/frameworks/base/media/java/android/media/ |
AudioDevicePortConfig.java | 29 AudioDevicePortConfig(AudioDevicePort devicePort, int samplingRate, int channelMask, 31 super((AudioPort)devicePort, samplingRate, channelMask, format, gain); 35 this(config.port(), config.samplingRate(), config.channelMask(), config.format(),
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AudioMixPortConfig.java | 29 AudioMixPortConfig(AudioMixPort mixPort, int samplingRate, int channelMask, int format, 31 super((AudioPort)mixPort, samplingRate, channelMask, format, gain);
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AudioMixPort.java | 38 public AudioMixPortConfig buildConfig(int samplingRate, int channelMask, int format, 40 return new AudioMixPortConfig(this, samplingRate, channelMask, format, gain);
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AudioPortConfig.java | 48 AudioPortConfig(AudioPort port, int samplingRate, int channelMask, int format, 51 mSamplingRate = samplingRate; 68 public int samplingRate() {
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AudioDevicePort.java | 73 public AudioDevicePortConfig buildConfig(int samplingRate, int channelMask, int format, 75 return new AudioDevicePortConfig(this, samplingRate, channelMask, format, gain);
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AudioPort.java | 151 public AudioPortConfig buildConfig(int samplingRate, int channelMask, int format, 153 return new AudioPortConfig(this, samplingRate, channelMask, format, gain);
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/external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/adaptivestreaming/ |
AudioQuality.java | 23 long 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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/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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playback_sample.py | 46 samplingRate = 44100 48 freq = calc_freq(hostRecording, samplingRate) 61 def calc_freq(recording, samplingRate):
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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...] |
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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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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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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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...] |
playback_thd.py | 50 samplingRate = 44100 52 thd = calc_thd(hostRecording, signalFrequency, samplingRate, 0.02) * 100
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/external/aac/libMpegTPEnc/src/ |
tpenc_adif.h | 101 INT samplingRate;
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/frameworks/base/media/java/android/media/tv/ |
ITvInputHardware.aidl | 57 * @param samplingRate desired sampling rate. Use default when it's 0. 62 void overrideAudioSink(int audioType, String audioAddress, int samplingRate, int channelMask,
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/cts/suite/audio_quality/lib/include/audio/ |
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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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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/cts/suite/audio_quality/lib/src/audio/ |
AudioRecordingLocal.cpp | 41 bool AudioRecordingLocal::doPrepare(AudioHardware::SamplingRate samplingRate, int samplesInOneGo) 49 config.rate = samplingRate;
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AudioRemote.cpp | 21 bool AudioRemote::prepare(AudioHardware::SamplingRate samplingRate, int volume, int mode) 27 mSamplingRate = samplingRate;
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/external/aac/libAACdec/src/ |
channelinfo.cpp | 236 UINT samplingRate 243 t->samplingRate = samplingRate;
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