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  /external/aac/libMpegTPEnc/src/
tpenc_adif.h 101 INT samplingRate;
  /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)
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)
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...]
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
  /external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/adaptivestreaming/
AudioQuality.java 23 long 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 93 static status_t getOutputSamplingRate(uint32_t* samplingRate,
101 uint32_t* samplingRate);
158 : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) {}
160 uint32_t samplingRate;
197 uint32_t samplingRate = 0,
210 uint32_t samplingRate = 0,
  /frameworks/av/media/libstagefright/codecs/mp3dec/include/
pvmp3decoder_api.h 196 int32 samplingRate;
  /cts/tests/tests/media/src/android/media/cts/
VisualizerTest.java 83 int samplingRate = mVisualizer.getSamplingRate();
284 Visualizer visualizer, byte[] waveform, int samplingRate) {
296 Visualizer visualizer, byte[] fft, int samplingRate) {
  /external/aac/libAACdec/src/
channelinfo.h 145 UINT samplingRate;
298 AAC_DECODER_ERROR getSamplingRateInfo(SamplingRateInfo *t, UINT samplesPerFrame, UINT samplingRateIndex, UINT samplingRate);
  /external/chromium_org/third_party/WebKit/Source/devtools/front_end/
CPUProfileView.js 580 var samplingRate = profile.totalHitCount / durationMs;
583 node.selfTime = node.hitCount * samplingRate;
    [all...]
  /frameworks/av/media/libmedia/
IAudioPolicyService.cpp 127 uint32_t samplingRate,
136 data.writeInt32(samplingRate);
186 uint32_t samplingRate,
194 data.writeInt32(samplingRate);
456 uint32_t samplingRate = data.readInt32();
467 samplingRate,
508 uint32_t samplingRate = data.readInt32();
513 samplingRate,
IAudioFlinger.cpp 393 uint32_t samplingRate = pSamplingRate != NULL ? *pSamplingRate : 0;
401 data.writeInt32(samplingRate);
417 samplingRate = reply.readInt32();
418 if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
474 uint32_t samplingRate = pSamplingRate != NULL ? *pSamplingRate : 0;
482 data.writeInt32(samplingRate);
489 samplingRate = reply.readInt32();
490 if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
    [all...]
  /frameworks/av/media/libstagefright/mpeg2ts/
ESQueue.cpp 654 int samplingRate, numChannels, bitrate, numSamples;
656 header, &frameSize, &samplingRate, &numChannels,
699 mFormat->setInt32(kKeySampleRate, samplingRate);
  /frameworks/base/media/java/android/media/audiofx/
Visualizer.java 552 * @param samplingRate sampling rate of the audio visualized.
554 void onWaveFormDataCapture(Visualizer visualizer, byte[] waveform, int samplingRate);
563 * @param samplingRate sampling rate of the audio visualized.
565 void onFftDataCapture(Visualizer visualizer, byte[] fft, int samplingRate);
665 int samplingRate = msg.arg1;
669 l.onWaveFormDataCapture(mVisualizer, data, samplingRate);
672 l.onFftDataCapture(mVisualizer, data, samplingRate);
  /frameworks/base/media/tests/MediaFrameworkTest/src/com/android/mediaframeworktest/functional/audio/
MediaVisualizerTest.java 138 int samplingRate = mVisualizer.getSamplingRate();
140 samplingRate >= MIN_SAMPLING_RATE && samplingRate <= MAX_SAMPLING_RATE);
584 Visualizer visualizer, byte[] waveform, int samplingRate) {
596 Visualizer visualizer, byte[] fft, int samplingRate) {
  /cts/suite/audio_quality/client/src/com/android/cts/audiotest/
AudioProtocol.java 237 final int samplingRate = mDataBuffer.getInt(1 * 4);
247 if (samplingRate != 44100) {
262 int bufferSize = AudioTrack.getMinBufferSize(samplingRate,
272 mPlayback = new AudioTrack(type, samplingRate,
333 final int samplingRate = mDataBuffer.getInt(0);
339 if (samplingRate != 44100) {
351 int minBufferSize = AudioRecord.getMinBufferSize(samplingRate,
354 mRecord = new AudioRecord(type, samplingRate,
  /external/aac/libAACenc/src/
aacenc_lib.cpp 240 ULONG samplingRate; /*!< Encoder output sampling rate. */
280 * \param samplingRate Audio signal sampling rate.
290 const ULONG samplingRate,
302 if ( (samplingRate <= eldSbrAutoConfigTab[i].samplingRate)
506 cc->samplingRate = hAacConfig->sampleRate;
644 const INT samplingRate,
661 coreSamplingRate = samplingRate >> (sbrEncoder_IsSingleRatePossible(aot) ? (sbrDownSampleRate-1):1);
663 coreSamplingRate = samplingRate;
    [all...]
aacenc_tns.cpp 168 INT samplingRate;
245 const INT samplingRate,
303 if (sampleRate >= pMaxBandsTab[i].samplingRate) {
    [all...]
  /frameworks/av/media/libeffects/preprocessing/
PreProcessing.cpp 108 uint32_t samplingRate; // sampling rate at effect process interface
799 session->samplingRate = kPreprocDefaultSr;
885 if (config->inputCfg.samplingRate != config->outputCfg.samplingRate ||
892 config->inputCfg.samplingRate, config->inputCfg.channels);
897 if (session->samplingRate != config->inputCfg.samplingRate ||
907 if (config->inputCfg.samplingRate >= 32000 && !(session->createdMsk & (1 << PREPROC_AEC))) {
910 if (config->inputCfg.samplingRate >= 16000) {
912 } else if (config->inputCfg.samplingRate >= 8000)
    [all...]
  /hardware/libhardware/include/hardware/
audio_effect.h     [all...]
  /hardware/libhardware_legacy/audio/
AudioPolicyManagerBase.cpp 497 uint32_t samplingRate,
509 if (profile->isCompatibleProfile(device, samplingRate, format,
517 if (profile->isCompatibleProfile(device, samplingRate, format,
531 uint32_t samplingRate,
541 ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
542 device, stream, samplingRate, format, channelMask, flags);
546 ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
586 samplingRate,
598 if ((samplingRate == outputDesc->mSamplingRate) &&
613 outputDesc->mSamplingRate = samplingRate;
    [all...]
  /external/aac/libSYS/include/
FDK_audio.h 334 INT samplingRate; /**< Sampling rate. */

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