| /external/webrtc/webrtc/common_audio/ |
| D | audio_util_unittest.cc | 110 const int kNumChannels = 2; in TEST() local
111 const size_t kLength = kSamplesPerChannel * kNumChannels; in TEST()
114 Deinterleave(kInterleaved, kSamplesPerChannel, kNumChannels, deinterleaved); in TEST()
121 Interleave(deinterleaved, kSamplesPerChannel, kNumChannels, interleaved); in TEST()
128 const int kNumChannels = 1; in TEST() local
131 Deinterleave(kInterleaved, kSamplesPerChannel, kNumChannels, deinterleaved); in TEST()
135 Interleave(deinterleaved, kSamplesPerChannel, kNumChannels, interleaved); in TEST()
142 const int kNumChannels = 1; in TEST() local
143 const int16_t interleaved[kNumChannels * kNumFrames] = {1, 2, -1, -3}; in TEST()
146 DownmixInterleavedToMono(interleaved, kNumFrames, kNumChannels, in TEST()
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| D | audio_ring_buffer_unittest.cc | 93 const size_t kNumChannels = 1; in TEST_F() local
96 ChannelBuffer<float> input(kNumFrames, kNumChannels); in TEST_F()
98 AudioRingBuffer buf(kNumChannels, kNumFrames); in TEST_F()
99 buf.Write(input.channels(), kNumChannels, kNumFrames); in TEST_F()
102 ChannelBuffer<float> output(1, kNumChannels); in TEST_F()
103 buf.Read(output.channels(), kNumChannels, 1); in TEST_F()
106 buf.Read(output.channels(), kNumChannels, 1); in TEST_F()
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| D | wav_file_unittest.cc | 139 static const size_t kNumChannels = 2; in TEST() local
140 static const size_t kNumSamples = 3 * kSampleRate * kNumChannels; in TEST()
142 for (size_t i = 0; i < kNumSamples; i += kNumChannels) { in TEST()
145 const double t = static_cast<double>(i) / (kNumChannels * kSampleRate); in TEST()
152 WavWriter w(outfile, kSampleRate, kNumChannels); in TEST()
154 EXPECT_EQ(kNumChannels, w.num_channels()); in TEST()
165 EXPECT_EQ(kNumChannels, r.num_channels()); in TEST()
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| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| D | time_stretch_unittest.cc | 29 const size_t kNumChannels = 1; variable
35 BackgroundNoise bgn(kNumChannels); in TEST()
36 Accelerate accelerate(kSampleRate, kNumChannels, bgn); in TEST()
38 kSampleRate, kNumChannels, bgn, kOverlapSamples); in TEST()
44 BackgroundNoise bgn(kNumChannels); in TEST()
48 accelerate_factory.Create(kSampleRate, kNumChannels, bgn); in TEST()
54 kSampleRate, kNumChannels, bgn, kOverlapSamples); in TEST()
67 background_noise_(kNumChannels) { in TimeStretchTest()
79 Accelerate accelerate(sample_rate_hz_, kNumChannels, background_noise_); in TestAccelerate()
82 AudioMultiVector output(kNumChannels); in TestAccelerate()
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| /external/webrtc/webrtc/common_audio/vad/ |
| D | vad_filterbank_unittest.cc | 29 static const int16_t kFeatures[kNumValidFrameLengths * kNumChannels] = { in TEST_F()
34 static const int16_t kOffsetVector[kNumChannels] = { in TEST_F()
36 int16_t features[kNumChannels]; in TEST_F()
52 for (int k = 0; k < kNumChannels; ++k) { in TEST_F()
53 EXPECT_EQ(kFeatures[k + frame_length_index * kNumChannels], in TEST_F()
68 for (int k = 0; k < kNumChannels; ++k) { in TEST_F()
84 for (int k = 0; k < kNumChannels; ++k) { in TEST_F()
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| D | vad_core.c | 20 static const int16_t kSpectrumWeight[kNumChannels] = { 6, 8, 10, 12, 14, 16 };
25 static const int16_t kMinimumDifference[kNumChannels] = {
28 static const int16_t kMaximumSpeech[kNumChannels] = {
33 static const int16_t kMaximumNoise[kNumChannels] = {
107 data[k * kNumChannels] += offset; in WeightedAverage()
108 weighted_average += data[k * kNumChannels] * weights[k * kNumChannels]; in WeightedAverage()
178 for (channel = 0; channel < kNumChannels; channel++) { in GmmProbability()
185 gaussian = channel + k * kNumChannels; in GmmProbability()
247 ngprvec[channel + kNumChannels] = 16384 - ngprvec[channel]; in GmmProbability()
261 sgprvec[channel + kNumChannels] = 16384 - sgprvec[channel]; in GmmProbability()
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| D | vad_core.h | 22 enum { kNumChannels = 6 }; // Number of frequency bands (named channels). enumerator
24 enum { kTableSize = kNumChannels * kNumGaussians };
42 int16_t index_vector[16 * kNumChannels];
43 int16_t low_value_vector[16 * kNumChannels];
45 int16_t mean_value[kNumChannels];
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| D | vad_sp_unittest.cc | 64 for (int j = 0; j < kNumChannels; ++j) { in TEST_F()
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| D | vad_sp.c | 75 assert(channel < kNumChannels); in WebRtcVad_FindMinimum()
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| D | vad_filterbank.c | 265 assert(4 < kNumChannels - 1); // Checking maximum |frequency_band|. in WebRtcVad_CalculateFeatures()
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| /external/webrtc/webrtc/common_audio/resampler/ |
| D | resampler_unittest.cc | 20 const int kNumChannels[] = {1, 2}; variable
21 const size_t kNumChannelsSize = sizeof(kNumChannels) / sizeof(*kNumChannels);
79 << ", channels: " << kNumChannels[k]; in TEST_F()
82 EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kNumChannels[k])); in TEST_F()
84 EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kNumChannels[k])); in TEST_F()
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| /external/webrtc/webrtc/modules/audio_processing/intelligibility/test/ |
| D | intelligibility_proc.cc | 71 const size_t kNumChannels = 1; variable
125 enh.AnalyzeCaptureAudio(&noise_cursor, FLAGS_sample_rate, kNumChannels); in void_main()
126 enh.ProcessRenderAudio(&clear_cursor, FLAGS_sample_rate, kNumChannels); in void_main()
135 WavWriter out_file(temp_out_filename, FLAGS_sample_rate, kNumChannels); in void_main()
141 WavWriter out_file(FLAGS_out_file, FLAGS_sample_rate, kNumChannels); in void_main()
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| /external/webrtc/webrtc/modules/audio_device/android/ |
| D | audio_common.h | 17 const int kNumChannels = 1; variable
20 const size_t kBytesPerFrame = kNumChannels * (16 / 8);
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| D | opensles_common.cc | 17 using webrtc::kNumChannels;
24 configuration.numChannels = kNumChannels; in CreatePcmConfiguration()
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| /external/webrtc/webrtc/modules/audio_processing/vad/ |
| D | voice_activity_detector.cc | 21 const size_t kNumChannels = 1; variable
46 resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels), in ProcessChunk()
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| /external/webrtc/webrtc/modules/audio_processing/transient/ |
| D | transient_suppressor_unittest.cc | 19 static const int kNumChannels = 1; in TEST() local
22 ts.Initialize(ts::kSampleRate16kHz, ts::kSampleRate16kHz, kNumChannels); in TEST()
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| /external/webrtc/webrtc/modules/audio_processing/intelligibility/ |
| D | intelligibility_enhancer_unittest.cc | 79 const int kNumChannels = 1; variable
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/modules/audio_processing/agc/ |
| D | agc_manager_direct_unittest.cc | 33 const int kNumChannels = 1; variable
90 manager_.AnalyzePreProcess(nullptr, kNumChannels, kSamplesPerChannel); in CallPreProc()
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| /external/webrtc/webrtc/modules/audio_processing/test/ |
| D | audio_processing_unittest.cc | 2363 const size_t kNumChannels[] = {1, 2}; in SetUpTestCase() local
2365 for (size_t j = 0; j < arraysize(kNumChannels); ++j) { in SetUpTestCase()
2366 for (size_t k = 0; k < arraysize(kNumChannels); ++k) { in SetUpTestCase()
2369 kNativeRates[i], kNumChannels[j], kNumChannels[j], in SetUpTestCase()
2370 kNumChannels[k], kNumChannels[k], "ref"); in SetUpTestCase()
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