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Diffstat (limited to 'libAACdec/src/usacdec_fac.cpp')
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diff --git a/libAACdec/src/usacdec_fac.cpp b/libAACdec/src/usacdec_fac.cpp new file mode 100644 index 0000000..71ce4a9 --- /dev/null +++ b/libAACdec/src/usacdec_fac.cpp @@ -0,0 +1,743 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC decoder library ****************************** + + Author(s): Manuel Jander + + Description: USAC FAC + +*******************************************************************************/ + +#include "usacdec_fac.h" + +#include "usacdec_const.h" +#include "usacdec_lpc.h" +#include "usacdec_acelp.h" +#include "usacdec_rom.h" +#include "dct.h" +#include "FDK_tools_rom.h" +#include "mdct.h" + +#define SPEC_FAC(ptr, i, gl) ((ptr) + ((i) * (gl))) + +FIXP_DBL *CLpd_FAC_GetMemory(CAacDecoderChannelInfo *pAacDecoderChannelInfo, + UCHAR mod[NB_DIV], int *pState) { + FIXP_DBL *ptr; + int i; + int k = 0; + int max_windows = 8; + + FDK_ASSERT(*pState >= 0 && *pState < max_windows); + + /* Look for free space to store FAC data. 2 FAC data blocks fit into each TCX + * spectral data block. */ + for (i = *pState; i < max_windows; i++) { + if (mod[i >> 1] == 0) { + break; + } + } + + *pState = i + 1; + + if (i == max_windows) { + ptr = pAacDecoderChannelInfo->data.usac.fac_data0; + } else { + FDK_ASSERT(mod[(i >> 1)] == 0); + ptr = SPEC_FAC(pAacDecoderChannelInfo->pSpectralCoefficient, i, + pAacDecoderChannelInfo->granuleLength << k); + } + + return ptr; +} + +int CLpd_FAC_Read(HANDLE_FDK_BITSTREAM hBs, FIXP_DBL *pFac, UCHAR *pFacScale, + int length, int use_gain, int frame) { + FIXP_DBL fac_gain; + int fac_gain_e = 0; + + if (use_gain) { + CLpd_DecodeGain(&fac_gain, &fac_gain_e, FDKreadBits(hBs, 7)); + } + + if (CLpc_DecodeAVQ(hBs, pFac, 1, 1, length) != 0) { + return -1; + } + + { + int scale; + + scale = getScalefactor(pFac, length); + scaleValues(pFac, length, scale); + pFacScale[frame] = DFRACT_BITS - 1 - scale; + } + + if (use_gain) { + int i; + + pFacScale[frame] += fac_gain_e; + + for (i = 0; i < length; i++) { + pFac[i] = fMult(pFac[i], fac_gain); + } + } + return 0; +} + +/** + * \brief Apply synthesis filter with zero input to x. The overall filter gain + * is 1.0. + * \param a LPC filter coefficients. + * \param length length of the input/output data vector x. + * \param x input/output vector, where the synthesis filter is applied in place. + */ +static void Syn_filt_zero(const FIXP_LPC a[], const INT a_exp, INT length, + FIXP_DBL x[]) { + int i, j; + FIXP_DBL L_tmp; + + for (i = 0; i < length; i++) { + L_tmp = (FIXP_DBL)0; + + for (j = 0; j < fMin(i, M_LP_FILTER_ORDER); j++) { + L_tmp -= fMultDiv2(a[j], x[i - (j + 1)]); + } + + L_tmp = scaleValue(L_tmp, a_exp + 1); + + x[i] = scaleValueSaturate((x[i] >> 1) + (L_tmp >> 1), + 1); /* Avoid overflow issues and saturate. */ + } +} + +/* Table is also correct for coreCoderFrameLength = 768. Factor 3/4 is canceled + out: gainFac = 0.5 * sqrt(fac_length/lFrame) +*/ +static const FIXP_DBL gainFac[4] = {0x40000000, 0x2d413ccd, 0x20000000, + 0x16a09e66}; + +void CFac_ApplyGains(FIXP_DBL fac_data[LFAC], const INT fac_length, + const FIXP_DBL tcx_gain, const FIXP_DBL alfd_gains[], + const INT mod) { + FIXP_DBL facFactor; + int i; + + FDK_ASSERT((fac_length == 128) || (fac_length == 96)); + + /* 2) Apply gain factor to FAC data */ + facFactor = fMult(gainFac[mod], tcx_gain); + for (i = 0; i < fac_length; i++) { + fac_data[i] = fMult(fac_data[i], facFactor); + } + + /* 3) Apply spectrum deshaping using alfd_gains */ + for (i = 0; i < fac_length / 4; i++) { + int k; + + k = i >> (3 - mod); + fac_data[i] = fMult(fac_data[i], alfd_gains[k]) + << 1; /* alfd_gains is scaled by one bit. */ + } +} + +static void CFac_CalcFacSignal(FIXP_DBL *pOut, FIXP_DBL *pFac, + const int fac_scale, const int fac_length, + const FIXP_LPC A[M_LP_FILTER_ORDER], + const INT A_exp, const int fAddZir, + const int isFdFac) { + FIXP_LPC wA[M_LP_FILTER_ORDER]; + FIXP_DBL tf_gain = (FIXP_DBL)0; + int wlength; + int scale = fac_scale; + + /* obtain tranform gain. */ + imdct_gain(&tf_gain, &scale, isFdFac ? 0 : fac_length); + + /* 4) Compute inverse DCT-IV of FAC data. Output scale of DCT IV is 16 bits. + */ + dct_IV(pFac, fac_length, &scale); + /* dct_IV scale = log2(fac_length). "- 7" is a factor of 2/128 */ + if (tf_gain != (FIXP_DBL)0) { /* non-radix 2 transform gain */ + int i; + + for (i = 0; i < fac_length; i++) { + pFac[i] = fMult(tf_gain, pFac[i]); + } + } + scaleValuesSaturate(pOut, pFac, fac_length, + scale); /* Avoid overflow issues and saturate. */ + + E_LPC_a_weight(wA, A, M_LP_FILTER_ORDER); + + /* We need the output of the IIR filter to be longer than "fac_length". + For this reason we run it with zero input appended to the end of the input + sequence, i.e. we generate its ZIR and extend the output signal.*/ + FDKmemclear(pOut + fac_length, fac_length * sizeof(FIXP_DBL)); + wlength = 2 * fac_length; + + /* 5) Apply weighted synthesis filter to FAC data, including optional Zir (5. + * item 4). */ + Syn_filt_zero(wA, A_exp, wlength, pOut); +} + +INT CLpd_FAC_Mdct2Acelp(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pFac, + const int fac_scale, FIXP_LPC *A, INT A_exp, + INT nrOutSamples, const INT fac_length, + const INT isFdFac, UCHAR prevWindowShape) { + FIXP_DBL *pOvl; + FIXP_DBL *pOut0; + const FIXP_WTP *pWindow; + int i, fl, nrSamples = 0; + + FDK_ASSERT(fac_length <= 1024 / (4 * 2)); + + fl = fac_length * 2; + + pWindow = FDKgetWindowSlope(fl, prevWindowShape); + + /* Adapt window slope length in case of frame loss. */ + if (hMdct->prev_fr != fl) { + int nl = 0; + imdct_adapt_parameters(hMdct, &fl, &nl, fac_length, pWindow, nrOutSamples); + FDK_ASSERT(nl == 0); + } + + if (nrSamples < nrOutSamples) { + pOut0 = output; + nrSamples += hMdct->ov_offset; + /* Purge buffered output. */ + FDKmemcpy(pOut0, hMdct->overlap.time, hMdct->ov_offset * sizeof(pOut0[0])); + hMdct->ov_offset = 0; + } + + pOvl = hMdct->overlap.freq + hMdct->ov_size - 1; + + if (nrSamples >= nrOutSamples) { + pOut0 = hMdct->overlap.time + hMdct->ov_offset; + hMdct->ov_offset += hMdct->prev_nr + fl / 2; + } else { + pOut0 = output + nrSamples; + nrSamples += hMdct->prev_nr + fl / 2; + } + if (hMdct->prevPrevAliasSymmetry == 0) { + for (i = 0; i < hMdct->prev_nr; i++) { + FIXP_DBL x = -(*pOvl--); + *pOut0 = IMDCT_SCALE_DBL(x); + pOut0++; + } + } else { + for (i = 0; i < hMdct->prev_nr; i++) { + FIXP_DBL x = (*pOvl--); + *pOut0 = IMDCT_SCALE_DBL(x); + pOut0++; + } + } + hMdct->prev_nr = 0; + + { + if (pFac != NULL) { + /* Note: The FAC gain might have been applied directly after bit stream + * parse in this case. */ + CFac_CalcFacSignal(pOut0, pFac, fac_scale, fac_length, A, A_exp, 0, + isFdFac); + } else { + /* Clear buffer because of the overlap and ADD! */ + FDKmemclear(pOut0, fac_length * sizeof(FIXP_DBL)); + } + } + + i = 0; + + if (hMdct->prevPrevAliasSymmetry == 0) { + for (; i < fl / 2; i++) { + FIXP_DBL x0; + + /* Overlap Add */ + x0 = -fMult(*pOvl--, pWindow[i].v.re); + + *pOut0 += IMDCT_SCALE_DBL(x0); + pOut0++; + } + } else { + for (; i < fl / 2; i++) { + FIXP_DBL x0; + + /* Overlap Add */ + x0 = fMult(*pOvl--, pWindow[i].v.re); + + *pOut0 += IMDCT_SCALE_DBL(x0); + pOut0++; + } + } + if (hMdct->pFacZir != + 0) { /* this should only happen for ACELP -> TCX20 -> ACELP transition */ + FIXP_DBL *pOut = pOut0 - fl / 2; /* fl/2 == fac_length */ + for (i = 0; i < fl / 2; i++) { + pOut[i] += IMDCT_SCALE_DBL(hMdct->pFacZir[i]); + } + hMdct->pFacZir = NULL; + } + + hMdct->prev_fr = 0; + hMdct->prev_nr = 0; + hMdct->prev_tl = 0; + hMdct->prevPrevAliasSymmetry = hMdct->prevAliasSymmetry; + + return nrSamples; +} + +INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec, + const SHORT spec_scale[], const int nSpec, + FIXP_DBL *pFac, const int fac_scale, + const INT fac_length, INT noOutSamples, const INT tl, + const FIXP_WTP *wrs, const INT fr, FIXP_LPC A[16], + INT A_exp, CAcelpStaticMem *acelp_mem, + const FIXP_DBL gain, const int last_frame_lost, + const int isFdFac, const UCHAR last_lpd_mode, + const int k, int currAliasingSymmetry) { + FIXP_DBL *pCurr, *pOvl, *pSpec; + const FIXP_WTP *pWindow; + const FIXP_WTB *FacWindowZir_conceal; + UCHAR doFacZirConceal = 0; + int doDeemph = 1; + const FIXP_WTB *FacWindowZir, *FacWindowSynth; + FIXP_DBL *pOut0 = output, *pOut1; + int w, i, fl, nl, nr, f_len, nrSamples = 0, s = 0, scale, total_gain_e; + FIXP_DBL *pF, *pFAC_and_FAC_ZIR = NULL; + FIXP_DBL total_gain = gain; + + FDK_ASSERT(fac_length <= 1024 / (4 * 2)); + switch (fac_length) { + /* coreCoderFrameLength = 1024 */ + case 128: + pWindow = SineWindow256; + FacWindowZir = FacWindowZir128; + FacWindowSynth = FacWindowSynth128; + break; + case 64: + pWindow = SineWindow128; + FacWindowZir = FacWindowZir64; + FacWindowSynth = FacWindowSynth64; + break; + case 32: + pWindow = SineWindow64; + FacWindowZir = FacWindowZir32; + FacWindowSynth = FacWindowSynth32; + break; + /* coreCoderFrameLength = 768 */ + case 96: + pWindow = SineWindow192; + FacWindowZir = FacWindowZir96; + FacWindowSynth = FacWindowSynth96; + break; + case 48: + pWindow = SineWindow96; + FacWindowZir = FacWindowZir48; + FacWindowSynth = FacWindowSynth48; + break; + default: + FDK_ASSERT(0); + return 0; + } + + FacWindowZir_conceal = FacWindowSynth; + /* Derive NR and NL */ + fl = fac_length * 2; + nl = (tl - fl) >> 1; + nr = (tl - fr) >> 1; + + if (noOutSamples > nrSamples) { + /* Purge buffered output. */ + FDKmemcpy(pOut0, hMdct->overlap.time, hMdct->ov_offset * sizeof(pOut0[0])); + nrSamples = hMdct->ov_offset; + hMdct->ov_offset = 0; + } + + if (nrSamples >= noOutSamples) { + pOut1 = hMdct->overlap.time + hMdct->ov_offset; + if (hMdct->ov_offset < fac_length) { + pOut0 = output + nrSamples; + } else { + pOut0 = pOut1; + } + hMdct->ov_offset += fac_length + nl; + } else { + pOut1 = output + nrSamples; + pOut0 = output + nrSamples; + } + + { + pFAC_and_FAC_ZIR = CLpd_ACELP_GetFreeExcMem(acelp_mem, 2 * fac_length); + { + const FIXP_DBL *pTmp1, *pTmp2; + + doFacZirConceal |= ((last_frame_lost != 0) && (k == 0)); + doDeemph &= (last_lpd_mode != 4); + if (doFacZirConceal) { + /* ACELP contribution in concealment case: + Use ZIR with a modified ZIR window to preserve some more energy. + Dont use FAC, which contains wrong information for concealed frame + Dont use last ACELP samples, but double ZIR, instead (afterwards) */ + FDKmemclear(pFAC_and_FAC_ZIR, 2 * fac_length * sizeof(FIXP_DBL)); + FacWindowSynth = (FIXP_WTB *)pFAC_and_FAC_ZIR; + FacWindowZir = FacWindowZir_conceal; + } else { + CFac_CalcFacSignal(pFAC_and_FAC_ZIR, pFac, fac_scale + s, fac_length, A, + A_exp, 1, isFdFac); + } + /* 6) Get windowed past ACELP samples and ACELP ZIR signal */ + + /* + * Get ACELP ZIR (pFac[]) and ACELP past samples (pOut0[]) and add them + * to the FAC synth signal contribution on pOut1[]. + */ + { + { + CLpd_Acelp_Zir(A, A_exp, acelp_mem, fac_length, pFac, doDeemph); + + pTmp1 = pOut0; + pTmp2 = pFac; + } + + for (i = 0, w = 0; i < fac_length; i++) { + FIXP_DBL x; + /* Div2 is compensated by table scaling */ + x = fMultDiv2(pTmp2[i], FacWindowZir[w]); + x += fMultDiv2(pTmp1[-i - 1], FacWindowSynth[w]); + x += pFAC_and_FAC_ZIR[i]; + pOut1[i] = x; + + w++; + } + } + + if (doFacZirConceal) { + /* ZIR is the only ACELP contribution, so double it */ + scaleValues(pOut1, fac_length, 1); + } + } + } + + if (nrSamples < noOutSamples) { + nrSamples += fac_length + nl; + } + + /* Obtain transform gain */ + total_gain = gain; + total_gain_e = 0; + imdct_gain(&total_gain, &total_gain_e, tl); + + /* IMDCT overlap add */ + scale = total_gain_e; + pSpec = _pSpec; + + /* Note:when comming from an LPD frame (TCX/ACELP) the previous alisaing + * symmetry must always be 0 */ + if (currAliasingSymmetry == 0) { + dct_IV(pSpec, tl, &scale); + } else { + FIXP_DBL _tmp[1024 + ALIGNMENT_DEFAULT / sizeof(FIXP_DBL)]; + FIXP_DBL *tmp = (FIXP_DBL *)ALIGN_PTR(_tmp); + C_ALLOC_ALIGNED_REGISTER(tmp, sizeof(_tmp)); + dst_III(pSpec, tmp, tl, &scale); + C_ALLOC_ALIGNED_UNREGISTER(tmp); + } + + /* Optional scaling of time domain - no yet windowed - of current spectrum */ + if (total_gain != (FIXP_DBL)0) { + scaleValuesWithFactor(pSpec, total_gain, tl, spec_scale[0] + scale); + } else { + scaleValues(pSpec, tl, spec_scale[0] + scale); + } + + pOut1 += fl / 2 - 1; + pCurr = pSpec + tl - fl / 2; + + for (i = 0; i < fl / 2; i++) { + FIXP_DBL x1; + + /* FAC signal is already on pOut1, because of that the += operator. */ + x1 = fMult(*pCurr++, pWindow[i].v.re); + FDK_ASSERT((pOut1 >= hMdct->overlap.time && + pOut1 < hMdct->overlap.time + hMdct->ov_size) || + (pOut1 >= output && pOut1 < output + 1024)); + *pOut1 += IMDCT_SCALE_DBL(-x1); + pOut1--; + } + + /* NL output samples TL/2+FL/2..TL. - current[FL/2..0] */ + pOut1 += (fl / 2) + 1; + + pFAC_and_FAC_ZIR += fac_length; /* set pointer to beginning of FAC ZIR */ + + if (nl == 0) { + /* save pointer to write FAC ZIR data later */ + hMdct->pFacZir = pFAC_and_FAC_ZIR; + } else { + FDK_ASSERT(nl >= fac_length); + /* FAC ZIR will be added now ... */ + hMdct->pFacZir = NULL; + } + + pF = pFAC_and_FAC_ZIR; + f_len = fac_length; + + pCurr = pSpec + tl - fl / 2 - 1; + for (i = 0; i < nl; i++) { + FIXP_DBL x = -(*pCurr--); + /* 5) (item 4) Synthesis filter Zir component, FAC ZIR (another one). */ + if (i < f_len) { + x += *pF++; + } + + FDK_ASSERT((pOut1 >= hMdct->overlap.time && + pOut1 < hMdct->overlap.time + hMdct->ov_size) || + (pOut1 >= output && pOut1 < output + 1024)); + *pOut1 = IMDCT_SCALE_DBL(x); + pOut1++; + } + + hMdct->prev_nr = nr; + hMdct->prev_fr = fr; + hMdct->prev_wrs = wrs; + hMdct->prev_tl = tl; + hMdct->prevPrevAliasSymmetry = hMdct->prevAliasSymmetry; + hMdct->prevAliasSymmetry = currAliasingSymmetry; + fl = fr; + nl = nr; + + pOvl = pSpec + tl / 2 - 1; + pOut0 = pOut1; + + for (w = 1; w < nSpec; w++) /* for ACELP -> FD short */ + { + const FIXP_WTP *pWindow_prev; + + /* Setup window pointers */ + pWindow_prev = hMdct->prev_wrs; + + /* Current spectrum */ + pSpec = _pSpec + w * tl; + + scale = total_gain_e; + + /* For the second, third, etc. short frames the alisaing symmetry is equal, + * either (0,0) or (1,1) */ + if (currAliasingSymmetry == 0) { + /* DCT IV of current spectrum */ + dct_IV(pSpec, tl, &scale); + } else { + dst_IV(pSpec, tl, &scale); + } + + /* Optional scaling of time domain - no yet windowed - of current spectrum + */ + /* and de-scale current spectrum signal (time domain, no yet windowed) */ + if (total_gain != (FIXP_DBL)0) { + scaleValuesWithFactor(pSpec, total_gain, tl, spec_scale[w] + scale); + } else { + scaleValues(pSpec, tl, spec_scale[w] + scale); + } + + if (noOutSamples <= nrSamples) { + /* Divert output first half to overlap buffer if we already got enough + * output samples. */ + pOut0 = hMdct->overlap.time + hMdct->ov_offset; + hMdct->ov_offset += hMdct->prev_nr + fl / 2; + } else { + /* Account output samples */ + nrSamples += hMdct->prev_nr + fl / 2; + } + + /* NR output samples 0 .. NR. -overlap[TL/2..TL/2-NR] */ + for (i = 0; i < hMdct->prev_nr; i++) { + FIXP_DBL x = -(*pOvl--); + *pOut0 = IMDCT_SCALE_DBL(x); + pOut0++; + } + + if (noOutSamples <= nrSamples) { + /* Divert output second half to overlap buffer if we already got enough + * output samples. */ + pOut1 = hMdct->overlap.time + hMdct->ov_offset + fl / 2 - 1; + hMdct->ov_offset += fl / 2 + nl; + } else { + pOut1 = pOut0 + (fl - 1); + nrSamples += fl / 2 + nl; + } + + /* output samples before window crossing point NR .. TL/2. + * -overlap[TL/2-NR..TL/2-NR-FL/2] + current[NR..TL/2] */ + /* output samples after window crossing point TL/2 .. TL/2+FL/2. + * -overlap[0..FL/2] - current[TL/2..FL/2] */ + pCurr = pSpec + tl - fl / 2; + if (currAliasingSymmetry == 0) { + for (i = 0; i < fl / 2; i++) { + FIXP_DBL x0, x1; + + cplxMult(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]); + *pOut0 = IMDCT_SCALE_DBL(x0); + *pOut1 = IMDCT_SCALE_DBL(-x1); + pOut0++; + pOut1--; + } + } else { + if (hMdct->prevPrevAliasSymmetry == 0) { + /* Jump DST II -> DST IV for the second window */ + for (i = 0; i < fl / 2; i++) { + FIXP_DBL x0, x1; + + cplxMult(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]); + *pOut0 = IMDCT_SCALE_DBL(x0); + *pOut1 = IMDCT_SCALE_DBL(x1); + pOut0++; + pOut1--; + } + } else { + /* Jump DST IV -> DST IV from the second window on */ + for (i = 0; i < fl / 2; i++) { + FIXP_DBL x0, x1; + + cplxMult(&x1, &x0, *pCurr++, *pOvl--, pWindow_prev[i]); + *pOut0 = IMDCT_SCALE_DBL(x0); + *pOut1 = IMDCT_SCALE_DBL(x1); + pOut0++; + pOut1--; + } + } + } + + if (hMdct->pFacZir != 0) { + /* add FAC ZIR of previous ACELP -> mdct transition */ + FIXP_DBL *pOut = pOut0 - fl / 2; + FDK_ASSERT(fl / 2 <= 128); + for (i = 0; i < fl / 2; i++) { + pOut[i] += IMDCT_SCALE_DBL(hMdct->pFacZir[i]); + } + hMdct->pFacZir = NULL; + } + pOut0 += (fl / 2); + + /* NL output samples TL/2+FL/2..TL. - current[FL/2..0] */ + pOut1 += (fl / 2) + 1; + pCurr = pSpec + tl - fl / 2 - 1; + for (i = 0; i < nl; i++) { + FIXP_DBL x = -(*pCurr--); + *pOut1 = IMDCT_SCALE_DBL(x); + pOut1++; + } + + /* Set overlap source pointer for next window pOvl = pSpec + tl/2 - 1; */ + pOvl = pSpec + tl / 2 - 1; + + /* Previous window values. */ + hMdct->prev_nr = nr; + hMdct->prev_fr = fr; + hMdct->prev_tl = tl; + hMdct->prev_wrs = pWindow_prev; + hMdct->prevPrevAliasSymmetry = hMdct->prevAliasSymmetry; + hMdct->prevAliasSymmetry = currAliasingSymmetry; + } + + /* Save overlap */ + + pOvl = hMdct->overlap.freq + hMdct->ov_size - tl / 2; + FDK_ASSERT(pOvl >= hMdct->overlap.time + hMdct->ov_offset); + FDK_ASSERT(tl / 2 <= hMdct->ov_size); + for (i = 0; i < tl / 2; i++) { + pOvl[i] = _pSpec[i + (w - 1) * tl]; + } + + return nrSamples; +} |