/external/aac/libMpegTPEnc/src/ |
tpenc_adif.h | 101 INT samplingRate;
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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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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...] |
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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/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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/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,
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/frameworks/av/media/libstagefright/codecs/mp3dec/include/ |
pvmp3decoder_api.h | 196 int32 samplingRate;
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/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) {
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/external/aac/libAACdec/src/ |
channelinfo.h | 145 UINT samplingRate; 298 AAC_DECODER_ERROR getSamplingRateInfo(SamplingRateInfo *t, UINT samplesPerFrame, UINT samplingRateIndex, UINT samplingRate);
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/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,
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IAudioFlinger.cpp | 395 uint32_t samplingRate = pSamplingRate != NULL ? *pSamplingRate : 0; 403 data.writeInt32(samplingRate); 419 samplingRate = reply.readInt32(); 420 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 476 uint32_t samplingRate = pSamplingRate != NULL ? *pSamplingRate : 0; 484 data.writeInt32(samplingRate); 491 samplingRate = reply.readInt32(); 492 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);
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/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);
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/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) {
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/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,
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/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", 595 samplingRate, 609 if ((samplingRate == outputDesc->mSamplingRate) && 624 outputDesc->mSamplingRate = samplingRate; [all...] |
/external/aac/libSYS/include/ |
FDK_audio.h | 334 INT samplingRate; /**< Sampling rate. */
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