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| 1 // Copyright 2014 The Chromium Authors. All rights reserved. |
| 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. |
| 4 |
| 5 #include "components/copresence/mediums/audio/audio_recorder.h" |
| 6 |
| 7 #include "base/bind.h" |
| 8 #include "base/memory/aligned_memory.h" |
| 9 #include "base/run_loop.h" |
| 10 #include "components/copresence/public/copresence_constants.h" |
| 11 #include "components/copresence/test/audio_test_support.h" |
| 12 #include "content/public/test/test_browser_thread_bundle.h" |
| 13 #include "media/audio/audio_manager.h" |
| 14 #include "media/audio/audio_manager_base.h" |
| 15 #include "media/base/audio_bus.h" |
| 16 #include "testing/gtest/include/gtest/gtest.h" |
| 17 |
| 18 namespace { |
| 19 |
| 20 class TestAudioInputStream : public media::AudioInputStream { |
| 21 public: |
| 22 TestAudioInputStream(const media::AudioParameters& params, |
| 23 const std::vector<float*> channel_data, |
| 24 size_t samples) |
| 25 : callback_(NULL), params_(params) { |
| 26 buffer_ = media::AudioBus::CreateWrapper(2); |
| 27 for (size_t i = 0; i < channel_data.size(); ++i) |
| 28 buffer_->SetChannelData(i, channel_data[i]); |
| 29 buffer_->set_frames(samples); |
| 30 } |
| 31 |
| 32 virtual ~TestAudioInputStream() {} |
| 33 |
| 34 virtual bool Open() OVERRIDE { return true; } |
| 35 virtual void Start(AudioInputCallback* callback) OVERRIDE { |
| 36 DCHECK(callback); |
| 37 callback_ = callback; |
| 38 media::AudioManager::Get()->GetTaskRunner()->PostTask( |
| 39 FROM_HERE, |
| 40 base::Bind(&TestAudioInputStream::SimulateRecording, |
| 41 base::Unretained(this))); |
| 42 } |
| 43 virtual void Stop() OVERRIDE {} |
| 44 virtual void Close() OVERRIDE {} |
| 45 virtual double GetMaxVolume() OVERRIDE { return 1.0; } |
| 46 virtual void SetVolume(double volume) OVERRIDE {} |
| 47 virtual double GetVolume() OVERRIDE { return 1.0; } |
| 48 virtual void SetAutomaticGainControl(bool enabled) OVERRIDE {} |
| 49 virtual bool GetAutomaticGainControl() OVERRIDE { return true; } |
| 50 |
| 51 private: |
| 52 void SimulateRecording() { |
| 53 const int fpb = params_.frames_per_buffer(); |
| 54 for (int i = 0; i < buffer_->frames() / fpb; ++i) { |
| 55 scoped_ptr<media::AudioBus> source = media::AudioBus::Create(2, fpb); |
| 56 buffer_->CopyPartialFramesTo(i * fpb, fpb, 0, source.get()); |
| 57 callback_->OnData(this, source.get(), fpb, 1.0); |
| 58 } |
| 59 } |
| 60 |
| 61 AudioInputCallback* callback_; |
| 62 media::AudioParameters params_; |
| 63 scoped_ptr<media::AudioBus> buffer_; |
| 64 |
| 65 DISALLOW_COPY_AND_ASSIGN(TestAudioInputStream); |
| 66 }; |
| 67 |
| 68 } // namespace |
| 69 |
| 70 namespace copresence { |
| 71 |
| 72 class AudioRecorderTest : public testing::Test { |
| 73 public: |
| 74 AudioRecorderTest() : total_samples_(0), recorder_(NULL) { |
| 75 if (!media::AudioManager::Get()) |
| 76 media::AudioManager::CreateForTesting(); |
| 77 } |
| 78 |
| 79 virtual ~AudioRecorderTest() { |
| 80 DeleteRecorder(); |
| 81 for (size_t i = 0; i < channel_data_.size(); ++i) |
| 82 base::AlignedFree(channel_data_[i]); |
| 83 } |
| 84 |
| 85 void CreateSimpleRecorder() { |
| 86 DeleteRecorder(); |
| 87 recorder_ = new AudioRecorder( |
| 88 base::Bind(&AudioRecorderTest::DecodeSamples, base::Unretained(this))); |
| 89 recorder_->Initialize(); |
| 90 } |
| 91 |
| 92 void CreateRecorder(size_t channels, |
| 93 size_t sample_rate, |
| 94 size_t bits_per_sample, |
| 95 size_t samples) { |
| 96 DeleteRecorder(); |
| 97 params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
| 98 kDefaultChannelLayout, |
| 99 channels, |
| 100 2, |
| 101 sample_rate, |
| 102 bits_per_sample, |
| 103 4096); |
| 104 |
| 105 channel_data_.clear(); |
| 106 channel_data_.push_back(GenerateSamples(0x1337, samples)); |
| 107 channel_data_.push_back(GenerateSamples(0x7331, samples)); |
| 108 |
| 109 total_samples_ = samples; |
| 110 |
| 111 recorder_ = new AudioRecorder( |
| 112 base::Bind(&AudioRecorderTest::DecodeSamples, base::Unretained(this))); |
| 113 recorder_->set_input_stream_for_testing( |
| 114 new TestAudioInputStream(params_, channel_data_, samples)); |
| 115 recorder_->set_params_for_testing(new media::AudioParameters(params_)); |
| 116 recorder_->Initialize(); |
| 117 } |
| 118 |
| 119 void DeleteRecorder() { |
| 120 if (!recorder_) |
| 121 return; |
| 122 recorder_->Finalize(); |
| 123 recorder_ = NULL; |
| 124 } |
| 125 |
| 126 void RecordAndVerifySamples() { |
| 127 received_samples_.clear(); |
| 128 run_loop_.reset(new base::RunLoop()); |
| 129 recorder_->Record(); |
| 130 run_loop_->Run(); |
| 131 } |
| 132 |
| 133 void DecodeSamples(const std::string& samples) { |
| 134 received_samples_ += samples; |
| 135 // We expect one less decode than our total samples would ideally have |
| 136 // triggered since we process data in 4k chunks. So our sample processing |
| 137 // will never rarely be perfectly aligned with 0.5s worth of samples, hence |
| 138 // we will almost always run with a buffer of leftover samples that will |
| 139 // not get sent to this callback since the recorder will be waiting for |
| 140 // more data. |
| 141 const size_t decode_buffer = params_.sample_rate() / 2; // 0.5s |
| 142 const size_t expected_samples = |
| 143 (total_samples_ / decode_buffer - 1) * decode_buffer; |
| 144 const size_t expected_samples_size = |
| 145 expected_samples * sizeof(float) * params_.channels(); |
| 146 if (received_samples_.size() == expected_samples_size) { |
| 147 VerifySamples(); |
| 148 run_loop_->Quit(); |
| 149 } |
| 150 } |
| 151 |
| 152 void VerifySamples() { |
| 153 int differences = 0; |
| 154 |
| 155 float* buffer_view = |
| 156 reinterpret_cast<float*>(string_as_array(&received_samples_)); |
| 157 const int channels = params_.channels(); |
| 158 const int frames = |
| 159 received_samples_.size() / sizeof(float) / params_.channels(); |
| 160 for (int ch = 0; ch < channels; ++ch) { |
| 161 for (int si = 0, di = ch; si < frames; ++si, di += channels) |
| 162 differences += (buffer_view[di] != channel_data_[ch][si]); |
| 163 } |
| 164 |
| 165 ASSERT_EQ(0, differences); |
| 166 } |
| 167 |
| 168 protected: |
| 169 float* GenerateSamples(int random_seed, size_t size) { |
| 170 float* samples = static_cast<float*>(base::AlignedAlloc( |
| 171 size * sizeof(float), media::AudioBus::kChannelAlignment)); |
| 172 PopulateSamples(0x1337, size, samples); |
| 173 return samples; |
| 174 } |
| 175 bool IsRecording() { |
| 176 recorder_->FlushAudioLoopForTesting(); |
| 177 return recorder_->is_recording_; |
| 178 } |
| 179 |
| 180 std::vector<float*> channel_data_; |
| 181 media::AudioParameters params_; |
| 182 size_t total_samples_; |
| 183 |
| 184 AudioRecorder* recorder_; |
| 185 |
| 186 std::string received_samples_; |
| 187 |
| 188 scoped_ptr<base::RunLoop> run_loop_; |
| 189 content::TestBrowserThreadBundle thread_bundle_; |
| 190 }; |
| 191 |
| 192 #if defined(OS_WIN) || defined(OS_MACOSX) |
| 193 // Windows does not let us use non-OS params. The tests need to be rewritten to |
| 194 // use the params provided to us by the audio manager rather than setting our |
| 195 // own params. |
| 196 #define MAYBE_BasicRecordAndStop DISABLED_BasicRecordAndStop |
| 197 #define MAYBE_OutOfOrderRecordAndStopMultiple DISABLED_OutOfOrderRecordAndStopMu
ltiple |
| 198 #define MAYBE_RecordingEndToEnd DISABLED_RecordingEndToEnd |
| 199 #else |
| 200 #define MAYBE_BasicRecordAndStop BasicRecordAndStop |
| 201 #define MAYBE_OutOfOrderRecordAndStopMultiple OutOfOrderRecordAndStopMultiple |
| 202 #define MAYBE_RecordingEndToEnd RecordingEndToEnd |
| 203 #endif |
| 204 |
| 205 TEST_F(AudioRecorderTest, MAYBE_BasicRecordAndStop) { |
| 206 CreateSimpleRecorder(); |
| 207 |
| 208 recorder_->Record(); |
| 209 EXPECT_TRUE(IsRecording()); |
| 210 recorder_->Stop(); |
| 211 EXPECT_FALSE(IsRecording()); |
| 212 recorder_->Record(); |
| 213 |
| 214 EXPECT_TRUE(IsRecording()); |
| 215 recorder_->Stop(); |
| 216 EXPECT_FALSE(IsRecording()); |
| 217 recorder_->Record(); |
| 218 |
| 219 EXPECT_TRUE(IsRecording()); |
| 220 recorder_->Stop(); |
| 221 EXPECT_FALSE(IsRecording()); |
| 222 |
| 223 DeleteRecorder(); |
| 224 } |
| 225 |
| 226 TEST_F(AudioRecorderTest, MAYBE_OutOfOrderRecordAndStopMultiple) { |
| 227 CreateSimpleRecorder(); |
| 228 |
| 229 recorder_->Stop(); |
| 230 recorder_->Stop(); |
| 231 recorder_->Stop(); |
| 232 EXPECT_FALSE(IsRecording()); |
| 233 |
| 234 recorder_->Record(); |
| 235 recorder_->Record(); |
| 236 EXPECT_TRUE(IsRecording()); |
| 237 |
| 238 recorder_->Stop(); |
| 239 recorder_->Stop(); |
| 240 EXPECT_FALSE(IsRecording()); |
| 241 |
| 242 DeleteRecorder(); |
| 243 } |
| 244 |
| 245 TEST_F(AudioRecorderTest, MAYBE_RecordingEndToEnd) { |
| 246 const int kNumSamples = 48000 * 3; |
| 247 CreateRecorder( |
| 248 kDefaultChannels, kDefaultSampleRate, kDefaultBitsPerSample, kNumSamples); |
| 249 |
| 250 RecordAndVerifySamples(); |
| 251 |
| 252 DeleteRecorder(); |
| 253 } |
| 254 |
| 255 // TODO(rkc): Add tests with recording different sample rates. |
| 256 |
| 257 } // namespace copresence |
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