| /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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