diff options
Diffstat (limited to 'fdk-aac/libAACdec/src/block.cpp')
| -rw-r--r-- | fdk-aac/libAACdec/src/block.cpp | 1260 |
1 files changed, 1260 insertions, 0 deletions
diff --git a/fdk-aac/libAACdec/src/block.cpp b/fdk-aac/libAACdec/src/block.cpp new file mode 100644 index 0000000..b3d09a6 --- /dev/null +++ b/fdk-aac/libAACdec/src/block.cpp @@ -0,0 +1,1260 @@ +/* ----------------------------------------------------------------------------- +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): Josef Hoepfl + + Description: long/short-block decoding + +*******************************************************************************/ + +#include "block.h" + +#include "aac_rom.h" +#include "FDK_bitstream.h" +#include "scale.h" +#include "FDK_tools_rom.h" + +#include "usacdec_fac.h" +#include "usacdec_lpd.h" +#include "usacdec_lpc.h" +#include "FDK_trigFcts.h" + +#include "ac_arith_coder.h" + +#include "aacdec_hcr.h" +#include "rvlc.h" + +#if defined(__arm__) +#include "arm/block_arm.cpp" +#endif + +/*! + \brief Read escape sequence of codeword + + The function reads the escape sequence from the bitstream, + if the absolute value of the quantized coefficient has the + value 16. + A limitation is implemented to maximal 21 bits according to + ISO/IEC 14496-3:2009(E) 4.6.3.3. + This limits the escape prefix to a maximum of eight 1's. + If more than eight 1's are read, MAX_QUANTIZED_VALUE + 1 is + returned, independent of the sign of parameter q. + + \return quantized coefficient +*/ +LONG CBlock_GetEscape(HANDLE_FDK_BITSTREAM bs, /*!< pointer to bitstream */ + const LONG q) /*!< quantized coefficient */ +{ + if (fAbs(q) != 16) return (q); + + LONG i, off; + for (i = 4; i < 13; i++) { + if (FDKreadBit(bs) == 0) break; + } + + if (i == 13) return (MAX_QUANTIZED_VALUE + 1); + + off = FDKreadBits(bs, i); + i = off + (1 << i); + + if (q < 0) i = -i; + + return i; +} + +AAC_DECODER_ERROR CBlock_ReadScaleFactorData( + CAacDecoderChannelInfo *pAacDecoderChannelInfo, HANDLE_FDK_BITSTREAM bs, + UINT flags) { + int temp; + int band; + int group; + int position = 0; /* accu for intensity delta coding */ + int factor = pAacDecoderChannelInfo->pDynData->RawDataInfo + .GlobalGain; /* accu for scale factor delta coding */ + UCHAR *pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook; + SHORT *pScaleFactor = pAacDecoderChannelInfo->pDynData->aScaleFactor; + const CodeBookDescription *hcb = &AACcodeBookDescriptionTable[BOOKSCL]; + + const USHORT(*CodeBook)[HuffmanEntries] = hcb->CodeBook; + + int ScaleFactorBandsTransmitted = + GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); + for (group = 0; group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); + group++) { + for (band = 0; band < ScaleFactorBandsTransmitted; band++) { + switch (pCodeBook[band]) { + case ZERO_HCB: /* zero book */ + pScaleFactor[band] = 0; + break; + + default: /* decode scale factor */ + if (!((flags & (AC_USAC | AC_RSVD50 | AC_RSV603DA)) && band == 0 && + group == 0)) { + temp = CBlock_DecodeHuffmanWordCB(bs, CodeBook); + factor += temp - 60; /* MIDFAC 1.5 dB */ + } + pScaleFactor[band] = factor - 100; + break; + + case INTENSITY_HCB: /* intensity steering */ + case INTENSITY_HCB2: + temp = CBlock_DecodeHuffmanWordCB(bs, CodeBook); + position += temp - 60; + pScaleFactor[band] = position - 100; + break; + + case NOISE_HCB: /* PNS */ + if (flags & (AC_MPEGD_RES | AC_USAC | AC_RSVD50 | AC_RSV603DA)) { + return AAC_DEC_PARSE_ERROR; + } + CPns_Read(&pAacDecoderChannelInfo->data.aac.PnsData, bs, hcb, + pAacDecoderChannelInfo->pDynData->aScaleFactor, + pAacDecoderChannelInfo->pDynData->RawDataInfo.GlobalGain, + band, group); + break; + } + } + pCodeBook += 16; + pScaleFactor += 16; + } + + return AAC_DEC_OK; +} + +void CBlock_ScaleSpectralData(CAacDecoderChannelInfo *pAacDecoderChannelInfo, + UCHAR maxSfbs, + SamplingRateInfo *pSamplingRateInfo) { + int band; + int window; + const SHORT *RESTRICT pSfbScale = pAacDecoderChannelInfo->pDynData->aSfbScale; + SHORT *RESTRICT pSpecScale = pAacDecoderChannelInfo->specScale; + int groupwin, group; + const SHORT *RESTRICT BandOffsets = GetScaleFactorBandOffsets( + &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); + SPECTRAL_PTR RESTRICT pSpectralCoefficient = + pAacDecoderChannelInfo->pSpectralCoefficient; + + FDKmemclear(pSpecScale, 8 * sizeof(SHORT)); + + for (window = 0, group = 0; + group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++) { + for (groupwin = 0; groupwin < GetWindowGroupLength( + &pAacDecoderChannelInfo->icsInfo, group); + groupwin++, window++) { + int SpecScale_window = pSpecScale[window]; + FIXP_DBL *pSpectrum = SPEC(pSpectralCoefficient, window, + pAacDecoderChannelInfo->granuleLength); + + /* find scaling for current window */ + for (band = 0; band < maxSfbs; band++) { + SpecScale_window = + fMax(SpecScale_window, (int)pSfbScale[window * 16 + band]); + } + + if (pAacDecoderChannelInfo->pDynData->TnsData.Active && + pAacDecoderChannelInfo->pDynData->TnsData.NumberOfFilters[window] > + 0) { + int filter_index, SpecScale_window_tns; + int tns_start, tns_stop; + + /* Find max scale of TNS bands */ + SpecScale_window_tns = 0; + tns_start = GetMaximumTnsBands(&pAacDecoderChannelInfo->icsInfo, + pSamplingRateInfo->samplingRateIndex); + tns_stop = 0; + for (filter_index = 0; + filter_index < (int)pAacDecoderChannelInfo->pDynData->TnsData + .NumberOfFilters[window]; + filter_index++) { + for (band = pAacDecoderChannelInfo->pDynData->TnsData + .Filter[window][filter_index] + .StartBand; + band < pAacDecoderChannelInfo->pDynData->TnsData + .Filter[window][filter_index] + .StopBand; + band++) { + SpecScale_window_tns = + fMax(SpecScale_window_tns, (int)pSfbScale[window * 16 + band]); + } + /* Find TNS line boundaries for all TNS filters */ + tns_start = + fMin(tns_start, (int)pAacDecoderChannelInfo->pDynData->TnsData + .Filter[window][filter_index] + .StartBand); + tns_stop = + fMax(tns_stop, (int)pAacDecoderChannelInfo->pDynData->TnsData + .Filter[window][filter_index] + .StopBand); + } + SpecScale_window_tns = SpecScale_window_tns + + pAacDecoderChannelInfo->pDynData->TnsData.GainLd; + FDK_ASSERT(tns_stop >= tns_start); + /* Consider existing headroom of all MDCT lines inside the TNS bands. */ + SpecScale_window_tns -= + getScalefactor(pSpectrum + BandOffsets[tns_start], + BandOffsets[tns_stop] - BandOffsets[tns_start]); + if (SpecScale_window <= 17) { + SpecScale_window_tns++; + } + /* Add enough mantissa head room such that the spectrum is still + representable after applying TNS. */ + SpecScale_window = fMax(SpecScale_window, SpecScale_window_tns); + } + + /* store scaling of current window */ + pSpecScale[window] = SpecScale_window; + +#ifdef FUNCTION_CBlock_ScaleSpectralData_func1 + + CBlock_ScaleSpectralData_func1(pSpectrum, maxSfbs, BandOffsets, + SpecScale_window, pSfbScale, window); + +#else /* FUNCTION_CBlock_ScaleSpectralData_func1 */ + for (band = 0; band < maxSfbs; band++) { + int scale = fMin(DFRACT_BITS - 1, + SpecScale_window - pSfbScale[window * 16 + band]); + if (scale) { + FDK_ASSERT(scale > 0); + + /* following relation can be used for optimizations: + * (BandOffsets[i]%4) == 0 for all i */ + int max_index = BandOffsets[band + 1]; + DWORD_ALIGNED(pSpectrum); + for (int index = BandOffsets[band]; index < max_index; index++) { + pSpectrum[index] >>= scale; + } + } + } +#endif /* FUNCTION_CBlock_ScaleSpectralData_func1 */ + } + } +} + +AAC_DECODER_ERROR CBlock_ReadSectionData( + HANDLE_FDK_BITSTREAM bs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, + const SamplingRateInfo *pSamplingRateInfo, const UINT flags) { + int top, band; + int sect_len, sect_len_incr; + int group; + UCHAR sect_cb; + UCHAR *pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook; + /* HCR input (long) */ + SHORT *pNumLinesInSec = + pAacDecoderChannelInfo->pDynData->specificTo.aac.aNumLineInSec4Hcr; + int numLinesInSecIdx = 0; + UCHAR *pHcrCodeBook = + pAacDecoderChannelInfo->pDynData->specificTo.aac.aCodeBooks4Hcr; + const SHORT *BandOffsets = GetScaleFactorBandOffsets( + &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); + pAacDecoderChannelInfo->pDynData->specificTo.aac.numberSection = 0; + AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK; + + FDKmemclear(pCodeBook, sizeof(UCHAR) * (8 * 16)); + + const int nbits = + (IsLongBlock(&pAacDecoderChannelInfo->icsInfo) == 1) ? 5 : 3; + + int sect_esc_val = (1 << nbits) - 1; + + UCHAR ScaleFactorBandsTransmitted = + GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); + for (group = 0; group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); + group++) { + for (band = 0; band < ScaleFactorBandsTransmitted;) { + sect_len = 0; + if (flags & AC_ER_VCB11) { + sect_cb = (UCHAR)FDKreadBits(bs, 5); + } else + sect_cb = (UCHAR)FDKreadBits(bs, 4); + + if (((flags & AC_ER_VCB11) == 0) || (sect_cb < 11) || + ((sect_cb > 11) && (sect_cb < 16))) { + sect_len_incr = FDKreadBits(bs, nbits); + while (sect_len_incr == sect_esc_val) { + sect_len += sect_esc_val; + sect_len_incr = FDKreadBits(bs, nbits); + } + } else { + sect_len_incr = 1; + } + + sect_len += sect_len_incr; + + top = band + sect_len; + + if (flags & AC_ER_HCR) { + /* HCR input (long) -- collecting sideinfo (for HCR-_long_ only) */ + if (numLinesInSecIdx >= MAX_SFB_HCR) { + return AAC_DEC_PARSE_ERROR; + } + if (top > (int)GetNumberOfScaleFactorBands( + &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo)) { + return AAC_DEC_PARSE_ERROR; + } + pNumLinesInSec[numLinesInSecIdx] = BandOffsets[top] - BandOffsets[band]; + numLinesInSecIdx++; + if (sect_cb == BOOKSCL) { + return AAC_DEC_INVALID_CODE_BOOK; + } else { + *pHcrCodeBook++ = sect_cb; + } + pAacDecoderChannelInfo->pDynData->specificTo.aac.numberSection++; + } + + /* Check spectral line limits */ + if (IsLongBlock(&(pAacDecoderChannelInfo->icsInfo))) { + if (top > 64) { + return AAC_DEC_DECODE_FRAME_ERROR; + } + } else { /* short block */ + if (top + group * 16 > (8 * 16)) { + return AAC_DEC_DECODE_FRAME_ERROR; + } + } + + /* Check if decoded codebook index is feasible */ + if ((sect_cb == BOOKSCL) || + ((sect_cb == INTENSITY_HCB || sect_cb == INTENSITY_HCB2) && + pAacDecoderChannelInfo->pDynData->RawDataInfo.CommonWindow == 0)) { + return AAC_DEC_INVALID_CODE_BOOK; + } + + /* Store codebook index */ + for (; band < top; band++) { + pCodeBook[group * 16 + band] = sect_cb; + } + } + } + + return ErrorStatus; +} + +/* mso: provides a faster way to i-quantize a whole band in one go */ + +/** + * \brief inverse quantize one sfb. Each value of the sfb is processed according + * to the formula: spectrum[i] = Sign(spectrum[i]) * Matissa(spectrum[i])^(4/3) + * * 2^(lsb/4). + * \param spectrum pointer to first line of the sfb to be inverse quantized. + * \param noLines number of lines belonging to the sfb. + * \param lsb last 2 bits of the scale factor of the sfb. + * \param scale max allowed shift scale for the sfb. + */ +static inline void InverseQuantizeBand( + FIXP_DBL *RESTRICT spectrum, const FIXP_DBL *RESTRICT InverseQuantTabler, + const FIXP_DBL *RESTRICT MantissaTabler, + const SCHAR *RESTRICT ExponentTabler, INT noLines, INT scale) { + scale = scale + 1; /* +1 to compensate fMultDiv2 shift-right in loop */ + + FIXP_DBL *RESTRICT ptr = spectrum; + FIXP_DBL signedValue; + + for (INT i = noLines; i--;) { + if ((signedValue = *ptr++) != FL2FXCONST_DBL(0)) { + FIXP_DBL value = fAbs(signedValue); + UINT freeBits = CntLeadingZeros(value); + UINT exponent = 32 - freeBits; + + UINT x = (UINT)(LONG)value << (INT)freeBits; + x <<= 1; /* shift out sign bit to avoid masking later on */ + UINT tableIndex = x >> 24; + x = (x >> 20) & 0x0F; + + UINT r0 = (UINT)(LONG)InverseQuantTabler[tableIndex + 0]; + UINT r1 = (UINT)(LONG)InverseQuantTabler[tableIndex + 1]; + UINT temp = (r1 - r0) * x + (r0 << 4); + + value = fMultDiv2((FIXP_DBL)temp, MantissaTabler[exponent]); + + /* + 1 compensates fMultDiv2() */ + scaleValueInPlace(&value, scale + ExponentTabler[exponent]); + + signedValue = (signedValue < (FIXP_DBL)0) ? -value : value; + ptr[-1] = signedValue; + } + } +} + +static inline FIXP_DBL maxabs_D(const FIXP_DBL *pSpectralCoefficient, + const int noLines) { + /* Find max spectral line value of the current sfb */ + FIXP_DBL locMax = (FIXP_DBL)0; + int i; + + DWORD_ALIGNED(pSpectralCoefficient); + + for (i = noLines; i-- > 0;) { + /* Expensive memory access */ + locMax = fMax(fixp_abs(pSpectralCoefficient[i]), locMax); + } + + return locMax; +} + +AAC_DECODER_ERROR CBlock_InverseQuantizeSpectralData( + CAacDecoderChannelInfo *pAacDecoderChannelInfo, + SamplingRateInfo *pSamplingRateInfo, UCHAR *band_is_noise, + UCHAR active_band_search) { + int window, group, groupwin, band; + int ScaleFactorBandsTransmitted = + GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); + UCHAR *RESTRICT pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook; + SHORT *RESTRICT pSfbScale = pAacDecoderChannelInfo->pDynData->aSfbScale; + SHORT *RESTRICT pScaleFactor = pAacDecoderChannelInfo->pDynData->aScaleFactor; + const SHORT *RESTRICT BandOffsets = GetScaleFactorBandOffsets( + &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); + const SHORT total_bands = + GetScaleFactorBandsTotal(&pAacDecoderChannelInfo->icsInfo); + + FDKmemclear(pAacDecoderChannelInfo->pDynData->aSfbScale, + (8 * 16) * sizeof(SHORT)); + + for (window = 0, group = 0; + group < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++) { + for (groupwin = 0; groupwin < GetWindowGroupLength( + &pAacDecoderChannelInfo->icsInfo, group); + groupwin++, window++) { + /* inverse quantization */ + for (band = 0; band < ScaleFactorBandsTransmitted; band++) { + FIXP_DBL *pSpectralCoefficient = + SPEC(pAacDecoderChannelInfo->pSpectralCoefficient, window, + pAacDecoderChannelInfo->granuleLength) + + BandOffsets[band]; + FIXP_DBL locMax; + + const int noLines = BandOffsets[band + 1] - BandOffsets[band]; + const int bnds = group * 16 + band; + + if ((pCodeBook[bnds] == ZERO_HCB) || + (pCodeBook[bnds] == INTENSITY_HCB) || + (pCodeBook[bnds] == INTENSITY_HCB2)) + continue; + + if (pCodeBook[bnds] == NOISE_HCB) { + /* Leave headroom for PNS values. + 1 because ceil(log2(2^(0.25*3))) = + 1, worst case of additional headroom required because of the + scalefactor. */ + pSfbScale[window * 16 + band] = (pScaleFactor[bnds] >> 2) + 1; + continue; + } + + locMax = maxabs_D(pSpectralCoefficient, noLines); + + if (active_band_search) { + if (locMax != FIXP_DBL(0)) { + band_is_noise[group * 16 + band] = 0; + } + } + + /* Cheap robustness improvement - Do not remove!!! */ + if (fixp_abs(locMax) > (FIXP_DBL)MAX_QUANTIZED_VALUE) { + return AAC_DEC_PARSE_ERROR; + } + + /* Added by Youliy Ninov: + The inverse quantization operation is given by (ISO/IEC 14496-3:2009(E)) + by: + + x_invquant=Sign(x_quant). abs(x_quant)^(4/3) + + We apply a gain, derived from the scale factor for the particular sfb, + according to the following function: + + gain=2^(0.25*ScaleFactor) + + So, after scaling we have: + + x_rescale=gain*x_invquant=Sign(x_quant)*2^(0.25*ScaleFactor)*abs(s_quant)^(4/3) + + We could represent the ScaleFactor as: + + ScaleFactor= (ScaleFactor >> 2)*4 + ScaleFactor %4 + + When we substitute it we get: + + x_rescale=Sign(x_quant)*2^(ScaleFactor>>2)* ( + 2^(0.25*(ScaleFactor%4))*abs(s_quant)^(4/3)) + + When we set: msb=(ScaleFactor>>2) and lsb=(ScaleFactor%4), we obtain: + + x_rescale=Sign(x_quant)*(2^msb)* ( 2^(lsb/4)*abs(s_quant)^(4/3)) + + The rescaled output can be represented by: + mantissa : Sign(x_quant)*( 2^(lsb/4)*abs(s_quant)^(4/3)) + exponent :(2^msb) + + */ + + int msb = pScaleFactor[bnds] >> 2; + + /* Inverse quantize band only if it is not empty */ + if (locMax != FIXP_DBL(0)) { + int lsb = pScaleFactor[bnds] & 0x03; + + int scale = EvaluatePower43(&locMax, lsb); + + scale = CntLeadingZeros(locMax) - scale - 2; + + pSfbScale[window * 16 + band] = msb - scale; + InverseQuantizeBand(pSpectralCoefficient, InverseQuantTable, + MantissaTable[lsb], ExponentTable[lsb], noLines, + scale); + } else { + pSfbScale[window * 16 + band] = msb; + } + + } /* for (band=0; band < ScaleFactorBandsTransmitted; band++) */ + + /* Make sure the array is cleared to the end */ + SHORT start_clear = BandOffsets[ScaleFactorBandsTransmitted]; + SHORT end_clear = BandOffsets[total_bands]; + int diff_clear = (int)(end_clear - start_clear); + FIXP_DBL *pSpectralCoefficient = + SPEC(pAacDecoderChannelInfo->pSpectralCoefficient, window, + pAacDecoderChannelInfo->granuleLength) + + start_clear; + FDKmemclear(pSpectralCoefficient, diff_clear * sizeof(FIXP_DBL)); + + } /* for (groupwin=0; groupwin < + GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo,group); + groupwin++, window++) */ + } /* for (window=0, group=0; group < + GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); group++)*/ + + return AAC_DEC_OK; +} + +AAC_DECODER_ERROR CBlock_ReadSpectralData( + HANDLE_FDK_BITSTREAM bs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, + const SamplingRateInfo *pSamplingRateInfo, const UINT flags) { + int index, i; + const SHORT *RESTRICT BandOffsets = GetScaleFactorBandOffsets( + &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); + + SPECTRAL_PTR pSpectralCoefficient = + pAacDecoderChannelInfo->pSpectralCoefficient; + + FDK_ASSERT(BandOffsets != NULL); + + FDKmemclear(pSpectralCoefficient, sizeof(SPECTRUM)); + + if ((flags & AC_ER_HCR) == 0) { + int group; + int groupoffset; + UCHAR *pCodeBook = pAacDecoderChannelInfo->pDynData->aCodeBook; + int ScaleFactorBandsTransmitted = + GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); + int granuleLength = pAacDecoderChannelInfo->granuleLength; + + groupoffset = 0; + + /* plain huffman decoder short */ + int max_group = GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); + + for (group = 0; group < max_group; group++) { + int max_groupwin = + GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo, group); + int band; + + int bnds = group * 16; + + int bandOffset1 = BandOffsets[0]; + for (band = 0; band < ScaleFactorBandsTransmitted; band++, bnds++) { + UCHAR currentCB = pCodeBook[bnds]; + int bandOffset0 = bandOffset1; + bandOffset1 = BandOffsets[band + 1]; + + /* patch to run plain-huffman-decoder with vcb11 input codebooks + * (LAV-checking might be possible below using the virtual cb and a + * LAV-table) */ + if ((currentCB >= 16) && (currentCB <= 31)) { + pCodeBook[bnds] = currentCB = 11; + } + if (((currentCB != ZERO_HCB) && (currentCB != NOISE_HCB) && + (currentCB != INTENSITY_HCB) && (currentCB != INTENSITY_HCB2))) { + const CodeBookDescription *hcb = + &AACcodeBookDescriptionTable[currentCB]; + int step = hcb->Dimension; + int offset = hcb->Offset; + int bits = hcb->numBits; + int mask = (1 << bits) - 1; + const USHORT(*CodeBook)[HuffmanEntries] = hcb->CodeBook; + int groupwin; + + FIXP_DBL *mdctSpectrum = + &pSpectralCoefficient[groupoffset * granuleLength]; + + if (offset == 0) { + for (groupwin = 0; groupwin < max_groupwin; groupwin++) { + for (index = bandOffset0; index < bandOffset1; index += step) { + int idx = CBlock_DecodeHuffmanWordCB(bs, CodeBook); + for (i = 0; i < step; i++, idx >>= bits) { + FIXP_DBL tmp = (FIXP_DBL)((idx & mask) - offset); + if (tmp != FIXP_DBL(0)) tmp = (FDKreadBit(bs)) ? -tmp : tmp; + mdctSpectrum[index + i] = tmp; + } + + if (currentCB == ESCBOOK) { + for (int j = 0; j < 2; j++) + mdctSpectrum[index + j] = (FIXP_DBL)CBlock_GetEscape( + bs, (LONG)mdctSpectrum[index + j]); + } + } + mdctSpectrum += granuleLength; + } + } else { + for (groupwin = 0; groupwin < max_groupwin; groupwin++) { + for (index = bandOffset0; index < bandOffset1; index += step) { + int idx = CBlock_DecodeHuffmanWordCB(bs, CodeBook); + for (i = 0; i < step; i++, idx >>= bits) { + mdctSpectrum[index + i] = (FIXP_DBL)((idx & mask) - offset); + } + if (currentCB == ESCBOOK) { + for (int j = 0; j < 2; j++) + mdctSpectrum[index + j] = (FIXP_DBL)CBlock_GetEscape( + bs, (LONG)mdctSpectrum[index + j]); + } + } + mdctSpectrum += granuleLength; + } + } + } + } + groupoffset += max_groupwin; + } + /* plain huffman decoding (short) finished */ + } + + /* HCR - Huffman Codeword Reordering short */ + else /* if ( flags & AC_ER_HCR ) */ + + { + H_HCR_INFO hHcr = &pAacDecoderChannelInfo->pComData->overlay.aac.erHcrInfo; + + int hcrStatus = 0; + + /* advanced Huffman decoding starts here (HCR decoding :) */ + if (pAacDecoderChannelInfo->pDynData->specificTo.aac + .lenOfReorderedSpectralData != 0) { + /* HCR initialization short */ + hcrStatus = HcrInit(hHcr, pAacDecoderChannelInfo, pSamplingRateInfo, bs); + + if (hcrStatus != 0) { + return AAC_DEC_DECODE_FRAME_ERROR; + } + + /* HCR decoding short */ + hcrStatus = + HcrDecoder(hHcr, pAacDecoderChannelInfo, pSamplingRateInfo, bs); + + if (hcrStatus != 0) { +#if HCR_ERROR_CONCEALMENT + HcrMuteErroneousLines(hHcr); +#else + return AAC_DEC_DECODE_FRAME_ERROR; +#endif /* HCR_ERROR_CONCEALMENT */ + } + + FDKpushFor(bs, pAacDecoderChannelInfo->pDynData->specificTo.aac + .lenOfReorderedSpectralData); + } + } + /* HCR - Huffman Codeword Reordering short finished */ + + if (IsLongBlock(&pAacDecoderChannelInfo->icsInfo) && + !(flags & (AC_ELD | AC_SCALABLE))) { + /* apply pulse data */ + CPulseData_Apply( + &pAacDecoderChannelInfo->pDynData->specificTo.aac.PulseData, + GetScaleFactorBandOffsets(&pAacDecoderChannelInfo->icsInfo, + pSamplingRateInfo), + SPEC_LONG(pSpectralCoefficient)); + } + + return AAC_DEC_OK; +} + +static const FIXP_SGL noise_level_tab[8] = { + /* FDKpow(2, (float)(noise_level-14)/3.0f) * 2; (*2 to compensate for + fMultDiv2) noise_level_tab(noise_level==0) == 0 by definition + */ + FX_DBL2FXCONST_SGL(0x00000000 /*0x0a145173*/), + FX_DBL2FXCONST_SGL(0x0cb2ff5e), + FX_DBL2FXCONST_SGL(0x10000000), + FX_DBL2FXCONST_SGL(0x1428a2e7), + FX_DBL2FXCONST_SGL(0x1965febd), + FX_DBL2FXCONST_SGL(0x20000000), + FX_DBL2FXCONST_SGL(0x28514606), + FX_DBL2FXCONST_SGL(0x32cbfd33)}; + +void CBlock_ApplyNoise(CAacDecoderChannelInfo *pAacDecoderChannelInfo, + SamplingRateInfo *pSamplingRateInfo, ULONG *nfRandomSeed, + UCHAR *band_is_noise) { + const SHORT *swb_offset = GetScaleFactorBandOffsets( + &pAacDecoderChannelInfo->icsInfo, pSamplingRateInfo); + int g, win, gwin, sfb, noiseFillingStartOffset, nfStartOffset_sfb; + + /* Obtain noise level and scale factor offset. */ + int noise_level = pAacDecoderChannelInfo->pDynData->specificTo.usac + .fd_noise_level_and_offset >> + 5; + const FIXP_SGL noiseVal_pos = noise_level_tab[noise_level]; + + /* noise_offset can change even when noise_level=0. Neccesary for IGF stereo + * filling */ + const int noise_offset = (pAacDecoderChannelInfo->pDynData->specificTo.usac + .fd_noise_level_and_offset & + 0x1f) - + 16; + + int max_sfb = + GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo->icsInfo); + + noiseFillingStartOffset = + (GetWindowSequence(&pAacDecoderChannelInfo->icsInfo) == BLOCK_SHORT) + ? 20 + : 160; + if (pAacDecoderChannelInfo->granuleLength == 96) { + noiseFillingStartOffset = + (3 * noiseFillingStartOffset) / + 4; /* scale offset with 3/4 for coreCoderFrameLength == 768 */ + } + + /* determine sfb from where on noise filling is applied */ + for (sfb = 0; swb_offset[sfb] < noiseFillingStartOffset; sfb++) + ; + nfStartOffset_sfb = sfb; + + /* if (noise_level!=0) */ + { + for (g = 0, win = 0; g < GetWindowGroups(&pAacDecoderChannelInfo->icsInfo); + g++) { + int windowGroupLength = + GetWindowGroupLength(&pAacDecoderChannelInfo->icsInfo, g); + for (sfb = nfStartOffset_sfb; sfb < max_sfb; sfb++) { + int bin_start = swb_offset[sfb]; + int bin_stop = swb_offset[sfb + 1]; + + int flagN = band_is_noise[g * 16 + sfb]; + + /* if all bins of one sfb in one window group are zero modify the scale + * factor by noise_offset */ + if (flagN) { + /* Change scaling factors for empty signal bands */ + pAacDecoderChannelInfo->pDynData->aScaleFactor[g * 16 + sfb] += + noise_offset; + /* scale factor "sf" implied gain "g" is g = 2^(sf/4) */ + for (gwin = 0; gwin < windowGroupLength; gwin++) { + pAacDecoderChannelInfo->pDynData + ->aSfbScale[(win + gwin) * 16 + sfb] += (noise_offset >> 2); + } + } + + ULONG seed = *nfRandomSeed; + /* + 1 because exponent of MantissaTable[lsb][0] is always 1. */ + int scale = + (pAacDecoderChannelInfo->pDynData->aScaleFactor[g * 16 + sfb] >> + 2) + + 1; + int lsb = + pAacDecoderChannelInfo->pDynData->aScaleFactor[g * 16 + sfb] & 3; + FIXP_DBL mantissa = MantissaTable[lsb][0]; + + for (gwin = 0; gwin < windowGroupLength; gwin++) { + FIXP_DBL *pSpec = + SPEC(pAacDecoderChannelInfo->pSpectralCoefficient, win + gwin, + pAacDecoderChannelInfo->granuleLength); + + int scale1 = scale - pAacDecoderChannelInfo->pDynData + ->aSfbScale[(win + gwin) * 16 + sfb]; + FIXP_DBL scaled_noiseVal_pos = + scaleValue(fMultDiv2(noiseVal_pos, mantissa), scale1); + FIXP_DBL scaled_noiseVal_neg = -scaled_noiseVal_pos; + + /* If the whole band is zero, just fill without checking */ + if (flagN) { + for (int bin = bin_start; bin < bin_stop; bin++) { + seed = (ULONG)( + (UINT64)seed * 69069 + + 5); /* Inlined: UsacRandomSign - origin in usacdec_lpd.h */ + pSpec[bin] = + (seed & 0x10000) ? scaled_noiseVal_neg : scaled_noiseVal_pos; + } /* for (bin...) */ + } + /*If band is sparsely filled, check for 0 and fill */ + else { + for (int bin = bin_start; bin < bin_stop; bin++) { + if (pSpec[bin] == (FIXP_DBL)0) { + seed = (ULONG)( + (UINT64)seed * 69069 + + 5); /* Inlined: UsacRandomSign - origin in usacdec_lpd.h */ + pSpec[bin] = (seed & 0x10000) ? scaled_noiseVal_neg + : scaled_noiseVal_pos; + } + } /* for (bin...) */ + } + + } /* for (gwin...) */ + *nfRandomSeed = seed; + } /* for (sfb...) */ + win += windowGroupLength; + } /* for (g...) */ + + } /* ... */ +} + +AAC_DECODER_ERROR CBlock_ReadAcSpectralData( + HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, + const SamplingRateInfo *pSamplingRateInfo, const UINT frame_length, + const UINT flags) { + AAC_DECODER_ERROR errorAAC = AAC_DEC_OK; + ARITH_CODING_ERROR error = ARITH_CODER_OK; + int arith_reset_flag, lg, numWin, win, winLen; + const SHORT *RESTRICT BandOffsets; + + /* number of transmitted spectral coefficients */ + BandOffsets = GetScaleFactorBandOffsets(&pAacDecoderChannelInfo->icsInfo, + pSamplingRateInfo); + lg = BandOffsets[GetScaleFactorBandsTransmitted( + &pAacDecoderChannelInfo->icsInfo)]; + + numWin = GetWindowsPerFrame(&pAacDecoderChannelInfo->icsInfo); + winLen = (IsLongBlock(&pAacDecoderChannelInfo->icsInfo)) + ? (int)frame_length + : (int)frame_length / numWin; + + if (flags & AC_INDEP) { + arith_reset_flag = 1; + } else { + arith_reset_flag = (USHORT)FDKreadBits(hBs, 1); + } + + for (win = 0; win < numWin; win++) { + error = + CArco_DecodeArithData(pAacDecoderStaticChannelInfo->hArCo, hBs, + SPEC(pAacDecoderChannelInfo->pSpectralCoefficient, + win, pAacDecoderChannelInfo->granuleLength), + lg, winLen, arith_reset_flag && (win == 0)); + if (error != ARITH_CODER_OK) { + goto bail; + } + } + +bail: + if (error == ARITH_CODER_ERROR) { + errorAAC = AAC_DEC_PARSE_ERROR; + } + + return errorAAC; +} + +void ApplyTools(CAacDecoderChannelInfo *pAacDecoderChannelInfo[], + const SamplingRateInfo *pSamplingRateInfo, const UINT flags, + const UINT elFlags, const int channel, + const int common_window) { + if (!(flags & (AC_USAC | AC_RSVD50 | AC_MPEGD_RES | AC_RSV603DA))) { + CPns_Apply(&pAacDecoderChannelInfo[channel]->data.aac.PnsData, + &pAacDecoderChannelInfo[channel]->icsInfo, + pAacDecoderChannelInfo[channel]->pSpectralCoefficient, + pAacDecoderChannelInfo[channel]->specScale, + pAacDecoderChannelInfo[channel]->pDynData->aScaleFactor, + pSamplingRateInfo, + pAacDecoderChannelInfo[channel]->granuleLength, channel); + } + + UCHAR nbands = + GetScaleFactorBandsTransmitted(&pAacDecoderChannelInfo[channel]->icsInfo); + + CTns_Apply(&pAacDecoderChannelInfo[channel]->pDynData->TnsData, + &pAacDecoderChannelInfo[channel]->icsInfo, + pAacDecoderChannelInfo[channel]->pSpectralCoefficient, + pSamplingRateInfo, pAacDecoderChannelInfo[channel]->granuleLength, + nbands, (elFlags & AC_EL_ENHANCED_NOISE) ? 1 : 0, flags); +} + +static int getWindow2Nr(int length, int shape) { + int nr = 0; + + if (shape == 2) { + /* Low Overlap, 3/4 zeroed */ + nr = (length * 3) >> 2; + } + + return nr; +} + +FIXP_DBL get_gain(const FIXP_DBL *x, const FIXP_DBL *y, int n) { + FIXP_DBL corr = (FIXP_DBL)0; + FIXP_DBL ener = (FIXP_DBL)1; + + int headroom_x = getScalefactor(x, n); + int headroom_y = getScalefactor(y, n); + + /*Calculate the normalization necessary due to addition*/ + /* Check for power of two /special case */ + INT width_shift = (INT)(fNormz((FIXP_DBL)n)); + /* Get the number of bits necessary minus one, because we need one sign bit + * only */ + width_shift = 31 - width_shift; + + for (int i = 0; i < n; i++) { + corr += + fMultDiv2((x[i] << headroom_x), (y[i] << headroom_y)) >> width_shift; + ener += fPow2Div2((y[i] << headroom_y)) >> width_shift; + } + + int exp_corr = (17 - headroom_x) + (17 - headroom_y) + width_shift + 1; + int exp_ener = ((17 - headroom_y) << 1) + width_shift + 1; + + int temp_exp = 0; + FIXP_DBL output = fDivNormSigned(corr, ener, &temp_exp); + + int output_exp = (exp_corr - exp_ener) + temp_exp; + + INT output_shift = 17 - output_exp; + output_shift = fMin(output_shift, 31); + + output = scaleValue(output, -output_shift); + + return output; +} + +void CBlock_FrequencyToTime( + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, + CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[], + const SHORT frameLen, const int frameOk, FIXP_DBL *pWorkBuffer1, + UINT elFlags, INT elCh) { + int fr, fl, tl, nSpec; + +#if defined(FDK_ASSERT_ENABLE) + LONG nSamples; +#endif + + /* Determine left slope length (fl), right slope length (fr) and transform + length (tl). USAC: The slope length may mismatch with the previous frame in + case of LPD / FD transitions. The adjustment is handled by the imdct + implementation. + */ + tl = frameLen; + nSpec = 1; + + switch (pAacDecoderChannelInfo->icsInfo.WindowSequence) { + default: + case BLOCK_LONG: + fl = frameLen; + fr = frameLen - + getWindow2Nr(frameLen, + GetWindowShape(&pAacDecoderChannelInfo->icsInfo)); + /* New startup needs differentiation between sine shape and low overlap + shape. This is a special case for the LD-AAC transformation windows, + because the slope length can be different while using the same window + sequence. */ + if (pAacDecoderStaticChannelInfo->IMdct.prev_tl == 0) { + fl = fr; + } + break; + case BLOCK_STOP: + fl = frameLen >> 3; + fr = frameLen; + break; + case BLOCK_START: /* or StopStartSequence */ + fl = frameLen; + fr = frameLen >> 3; + break; + case BLOCK_SHORT: + fl = fr = frameLen >> 3; + tl >>= 3; + nSpec = 8; + break; + } + + { + int last_frame_lost = pAacDecoderStaticChannelInfo->last_lpc_lost; + + if (pAacDecoderStaticChannelInfo->last_core_mode == LPD) { + INT fac_FB = 1; + if (elFlags & AC_EL_FULLBANDLPD) { + fac_FB = 2; + } + + FIXP_DBL *synth; + + /* Keep some free space at the beginning of the buffer. To be used for + * past data */ + if (!(elFlags & AC_EL_LPDSTEREOIDX)) { + synth = pWorkBuffer1 + ((PIT_MAX_MAX - (1 * L_SUBFR)) * fac_FB); + } else { + synth = pWorkBuffer1 + PIT_MAX_MAX * fac_FB; + } + + int fac_length = + (pAacDecoderChannelInfo->icsInfo.WindowSequence == BLOCK_SHORT) + ? (frameLen >> 4) + : (frameLen >> 3); + + INT pitch[NB_SUBFR_SUPERFR + SYN_SFD]; + FIXP_DBL pit_gain[NB_SUBFR_SUPERFR + SYN_SFD]; + + int nbDiv = (elFlags & AC_EL_FULLBANDLPD) ? 2 : 4; + int lFrame = (elFlags & AC_EL_FULLBANDLPD) ? frameLen / 2 : frameLen; + int nbSubfr = + lFrame / (nbDiv * L_SUBFR); /* number of subframes per division */ + int LpdSfd = (nbDiv * nbSubfr) >> 1; + int SynSfd = LpdSfd - BPF_SFD; + + FDKmemclear( + pitch, + sizeof( + pitch)); // added to prevent ferret errors in bass_pf_1sf_delay + FDKmemclear(pit_gain, sizeof(pit_gain)); + + /* FAC case */ + if (pAacDecoderStaticChannelInfo->last_lpd_mode == 0 || + pAacDecoderStaticChannelInfo->last_lpd_mode == 4) { + FIXP_DBL fac_buf[LFAC]; + FIXP_LPC *A = pAacDecoderChannelInfo->data.usac.lp_coeff[0]; + + if (!frameOk || last_frame_lost || + (pAacDecoderChannelInfo->data.usac.fac_data[0] == NULL)) { + FDKmemclear(fac_buf, + pAacDecoderChannelInfo->granuleLength * sizeof(FIXP_DBL)); + pAacDecoderChannelInfo->data.usac.fac_data[0] = fac_buf; + pAacDecoderChannelInfo->data.usac.fac_data_e[0] = 0; + } + + INT A_exp; /* linear prediction coefficients exponent */ + { + for (int i = 0; i < M_LP_FILTER_ORDER; i++) { + A[i] = FX_DBL2FX_LPC(fixp_cos( + fMult(pAacDecoderStaticChannelInfo->lpc4_lsf[i], + FL2FXCONST_SGL((1 << LSPARG_SCALE) * M_PI / 6400.0)), + LSF_SCALE - LSPARG_SCALE)); + } + + E_LPC_f_lsp_a_conversion(A, A, &A_exp); + } + +#if defined(FDK_ASSERT_ENABLE) + nSamples = +#endif + CLpd_FAC_Acelp2Mdct( + &pAacDecoderStaticChannelInfo->IMdct, synth, + SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient), + pAacDecoderChannelInfo->specScale, nSpec, + pAacDecoderChannelInfo->data.usac.fac_data[0], + pAacDecoderChannelInfo->data.usac.fac_data_e[0], fac_length, + frameLen, tl, + FDKgetWindowSlope( + fr, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), + fr, A, A_exp, &pAacDecoderStaticChannelInfo->acelp, + (FIXP_DBL)0, /* FAC gain has already been applied. */ + (last_frame_lost || !frameOk), 1, + pAacDecoderStaticChannelInfo->last_lpd_mode, 0, + pAacDecoderChannelInfo->currAliasingSymmetry); + + } else { +#if defined(FDK_ASSERT_ENABLE) + nSamples = +#endif + imlt_block( + &pAacDecoderStaticChannelInfo->IMdct, synth, + SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient), + pAacDecoderChannelInfo->specScale, nSpec, frameLen, tl, + FDKgetWindowSlope( + fl, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), + fl, + FDKgetWindowSlope( + fr, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), + fr, (FIXP_DBL)0, + pAacDecoderChannelInfo->currAliasingSymmetry + ? MLT_FLAG_CURR_ALIAS_SYMMETRY + : 0); + } + FDK_ASSERT(nSamples == frameLen); + + /* The "if" clause is entered both for fullbandLpd mono and + * non-fullbandLpd*. The "else"-> just for fullbandLpd stereo*/ + if (!(elFlags & AC_EL_LPDSTEREOIDX)) { + FDKmemcpy(pitch, pAacDecoderStaticChannelInfo->old_T_pf, + SynSfd * sizeof(INT)); + FDKmemcpy(pit_gain, pAacDecoderStaticChannelInfo->old_gain_pf, + SynSfd * sizeof(FIXP_DBL)); + + for (int i = SynSfd; i < LpdSfd + 3; i++) { + pitch[i] = L_SUBFR; + pit_gain[i] = (FIXP_DBL)0; + } + + if (pAacDecoderStaticChannelInfo->last_lpd_mode == 0) { + pitch[SynSfd] = pitch[SynSfd - 1]; + pit_gain[SynSfd] = pit_gain[SynSfd - 1]; + if (IsLongBlock(&pAacDecoderChannelInfo->icsInfo)) { + pitch[SynSfd + 1] = pitch[SynSfd]; + pit_gain[SynSfd + 1] = pit_gain[SynSfd]; + } + } + + /* Copy old data to the beginning of the buffer */ + { + FDKmemcpy( + pWorkBuffer1, pAacDecoderStaticChannelInfo->old_synth, + ((PIT_MAX_MAX - (1 * L_SUBFR)) * fac_FB) * sizeof(FIXP_DBL)); + } + + FIXP_DBL *p2_synth = pWorkBuffer1 + (PIT_MAX_MAX * fac_FB); + + /* recalculate pitch gain to allow postfilering on FAC area */ + for (int i = 0; i < SynSfd + 2; i++) { + int T = pitch[i]; + FIXP_DBL gain = pit_gain[i]; + + if (gain > (FIXP_DBL)0) { + gain = get_gain(&p2_synth[i * L_SUBFR * fac_FB], + &p2_synth[(i * L_SUBFR * fac_FB) - fac_FB * T], + L_SUBFR * fac_FB); + pit_gain[i] = gain; + } + } + + bass_pf_1sf_delay(p2_synth, pitch, pit_gain, frameLen, + (LpdSfd + 2) * L_SUBFR + BPF_SFD * L_SUBFR, + frameLen - (LpdSfd + 4) * L_SUBFR, outSamples, + pAacDecoderStaticChannelInfo->mem_bpf); + } + + } else /* last_core_mode was not LPD */ + { + FIXP_DBL *tmp = + pAacDecoderChannelInfo->pComStaticData->pWorkBufferCore1->mdctOutTemp; +#if defined(FDK_ASSERT_ENABLE) + nSamples = +#endif + imlt_block(&pAacDecoderStaticChannelInfo->IMdct, tmp, + SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient), + pAacDecoderChannelInfo->specScale, nSpec, frameLen, tl, + FDKgetWindowSlope( + fl, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), + fl, + FDKgetWindowSlope( + fr, GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), + fr, (FIXP_DBL)0, + pAacDecoderChannelInfo->currAliasingSymmetry + ? MLT_FLAG_CURR_ALIAS_SYMMETRY + : 0); + + scaleValuesSaturate(outSamples, tmp, frameLen, MDCT_OUT_HEADROOM); + } + } + + FDK_ASSERT(nSamples == frameLen); + + pAacDecoderStaticChannelInfo->last_core_mode = + (pAacDecoderChannelInfo->icsInfo.WindowSequence == BLOCK_SHORT) ? FD_SHORT + : FD_LONG; + pAacDecoderStaticChannelInfo->last_lpd_mode = 255; +} + +#include "ldfiltbank.h" +void CBlock_FrequencyToTimeLowDelay( + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, + CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[], + const short frameLen) { + InvMdctTransformLowDelay_fdk( + SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient), + pAacDecoderChannelInfo->specScale[0], outSamples, + pAacDecoderStaticChannelInfo->pOverlapBuffer, frameLen); +} |