| /external/webrtc/webrtc/modules/audio_device/android/ |
| audio_common.h | 17 const int kNumChannels = 1;
20 const size_t kBytesPerFrame = kNumChannels * (16 / 8);
|
| opensles_common.cc | 17 using webrtc::kNumChannels;
24 configuration.numChannels = kNumChannels;
|
| /external/webrtc/webrtc/common_audio/ |
| audio_util_unittest.cc | 110 const int kNumChannels = 2;
111 const size_t kLength = kSamplesPerChannel * kNumChannels;
114 Deinterleave(kInterleaved, kSamplesPerChannel, kNumChannels, deinterleaved);
121 Interleave(deinterleaved, kSamplesPerChannel, kNumChannels, interleaved);
128 const int kNumChannels = 1;
131 Deinterleave(kInterleaved, kSamplesPerChannel, kNumChannels, deinterleaved);
135 Interleave(deinterleaved, kSamplesPerChannel, kNumChannels, interleaved);
142 const int kNumChannels = 1;
143 const int16_t interleaved[kNumChannels * kNumFrames] = {1, 2, -1, -3};
146 DownmixInterleavedToMono(interleaved, kNumFrames, kNumChannels,
[all...] |
| audio_ring_buffer_unittest.cc | 93 const size_t kNumChannels = 1;
96 ChannelBuffer<float> input(kNumFrames, kNumChannels);
98 AudioRingBuffer buf(kNumChannels, kNumFrames);
99 buf.Write(input.channels(), kNumChannels, kNumFrames);
102 ChannelBuffer<float> output(1, kNumChannels);
103 buf.Read(output.channels(), kNumChannels, 1);
106 buf.Read(output.channels(), kNumChannels, 1);
|
| wav_file_unittest.cc | 139 static const size_t kNumChannels = 2;
140 static const size_t kNumSamples = 3 * kSampleRate * kNumChannels;
142 for (size_t i = 0; i < kNumSamples; i += kNumChannels) {
145 const double t = static_cast<double>(i) / (kNumChannels * kSampleRate);
152 WavWriter w(outfile, kSampleRate, kNumChannels);
154 EXPECT_EQ(kNumChannels, w.num_channels());
165 EXPECT_EQ(kNumChannels, r.num_channels());
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| time_stretch_unittest.cc | 29 const size_t kNumChannels = 1;
35 BackgroundNoise bgn(kNumChannels);
36 Accelerate accelerate(kSampleRate, kNumChannels, bgn);
38 kSampleRate, kNumChannels, bgn, kOverlapSamples);
44 BackgroundNoise bgn(kNumChannels);
48 accelerate_factory.Create(kSampleRate, kNumChannels, bgn);
54 kSampleRate, kNumChannels, bgn, kOverlapSamples);
67 background_noise_(kNumChannels) {
79 Accelerate accelerate(sample_rate_hz_, kNumChannels, background_noise_);
82 AudioMultiVector output(kNumChannels);
[all...] |
| /external/webrtc/webrtc/common_audio/vad/ |
| vad_filterbank_unittest.cc | 29 static const int16_t kFeatures[kNumValidFrameLengths * kNumChannels] = {
34 static const int16_t kOffsetVector[kNumChannels] = {
36 int16_t features[kNumChannels];
52 for (int k = 0; k < kNumChannels; ++k) {
53 EXPECT_EQ(kFeatures[k + frame_length_index * kNumChannels],
68 for (int k = 0; k < kNumChannels; ++k) {
84 for (int k = 0; k < kNumChannels; ++k) {
|
| vad_core.h | 22 enum { kNumChannels = 6 }; // Number of frequency bands (named channels).
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];
|
| 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;
108 weighted_average += data[k * kNumChannels] * weights[k * kNumChannels];
118 // - features [i] : Feature vector of length |kNumChannels|
178 for (channel = 0; channel < kNumChannels; channel++) {
185 gaussian = channel + k * kNumChannels;
[all...] |
| vad_sp_unittest.cc | 64 for (int j = 0; j < kNumChannels; ++j) {
|
| vad_sp.c | 75 assert(channel < kNumChannels);
|
| vad_filterbank.c | 265 assert(4 < kNumChannels - 1); // Checking maximum |frequency_band|.
|
| /external/webrtc/webrtc/modules/audio_processing/intelligibility/test/ |
| intelligibility_proc.cc | 71 const size_t kNumChannels = 1;
125 enh.AnalyzeCaptureAudio(&noise_cursor, FLAGS_sample_rate, kNumChannels);
126 enh.ProcessRenderAudio(&clear_cursor, FLAGS_sample_rate, kNumChannels);
135 WavWriter out_file(temp_out_filename, FLAGS_sample_rate, kNumChannels);
141 WavWriter out_file(FLAGS_out_file, FLAGS_sample_rate, kNumChannels);
|
| /external/webrtc/webrtc/common_audio/resampler/ |
| resampler_unittest.cc | 20 const int kNumChannels[] = {1, 2};
21 const size_t kNumChannelsSize = sizeof(kNumChannels) / sizeof(*kNumChannels);
79 << ", channels: " << kNumChannels[k];
82 EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kNumChannels[k]));
84 EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kNumChannels[k]));
|
| /external/webrtc/webrtc/modules/audio_processing/transient/ |
| transient_suppressor_unittest.cc | 19 static const int kNumChannels = 1;
22 ts.Initialize(ts::kSampleRate16kHz, ts::kSampleRate16kHz, kNumChannels);
|
| /external/webrtc/webrtc/modules/audio_processing/vad/ |
| voice_activity_detector.cc | 21 const size_t kNumChannels = 1;
46 resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels),
|
| /external/webrtc/webrtc/modules/audio_processing/intelligibility/ |
| intelligibility_enhancer_unittest.cc | 79 const int kNumChannels = 1;
102 enh_->AnalyzeCaptureAudio(&noise_cursor, kSampleRate, kNumChannels);
103 enh_->ProcessRenderAudio(&clear_cursor, kSampleRate, kNumChannels);
|
| /frameworks/av/cmds/stagefright/ |
| record.cpp | 299 const int32_t kNumChannels = 2;
300 sp<MediaSource> audioSource = new SineSource(kSampleRate, kNumChannels);
317 encMeta->setInt32("channel-count", kNumChannels);
|
| /external/webrtc/webrtc/modules/audio_processing/agc/ |
| agc_manager_direct_unittest.cc | 33 const int kNumChannels = 1;
90 manager_.AnalyzePreProcess(nullptr, kNumChannels, kSamplesPerChannel);
|
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| audio_processing_unittest.cc | [all...] |
