| /cts/suite/audio_quality/test_description/processing/ |
| D | calc_thd.py | 23 def calc_thd(data, signalFrequency, samplingRate, frequencyMargin): argument
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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| D | check_spectrum_playback.py | 38 def do_check_spectrum_playback(hostData, samplingRate, fLow, fHigh, margainLow, margainHigh): argument
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, fLow,\
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| D | check_spectrum.py | 39 def do_check_spectrum(hostData, DUTData, samplingRate, fLow, fHigh, margainLow, margainHigh): argument
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, fHigh,\
[all …]
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| 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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| D | gen_random.py | 30 def do_gen_random(peakAmpl, durationInMSec, samplingRate, fHigh, stereo=True): argument
31 samples = durationInMSec * samplingRate / 1000
36 iHigh = freqSamples * fHigh * 2 / samplingRate + 1
94 samplingRate = 44100 variable
98 result = do_gen_random(peakAmplitude, durationInMSec, samplingRate, fHigh)
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| D | playback_sample.py | 46 samplingRate = 44100
48 freq = calc_freq(hostRecording, samplingRate)
61 def calc_freq(recording, samplingRate): argument
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| 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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| D | playback_thd.py | 50 samplingRate = 44100
52 thd = calc_thd(hostRecording, signalFrequency, samplingRate, 0.02) * 100
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| /cts/apps/CtsVerifier/src/com/android/cts/verifier/audio/ |
| D | Correlation.java | 35 public void init(int blockSize, int samplingRate) { in init() argument
37 mSamplingRate = samplingRate; in init()
40 public boolean computeCorrelation(double [] data, int samplingRate) { in computeCorrelation() argument
43 mSamplingRate = samplingRate; in computeCorrelation()
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| D | NativeAudioThread.java | 70 public void setParams(int samplingRate, int playBufferInBytes, int recBufferInBytes, in setParams() argument
72 mSamplingRate = samplingRate; in setParams()
92 public native long slesInit(int samplingRate, int frameCount, int micSource, in slesInit() argument
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| /cts/apps/CtsVerifier/jni/audio_loopback/ |
| D | sles.h | 80 int slesInit(sles_data ** ppSles, int samplingRate, int frameCount,
89 int slesCreateServer(sles_data *pSles, int samplingRate, int frameCount, int micSource, int numFram…
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| D | jni_sles.c | 25 (JNIEnv *env __unused, jobject obj __unused, jint samplingRate, jint frameCount, in Java_com_android_cts_verifier_audio_NativeAudioThread_slesInit() argument
30 if (slesInit(&pSles, samplingRate, frameCount, micSource, numFramesToIgnore) != SLES_FAIL) { in Java_com_android_cts_verifier_audio_NativeAudioThread_slesInit()
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| D | sles.cpp | 38 int slesInit(sles_data ** ppSles, int samplingRate, int frameCount, int micSource, in slesInit() argument
48 SLES_PRINTF("creating server. Sampling rate =%d, frame count = %d",samplingRate, in slesInit()
50 status = slesCreateServer(pSles, samplingRate, frameCount, micSource, in slesInit()
215 int slesCreateServer(sles_data *pSles, int samplingRate, int frameCount, in slesCreateServer() argument
265 pSles->sampleRate = samplingRate;//48000; // -s# in slesCreateServer()
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| /cts/suite/audio_quality/lib/src/audio/ |
| D | AudioRemote.cpp | 21 bool AudioRemote::prepare(AudioHardware::SamplingRate samplingRate, int volume, int mode) in prepare() argument
27 mSamplingRate = samplingRate; in prepare()
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| D | AudioPlaybackLocal.cpp | 54 bool AudioPlaybackLocal::doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo) in doPrepare() argument
62 config.rate = samplingRate; in doPrepare()
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| D | AudioRecordingLocal.cpp | 41 bool AudioRecordingLocal::doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo) in doPrepare() argument
49 config.rate = samplingRate; in doPrepare()
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| D | AudioLocal.cpp | 20 bool AudioLocal::prepare(AudioHardware::SamplingRate samplingRate, int gain, int /*mode*/) in prepare() argument
37 mSamplingRate = samplingRate; in prepare()
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| D | AudioSignalFactory.cpp | 24 int maxPositive, AudioHardware::SamplingRate samplingRate, int signalFreq, in generateSineWave() argument
32 double multiplier = 2.0 * M_PI * (double)signalFreq / samplingRate; in generateSineWave()
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| /cts/suite/audio_quality/client/src/com/android/cts/audiotest/ |
| D | AudioProtocol.java | 237 final int samplingRate = mDataBuffer.getInt(1 * 4); in handleStartPlayback() local
247 if (samplingRate != 44100) { in handleStartPlayback()
262 int bufferSize = AudioTrack.getMinBufferSize(samplingRate, in handleStartPlayback()
272 mPlayback = new AudioTrack(type, samplingRate, in handleStartPlayback()
333 final int samplingRate = mDataBuffer.getInt(0); in handleStartRecording() local
339 if (samplingRate != 44100) { in handleStartRecording()
351 int minBufferSize = AudioRecord.getMinBufferSize(samplingRate, in handleStartRecording()
354 mRecord = new AudioRecord(type, samplingRate, in handleStartRecording()
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| /cts/suite/audio_quality/lib/include/audio/ |
| D | AudioSignalFactory.h | 31 int maxPositive, AudioHardware::SamplingRate samplingRate, int signalFreq, int samples,
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| D | AudioHardware.h | 64 virtual bool prepare(SamplingRate samplingRate, int volume, int mode = EModeVoice) = 0;
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| D | AudioRemote.h | 29 virtual bool prepare(AudioHardware::SamplingRate samplingRate, int volume,
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| D | AudioLocal.h | 36 virtual bool prepare(AudioHardware::SamplingRate samplingRate, int gain,
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| /cts/suite/audio_quality/test/ |
| D | AudioLocalTest.cpp | 48 virtual bool doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo) { in doPrepare() argument
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| D | AudioSignalFactoryTest.cpp | 26 AudioHardware::SamplingRate samplingRate, int signalFreq, int samples) { in testSignalBasic() argument
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