile(file_path, "wb"); | 95 binary_file_ = base::OpenFile(file_path, "wb"); |
| 101 DLOG_IF(ERROR, !binary_file_) << "Failed to open binary PCM data file."; | 96 DLOG_IF(ERROR, !binary_file_) << "Failed to open binary PCM data file."; |
| 102 VLOG(0) << ">> Output file: " << file_path.value() << " has been created."; | 97 VLOG(0) << ">> Output file: " << file_path.value() << " has been created."; |
| 103 VLOG(0) << "bits_per_sample_:" << bits_per_sample_; | |
| 104 } | 98 } |
| 105 | 99 |
| 106 virtual ~WriteToFileAudioSink() { | 100 virtual ~WriteToFileAudioSink() { |
| 107 size_t bytes_written = 0; | 101 size_t bytes_written = 0; |
| 108 while (bytes_written < bytes_to_write_) { | 102 while (bytes_written < bytes_to_write_) { |
| 109 const uint8* chunk; | 103 const uint8* chunk; |
| 110 int chunk_size; | 104 int chunk_size; |
| 111 | 105 |
| 112 // Stop writing if no more data is available. | 106 // Stop writing if no more data is available. |
| 113 if (!buffer_.GetCurrentChunk(&chunk, &chunk_size)) | 107 if (!buffer_.GetCurrentChunk(&chunk, &chunk_size)) |
| 114 break; | 108 break; |
| 115 | 109 |
| 116 // Write recorded data chunk to the file and prepare for next chunk. | 110 // Write recorded data chunk to the file and prepare for next chunk. |
| 117 fwrite(chunk, 1, chunk_size, binary_file_); | 111 fwrite(chunk, 1, chunk_size, binary_file_); |
| 118 buffer_.Seek(chunk_size); | 112 buffer_.Seek(chunk_size); |
| 119 bytes_written += chunk_size; | 113 bytes_written += chunk_size; |
| 120 } | 114 } |
| 121 base::CloseFile(binary_file_); | 115 base::CloseFile(binary_file_); |
| 122 } | 116 } |
| 123 | 117 |
| 124 // AudioInputStream::AudioInputCallback implementation. | 118 // AudioInputStream::AudioInputCallback implementation. |
| 125 virtual void OnData(AudioInputStream* stream, | 119 virtual void OnData(AudioInputStream* stream, |
| 126 const AudioBus* src, | 120 const uint8* src, |
| 121 uint32 size, |
| 127 uint32 hardware_delay_bytes, | 122 uint32 hardware_delay_bytes, |
| 128 double volume) { | 123 double volume) { |
| 129 EXPECT_EQ(bits_per_sample_, 16); | |
| 130 const int num_samples = src->frames() * src->channels(); | |
| 131 scoped_ptr<int16> interleaved(new int16[num_samples]); | |
| 132 const int bytes_per_sample = sizeof(*interleaved); | |
| 133 src->ToInterleaved(src->frames(), bytes_per_sample, interleaved.get()); | |
| 134 | |
| 135 // Store data data in a temporary buffer to avoid making blocking | 124 // Store data data in a temporary buffer to avoid making blocking |
| 136 // fwrite() calls in the audio callback. The complete buffer will be | 125 // fwrite() calls in the audio callback. The complete buffer will be |
| 137 // written to file in the destructor. | 126 // written to file in the destructor. |
| 138 const int size = bytes_per_sample * num_samples; | 127 if (buffer_.Append(src, size)) { |
| 139 if (buffer_.Append((const uint8*)interleaved.get(), size)) { | |
| 140 bytes_to_write_ += size; | 128 bytes_to_write_ += size; |
| 141 } | 129 } |
| 142 } | 130 } |
| 143 | 131 |
| 144 virtual void OnError(AudioInputStream* stream) {} | 132 virtual void OnError(AudioInputStream* stream) {} |
| 145 | 133 |
| 146 private: | 134 private: |
| 147 int bits_per_sample_; | |
| 148 media::SeekableBuffer buffer_; | 135 media::SeekableBuffer buffer_; |
| 149 FILE* binary_file_; | 136 FILE* binary_file_; |
| 150 size_t bytes_to_write_; | 137 size_t bytes_to_write_; |
| 151 }; | 138 }; |
| 152 | 139 |
| 153 // Convenience method which ensures that we are not running on the build | 140 // Convenience method which ensures that we are not running on the build |
| 154 // bots and that at least one valid input device can be found. We also | 141 // bots and that at least one valid input device can be found. We also |
| 155 // verify that we are not running on XP since the low-latency (WASAPI- | 142 // verify that we are not running on XP since the low-latency (WASAPI- |
| 156 // based) version requires Windows Vista or higher. | 143 // based) version requires Windows Vista or higher. |
| 157 static bool CanRunAudioTests(AudioManager* audio_man) { | 144 static bool CanRunAudioTests(AudioManager* audio_man) { |
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| 382 | 369 |
| 383 MockAudioInputCallback sink; | 370 MockAudioInputCallback sink; |
| 384 | 371 |
| 385 // Derive the expected size in bytes of each recorded packet. | 372 // Derive the expected size in bytes of each recorded packet. |
| 386 uint32 bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() * | 373 uint32 bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() * |
| 387 (aisw.bits_per_sample() / 8); | 374 (aisw.bits_per_sample() / 8); |
| 388 | 375 |
| 389 // We use 10ms packets and will run the test until ten packets are received. | 376 // We use 10ms packets and will run the test until ten packets are received. |
| 390 // All should contain valid packets of the same size and a valid delay | 377 // All should contain valid packets of the same size and a valid delay |
| 391 // estimate. | 378 // estimate. |
| 392 EXPECT_CALL(sink, OnData(ais.get(), NotNull(), Gt(bytes_per_packet), _)) | 379 EXPECT_CALL(sink, OnData( |
| 380 ais.get(), NotNull(), bytes_per_packet, Gt(bytes_per_packet), _)) |
| 393 .Times(AtLeast(10)) | 381 .Times(AtLeast(10)) |
| 394 .WillRepeatedly(CheckCountAndPostQuitTask(&count, 10, &loop)); | 382 .WillRepeatedly(CheckCountAndPostQuitTask(&count, 10, &loop)); |
| 395 ais->Start(&sink); | 383 ais->Start(&sink); |
| 396 loop.Run(); | 384 loop.Run(); |
| 397 ais->Stop(); | 385 ais->Stop(); |
| 398 | 386 |
| 399 // Store current packet size (to be used in the subsequent tests). | 387 // Store current packet size (to be used in the subsequent tests). |
| 400 int frames_per_buffer_10ms = aisw.frames_per_buffer(); | 388 int frames_per_buffer_10ms = aisw.frames_per_buffer(); |
| 401 | 389 |
| 402 ais.Close(); | 390 ais.Close(); |
| 403 | 391 |
| 404 // 20 ms packet size. | 392 // 20 ms packet size. |
| 405 | 393 |
| 406 count = 0; | 394 count = 0; |
| 407 ais.Reset(aisw.Create(2 * frames_per_buffer_10ms)); | 395 ais.Reset(aisw.Create(2 * frames_per_buffer_10ms)); |
| 408 EXPECT_TRUE(ais->Open()); | 396 EXPECT_TRUE(ais->Open()); |
| 409 bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() * | 397 bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() * |
| 410 (aisw.bits_per_sample() / 8); | 398 (aisw.bits_per_sample() / 8); |
| 411 | 399 |
| 412 EXPECT_CALL(sink, OnData(ais.get(), NotNull(), Gt(bytes_per_packet), _)) | 400 EXPECT_CALL(sink, OnData( |
| 401 ais.get(), NotNull(), bytes_per_packet, Gt(bytes_per_packet), _)) |
| 413 .Times(AtLeast(10)) | 402 .Times(AtLeast(10)) |
| 414 .WillRepeatedly(CheckCountAndPostQuitTask(&count, 10, &loop)); | 403 .WillRepeatedly(CheckCountAndPostQuitTask(&count, 10, &loop)); |
| 415 ais->Start(&sink); | 404 ais->Start(&sink); |
| 416 loop.Run(); | 405 loop.Run(); |
| 417 ais->Stop(); | 406 ais->Stop(); |
| 418 ais.Close(); | 407 ais.Close(); |
| 419 | 408 |
| 420 // 5 ms packet size. | 409 // 5 ms packet size. |
| 421 | 410 |
| 422 count = 0; | 411 count = 0; |
| 423 ais.Reset(aisw.Create(frames_per_buffer_10ms / 2)); | 412 ais.Reset(aisw.Create(frames_per_buffer_10ms / 2)); |
| 424 EXPECT_TRUE(ais->Open()); | 413 EXPECT_TRUE(ais->Open()); |
| 425 bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() * | 414 bytes_per_packet = aisw.channels() * aisw.frames_per_buffer() * |
| 426 (aisw.bits_per_sample() / 8); | 415 (aisw.bits_per_sample() / 8); |
| 427 | 416 |
| 428 EXPECT_CALL(sink, OnData(ais.get(), NotNull(), Gt(bytes_per_packet), _)) | 417 EXPECT_CALL(sink, OnData( |
| 418 ais.get(), NotNull(), bytes_per_packet, Gt(bytes_per_packet), _)) |
| 429 .Times(AtLeast(10)) | 419 .Times(AtLeast(10)) |
| 430 .WillRepeatedly(CheckCountAndPostQuitTask(&count, 10, &loop)); | 420 .WillRepeatedly(CheckCountAndPostQuitTask(&count, 10, &loop)); |
| 431 ais->Start(&sink); | 421 ais->Start(&sink); |
| 432 loop.Run(); | 422 loop.Run(); |
| 433 ais->Stop(); | 423 ais->Stop(); |
| 434 ais.Close(); | 424 ais.Close(); |
| 435 } | 425 } |
| 436 | 426 |
| 437 // Test that we can capture a stream in loopback. | 427 // Test that we can capture loopback stream. |
| 438 TEST(WinAudioInputTest, WASAPIAudioInputStreamLoopback) { | 428 TEST(WinAudioInputTest, WASAPIAudioInputStreamLoopback) { |
| 439 scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting()); | 429 scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting()); |
| 440 if (!audio_manager->HasAudioOutputDevices() || !CoreAudioUtil::IsSupported()) | 430 if (!audio_manager->HasAudioOutputDevices() || !CoreAudioUtil::IsSupported()) |
| 441 return; | 431 return; |
| 442 | 432 |
| 443 AudioParameters params = audio_manager->GetInputStreamParameters( | 433 AudioParameters params = audio_manager->GetInputStreamParameters( |
| 444 AudioManagerBase::kLoopbackInputDeviceId); | 434 AudioManagerBase::kLoopbackInputDeviceId); |
| 445 EXPECT_EQ(params.effects(), 0); | 435 EXPECT_EQ(params.effects(), 0); |
| 446 | 436 |
| 447 AudioParameters output_params = | 437 AudioParameters output_params = |
| 448 audio_manager->GetOutputStreamParameters(std::string()); | 438 audio_manager->GetOutputStreamParameters(std::string()); |
| 449 EXPECT_EQ(params.sample_rate(), output_params.sample_rate()); | 439 EXPECT_EQ(params.sample_rate(), output_params.sample_rate()); |
| 450 EXPECT_EQ(params.channel_layout(), output_params.channel_layout()); | 440 EXPECT_EQ(params.channel_layout(), output_params.channel_layout()); |
| 451 | 441 |
| 452 ScopedAudioInputStream stream(audio_manager->MakeAudioInputStream( | 442 ScopedAudioInputStream stream(audio_manager->MakeAudioInputStream( |
| 453 params, AudioManagerBase::kLoopbackInputDeviceId)); | 443 params, AudioManagerBase::kLoopbackInputDeviceId)); |
| 454 ASSERT_TRUE(stream->Open()); | 444 ASSERT_TRUE(stream->Open()); |
| 455 FakeAudioInputCallback sink; | 445 FakeAudioInputCallback sink; |
| 456 stream->Start(&sink); | 446 stream->Start(&sink); |
| 457 ASSERT_FALSE(sink.error()); | 447 ASSERT_FALSE(sink.error()); |
| 458 | 448 |
| 459 sink.WaitForData(); | 449 sink.WaitForData(); |
| 460 stream.Close(); | 450 stream.Close(); |
| 461 | 451 |
| 462 EXPECT_GT(sink.num_received_audio_frames(), 0); | 452 EXPECT_FALSE(sink.received_data().empty()); |
| 463 EXPECT_FALSE(sink.error()); | 453 EXPECT_FALSE(sink.error()); |
| 464 } | 454 } |
| 465 | 455 |
| 466 // This test is intended for manual tests and should only be enabled | 456 // This test is intended for manual tests and should only be enabled |
| 467 // when it is required to store the captured data on a local file. | 457 // when it is required to store the captured data on a local file. |
| 468 // By default, GTest will print out YOU HAVE 1 DISABLED TEST. | 458 // By default, GTest will print out YOU HAVE 1 DISABLED TEST. |
| 469 // To include disabled tests in test execution, just invoke the test program | 459 // To include disabled tests in test execution, just invoke the test program |
| 470 // with --gtest_also_run_disabled_tests or set the GTEST_ALSO_RUN_DISABLED_TESTS | 460 // with --gtest_also_run_disabled_tests or set the GTEST_ALSO_RUN_DISABLED_TESTS |
| 471 // environment variable to a value greater than 0. | 461 // environment variable to a value greater than 0. |
| 472 TEST(WinAudioInputTest, DISABLED_WASAPIAudioInputStreamRecordToFile) { | 462 TEST(WinAudioInputTest, DISABLED_WASAPIAudioInputStreamRecordToFile) { |
| 473 scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting()); | 463 scoped_ptr<AudioManager> audio_manager(AudioManager::CreateForTesting()); |
| 474 if (!CanRunAudioTests(audio_manager.get())) | 464 if (!CanRunAudioTests(audio_manager.get())) |
| 475 return; | 465 return; |
| 476 | 466 |
| 477 // Name of the output PCM file containing captured data. The output file | 467 // Name of the output PCM file containing captured data. The output file |
| 478 // will be stored in the directory containing 'media_unittests.exe'. | 468 // will be stored in the directory containing 'media_unittests.exe'. |
| 479 // Example of full name: \src\build\Debug\out_stereo_10sec.pcm. | 469 // Example of full name: \src\build\Debug\out_stereo_10sec.pcm. |
| 480 const char* file_name = "out_stereo_10sec.pcm"; | 470 const char* file_name = "out_stereo_10sec.pcm"; |
| 481 | 471 |
| 482 AudioInputStreamWrapper aisw(audio_manager.get()); | 472 AudioInputStreamWrapper aisw(audio_manager.get()); |
| 483 ScopedAudioInputStream ais(aisw.Create()); | 473 ScopedAudioInputStream ais(aisw.Create()); |
| 484 EXPECT_TRUE(ais->Open()); | 474 EXPECT_TRUE(ais->Open()); |
| 485 | 475 |
| 486 VLOG(0) << ">> Sample rate: " << aisw.sample_rate() << " [Hz]"; | 476 VLOG(0) << ">> Sample rate: " << aisw.sample_rate() << " [Hz]"; |
| 487 WriteToFileAudioSink file_sink(file_name, aisw.bits_per_sample()); | 477 WriteToFileAudioSink file_sink(file_name); |
| 488 VLOG(0) << ">> Speak into the default microphone while recording."; | 478 VLOG(0) << ">> Speak into the default microphone while recording."; |
| 489 ais->Start(&file_sink); | 479 ais->Start(&file_sink); |
| 490 base::PlatformThread::Sleep(TestTimeouts::action_timeout()); | 480 base::PlatformThread::Sleep(TestTimeouts::action_timeout()); |
| 491 ais->Stop(); | 481 ais->Stop(); |
| 492 VLOG(0) << ">> Recording has stopped."; | 482 VLOG(0) << ">> Recording has stopped."; |
| 493 ais.Close(); | 483 ais.Close(); |
| 494 } | 484 } |
| 495 | 485 |
| 496 } // namespace media | 486 } // namespace media |
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