/external/zxing/core/ |
H A D | core.jar | META-INF/ META-INF/MANIFEST.MF com/ com/google/ com/google/zxing/ com/google/zxing/aztec/ ... |
/external/webrtc/src/common_audio/signal_processing/ |
H A D | auto_correlation.c | 24 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
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H A D | complex_fft.c | 289 int i, j, l, k, istep, n, m, scale, shift; local 301 scale = 0; 317 scale++; 323 scale++; 424 return scale;
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/external/webrtc/src/common_audio/signal_processing/include/ |
H A D | signal_processing_library.h | 315 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.
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/interface/ |
H A D | isacfix.h | 547 * - scale : factor for rate change (0.4 ~=> half the rate, 1 no change). 558 float scale,
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
H A D | codec.h | 47 float scale); 138 WebRtc_Word16* __restrict scale); 152 WebRtc_Word16* __restrict scale); 168 WebRtc_Word16* __restrict scale);
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H A D | encode.c | 328 // 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 D | filters.c | 31 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
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H A D | filters_neon.c | 31 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
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H A D | isacfix.c | 464 float scale, 485 scale); 462 WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst, WebRtc_Word16 bweIndex, float scale, WebRtc_Word16 *encoded) argument
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H A D | lpc_masking_model.c | 81 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
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H A D | pitch_filter.c | 184 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);
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/external/webrtc/src/modules/audio_coding/codecs/isac/fix/test/ |
H A D | kenny.c | 142 float scale = (float)0.7; local 366 scale = (float)strtod( argv[i+1], NULL ); 587 scale,
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/external/webrtc/src/modules/audio_coding/codecs/isac/main/source/ |
H A D | codec.h | 47 float scale); 51 WebRtc_Word32 jitterInfo, float scale, enum ISACBandwidth bandwidth);
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H A D | encode.c | 444 /* 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 D | isac.c | 763 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 D | aec_core.c | 41 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 D | aec_core_sse2.c | 174 float scale = 2.0f / PART_LEN2; local 175 const __m128 scale_ps = _mm_load_ps1(&scale);
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/external/webrtc/src/modules/audio_processing/agc/ |
H A D | analog_agc.c | 47 /* 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 D | analog_agc.h | 74 WebRtc_Word16 analogTarget; // Digital reference level in ENV scale 115 WebRtc_Word16 scale; // Scale factor for internal volume levels member in struct:__anon33486
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/external/webrtc/src/modules/audio_processing/utility/ |
H A D | delay_estimator_wrapper.c | 55 // - 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
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/external/vixl/src/a64/ |
H A D | assembler-a64.h | 1657 static Instr FPScale(unsigned scale) { 1658 VIXL_ASSERT(is_uint6(scale)); 1659 return scale << FPScale_offset;
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/external/webp/src/dsp/ |
H A D | alpha_processing.c | 146 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);
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/external/webp/src/enc/ |
H A D | picture_csp.c | 107 const double scale = 1. / kGammaScale; local 113 const double x = scale * (v << kGammaTabFix);
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H A D | webpenc.c | 235 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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