| /external/webrtc/audio/ |
| D | audio_state_unittest.cc | 30 constexpr int kSampleRate = 16000; variable
61 int PreferredSampleRate() const /*override*/ { return kSampleRate; } in PreferredSampleRate()
143 constexpr int kSampleRate = 16000; in TEST() local
145 auto audio_data = Create10msTestData(kSampleRate, kNumChannels); in TEST()
148 &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels, in TEST()
149 kSampleRate, 0, 0, 0, false, new_mic_level); in TEST()
199 constexpr int kSampleRate = 16000; in TEST() local
201 auto audio_data = Create10msTestData(kSampleRate, kNumChannels); in TEST()
204 &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels, in TEST()
205 kSampleRate, 5, 0, 0, true, new_mic_level); in TEST()
[all …]
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| /external/webrtc/modules/audio_coding/neteq/ |
| D | time_stretch_unittest.cc | 32 const int kSampleRate = 8000; in TEST() local
33 const int kOverlapSamples = 5 * kSampleRate / 8000; in TEST()
35 Accelerate accelerate(kSampleRate, kNumChannels, bgn); in TEST()
36 PreemptiveExpand preemptive_expand(kSampleRate, kNumChannels, bgn, in TEST()
41 const int kSampleRate = 8000; in TEST() local
42 const int kOverlapSamples = 5 * kSampleRate / 8000; in TEST()
47 accelerate_factory.Create(kSampleRate, kNumChannels, bgn); in TEST()
53 kSampleRate, kNumChannels, bgn, kOverlapSamples); in TEST()
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| /external/tensorflow/tensorflow/lite/experimental/microfrontend/lib/ |
| D | filterbank_test.cc | 24 const int kSampleRate = 1000; variable
54 kSampleRate, kSpectrumSize)); in TF_LITE_MICRO_TEST()
65 kSampleRate, kSpectrumSize)); in TF_LITE_MICRO_TEST()
76 kSampleRate, kSpectrumSize)); in TF_LITE_MICRO_TEST()
93 kSampleRate, kSpectrumSize)); in TF_LITE_MICRO_TEST()
110 kSampleRate, kSpectrumSize)); in TF_LITE_MICRO_TEST()
127 kSampleRate, kSpectrumSize)); in TF_LITE_MICRO_TEST()
147 kSampleRate, kSpectrumSize)); in TF_LITE_MICRO_TEST()
185 kSampleRate, kSpectrumSize)); in TF_LITE_MICRO_TEST()
203 kSampleRate, kSpectrumSize)); in TF_LITE_MICRO_TEST()
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| D | window_test.cc | 22 const int kSampleRate = 1000; variable
49 WindowPopulateState(&config.config_, &state, kSampleRate)); in TF_LITE_MICRO_TEST()
68 WindowPopulateState(&config.config_, &state, kSampleRate)); in TF_LITE_MICRO_TEST()
87 WindowPopulateState(&config.config_, &state, kSampleRate)); in TF_LITE_MICRO_TEST()
110 WindowPopulateState(&config.config_, &state, kSampleRate)); in TF_LITE_MICRO_TEST()
126 WindowPopulateState(&config.config_, &state, kSampleRate)); in TF_LITE_MICRO_TEST()
153 WindowPopulateState(&config.config_, &state, kSampleRate)); in TF_LITE_MICRO_TEST()
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| D | frontend_test.cc | 22 const int kSampleRate = 1000; variable
63 FrontendPopulateState(&config.config_, &state, kSampleRate)); in TF_LITE_MICRO_TEST()
84 FrontendPopulateState(&config.config_, &state, kSampleRate)); in TF_LITE_MICRO_TEST()
109 FrontendPopulateState(&config.config_, &state, kSampleRate)); in TF_LITE_MICRO_TEST()
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| /external/tensorflow/tensorflow/core/kernels/ |
| D | summary_audio_op_test.cc | 79 const float kSampleRate = 44100.0f; in TEST_F() local
88 AddInputFromArray<float>(TensorShape({}), {kSampleRate}); in TEST_F()
113 const float kSampleRate = 44100.0f; in TEST_F() local
122 AddInputFromArray<float>(TensorShape({}), {kSampleRate}); in TEST_F()
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| /external/webrtc/modules/audio_device/ |
| D | fine_audio_buffer_unittest.cc | 30 const int kSampleRate = 44100; variable
32 const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
98 audio_device_buffer.SetPlayoutSampleRate(kSampleRate); in RunFineBufferTest()
100 audio_device_buffer.SetRecordingSampleRate(kSampleRate); in RunFineBufferTest()
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| /external/webrtc/common_audio/ |
| D | wav_file_unittest.cc | 106 constexpr int kSampleRate = 8000; in TEST() local
108 constexpr size_t kNumSamples = 3 * kSampleRate * kNumChannels; in TEST()
121 static_cast<double>(i) / (kNumChannels * kSampleRate); in TEST()
128 WavWriter w(outfile, kSampleRate, kNumChannels, wav_format); in TEST()
129 EXPECT_EQ(kSampleRate, w.sample_rate()); in TEST()
153 EXPECT_EQ(kSampleRate, r.sample_rate()); in TEST()
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| /external/webrtc/audio/test/ |
| D | audio_end_to_end_test.cc | 28 constexpr int kSampleRate = 48000; variable
52 return TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, kSampleRate); in CreateCapturer()
57 return TestAudioDeviceModule::CreateDiscardRenderer(kSampleRate); in CreateRenderer()
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| /external/webrtc/modules/audio_device/include/ |
| D | test_audio_device_unittest.cc | 39 static const int kSampleRate = kSamplesPerFrame * 100; in RunTest() local
40 EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate), in RunTest()
137 static const int kSampleRate = kSamplesPerFrame * 100; in TEST() local
138 EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate), in TEST()
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| /external/webrtc/modules/audio_device/android/ |
| D | audio_manager_unittest.cc | 220 const int kSampleRate = 48000; in TEST_F() local
226 AudioParameters params(kSampleRate, kChannels, kFramesPerBuffer); in TEST_F()
228 EXPECT_EQ(kSampleRate, params.sample_rate()); in TEST_F()
231 EXPECT_EQ(static_cast<size_t>(kSampleRate / 100), in TEST_F()
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| /external/brotli/c/enc/ |
| D | compress_fragment.c | 95 static const size_t kSampleRate = 29; in BuildAndStoreLiteralPrefixCode() local
96 for (i = 0; i < input_size; i += kSampleRate) { in BuildAndStoreLiteralPrefixCode()
99 histogram_total = (input_size + kSampleRate - 1) / kSampleRate; in BuildAndStoreLiteralPrefixCode()
375 static const size_t kSampleRate = 43; in ShouldMergeBlock() local
377 for (i = 0; i < len; i += kSampleRate) { in ShouldMergeBlock()
381 const size_t total = (len + kSampleRate - 1) / kSampleRate; in ShouldMergeBlock()
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| D | encode.c | 521 static const uint32_t kSampleRate = 13; in ShouldCompress() local
524 (double)bytes * kMinEntropy / kSampleRate; in ShouldCompress()
525 size_t t = (bytes + kSampleRate - 1) / kSampleRate; in ShouldCompress()
530 pos += kSampleRate; in ShouldCompress()
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| /external/libwebm/testing/ |
| D | test_util.h | 41 const int kSampleRate = 30; variable
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| D | mkvmuxer_tests.cc | 79 segment_.AddAudioTrack(kSampleRate, kChannels, kAudioTrackNumber)); in AddAudioTrack()
162 segment_.AddAudioTrack(kSampleRate, kChannels, kAudioTrackNumber)); in TEST_F()
166 EXPECT_EQ(kSampleRate, audio->sample_rate()); in TEST_F()
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| D | mkvparser_tests.cc | 162 EXPECT_EQ(kSampleRate, audio_track->GetSamplingRate()); in TEST_F()
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| /external/webrtc/modules/video_coding/ |
| D | fec_controller_unittest.cc | 29 kSampleRate = 90000 // RTP timestamps per second. enumerator
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| /external/webrtc/audio/voip/test/ |
| D | audio_egress_unittest.cc | 247 constexpr int kSampleRate = 8000; in TEST_F() local
250 egress_->RegisterTelephoneEventType(kPayloadType, kSampleRate); in TEST_F()
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| /external/webrtc/sdk/android/native_unittests/audio_device/ |
| D | audio_device_unittest.cc | 1035 const int kSampleRate = 48000; in TEST_F() local
1041 AudioParameters params(kSampleRate, kChannels, kFramesPerBuffer); in TEST_F()
1043 EXPECT_EQ(kSampleRate, params.sample_rate()); in TEST_F()
1046 EXPECT_EQ(static_cast<size_t>(kSampleRate / 100), in TEST_F()
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| /external/oboe/docs/ |
| D | GettingStarted.md | 264 ->setSampleRate(kSampleRate)
306 static int constexpr kSampleRate = 48000;
312 static double constexpr mPhaseIncrement = kFrequency * kTwoPi / (double) kSampleRate;
|