Searched refs:scale (Results 1 - 25 of 1847) sorted by last modified time

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/external/zxing/core/
H A Dcore.jarMETA-INF/ META-INF/MANIFEST.MF com/ com/google/ com/google/zxing/ com/google/zxing/aztec/ ...
/external/webrtc/src/common_audio/signal_processing/
H A Dauto_correlation.c24 int* scale)
45 // In order to avoid overflow when computing the sum we should scale the samples so that
138 *scale = scaling;
20 WebRtcSpl_AutoCorrelation(G_CONST WebRtc_Word16* in_vector, int in_vector_length, int order, WebRtc_Word32* result, int* scale) argument
H A Dcomplex_fft.c289 int i, j, l, k, istep, n, m, scale, shift; local
301 scale = 0;
317 scale++;
323 scale++;
424 return scale;
/external/webrtc/src/common_audio/signal_processing/include/
H A Dsignal_processing_library.h315 int* scale);
679 // Returns the # of bits required to scale the samples specified in the
1080 // - scale : The number of left shifts required to obtain the
1519 // The elements are in Q(-scale) domain, see more on Return
1534 // Return Value : The scale value that tells the number of left bit shifts
1537 // values. The scale parameter is always 0 or positive,
1538 // except if N>1024 (|stages|>10), which returns a scale
1583 // Return value : The scale parameter is always 0, except if N>1024,
1584 // which returns a scale value of -1, indicating error.
/external/webrtc/src/modules/audio_coding/codecs/isac/fix/interface/
H A Disacfix.h547 * - scale : factor for rate change (0.4 ~=> half the rate, 1 no change).
558 float scale,
/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/
H A Dcodec.h47 float scale);
138 WebRtc_Word16* __restrict scale);
152 WebRtc_Word16* __restrict scale);
168 WebRtc_Word16* __restrict scale);
H A Dencode.c328 // scale = 0.55 - (0.8 - bytesLeft / arithLenDFTBytes) * 5 / 6
329 // bytesLeft / arithLenDFTBytes below 0.2 will have a scale of zero and above 0.8 are treated as 0.8
333 // and the corresponding scale is chosen
348 // scale FFT coefficients to reduce the bit-rate
365 // scale the unquantized LPC gains and save the scaled version for the future use
488 float scale)
536 /* If scale < 1, rescale data to produce lower bitrate signal */
537 if ((0.0 < scale) && (scale < 1.0)) {
540 tmpLPCcoeffs_g[ii] = (WebRtc_Word32) ((scale) * (floa
486 WebRtcIsacfix_EncodeStoredData(ISACFIX_EncInst_t *ISACenc_obj, int BWnumber, float scale) argument
[all...]
H A Dfilters.c31 WebRtc_Word16* __restrict scale) {
63 *scale = scaling;
27 WebRtcIsacfix_AutocorrC(WebRtc_Word32* __restrict r, const WebRtc_Word16* __restrict x, WebRtc_Word16 N, WebRtc_Word16 order, WebRtc_Word16* __restrict scale) argument
H A Dfilters_neon.c31 WebRtc_Word16* __restrict scale) {
164 *scale = scaling;
26 WebRtcIsacfix_AutocorrNeon( WebRtc_Word32* __restrict r, const WebRtc_Word16* __restrict x, WebRtc_Word16 N, WebRtc_Word16 order, WebRtc_Word16* __restrict scale) argument
H A Disacfix.c464 float scale,
485 scale);
462 WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst, WebRtc_Word16 bweIndex, float scale, WebRtc_Word16 *encoded) argument
H A Dlpc_masking_model.c81 and with scale factor */
567 WebRtc_Word16 scale; local
657 WebRtcIsacfix_AutocorrFix(corrloQQ,DataLoQ6,WINLEN, ORDERLO+1, &scale);
658 QdomLO = 12-scale; // QdomLO is the Q-domain of corrloQQ
667 WebRtcIsacfix_AutocorrFix(corrhiQQ,DataHiQ6,WINLEN, ORDERHI, &scale);
669 QdomHI = 12-scale; // QdomHI is the Q-domain of corrhiQQ
H A Dpitch_filter.c184 WebRtc_Word16 scale; local
200 scale = 0;
249 scale++;
253 tmp2W32 = WEBRTC_SPL_RSHIFT_W32(tmp2W32, scale);
255 tmpW32 = WEBRTC_SPL_RSHIFT_W32(tmpW32, scale);
/external/webrtc/src/modules/audio_coding/codecs/isac/fix/test/
H A Dkenny.c142 float scale = (float)0.7; local
366 scale = (float)strtod( argv[i+1], NULL );
587 scale,
/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/
H A Dcodec.h47 float scale);
51 WebRtc_Word32 jitterInfo, float scale, enum ISACBandwidth bandwidth);
H A Dencode.c444 /* To be safe, we reduce the scale depending on
562 /* To be safe, we reduce the scale depending on the
1009 float scale) {
1040 if ((scale > 0.0) && (scale < 1.0)) {
1045 tmpLPCcoeffs_lo[ii] = scale * ISACSavedEnc_obj->LPCcoeffs_lo[ii];
1050 tmpLPCcoeffs_hi[ii] = scale * ISACSavedEnc_obj->LPCcoeffs_hi[ii];
1056 tmp_fre[ii] = (WebRtc_Word16)((scale) * (float)ISACSavedEnc_obj->fre[ii]);
1057 tmp_fim[ii] = (WebRtc_Word16)((scale) * (float)ISACSavedEnc_obj->fim[ii]);
1109 if (scale < 1.
1007 WebRtcIsac_EncodeStoredDataLb(const ISAC_SaveEncData_t* ISACSavedEnc_obj, Bitstr* ISACBitStr_obj, int BWnumber, float scale) argument
1134 WebRtcIsac_EncodeStoredDataUb( const ISACUBSaveEncDataStruct* ISACSavedEnc_obj, Bitstr* bitStream, WebRtc_Word32 jitterInfo, float scale, enum ISACBandwidth bandwidth) argument
[all...]
H A Disac.c763 float scale; local
806 scale = (float)pow(10, (gain1 - gain2) / 20.0);
807 /* Change the scale if this is a RCU bit-stream. */
808 scale = (isRCU) ? (scale * RCU_TRANSCODING_SCALE) : scale;
812 &iSACBitStreamInst, bweIndex, scale);
835 scale = (float)pow(10, (gain1 - gain2) / 20.0);
837 /* Change the scale if this is a RCU bit-stream. */
838 scale
[all...]
/external/webrtc/src/modules/audio_processing/aec/
H A Daec_core.c41 static const float cnScaleHband = (float)0.4; // scale for comfort noise in H band
341 float scale = 2.0f / PART_LEN2; local
343 fft[j] *= scale;
637 float scale; local
787 scale = 2.0f / PART_LEN2;
789 y[i] = fft[PART_LEN + i] * scale; // fft scaling
857 float scale, dtmp; local
1131 scale = 2.0f / PART_LEN2;
1133 fft[i] *= scale; // fft scaling
1140 fft[PART_LEN + i] *= scale; // ff
[all...]
H A Daec_core_sse2.c174 float scale = 2.0f / PART_LEN2; local
175 const __m128 scale_ps = _mm_load_ps1(&scale);
/external/webrtc/src/modules/audio_processing/agc/
H A Danalog_agc.c47 /* This is the target level for the analog part in ENV scale. To convert to RMS scale you
52 /* Offset between RMS scale (analog part) and ENV scale (digital part). This value actually
458 micLevelTmp = WEBRTC_SPL_LSHIFT_W32(micLevelIn, stt->scale);
529 *micLevelOut = WEBRTC_SPL_RSHIFT_W32(stt->micGainIdx, stt->scale);
553 /* Set analog target level in envelope dBOv scale */
770 inMicLevelTmp = WEBRTC_SPL_LSHIFT_W32(inMicLevel, stt->scale);
1236 *outMicLevel = WEBRTC_SPL_RSHIFT_W32(stt->micVol, stt->scale);
1237 if (*outMicLevel > WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog, stt->scale))
[all...]
H A Danalog_agc.h74 WebRtc_Word16 analogTarget; // Digital reference level in ENV scale
115 WebRtc_Word16 scale; // Scale factor for internal volume levels member in struct:__anon33486
/external/webrtc/src/modules/audio_processing/utility/
H A Ddelay_estimator_wrapper.c55 // - scale : Scale for smoothing (should be less than 1.0).
61 float scale,
63 assert(scale < 1.0f);
64 *mean_value += (new_value - *mean_value) * scale;
60 MeanEstimatorFloat(float new_value, float scale, float* mean_value) argument
/external/vixl/src/a64/
H A Dassembler-a64.h1657 static Instr FPScale(unsigned scale) {
1658 VIXL_ASSERT(is_uint6(scale));
1659 return scale << FPScale_offset;
/external/webp/src/dsp/
H A Dalpha_processing.c146 const uint32_t scale = GetScale(alpha, inverse); local
148 out |= Mult(argb >> 0, scale) << 0;
149 out |= Mult(argb >> 8, scale) << 8;
150 out |= Mult(argb >> 16, scale) << 16;
166 const uint32_t scale = GetScale(a, inverse); local
167 ptr[x] = Mult(ptr[x], scale);
/external/webp/src/enc/
H A Dpicture_csp.c107 const double scale = 1. / kGammaScale; local
113 const double x = scale * (v << kGammaTabFix);
H A Dwebpenc.c235 const float scale = 1.f + config->quality * 5.f / 100.f; // in [1,6] local
236 VP8TBufferInit(&enc->tokens_, (int)(mb_w * mb_h * 4 * scale));

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