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Diffstat (limited to 'fdk-aac/libAACdec/src/channelinfo.h')
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diff --git a/fdk-aac/libAACdec/src/channelinfo.h b/fdk-aac/libAACdec/src/channelinfo.h new file mode 100644 index 0000000..4523400 --- /dev/null +++ b/fdk-aac/libAACdec/src/channelinfo.h @@ -0,0 +1,564 @@ +/* ----------------------------------------------------------------------------- +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: individual channel stream info + +*******************************************************************************/ + +#ifndef CHANNELINFO_H +#define CHANNELINFO_H + +#include "common_fix.h" + +#include "aac_rom.h" +#include "aacdecoder_lib.h" +#include "FDK_bitstream.h" +#include "overlapadd.h" + +#include "mdct.h" +#include "stereo.h" +#include "pulsedata.h" +#include "aacdec_tns.h" + +#include "aacdec_pns.h" + +#include "aacdec_hcr_types.h" +#include "rvlc_info.h" + +#include "usacdec_acelp.h" +#include "usacdec_const.h" +#include "usacdec_rom.h" + +#include "ac_arith_coder.h" + +#include "conceal_types.h" + +#include "aacdec_drc_types.h" + +#define WB_SECTION_SIZE (1024 * 2) + +#define DRM_BS_BUFFER_SIZE \ + (512) /* size of the dynamic buffer which is used to reverse the bits of \ + the DRM SBR payload */ + +/* Output rendering mode */ +typedef enum { + AACDEC_RENDER_INVALID = 0, + AACDEC_RENDER_IMDCT, + AACDEC_RENDER_ELDFB, + AACDEC_RENDER_LPD, + AACDEC_RENDER_INTIMDCT +} AACDEC_RENDER_MODE; + +enum { MAX_QUANTIZED_VALUE = 8191 }; + +typedef enum { FD_LONG, FD_SHORT, LPD } USAC_COREMODE; + +typedef struct { + const SHORT *ScaleFactorBands_Long; + const SHORT *ScaleFactorBands_Short; + UCHAR NumberOfScaleFactorBands_Long; + UCHAR NumberOfScaleFactorBands_Short; + UINT samplingRateIndex; + UINT samplingRate; +} SamplingRateInfo; + +typedef struct { + UCHAR CommonWindow; + UCHAR GlobalGain; + +} CRawDataInfo; + +typedef struct { + UCHAR WindowGroupLength[8]; + UCHAR WindowGroups; + UCHAR Valid; + + UCHAR WindowShape; /* 0: sine window, 1: KBD, 2: low overlap */ + BLOCK_TYPE WindowSequence; /* mdct.h; 0: long, 1: start, 2: short, 3: stop */ + UCHAR MaxSfBands; + UCHAR max_sfb_ste; + UCHAR ScaleFactorGrouping; + + UCHAR TotalSfBands; + +} CIcsInfo; + +enum { + ZERO_HCB = 0, + ESCBOOK = 11, + NSPECBOOKS = ESCBOOK + 1, + BOOKSCL = NSPECBOOKS, + NOISE_HCB = 13, + INTENSITY_HCB2 = 14, + INTENSITY_HCB = 15, + LAST_HCB +}; + +/* This struct holds the persistent data shared by both channels of a CPE. + It needs to be allocated for each CPE. */ +typedef struct { + CJointStereoPersistentData jointStereoPersistentData; +} CpePersistentData; + +/* + * This struct must be allocated one for every channel and must be persistent. + */ +typedef struct { + FIXP_DBL *pOverlapBuffer; + mdct_t IMdct; + + CArcoData *hArCo; + + INT pnsCurrentSeed; + + /* LPD memory */ + FIXP_DBL old_synth[PIT_MAX_MAX - L_SUBFR]; + INT old_T_pf[SYN_SFD]; + FIXP_DBL old_gain_pf[SYN_SFD]; + FIXP_DBL mem_bpf[L_FILT + L_SUBFR]; + UCHAR + old_bpf_control_info; /* (1: enable, 0: disable) bpf for past superframe + */ + + USAC_COREMODE last_core_mode; /* core mode used by the decoder in previous + frame. (not signalled by the bitstream, see + CAacDecoderChannelInfo::core_mode_last !! ) + */ + UCHAR last_lpd_mode; /* LPD mode used by the decoder in last LPD subframe + (not signalled by the bitstream, see + CAacDecoderChannelInfo::lpd_mode_last !! ) */ + UCHAR last_last_lpd_mode; /* LPD mode used in second last LPD subframe + (not signalled by the bitstream) */ + UCHAR last_lpc_lost; /* Flag indicating that the previous LPC is lost */ + + FIXP_LPC + lpc4_lsf[M_LP_FILTER_ORDER]; /* Last LPC4 coefficients in LSF domain. */ + FIXP_LPC lsf_adaptive_mean[M_LP_FILTER_ORDER]; /* Adaptive mean of LPC + coefficients in LSF domain + for concealment. */ + FIXP_LPC lp_coeff_old[2][M_LP_FILTER_ORDER]; /* Last LPC coefficients in LP + domain. lp_coeff_old[0] is lpc4 (coeffs for + right folding point of last tcx frame), + lp_coeff_old[1] are coeffs for left folding + point of last tcx frame */ + INT lp_coeff_old_exp[2]; + + FIXP_SGL + oldStability; /* LPC coeff stability value from last frame (required for + TCX concealment). */ + UINT numLostLpdFrames; /* Number of consecutive lost subframes. */ + + /* TCX memory */ + FIXP_DBL last_tcx_gain; + INT last_tcx_gain_e; + FIXP_DBL last_alfd_gains[32]; /* Scaled by one bit. */ + SHORT last_tcx_pitch; + UCHAR last_tcx_noise_factor; + + /* ACELP memory */ + CAcelpStaticMem acelp; + + ULONG nfRandomSeed; /* seed value for USAC noise filling random generator */ + + CDrcChannelData drcData; + CConcealmentInfo concealmentInfo; + + CpePersistentData *pCpeStaticData; + +} CAacDecoderStaticChannelInfo; + +/* + * This union must be allocated for every element (up to 2 channels). + */ +typedef struct { + /* Common bit stream data */ + SHORT aScaleFactor[( + 8 * 16)]; /* Spectral scale factors for each sfb in each window. */ + SHORT aSfbScale[(8 * 16)]; /* could be free after ApplyTools() */ + UCHAR + aCodeBook[(8 * 16)]; /* section data: codebook for each window and sfb. */ + UCHAR band_is_noise[(8 * 16)]; + CTnsData TnsData; + CRawDataInfo RawDataInfo; + + shouldBeUnion { + struct { + CPulseData PulseData; + SHORT aNumLineInSec4Hcr[MAX_SFB_HCR]; /* needed once for all channels + except for Drm syntax */ + UCHAR + aCodeBooks4Hcr[MAX_SFB_HCR]; /* needed once for all channels except for + Drm syntax. Same as "aCodeBook" ? */ + SHORT lenOfReorderedSpectralData; + SCHAR lenOfLongestCodeword; + SCHAR numberSection; + SCHAR rvlcCurrentScaleFactorOK; + SCHAR rvlcIntensityUsed; + } aac; + struct { + UCHAR fd_noise_level_and_offset; + UCHAR tns_active; + UCHAR tns_on_lr; + UCHAR tcx_noise_factor[4]; + UCHAR tcx_global_gain[4]; + } usac; + } + specificTo; + +} CAacDecoderDynamicData; + +typedef shouldBeUnion { + UCHAR DrmBsBuffer[DRM_BS_BUFFER_SIZE]; + + /* Common signal data, can be used once the bit stream data from above is not + * used anymore. */ + FIXP_DBL mdctOutTemp[1024]; + + FIXP_DBL synth_buf[(PIT_MAX_MAX + SYN_DELAY + L_FRAME_PLUS)]; + + FIXP_DBL workBuffer[WB_SECTION_SIZE]; +} +CWorkBufferCore1; + +/* Common data referenced by all channels */ +typedef struct { + CAacDecoderDynamicData pAacDecoderDynamicData[2]; + + CPnsInterChannelData pnsInterChannelData; + INT pnsRandomSeed[(8 * 16)]; + + CJointStereoData jointStereoData; /* One for one element */ + + shouldBeUnion { + struct { + CErHcrInfo erHcrInfo; + CErRvlcInfo erRvlcInfo; + SHORT aRvlcScfEsc[RVLC_MAX_SFB]; /* needed once for all channels */ + SHORT aRvlcScfFwd[RVLC_MAX_SFB]; /* needed once for all channels */ + SHORT aRvlcScfBwd[RVLC_MAX_SFB]; /* needed once for all channels */ + } aac; + } + overlay; + +} CAacDecoderCommonData; + +typedef struct { + CWorkBufferCore1 *pWorkBufferCore1; + CCplxPredictionData *cplxPredictionData; +} CAacDecoderCommonStaticData; + +/* + * This struct must be allocated one for every channel of every element and must + * be persistent. Among its members, the following memory areas can be + * overwritten under the given conditions: + * - pSpectralCoefficient The memory pointed to can be overwritten after time + * signal rendering. + * - data can be overwritten after time signal rendering. + * - pDynData memory pointed to can be overwritten after each + * CChannelElement_Decode() call. + * - pComData->overlay memory pointed to can be overwritten after each + * CChannelElement_Decode() call.. + */ +typedef struct { + shouldBeUnion { + struct { + FIXP_DBL fac_data0[LFAC]; + SCHAR fac_data_e[4]; + FIXP_DBL + *fac_data[4]; /* Pointers to unused parts of pSpectralCoefficient */ + + UCHAR core_mode; /* current core mode */ + USAC_COREMODE + core_mode_last; /* previous core mode, signalled in the bitstream + (not done by the decoder, see + CAacDecoderStaticChannelInfo::last_core_mode !!)*/ + UCHAR lpd_mode_last; /* previous LPD mode, signalled in the bitstream + (not done by the decoder, see + CAacDecoderStaticChannelInfo::last_core_mode !!)*/ + UCHAR mod[4]; + UCHAR bpf_control_info; /* (1: enable, 0: disable) bpf for current + superframe */ + + FIXP_LPC lsp_coeff[5][M_LP_FILTER_ORDER]; /* linear prediction + coefficients in LSP domain */ + FIXP_LPC + lp_coeff[5][M_LP_FILTER_ORDER]; /* linear prediction coefficients in + LP domain */ + INT lp_coeff_exp[5]; + FIXP_LPC lsf_adaptive_mean_cand + [M_LP_FILTER_ORDER]; /* concealment: is copied to + CAacDecoderStaticChannelInfo->lsf_adaptive_mean once frame is + assumed to be correct*/ + FIXP_SGL aStability[4]; /* LPC coeff stability values required for ACELP + and TCX (concealment) */ + + CAcelpChannelData acelp[4]; + + FIXP_DBL tcx_gain[4]; + SCHAR tcx_gain_e[4]; + } usac; + + struct { + CPnsData PnsData; /* Not required for USAC */ + } aac; + } + data; + + SPECTRAL_PTR pSpectralCoefficient; /* Spectral coefficients of each window */ + SHORT specScale[8]; /* Scale shift values of each spectrum window */ + CIcsInfo icsInfo; + INT granuleLength; /* Size of smallest spectrum piece */ + UCHAR ElementInstanceTag; + + AACDEC_RENDER_MODE renderMode; /* Output signal rendering mode */ + + CAacDecoderDynamicData * + pDynData; /* Data required for one element and discarded after decoding */ + CAacDecoderCommonData + *pComData; /* Data required for one channel at a time during decode */ + CAacDecoderCommonStaticData *pComStaticData; /* Persistent data required for + one channel at a time during + decode */ + + int currAliasingSymmetry; /* required for RSVD60 MCT */ + +} CAacDecoderChannelInfo; + +/* channelinfo.cpp */ + +AAC_DECODER_ERROR getSamplingRateInfo(SamplingRateInfo *t, UINT samplesPerFrame, + UINT samplingRateIndex, + UINT samplingRate); + +/** + * \brief Read max SFB from bit stream and assign TotalSfBands according + * to the window sequence and sample rate. + * \param hBs bit stream handle as data source + * \param pIcsInfo IcsInfo structure to read the window sequence and store + * MaxSfBands and TotalSfBands + * \param pSamplingRateInfo read only + */ +AAC_DECODER_ERROR IcsReadMaxSfb(HANDLE_FDK_BITSTREAM hBs, CIcsInfo *pIcsInfo, + const SamplingRateInfo *pSamplingRateInfo); + +AAC_DECODER_ERROR IcsRead(HANDLE_FDK_BITSTREAM bs, CIcsInfo *pIcsInfo, + const SamplingRateInfo *SamplingRateInfoTable, + const UINT flags); + +/* stereo.cpp, only called from this file */ + +/*! + \brief Applies MS stereo. + + The function applies MS stereo. + + \param pAacDecoderChannelInfo aac channel info. + \param pScaleFactorBandOffsets pointer to scalefactor band offsets. + \param pWindowGroupLength pointer to window group length array. + \param windowGroups number of window groups. + \param scaleFactorBandsTransmittedL number of transmitted scalefactor bands in + left channel. \param scaleFactorBandsTransmittedR number of transmitted + scalefactor bands in right channel. May differ from + scaleFactorBandsTransmittedL only for USAC. \return none +*/ +void CJointStereo_ApplyMS( + CAacDecoderChannelInfo *pAacDecoderChannelInfo[2], + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[2], + FIXP_DBL *spectrumL, FIXP_DBL *spectrumR, SHORT *SFBleftScale, + SHORT *SFBrightScale, SHORT *specScaleL, SHORT *specScaleR, + const SHORT *pScaleFactorBandOffsets, const UCHAR *pWindowGroupLength, + const int windowGroups, const int max_sfb_ste_outside, + const int scaleFactorBandsTransmittedL, + const int scaleFactorBandsTransmittedR, FIXP_DBL *store_dmx_re_prev, + SHORT *store_dmx_re_prev_e, const int mainband_flag); + +/*! + \brief Applies intensity stereo + + The function applies intensity stereo. + + \param pAacDecoderChannelInfo aac channel info. + \param pScaleFactorBandOffsets pointer to scalefactor band offsets. + \param pWindowGroupLength pointer to window group length array. + \param windowGroups number of window groups. + \param scaleFactorBandsTransmitted number of transmitted scalefactor bands. + \return none +*/ +void CJointStereo_ApplyIS(CAacDecoderChannelInfo *pAacDecoderChannelInfo[2], + const short *pScaleFactorBandOffsets, + const UCHAR *pWindowGroupLength, + const int windowGroups, + const int scaleFactorBandsTransmitted); + +/* aacdec_pns.cpp */ +int CPns_IsPnsUsed(const CPnsData *pPnsData, const int group, const int band); + +void CPns_SetCorrelation(CPnsData *pPnsData, const int group, const int band, + const int outofphase); + +/****************** inline functions ******************/ + +inline UCHAR IsValid(const CIcsInfo *pIcsInfo) { return pIcsInfo->Valid; } + +inline UCHAR IsLongBlock(const CIcsInfo *pIcsInfo) { + return (pIcsInfo->WindowSequence != BLOCK_SHORT); +} + +inline UCHAR GetWindowShape(const CIcsInfo *pIcsInfo) { + return pIcsInfo->WindowShape; +} + +inline BLOCK_TYPE GetWindowSequence(const CIcsInfo *pIcsInfo) { + return pIcsInfo->WindowSequence; +} + +inline const SHORT *GetScaleFactorBandOffsets( + const CIcsInfo *pIcsInfo, const SamplingRateInfo *samplingRateInfo) { + if (IsLongBlock(pIcsInfo)) { + return samplingRateInfo->ScaleFactorBands_Long; + } else { + return samplingRateInfo->ScaleFactorBands_Short; + } +} + +inline UCHAR GetNumberOfScaleFactorBands( + const CIcsInfo *pIcsInfo, const SamplingRateInfo *samplingRateInfo) { + if (IsLongBlock(pIcsInfo)) { + return samplingRateInfo->NumberOfScaleFactorBands_Long; + } else { + return samplingRateInfo->NumberOfScaleFactorBands_Short; + } +} + +inline int GetWindowsPerFrame(const CIcsInfo *pIcsInfo) { + return (pIcsInfo->WindowSequence == BLOCK_SHORT) ? 8 : 1; +} + +inline UCHAR GetWindowGroups(const CIcsInfo *pIcsInfo) { + return pIcsInfo->WindowGroups; +} + +inline UCHAR GetWindowGroupLength(const CIcsInfo *pIcsInfo, const INT index) { + return pIcsInfo->WindowGroupLength[index]; +} + +inline const UCHAR *GetWindowGroupLengthTable(const CIcsInfo *pIcsInfo) { + return pIcsInfo->WindowGroupLength; +} + +inline UCHAR GetScaleFactorBandsTransmitted(const CIcsInfo *pIcsInfo) { + return pIcsInfo->MaxSfBands; +} + +inline UCHAR GetScaleMaxFactorBandsTransmitted(const CIcsInfo *pIcsInfo0, + const CIcsInfo *pIcsInfo1) { + return fMax(pIcsInfo0->MaxSfBands, pIcsInfo1->MaxSfBands); +} + +inline UCHAR GetScaleFactorBandsTotal(const CIcsInfo *pIcsInfo) { + return pIcsInfo->TotalSfBands; +} + +/* Note: This function applies to AAC-LC only ! */ +inline UCHAR GetMaximumTnsBands(const CIcsInfo *pIcsInfo, + const int samplingRateIndex) { + return tns_max_bands_tbl[samplingRateIndex][!IsLongBlock(pIcsInfo)]; +} + +#endif /* #ifndef CHANNELINFO_H */ |