| /external/srec/srec/include/ |
| fft.h | 2 * fft.h *
29 /* These are the data objects associated with the FFT and IFFT
36 int fft_perform_and_magsq(fft_info *fft);
37 void do_magsq(fft_info *fft);
39 void configure_fft(fft_info *fft, int size);
40 int place_sample_data(fft_info *fft, fftdata *seq, fftdata *smooth, int num);
41 void unconfigure_fft(fft_info *fft);
|
| /external/chromium_org/third_party/openmax_dl/dl/sp/src/test/ |
| gensig.c | 25 * Generate a test signal and compute the theoretical FFT.
28 * is saved in |x| with the corresponding FFT in |fft|. The size of
37 struct ComplexFloat* fft,
55 fft[0].Re = signal_value * size;
56 fft[0].Im = real_only ? 0 : signal_value * size;
59 fft[k].Re = fft[k].Im = 0;
75 fft[0].Re = factor * size * (size + 1) / 2;
76 fft[0].Im = 0
[all...] |
| gensig.h | 26 * Generate a test signal and corresponding FFT.
29 struct ComplexFloat* fft,
|
| /external/chromium_org/third_party/webrtc/common_audio/signal_processing/ |
| real_fft_unittest.cc | 21 // FFT order.
23 // Lengths for real FFT's time and frequency bufffers.
24 // For N-point FFT, the length requirements from API are N and N+2 respectively.
27 // For complex FFT's time and freq buffer. The implementation requires
46 RealFFT* fft = WebRtcSpl_CreateRealFFT(11); local
47 EXPECT_TRUE(fft == NULL);
48 fft = WebRtcSpl_CreateRealFFT(-1);
49 EXPECT_TRUE(fft == NULL);
57 // One common buffer for complex FFT's time and frequency data.
60 // Prepare the inputs to forward FFT's
68 RealFFT* fft = WebRtcSpl_CreateRealFFT(kOrder); local
[all...] |
| /external/eigen/unsupported/test/ |
| FFTW.cpp | 11 #include <unsupported/Eigen/FFT>
80 typedef typename FFT<T>::Complex Complex;
81 typedef typename FFT<T>::Scalar Scalar;
85 FFT<T> fft; local
93 fft.SetFlag(fft.HalfSpectrum );
94 fft.fwd( freqBuf,tbuf);
98 fft.ClearFlag(fft.HalfSpectrum )
146 FFT<T> fft; local
217 FFT<float> fft; local
[all...] |
| /external/eigen/bench/ |
| benchFFT.cpp | 17 #include <unsupported/Eigen/FFT>
51 FFT< Scalar > fft; local
54 fft.SetFlag(fft.Unscaled);
58 fft.SetFlag(fft.HalfSpectrum);
64 fft.fwd( outbuf , inbuf);
72 fft.fwd( outbuf , inbuf);
75 fft.inv(inbuf,outbuf)
[all...] |
| /external/webrtc/src/modules/audio_processing/aecm/ |
| aecm_core_neon.c | 37 static void WindowAndFFTNeon(WebRtc_Word16* fft,
52 // fft[j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((time_signal[i]
61 __asm__("vst2.16 {d20, d21}, [%0, :128]" : : "r"(&fft[j]) : "q10");
63 // fft[PART_LEN2 + j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
73 __asm__("vst2.16 {d20, d21}, [%0, :128]" : : "r"(&fft[PART_LEN2 + j]) : "q10");
76 WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
77 WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
81 __asm__("vld2.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&fft[j]) : "q10", "q11");
90 WebRtc_Word16* fft,
99 // We overwrite two more elements in fft[], but it's ok
[all...] |
| /external/srec/srec/cfront/ |
| spec_anl.c | 79 ** The "NEW" fft performs shifting operations in fixed point, to maximise
83 channel->shift += place_sample_data(&freqobj->fft, channel->prebuff,
89 write_scaled_frames(freqobj->fft.size, 1, freqobj->fft.real, D_FIXED, (float)(0x01 << channel->shift));
90 write_scaled_frames(freqobj->fft.size, 1, freqobj->fft.imag, D_FIXED, (float)(0x01 << channel->shift));
94 write_scaled_frames(freqobj->fft.size, 1, freqobj->fft.real, D_FIXED, (float)1 / (0x01 << -channel->shift));
95 write_scaled_frames(freqobj->fft.size, 1, freqobj->fft.imag, D_FIXED, (float)1 / (0x01 << -channel->shift))
[all...] |
| sp_fft.c | 27 ** DESCRIPTION: Split-Radix FFT
140 /* cannot do smaller than 4 point FFT */
356 Compute a four point FFT that requires no multiplications
404 Compute a two point FFT that requires no multiplications
596 /* do a complex FFT of half size using the even indexed data
655 ** FFT will grow data log2Length. In order to avoid data overflow,
682 /* compute the real input fft, the real valued first and last component of
688 /* After fft, we now have the data,
692 ** to get fft data, we then need to reverse-shift the fixed data by the
706 ** = fftdata magnitude/FFT lengt
[all...] |
| /external/aac/libFDK/src/ |
| FDK_hybrid.cpp | 94 #include "fft.h"
502 FIXP_DBL fft[8]; local
524 /* write to fft coefficient n' */
525 fft[FFT_IDX_R(0)] = ( fMult(p[10], ( fMultSub(fMultDiv2(cr[ 2], pQmfReal[pReadIdx[ 2]]), ci[ 2], pQmfImag[pReadIdx[ 2]]))) +
528 fft[FFT_IDX_I(0)] = ( fMult(p[10], ( fMultAdd(fMultDiv2(ci[ 2], pQmfReal[pReadIdx[ 2]]), cr[ 2], pQmfImag[pReadIdx[ 2]]))) +
533 fft[FFT_IDX_R(1)] = ( fMult(p[ 9], ( fMultSub(fMultDiv2(cr[ 3], pQmfReal[pReadIdx[ 3]]), ci[ 3], pQmfImag[pReadIdx[ 3]]))) +
536 fft[FFT_IDX_I(1)] = ( fMult(p[ 9], ( fMultAdd(fMultDiv2(ci[ 3], pQmfReal[pReadIdx[ 3]]), cr[ 3], pQmfImag[pReadIdx[ 3]]))) +
541 fft[FFT_IDX_R(2)] = ( fMult(p[12], ( fMultSub(fMultDiv2(cr[ 0], pQmfReal[pReadIdx[ 0]]), ci[ 0], pQmfImag[pReadIdx[ 0]]))) +
545 fft[FFT_IDX_I(2)] = ( fMult(p[12], ( fMultAdd(fMultDiv2(ci[ 0], pQmfReal[pReadIdx[ 0]]), cr[ 0], pQmfImag[pReadIdx[ 0]]))) +
550 fft[FFT_IDX_R(3)] = ( fMult(p[11], ( fMultSub(fMultDiv2(cr[ 1], pQmfReal[pReadIdx[ 1]]), ci[ 1], pQmf (…)
[all...] |
| /external/webrtc/src/modules/audio_processing/aec/ |
| aec_core.c | 287 //static void FilterAdaptationUnconstrained(aec_t *aec, float *fft,
311 static void FilterAdaptation(aec_t *aec, float *fft, float ef[2][PART_LEN1]) {
325 fft[2 * j] = MulRe(aec->xfBuf[0][xPos + j],
328 fft[2 * j + 1] = MulIm(aec->xfBuf[0][xPos + j],
332 fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
336 aec_rdft_inverse_128(fft);
337 memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
339 // fft scaling
343 fft[j] *= scale;
346 aec_rdft_forward_128(fft);
545 float fft[PART_LEN2]; local
639 float fft[PART_LEN2]; local
856 float fft[PART_LEN2]; local
[all...] |
| /cts/suite/audio_quality/test_description/processing/ |
| calc_thd.py | 19 import scipy.fftpack as fft namespace
25 fftData = abs(fft.fft(data * np.hanning(len(data))))
|
| gen_random.py | 20 import scipy.fftpack as fft namespace
34 fftData = fft.rfft(randomSignal)
47 filteredData = fft.irfft(fftData)
49 #plt.plot(freq, abs(fft.fft(filteredData)))
|
| /frameworks/av/include/media/ |
| Visualizer.h | 36 * - Frequency data: 8-bit magnitude FFT by using the getFft() method
39 * getCaptureSize() and setCaptureSize() methods. Note that the size of the FFT
84 // callback used to return periodic PCM or FFT captures to the application. Either one or both
85 // types of data are returned (PCM and FFT) according to flags indicated when installing the
92 uint8_t *fft,
129 // return a capture in FFT 8 bit signed format. The size of the capture is equal to
130 // getCaptureSize() but the length of the FFT is half of the size (both parts of the spectrum
132 status_t getFft(uint8_t *fft);
158 status_t doFft(uint8_t *fft, uint8_t *waveform);
|
| /external/chromium_org/third_party/webrtc/modules/audio_processing/aec/ |
| aec_core.c | 193 // static void FilterAdaptationUnconstrained(AecCore* aec, float *fft,
217 static void FilterAdaptation(AecCore* aec, float* fft, float ef[2][PART_LEN1]) {
231 fft[2 * j] = MulRe(aec->xfBuf[0][xPos + j],
235 fft[2 * j + 1] = MulIm(aec->xfBuf[0][xPos + j],
240 fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
245 aec_rdft_inverse_128(fft);
246 memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
248 // fft scaling
252 fft[j] *= scale;
255 aec_rdft_forward_128(fft);
796 float fft[PART_LEN2]; local
1026 float fft[PART_LEN2]; local
1534 float fft[PART_LEN2]; local
[all...] |
| aec_core_mips.c | 437 float* fft,
461 "addiu %[fft_tmp], %[fft], 0 \n\t"
515 "swc1 %[f8], 4(%[fft]) \n\t"
524 : [fft] "r" (fft)
528 aec_rdft_inverse_128(fft);
529 memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
531 // fft scaling
537 "addiu %[fft_tmp], %[fft], 0 \n\t"
572 : [scale] "f" (scale), [fft] "r" (fft
[all...] |
| aec_core_neon.c | 179 float* fft,
209 vst1q_f32(&fft[2 * j + 0], g_n_h.val[0]);
210 vst1q_f32(&fft[2 * j + 4], g_n_h.val[1]);
213 fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
218 aec_rdft_inverse_128(fft);
219 memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
221 // fft scaling
226 const float32x4_t fft_ps = vld1q_f32(&fft[j]);
228 vst1q_f32(&fft[j], fft_scale);
231 aec_rdft_forward_128(fft);
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_processing/aecm/ |
| aecm_core_neon.c | 47 int16_t* fft,
56 int16_t* p_fft = fft;
57 int16_t* p_fft_offset = &fft[PART_LEN2];
99 // Do forward FFT, then take only the first PART_LEN complex samples,
101 WebRtcSpl_RealForwardFFT(aecm->real_fft, (int16_t*)fft,
118 int16_t* fft,
125 assert((uintptr_t)fft % 16 == 0);
136 int16_t* p_fft = fft;
137 int16_t* p_fft_offset = &fft[PART_LEN4 - 6];
140 // We overwrite two more elements in fft[], but it's ok
[all...] |
| aecm_core_c.c | 66 int16_t* fft,
72 // FFT of signal
75 // transformation array |fft|
76 fft[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
80 fft[PART_LEN + i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(
86 // Do forward FFT, then take only the first PART_LEN complex samples,
88 WebRtcSpl_RealForwardFFT(aecm->real_fft, fft, (int16_t*)freq_signal);
95 int16_t* fft,
102 // Reuse |efw| for the inverse FFT output after transferring
103 // the contents to |fft|
179 int16_t *fft = (int16_t *) (((uintptr_t) fft_buf + 31) & ~31); local
315 int16_t* fft = (int16_t*) (((uintptr_t) fft_buf + 31) & ~ 31); local
[all...] |
| aecm_core_mips.c | 76 int16_t* fft,
88 memset(fft, 0, sizeof(int16_t) * PART_LEN4);
90 // FFT of signal
115 "addu %[store_ptr1], %[fft], %[f_coef] \n\t"
116 "addu %[store_ptr2], %[fft], %[s_coef] \n\t"
138 "addu %[store_ptr1], %[fft], %[f_coef] \n\t"
139 "addu %[store_ptr2], %[fft], %[s_coef] \n\t"
156 [hanning] "r" (WebRtcAecm_kSqrtHanning), [fft] "r" (fft)
160 WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1)
638 int16_t *fft = (int16_t *) (((uintptr_t) fft_buf + 31) & ~31); local
823 int16_t* fft = (int16_t*)(((uint32_t)fft_buf + 31) & ~ 31); local
[all...] |
| /frameworks/base/media/java/android/media/audiofx/ |
| Visualizer.java | 42 * <li>Frequency data: 8-bit magnitude FFT by using the {@link #getFft(byte[])} method</li>
96 * captured data. A low playback volume will lead to low sample and fft values, and vice-versa.
177 * PCM and FFT capture listener registered by client
443 * <p>The capture is an 8-bit magnitude FFT, the frequency range covered being 0 (DC) to half of
478 * @param fft array of bytes where the FFT should be returned
484 public int getFft(byte[] fft)
490 return native_getFft(fft);
557 * <p>Data in the fft buffer is valid only within the scope of the callback.
558 * Applications which needs access to the fft data after returning from the callbac
[all...] |
| /frameworks/av/media/libmedia/ |
| Visualizer.cpp | 286 status_t Visualizer::getFft(uint8_t *fft)
288 if (fft == NULL) {
300 status = doFft(fft, buf);
303 memset(fft, 0, mCaptureSize);
308 status_t Visualizer::doFft(uint8_t *fft, uint8_t *waveform)
326 fft[i] = tmp;
330 fft[i + 1] = tmp;
349 uint8_t fft[mCaptureSize]; local
351 status = doFft(fft, waveform);
365 fftPtr = fft;
[all...] |
| /external/aac/libSBRdec/src/ |
| psdec_hybrid.cpp | 87 #include "fft.h"
213 Implementation using a FFT of length 8
242 Try to split off FFT Modulation Term:
243 FFT(x[t], q) = sum(x[t+k]*exp(-j*2*pi/N *q * k))
255 n m *exp(-j*2*pi) | n' fft
272 now use fft modulation coefficients
273 m[6] = = fft[0]
274 m[7] = = fft[1]
275 m[8] = m[ 0] = fft[2]
276 m[9] = m[ 1] = fft[3
383 FIXP_DBL *fft = (FIXP_DBL *)ALIGN_PTR(_fft); local
[all...] |
| /frameworks/base/media/tests/EffectsTest/src/com/android/effectstest/ |
| VisualizerTest.java | 158 public void onFftDataCapture(Visualizer visualizer, byte[] fft, int samplingRate) {
160 if (fft.length > 0) {
161 Log.d(TAG, "onFftDataCapture(): "+fft[0]);
162 displayVal(R.id.fftMin, fft[0]);
163 displayVal(R.id.fftMax, fft[fft.length - 1]);
164 displayVal(R.id.fftCenter, fft[fft.length/2]);
|
| /external/libvorbis/lib/ |
| psytune.c | 329 float *fft=work[i]; local
335 /* fft and mdct transforms */
337 fft[j]=pcm[i][j]*=window[j];
339 drft_forward(&f_look,fft);
342 fft[0]*=scale;
343 fft[0]=todB(fft);
345 float temp=scale*FAST_HYPOT(fft[j],fft[j+1]);
346 temp=fft[(j+1)>>1]=todB(&temp)
361 float *fft=work[i]; local
[all...] |
