/cts/suite/audio_quality/test_description/processing/ |
calc_thd.py | 23 def calc_thd(data, signalFrequency, samplingRate, frequencyMargin): 28 baseI = fftLen * signalFrequency * 2 / samplingRate 49 samplingRate = 44100 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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calc_delay.py | 62 samplingRate = 44100 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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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 158 data = getattr(mod, "do_gen_random")(peakAmpl, durationInMSec, samplingRate, fHigh, [all...] |
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 124 data = getattr(mod, "do_gen_random")(peakAmpl, durationInMSec, samplingRate, fHigh,\ 127 (passFail, minVal, maxVal, amp) = do_check_spectrum_playback(data, samplingRate, fLow, [all...] |
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 98 result = do_gen_random(peakAmplitude, durationInMSec, samplingRate, fHigh)
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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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playback_thd.py | 50 samplingRate = 44100 52 thd = calc_thd(hostRecording, signalFrequency, samplingRate, 0.02) * 100
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/cts/suite/audio_quality/lib/include/audio/ |
AudioLocal.h | 36 virtual bool prepare(AudioHardware::SamplingRate samplingRate, int gain, 46 virtual bool doPrepare(AudioHardware::SamplingRate, int samplesInOneGo) = 0; 71 AudioHardware::SamplingRate mSamplingRate;
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AudioRemote.h | 29 virtual bool prepare(AudioHardware::SamplingRate samplingRate, int volume, 38 AudioHardware::SamplingRate mSamplingRate;
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AudioSignalFactory.h | 31 int maxPositive, AudioHardware::SamplingRate samplingRate, int signalFreq, int samples,
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AudioHardware.h | 33 enum SamplingRate { 64 virtual bool prepare(SamplingRate samplingRate, int volume, int mode = EModeVoice) = 0;
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AudioPlaybackLocal.h | 33 bool doPrepare(AudioHardware::SamplingRate, int samplesInOneGo);
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/frameworks/av/libvideoeditor/lvpp/ |
DummyAudioSource.cpp | 32 int32_t samplingRate, int32_t channelCount, 36 return new DummyAudioSource(samplingRate, 44 int32_t samplingRate, int32_t channelCount, 46 : mSamplingRate(samplingRate), 59 ALOGV("samplingRate = %d", samplingRate);
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DummyAudioSource.h | 32 int32_t samplingRate, int32_t channelCount, 60 int32_t samplingRate, int32_t channelCount,
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/cts/suite/audio_quality/lib/src/audio/ |
AudioRemote.cpp | 21 bool AudioRemote::prepare(AudioHardware::SamplingRate samplingRate, int volume, int mode) 27 mSamplingRate = samplingRate;
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AudioPlaybackLocal.cpp | 54 bool AudioPlaybackLocal::doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo) 62 config.rate = samplingRate;
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AudioRecordingLocal.cpp | 41 bool AudioRecordingLocal::doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo) 49 config.rate = samplingRate;
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AudioSignalFactory.cpp | 24 int maxPositive, AudioHardware::SamplingRate samplingRate, int signalFreq, 32 double multiplier = 2.0 * M_PI * (double)signalFreq / samplingRate;
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AudioLocal.cpp | 20 bool AudioLocal::prepare(AudioHardware::SamplingRate samplingRate, int gain, int /*mode*/) 37 mSamplingRate = samplingRate;
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/cts/suite/audio_quality/test/ |
AudioLocalTest.cpp | 30 AudioHardware::SamplingRate mSamplingRate; 48 virtual bool doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo) {
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AudioSignalFactoryTest.cpp | 26 AudioHardware::SamplingRate samplingRate, int signalFreq, int samples) {
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/external/v8/tools/ |
profile_view.js | 32 * @param {number} samplingRate Number of ms between profiler ticks. 35 function ViewBuilder(samplingRate) { 36 this.samplingRate = samplingRate; 50 var samplingRate = this.samplingRate; 53 var totalWeight = node.totalWeight * samplingRate; 54 var selfWeight = node.selfWeight * samplingRate;
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/frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
tns_param.h | 32 Word32 samplingRate; 50 void GetTnsMaxBands(Word32 samplingRate, Word16 blockType, Word16* tnsMaxSfb);
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/frameworks/av/include/media/ |
AudioSystem.h | 88 static status_t getOutputSamplingRate(int* samplingRate, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); 93 int* samplingRate); 106 static status_t getOutputSamplingRate(int* samplingRate, int stream = AUDIO_STREAM_DEFAULT); 152 : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channels(0), frameCount(0), latency(0) {} 154 uint32_t samplingRate; 189 uint32_t samplingRate = 0, 201 uint32_t samplingRate = 0,
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/frameworks/av/media/libmedia/ |
AudioSystem.cpp | 209 status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType) { 210 return getOutputSamplingRate(samplingRate, (audio_stream_type_t)streamType); 213 status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type_t streamType) 226 return getSamplingRate(output, streamType, samplingRate); 231 int* samplingRate) 242 *samplingRate = af->sampleRate(output); 245 *samplingRate = outputDesc->samplingRate; 249 ALOGV("getSamplingRate() streamType %d, output %d, sampling rate %d", streamType, output, *samplingRate); 452 ALOGV("ioConfigChanged() new output samplingRate %d, format %d channels %d frameCount %d latency %d" [all...] |