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author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
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committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
commit | 0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch) | |
tree | d07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libAACdec/src/block.h | |
parent | efe406d9724f959c8bc2a31802559ca6d41fd897 (diff) | |
download | ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.gz ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.bz2 ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.zip |
Include patched FDK-AAC in the repository
The initial idea was to get the DAB+ patch into upstream, but since
that follows the android source releases, there is no place for a custom
DAB+ patch there.
So instead of having to maintain a patched fdk-aac that has to have the
same .so version as the distribution package on which it is installed,
we prefer having a separate fdk-aac-dab library to avoid collision.
At that point, there's no reason to keep fdk-aac in a separate
repository, as odr-audioenc is the only tool that needs DAB+ encoding
support. Including it here simplifies installation, and makes it
consistent with toolame-dab, also shipped in this repository.
DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop,
welle.io and others) can be done using upstream FDK-AAC.
Diffstat (limited to 'fdk-aac/libAACdec/src/block.h')
-rw-r--r-- | fdk-aac/libAACdec/src/block.h | 345 |
1 files changed, 345 insertions, 0 deletions
diff --git a/fdk-aac/libAACdec/src/block.h b/fdk-aac/libAACdec/src/block.h new file mode 100644 index 0000000..f0f56cd --- /dev/null +++ b/fdk-aac/libAACdec/src/block.h @@ -0,0 +1,345 @@ +/* ----------------------------------------------------------------------------- +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 + +*******************************************************************************/ + +#ifndef BLOCK_H +#define BLOCK_H + +#include "common_fix.h" + +#include "channelinfo.h" +#include "FDK_bitstream.h" + +/* PNS (of block) */ +void CPns_Read(CPnsData *pPnsData, HANDLE_FDK_BITSTREAM bs, + const CodeBookDescription *hcb, SHORT *pScaleFactor, + UCHAR global_gain, int band, int group); + +void CPns_Apply(const CPnsData *pPnsData, const CIcsInfo *pIcsInfo, + SPECTRAL_PTR pSpectrum, const SHORT *pSpecScale, + const SHORT *pScaleFactor, + const SamplingRateInfo *pSamplingRateInfo, + const INT granuleLength, const int channel); + +void CBlock_ApplyNoise(CAacDecoderChannelInfo *pAacDecoderChannelInfo, + SamplingRateInfo *pSamplingRateInfo, ULONG *nfRandomSeed, + UCHAR *band_is_noise); + +/* TNS (of block) */ +/*! + \brief Read tns data-present flag from bitstream + + The function reads the data-present flag for tns from + the bitstream. + + \return none +*/ +void CTns_ReadDataPresentFlag(HANDLE_FDK_BITSTREAM bs, CTnsData *pTnsData); + +void CTns_ReadDataPresentUsac(HANDLE_FDK_BITSTREAM hBs, CTnsData *pTnsData0, + CTnsData *pTnsData1, UCHAR *ptns_on_lr, + const CIcsInfo *pIcsInfo, const UINT flags, + const UINT elFlags, const int fCommonWindow); + +AAC_DECODER_ERROR CTns_Read(HANDLE_FDK_BITSTREAM bs, CTnsData *pTnsData, + const CIcsInfo *pIcsInfo, const UINT flags); + +void CTns_Apply(CTnsData *RESTRICT pTnsData, /*!< pointer to aac decoder info */ + const CIcsInfo *pIcsInfo, SPECTRAL_PTR pSpectralCoefficient, + const SamplingRateInfo *pSamplingRateInfo, + const INT granuleLength, const UCHAR nbands, + const UCHAR igf_active, const UINT flags); + +/* Block */ + +LONG CBlock_GetEscape(HANDLE_FDK_BITSTREAM bs, const LONG q); + +/** + * \brief Read scale factor data. See chapter 4.6.2.3.2 of ISO/IEC 14496-3. + * The SF_OFFSET = 100 value referenced in chapter 4.6.2.3.3 is already + * substracted from the scale factor values. Also includes PNS data reading. + * \param bs bit stream handle data source + * \param pAacDecoderChannelInfo channel context info were decoded data is + * stored into. + * \param flags the decoder flags. + */ +AAC_DECODER_ERROR CBlock_ReadScaleFactorData( + CAacDecoderChannelInfo *pAacDecoderChannelInfo, HANDLE_FDK_BITSTREAM bs, + const UINT flags); + +/** + * \brief Read Huffman encoded spectral data. + * \param pAacDecoderChannelInfo channel context info. + * \param pSamplingRateInfo sampling rate info (sfb offsets). + * \param flags syntax flags. + */ +AAC_DECODER_ERROR CBlock_ReadSpectralData( + HANDLE_FDK_BITSTREAM bs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, + const SamplingRateInfo *pSamplingRateInfo, const UINT flags); + +/** + * \brief Read Arithmetic encoded spectral data. + * \param pAacDecoderChannelInfo channel context info. + * \param pAacDecoderStaticChannelInfo static channel context info. + * \param pSamplingRateInfo sampling rate info (sfb offsets). + * \param frame_length spectral window length. + * \param flags syntax flags. + * \return error code. + */ +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 CBlock_ReadSectionData( + HANDLE_FDK_BITSTREAM bs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, + const SamplingRateInfo *pSamplingRateInfo, const UINT flags); + +/** + * \brief find a common exponent (shift factor) for all sfb in each Spectral + * window, and store them into CAacDecoderChannelInfo::specScale. + * \param pAacDecoderChannelInfo channel context info. + * \param UCHAR maxSfbs maximum number of SFBs to be processed (might differ + * from pAacDecoderChannelInfo->icsInfo.MaxSfBands) + * \param pSamplingRateInfo sampling rate info (sfb offsets). + */ +void CBlock_ScaleSpectralData(CAacDecoderChannelInfo *pAacDecoderChannelInfo, + UCHAR maxSfbs, + SamplingRateInfo *pSamplingRateInfo); + +/** + * \brief Apply TNS and PNS tools. + */ +void ApplyTools(CAacDecoderChannelInfo *pAacDecoderChannelInfo[], + const SamplingRateInfo *pSamplingRateInfo, const UINT flags, + const UINT elFlags, const int channel, const int maybe_jstereo); + +/** + * \brief Transform MDCT spectral data into time domain + */ +void CBlock_FrequencyToTime( + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, + CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[], + const SHORT frameLen, const int frameOk, FIXP_DBL *pWorkBuffer1, + UINT elFlags, INT elCh); + +/** + * \brief Transform double lapped MDCT (AAC-ELD) spectral data into time domain. + */ +void CBlock_FrequencyToTimeLowDelay( + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, + CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[], + const short frameLen); + +AAC_DECODER_ERROR CBlock_InverseQuantizeSpectralData( + CAacDecoderChannelInfo *pAacDecoderChannelInfo, + SamplingRateInfo *pSamplingRateInfo, UCHAR *band_is_noise, + UCHAR active_band_search); + +/** + * \brief Calculate 2^(lsb/4) * value^(4/3) + * \param pValue pointer to quantized value. The inverse quantized result is + * stored back here. + * \param lsb 2 LSBs of the scale factor (scaleFactor % 4) applied as power 2 + * factor to the resulting inverse quantized value. + * \return the exponent of the result (mantissa) stored into *pValue. + */ +FDK_INLINE +int EvaluatePower43(FIXP_DBL *pValue, UINT lsb) { + FIXP_DBL value; + UINT freeBits; + UINT exponent; + + value = *pValue; + freeBits = fNormz(value); + exponent = DFRACT_BITS - freeBits; + FDK_ASSERT(exponent < 14); + + UINT x = (((int)value << freeBits) >> 19); + UINT tableIndex = (x & 0x0FFF) >> 4; + FIXP_DBL invQVal; + + x = x & 0x0F; + + UINT r0 = (LONG)InverseQuantTable[tableIndex + 0]; + UINT r1 = (LONG)InverseQuantTable[tableIndex + 1]; + USHORT nx = 16 - x; + UINT temp = (r0)*nx + (r1)*x; + invQVal = (FIXP_DBL)temp; + + FDK_ASSERT(lsb < 4); + *pValue = fMultDiv2(invQVal, MantissaTable[lsb][exponent]); + + /* + 1 compensates fMultDiv2(). */ + return ExponentTable[lsb][exponent] + 1; +} + +/* Recalculate gain */ +FIXP_DBL get_gain(const FIXP_DBL *x, const FIXP_DBL *y, int n); + +/** + * \brief determine the required shift scale for the given quantized value and + * scale (factor % 4) value. + */ +FDK_INLINE int GetScaleFromValue(FIXP_DBL value, unsigned int lsb) { + if (value != (FIXP_DBL)0) { + int scale = EvaluatePower43(&value, lsb); + return CntLeadingZeros(value) - scale - 2; + } else + return 0; /* Return zero, because its useless to scale a zero value, saves + workload and avoids scaling overshifts. */ +} + +/*! + \brief Read huffman codeword + + The function reads the huffman codeword from the bitstream and + returns the index value. + + \return index value +*/ +inline int CBlock_DecodeHuffmanWord( + HANDLE_FDK_BITSTREAM bs, /*!< pointer to bitstream */ + const CodeBookDescription *hcb) /*!< pointer to codebook description */ +{ + UINT val; + UINT index = 0; + const USHORT(*CodeBook)[HuffmanEntries] = hcb->CodeBook; + + while (1) { + val = CodeBook[index] + [FDKreadBits(bs, HuffmanBits)]; /* Expensive memory access */ + + if ((val & 1) == 0) { + index = val >> 2; + continue; + } else { + if (val & 2) { + FDKpushBackCache(bs, 1); + } + + val >>= 2; + break; + } + } + + return val; +} +inline int CBlock_DecodeHuffmanWordCB( + HANDLE_FDK_BITSTREAM bs, /*!< pointer to bitstream */ + const USHORT ( + *CodeBook)[HuffmanEntries]) /*!< pointer to codebook description */ +{ + UINT index = 0; + + while (1) { + index = CodeBook[index][FDKread2Bits(bs)]; /* Expensive memory access */ + if (index & 1) break; + index >>= 2; + } + if (index & 2) { + FDKpushBackCache(bs, 1); + } + return index >> 2; +} + +#endif /* #ifndef BLOCK_H */ |