| /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
|
| 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)
|
| 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
|
| playback_thd.py | 50 samplingRate = 44100
52 thd = calc_thd(hostRecording, signalFrequency, samplingRate, 0.02) * 100
|
| /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;
|
| AudioRemote.h | 29 virtual bool prepare(AudioHardware::SamplingRate samplingRate, int volume,
38 AudioHardware::SamplingRate mSamplingRate;
|
| AudioSignalFactory.h | 31 int maxPositive, AudioHardware::SamplingRate samplingRate, int signalFreq, int samples,
|
| AudioHardware.h | 33 enum SamplingRate {
64 virtual bool prepare(SamplingRate samplingRate, int volume, int mode = EModeVoice) = 0;
|
| AudioPlaybackLocal.h | 33 bool doPrepare(AudioHardware::SamplingRate, int samplesInOneGo);
|
| /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);
|
| DummyAudioSource.h | 32 int32_t samplingRate, int32_t channelCount,
60 int32_t samplingRate, int32_t channelCount,
|
| /cts/suite/audio_quality/lib/src/audio/ |
| AudioRemote.cpp | 21 bool AudioRemote::prepare(AudioHardware::SamplingRate samplingRate, int volume, int mode)
27 mSamplingRate = samplingRate;
|
| AudioPlaybackLocal.cpp | 54 bool AudioPlaybackLocal::doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo)
62 config.rate = samplingRate;
|
| AudioRecordingLocal.cpp | 41 bool AudioRecordingLocal::doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo)
49 config.rate = samplingRate;
|
| AudioSignalFactory.cpp | 24 int maxPositive, AudioHardware::SamplingRate samplingRate, int signalFreq,
32 double multiplier = 2.0 * M_PI * (double)signalFreq / samplingRate;
|
| AudioLocal.cpp | 20 bool AudioLocal::prepare(AudioHardware::SamplingRate samplingRate, int gain, int /*mode*/)
37 mSamplingRate = samplingRate;
|
| /cts/suite/audio_quality/test/ |
| AudioLocalTest.cpp | 30 AudioHardware::SamplingRate mSamplingRate;
48 virtual bool doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo) {
|
| AudioSignalFactoryTest.cpp | 26 AudioHardware::SamplingRate samplingRate, int signalFreq, int samples) {
|
| /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;
|
| /frameworks/av/media/libstagefright/codecs/aacenc/inc/ |
| tns_param.h | 32 Word32 samplingRate;
50 void GetTnsMaxBands(Word32 samplingRate, Word16 blockType, Word16* tnsMaxSfb);
|
| /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,
|
| /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...] |
