| /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/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(),
|
| AudioPortConfig.java | 48 AudioPortConfig(AudioPort port, int samplingRate, int channelMask, int format,
51 mSamplingRate = samplingRate;
68 public int samplingRate() {
|
| /frameworks/av/media/libstagefright/codecs/mp3dec/include/ |
| pvmp3decoder_api.h | 196 int32 samplingRate;
|
| /frameworks/av/services/audiopolicy/common/managerdefinitions/src/ |
| AudioPort.cpp | 160 uint32_t samplingRate = UINT_MAX;
162 if ((samplingRates[i] < samplingRate) && (samplingRates[i] > 0)) {
163 samplingRate = samplingRates[i];
166 pickedRate = (samplingRate == UINT_MAX) ? 0 : samplingRate;
298 void AudioPort::pickAudioProfile(uint32_t &samplingRate,
303 samplingRate = 0;
333 samplingRate = pickedSamplingRate;
339 samplingRate, channelMask, format);
|
| /cts/tests/tests/media/src/android/media/cts/ |
| VisualizerTest.java | 84 int samplingRate = mVisualizer.getSamplingRate();
504 Visualizer visualizer, byte[] waveform, int samplingRate) {
516 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/aac/libAACenc/src/ |
| aacenc_tns.cpp | 148 INT samplingRate;
225 const INT samplingRate,
283 if (sampleRate >= pMaxBandsTab[i].samplingRate) {
[all...] |
| aacenc_lib.cpp | 246 ULONG samplingRate; /*!< Encoder output sampling rate. */
286 * \param samplingRate Audio signal sampling rate.
296 const ULONG samplingRate,
308 if ( (samplingRate <= eldSbrAutoConfigTab[i].samplingRate)
490 cc->samplingRate = hAacConfig->sampleRate;
620 const INT samplingRate,
637 coreSamplingRate = samplingRate >> (sbrEncoder_IsSingleRatePossible(aot) ? (sbrDownSampleRate-1):1);
639 coreSamplingRate = samplingRate;
[all...] |
| /frameworks/base/media/java/android/media/audiofx/ |
| Visualizer.java | 553 * @param samplingRate sampling rate of the audio visualized.
555 void onWaveFormDataCapture(Visualizer visualizer, byte[] waveform, int samplingRate);
564 * @param samplingRate sampling rate of the audio visualized.
566 void onFftDataCapture(Visualizer visualizer, byte[] fft, int samplingRate);
666 int samplingRate = msg.arg1;
670 l.onWaveFormDataCapture(mVisualizer, data, samplingRate);
673 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,
|
| /frameworks/av/media/libmedia/ |
| IAudioPolicyService.cpp | 149 uint32_t samplingRate,
158 data.writeInt32(samplingRate);
178 uint32_t samplingRate,
215 data.writeInt32(samplingRate);
285 uint32_t samplingRate,
305 data.writeInt32(samplingRate);
[all...] |
| /frameworks/av/media/libstagefright/mpeg2ts/ |
| ESQueue.cpp | 154 unsigned samplingRate = samplingRateTable[fscod];
166 (*metaData)->setInt32(kKeySampleRate, samplingRate);
945 int samplingRate, numChannels, bitrate, numSamples;
947 header, &frameSize, &samplingRate, &numChannels,
997 mFormat->setInt32(kKeySampleRate, samplingRate);
[all...] |
| /frameworks/av/media/libeffects/preprocessing/ |
| PreProcessing.cpp | 109 uint32_t samplingRate; // sampling rate at effect process interface
854 session->samplingRate = kPreprocDefaultSr;
940 if (config->inputCfg.samplingRate != config->outputCfg.samplingRate ||
947 config->inputCfg.samplingRate, config->inputCfg.channels);
[all...] |
| /hardware/libhardware/include/hardware/ |
| audio_effect.h | [all...] |
| /hardware/libhardware_legacy/audio/ |
| AudioPolicyManagerBase.cpp | 505 uint32_t samplingRate,
517 if (profile->isCompatibleProfile(device, samplingRate, format,
525 if (profile->isCompatibleProfile(device, samplingRate, format,
539 uint32_t samplingRate,
549 ALOGV("getOutput() device %d, stream %d, samplingRate %d, format %x, channelMask %x, flags %x",
550 device, stream, samplingRate, format, channelMask, flags);
554 ALOGV("getOutput() test output mCurOutput %d, samplingRate %d, format %d, channelMask %x, mDirectOutput %d",
603 samplingRate,
617 if ((samplingRate == outputDesc->mSamplingRate) &&
632 outputDesc->mSamplingRate = samplingRate;
[all...] |
| /external/aac/libSYS/include/ |
| FDK_audio.h | 332 INT samplingRate; /**< Sampling rate. */
|
