| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| D | time_stretch_unittest.cc | 33 const int kSampleRate = 8000; in TEST() local
34 const int kOverlapSamples = 5 * kSampleRate / 8000; in TEST()
36 Accelerate accelerate(kSampleRate, kNumChannels, bgn); in TEST()
38 kSampleRate, kNumChannels, bgn, kOverlapSamples); in TEST()
42 const int kSampleRate = 8000; in TEST() local
43 const int kOverlapSamples = 5 * kSampleRate / 8000; in TEST()
48 accelerate_factory.Create(kSampleRate, kNumChannels, bgn); in TEST()
54 kSampleRate, kNumChannels, bgn, kOverlapSamples); in TEST()
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| /external/webrtc/webrtc/modules/audio_device/ |
| D | fine_audio_buffer_unittest.cc | 134 const int kSampleRate = 44100; in TEST() local
135 const int kSamplesPer10Ms = kSampleRate * 10 / 1000; in TEST()
137 RunFineBufferTest(kSampleRate, kFrameSizeSamples); in TEST()
141 const int kSampleRate = 44100; in TEST() local
142 const int kSamplesPer10Ms = kSampleRate * 10 / 1000; in TEST()
144 RunFineBufferTest(kSampleRate, kFrameSizeSamples); in TEST()
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| /external/webrtc/webrtc/modules/audio_processing/intelligibility/ |
| D | intelligibility_enhancer_unittest.cc | 78 const int kSampleRate = 1000; variable
80 const int kFragmentSize = kSampleRate / 100;
91 config_.sample_rate_hz = kSampleRate; in IntelligibilityEnhancerTest()
96 config_.sample_rate_hz = kSampleRate; in CheckUpdate()
102 enh_->AnalyzeCaptureAudio(&noise_cursor, kSampleRate, kNumChannels); in CheckUpdate()
103 enh_->ProcessRenderAudio(&clear_cursor, kSampleRate, kNumChannels); in CheckUpdate()
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| /external/webrtc/webrtc/common_audio/ |
| D | wav_file_unittest.cc | 138 static const int kSampleRate = 8000; in TEST() local
140 static const size_t kNumSamples = 3 * kSampleRate * kNumChannels; in TEST()
145 const double t = static_cast<double>(i) / (kNumChannels * kSampleRate); in TEST()
152 WavWriter w(outfile, kSampleRate, kNumChannels); in TEST()
153 EXPECT_EQ(kSampleRate, w.sample_rate()); in TEST()
164 EXPECT_EQ(kSampleRate, r.sample_rate()); in TEST()
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| /external/webrtc/webrtc/modules/audio_device/android/ |
| D | audio_manager_unittest.cc | 133 const int kSampleRate = 48000; in TEST_F() local
139 AudioParameters params(kSampleRate, kChannels, kFramesPerBuffer); in TEST_F()
141 EXPECT_EQ(kSampleRate, params.sample_rate()); in TEST_F()
144 EXPECT_EQ(static_cast<size_t>(kSampleRate / 100), in TEST_F()
|
| /external/webrtc/webrtc/modules/audio_processing/beamformer/ |
| D | covariance_matrix_generator_unittest.cc | 147 const int kSampleRate = 16000; in TEST() local
162 kSampleRate, in TEST()
185 const int kSampleRate = 42000; in TEST() local
200 kSampleRate, in TEST()
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| /external/webrtc/webrtc/modules/video_coding/ |
| D | media_optimization_unittest.cc | 21 kSampleRate = 90000 // RTP timestamps per second. enumerator
44 next_timestamp_ += frame_time_ms_ * kSampleRate / 1000; in AddFrameAndAdvanceTime()
|
| /external/brotli/enc/ |
| D | compress_fragment.c | 96 static const size_t kSampleRate = 29; in BuildAndStoreLiteralPrefixCode() local
97 for (i = 0; i < input_size; i += kSampleRate) { in BuildAndStoreLiteralPrefixCode()
100 histogram_total = (input_size + kSampleRate - 1) / kSampleRate; in BuildAndStoreLiteralPrefixCode()
376 static const size_t kSampleRate = 43; in ShouldMergeBlock() local
378 for (i = 0; i < len; i += kSampleRate) { in ShouldMergeBlock()
382 const size_t total = (len + kSampleRate - 1) / kSampleRate; in ShouldMergeBlock()
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| D | encode.c | 507 static const uint32_t kSampleRate = 13; in ShouldCompress() local
510 (double)bytes * kMinEntropy / kSampleRate; in ShouldCompress()
511 size_t t = (bytes + kSampleRate - 1) / kSampleRate; in ShouldCompress()
516 pos += kSampleRate; in ShouldCompress()
|