/cts/suite/audio_quality/test_description/processing/ |
H A D | calc_thd.py | 23 def calc_thd(data, signalFrequency, samplingRate, frequencyMargin): 28 baseI = fftLen * signalFrequency * 2 / samplingRate 49 samplingRate = 44100 variable 52 samples = float(samplingRate) * float(durationInSec) 54 time = index / samplingRate 55 multiplier = 2.0 * np.pi * signalFrequency / float(samplingRate) 57 thd = calc_thd(data, signalFrequency, samplingRate, 0.02)
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H A D | calc_delay.py | 62 samplingRate = 44100 variable 67 samples = float(samplingRate) * float(durationInSec) 69 time = index / samplingRate 70 multiplier = 2.0 * np.pi * signalFrequency / float(samplingRate) 72 DELAY = durationInSec / 2.0 * samplingRate
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H A D | check_spectrum_playback.py | 38 def do_check_spectrum_playback(hostData, samplingRate, fLow, fHigh, margainLow, margainHigh): 41 iLow = N * fLow / samplingRate + 1 # 1 for DC 44 iHigh = N * fHigh / samplingRate + 1 # 1 for DC 47 print fLow, iLow, fHigh, iHigh, samplingRate 49 Phh, freqs = plt.psd(hostData, NFFT=N, Fs=samplingRate, Fc=0, detrend=plt.mlab.detrend_none,\ 93 samplingRate = inputData[1] 99 samplingRate, fLow, fHigh, margainLow, margainHigh) 121 samplingRate = 44100 variable 124 data = getattr(mod, "do_gen_random")(peakAmpl, durationInMSec, samplingRate, fHigh,\ 127 (passFail, minVal, maxVal, amp) = do_check_spectrum_playback(data, samplingRate, fLo [all...] |
H A D | gen_random.py | 30 def do_gen_random(peakAmpl, durationInMSec, samplingRate, fHigh, stereo=True): 31 samples = durationInMSec * samplingRate / 1000 36 iHigh = freqSamples * fHigh * 2 / samplingRate + 1 48 #freq = np.linspace(0.0, samplingRate, num=len(fftData), endpoint=False) 94 samplingRate = 44100 variable 98 result = do_gen_random(peakAmplitude, durationInMSec, samplingRate, fHigh)
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H A D | playback_thd.py | 50 samplingRate = 44100 52 thd = calc_thd(hostRecording, signalFrequency, samplingRate, 0.02) * 100
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H A D | recording_thd.py | 60 samplingRate = 44100 65 thdHost = calc_thd(hostRecording[delay:delay+N], signalFrequency, samplingRate, 0.02) * 100 66 thdDevice = calc_thd(deviceRecording, signalFrequency, samplingRate, 0.02) * 100
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H A D | check_spectrum.py | 39 def do_check_spectrum(hostData, DUTData, samplingRate, fLow, fHigh, margainLow, margainHigh): 42 iLow = N * fLow / samplingRate + 1 # 1 for DC 45 iHigh = N * fHigh / samplingRate + 1 # 1 for DC 48 print fLow, iLow, fHigh, iHigh, samplingRate 50 Phh, freqs = plt.psd(hostData, NFFT=N, Fs=samplingRate, Fc=0, detrend=plt.mlab.detrend_none,\ 53 Pdd, freqs = plt.psd(DUTData, NFFT=N, Fs=samplingRate, Fc=0, detrend=plt.mlab.detrend_none,\ 113 samplingRate = inputData[2] 133 samplingRate, fLow, fHigh, margainLow, margainHigh) 155 samplingRate = 44100 variable 158 data = getattr(mod, "do_gen_random")(peakAmpl, durationInMSec, samplingRate, fHig [all...] |
/cts/suite/audio_quality/lib/include/audio/ |
H A D | AudioSignalFactory.h | 31 int maxPositive, AudioHardware::SamplingRate samplingRate, int signalFreq, int samples,
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H A D | AudioHardware.h | 64 virtual bool prepare(SamplingRate samplingRate, int volume, int mode = EModeVoice) = 0;
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H A D | AudioLocal.h | 36 virtual bool prepare(AudioHardware::SamplingRate samplingRate, int gain,
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H A D | AudioRemote.h | 29 virtual bool prepare(AudioHardware::SamplingRate samplingRate, int volume,
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/cts/suite/audio_quality/lib/src/audio/ |
H A D | AudioRecordingLocal.cpp | 41 bool AudioRecordingLocal::doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo) argument 49 config.rate = samplingRate;
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H A D | AudioRemote.cpp | 21 bool AudioRemote::prepare(AudioHardware::SamplingRate samplingRate, int volume, int mode) argument 27 mSamplingRate = samplingRate;
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H A D | AudioLocal.cpp | 20 bool AudioLocal::prepare(AudioHardware::SamplingRate samplingRate, int gain, int /*mode*/) argument 37 mSamplingRate = samplingRate;
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H A D | AudioPlaybackLocal.cpp | 54 bool AudioPlaybackLocal::doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo) argument 62 config.rate = samplingRate;
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H A D | AudioSignalFactory.cpp | 24 int maxPositive, AudioHardware::SamplingRate samplingRate, int signalFreq, 32 double multiplier = 2.0 * M_PI * (double)signalFreq / samplingRate; 23 generateSineWave(AudioHardware::BytesPerSample BPS, int maxPositive, AudioHardware::SamplingRate samplingRate, int signalFreq, int samples, bool stereo) argument
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/cts/suite/audio_quality/test/ |
H A D | AudioLocalTest.cpp | 48 virtual bool doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo) { argument
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H A D | AudioSignalFactoryTest.cpp | 26 AudioHardware::SamplingRate samplingRate, int signalFreq, int samples) { 25 testSignalBasic(android::sp<Buffer>& buffer, int maxPositive, AudioHardware::SamplingRate samplingRate, int signalFreq, int samples) argument
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/cts/suite/audio_quality/client/src/com/android/cts/audiotest/ |
H A D | AudioProtocol.java | 237 final int samplingRate = mDataBuffer.getInt(1 * 4); 247 if (samplingRate != 44100) { 262 int bufferSize = AudioTrack.getMinBufferSize(samplingRate, 272 mPlayback = new AudioTrack(type, samplingRate, 333 final int samplingRate = mDataBuffer.getInt(0); 339 if (samplingRate != 44100) { 351 int minBufferSize = AudioRecord.getMinBufferSize(samplingRate, 354 mRecord = new AudioRecord(type, samplingRate,
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/cts/tests/tests/media/src/android/media/cts/ |
H A D | VisualizerTest.java | 83 int samplingRate = mVisualizer.getSamplingRate(); 284 Visualizer visualizer, byte[] waveform, int samplingRate) { 296 Visualizer visualizer, byte[] fft, int samplingRate) {
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