/external/aac/libAACenc/src/ |
H A D | tonality.h | 100 void FDKaacEnc_CalculateFullTonality( FIXP_DBL *RESTRICT spectrum,
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H A D | tonality.cpp | 96 static void FDKaacEnc_CalcSfbTonality(FIXP_DBL *RESTRICT spectrum, 105 void FDKaacEnc_CalculateFullTonality(FIXP_DBL *RESTRICT spectrum, argument 128 FDKaacEnc_CalculateChaosMeasure(spectrum, 138 FDKaacEnc_CalcSfbTonality(spectrum, 161 static void FDKaacEnc_CalcSfbTonality(FIXP_DBL *RESTRICT spectrum, argument 179 FIXP_DBL tmp = (*spectrum++)<<shiftBits;
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H A D | tns_func.h | 114 FIXP_DBL *spectrum, 137 FIXP_DBL* spectrum,
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H A D | aacenc_tns.cpp | 496 Scales up spectrum lines in a given frequency section 498 \param scaled spectrum 499 \param original spectrum 517 /* Get highest value in given spectrum */ 523 /* Scale spectrum according to highest value */ 537 \param pointer to spectrum 545 const FIXP_DBL *spectrum, 557 result += (fPow2(spectrum[i])>>scale); 562 result += (fMult(spectrum[i], spectrum[ 544 FDKaacEnc_CalcAutoCorrValue( const FIXP_DBL *spectrum, const INT startLine, const INT stopLine, const INT lag, const INT scale ) argument 617 FDKaacEnc_MergedAutoCorrelation( const FIXP_DBL *spectrum, const FIXP_DBL acfWindow[MAX_NUM_OF_FILTERS][TNS_MAX_ORDER+3+1], const INT lpcStartLine[MAX_NUM_OF_FILTERS], const INT lpcStopLine, const INT maxOrder, const INT acfSplit[MAX_NUM_OF_FILTERS], FIXP_DBL *_rxx1, FIXP_DBL *_rxx2 ) argument 745 FDKaacEnc_TnsDetect( TNS_DATA *tnsData, const TNS_CONFIG *tC, TNS_INFO* tnsInfo, INT sfbCnt, FIXP_DBL *spectrum, INT subBlockNumber, INT blockType ) argument 1004 FDKaacEnc_TnsEncode( TNS_INFO* tnsInfo, TNS_DATA* tnsData, const INT numOfSfb, const TNS_CONFIG *tC, const INT lowPassLine, FIXP_DBL* spectrum, const INT subBlockNumber, const INT blockType ) argument [all...] |
/external/webrtc/src/modules/audio_processing/utility/ |
H A D | delay_estimator_wrapper.c | 25 // Pointers to mean values of spectrum. 34 // Binary spectrum based delay estimator 67 // Computes the binary spectrum by comparing the input |spectrum| with a 71 // - spectrum : Spectrum of which the binary spectrum should be 73 // - threshold_spectrum : Threshold spectrum with which the input 74 // spectrum is compared. 76 // - out : Binary spectrum. 78 static uint32_t BinarySpectrumFix(uint16_t* spectrum, argument 113 BinarySpectrumFloat(float* spectrum, SpectrumType* threshold_spectrum, int* threshold_initialized) argument [all...] |
/external/aac/libFDK/include/ |
H A D | mdct.h | 146 * \param spectrum pointer to where the resulting MDCT spectrum will be stored into. 157 FIXP_DBL *spectrum, 216 * \param spectrum pointer to the input MDCT spectra. 217 * \param scalefactors scale shift values of the input spectrum. 231 FIXP_DBL *spectrum,
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/external/aac/libAACdec/src/ |
H A D | aacdec_pns.cpp | 348 FIXP_DBL *spectrum = SPEC(pSpectrum, window, granuleLength); local 361 noise_e = GenerateRandomVector (spectrum + BandOffsets [band], bandWidth, 368 noise_e = GenerateRandomVector (spectrum + BandOffsets [band], bandWidth, 374 ScaleBand (spectrum + BandOffsets [band], bandWidth,
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H A D | conceal.cpp | 185 FIXP_DBL *spectrum, 194 FIXP_DBL *spectrum, 932 pConcealmentInfo->windowSequence = pIcsInfo->WindowSequence; /* Store for next frame (spectrum in concealment buffer can't be used at all) */ 1170 /* Expand first short spectrum */ 1210 /* scale spectrum according to concealment state */ 1261 pConcealmentInfo->windowSequence = pIcsInfo->WindowSequence; /* Store for next frame (spectrum in concealment buffer can't be used at all) */ 1311 FIXP_DBL *spectrum, 1340 enAccu += fPow2Div2(*(spectrum + line)) >> sfbScale; 1355 enAccu += (enAccu + (fPow2Div2(*(spectrum + line)) >> sfbScale)) >> 3; 1374 enAccu += fPow2Div2(*(spectrum 1310 CConcealment_CalcBandEnergy( FIXP_DBL *spectrum, const SamplingRateInfo *pSamplingRateInfo, const int blockType, CConcealmentExpandType expandType, int *sfbEnergy ) argument 1408 CConcealment_InterpolateBuffer( FIXP_DBL *spectrum, SHORT *pSpecScalePrv, SHORT *pSpecScaleAct, SHORT *pSpecScaleOut, int *enPrv, int *enAct, int sfbCnt, const SHORT *pSfbOffset ) argument [all...] |
H A D | block.cpp | 378 * formula: spectrum[i] = Sign(spectrum[i]) * Matissa(spectrum[i])^(4/3) * 2^(lsb/4). 379 * \param spectrum pointer to first line of the sfb to be inverse quantized. 385 void InverseQuantizeBand( FIXP_DBL * RESTRICT spectrum, argument 394 FIXP_DBL *ptr = spectrum;
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/external/aac/libFDK/src/ |
H A D | mdct.cpp | 251 FIXP_DBL *spectrum, 302 /* Current spectrum */ 303 pSpec = spectrum+w*tl; 305 /* DCT IV of current spectrum. */ 308 /* Optional scaling of time domain - no yet windowed - of current spectrum */ 309 /* and de-scale current spectrum signal (time domain, no yet windowed) */ 380 pOvl[i] = spectrum[i+(nSpec-1)*tl]; 248 imdct_block( H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *spectrum, const SHORT scalefactor[], const INT nSpec, const INT noOutSamples, const INT tl, const FIXP_WTP *wls, INT fl, const FIXP_WTP *wrs, const INT fr, FIXP_DBL gain ) argument
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/external/libvorbis/doc/ |
H A D | 01-introduction.tex | 190 Therefore it would be wasteful to encode a full-spectrum version of it 202 vector is a low-resolution representation of the audio spectrum for 231 The spectral residue is the fine structure of the audio spectrum 301 \item compute dot product of floor and residue, producing audio spectrum vector 302 \item inverse monolithic transform of audio spectrum vector, always an MDCT in Vorbis I 451 producing the finished audio spectrum of each channel. 463 the audio spectrum vector should represent a minimum of 120dB (\~{}21 481 The audio spectrum is converted back into time domain PCM audio via an
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H A D | 04-codec.tex | 538 vectors are the length \varname{[n]}/2 audio spectrum for each 551 the audio spectrum vector should represent a minimum of 120dB (\~21 569 Convert the audio spectrum vector of each channel back into time
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H A D | 08-residue.tex | 8 A residue vector represents the fine detail of the audio spectrum of
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