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      1 /* -----------------------------------------------------------------------------
      2 Software License for The Fraunhofer FDK AAC Codec Library for Android
      3 
      4  Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Frderung der angewandten
      5 Forschung e.V. All rights reserved.
      6 
      7  1.    INTRODUCTION
      8 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
      9 that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
     10 scheme for digital audio. This FDK AAC Codec software is intended to be used on
     11 a wide variety of Android devices.
     12 
     13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
     14 general perceptual audio codecs. AAC-ELD is considered the best-performing
     15 full-bandwidth communications codec by independent studies and is widely
     16 deployed. AAC has been standardized by ISO and IEC as part of the MPEG
     17 specifications.
     18 
     19 Patent licenses for necessary patent claims for the FDK AAC Codec (including
     20 those of Fraunhofer) may be obtained through Via Licensing
     21 (www.vialicensing.com) or through the respective patent owners individually for
     22 the purpose of encoding or decoding bit streams in products that are compliant
     23 with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
     24 Android devices already license these patent claims through Via Licensing or
     25 directly from the patent owners, and therefore FDK AAC Codec software may
     26 already be covered under those patent licenses when it is used for those
     27 licensed purposes only.
     28 
     29 Commercially-licensed AAC software libraries, including floating-point versions
     30 with enhanced sound quality, are also available from Fraunhofer. Users are
     31 encouraged to check the Fraunhofer website for additional applications
     32 information and documentation.
     33 
     34 2.    COPYRIGHT LICENSE
     35 
     36 Redistribution and use in source and binary forms, with or without modification,
     37 are permitted without payment of copyright license fees provided that you
     38 satisfy the following conditions:
     39 
     40 You must retain the complete text of this software license in redistributions of
     41 the FDK AAC Codec or your modifications thereto in source code form.
     42 
     43 You must retain the complete text of this software license in the documentation
     44 and/or other materials provided with redistributions of the FDK AAC Codec or
     45 your modifications thereto in binary form. You must make available free of
     46 charge copies of the complete source code of the FDK AAC Codec and your
     47 modifications thereto to recipients of copies in binary form.
     48 
     49 The name of Fraunhofer may not be used to endorse or promote products derived
     50 from this library without prior written permission.
     51 
     52 You may not charge copyright license fees for anyone to use, copy or distribute
     53 the FDK AAC Codec software or your modifications thereto.
     54 
     55 Your modified versions of the FDK AAC Codec must carry prominent notices stating
     56 that you changed the software and the date of any change. For modified versions
     57 of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
     58 must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
     59 AAC Codec Library for Android."
     60 
     61 3.    NO PATENT LICENSE
     62 
     63 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
     64 limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
     65 Fraunhofer provides no warranty of patent non-infringement with respect to this
     66 software.
     67 
     68 You may use this FDK AAC Codec software or modifications thereto only for
     69 purposes that are authorized by appropriate patent licenses.
     70 
     71 4.    DISCLAIMER
     72 
     73 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
     74 holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
     75 including but not limited to the implied warranties of merchantability and
     76 fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
     77 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
     78 or consequential damages, including but not limited to procurement of substitute
     79 goods or services; loss of use, data, or profits, or business interruption,
     80 however caused and on any theory of liability, whether in contract, strict
     81 liability, or tort (including negligence), arising in any way out of the use of
     82 this software, even if advised of the possibility of such damage.
     83 
     84 5.    CONTACT INFORMATION
     85 
     86 Fraunhofer Institute for Integrated Circuits IIS
     87 Attention: Audio and Multimedia Departments - FDK AAC LL
     88 Am Wolfsmantel 33
     89 91058 Erlangen, Germany
     90 
     91 www.iis.fraunhofer.de/amm
     92 amm-info (at) iis.fraunhofer.de
     93 ----------------------------------------------------------------------------- */
     94 
     95 /**************************** SBR encoder library ******************************
     96 
     97    Author(s):   Markus Multrus
     98 
     99    Description: PS Wrapper, Downmix header file
    100 
    101 *******************************************************************************/
    102 
    103 #ifndef PS_MAIN_H
    104 #define PS_MAIN_H
    105 
    106 /* Includes ******************************************************************/
    107 
    108 #include "sbr_def.h"
    109 #include "qmf.h"
    110 #include "ps_encode.h"
    111 #include "FDK_bitstream.h"
    112 #include "FDK_hybrid.h"
    113 
    114 /* Data Types ****************************************************************/
    115 typedef enum {
    116   PSENC_STEREO_BANDS_INVALID = 0,
    117   PSENC_STEREO_BANDS_10 = 10,
    118   PSENC_STEREO_BANDS_20 = 20
    119 
    120 } PSENC_STEREO_BANDS_CONFIG;
    121 
    122 typedef enum {
    123   PSENC_NENV_1 = 1,
    124   PSENC_NENV_2 = 2,
    125   PSENC_NENV_4 = 4,
    126   PSENC_NENV_DEFAULT = PSENC_NENV_2,
    127   PSENC_NENV_MAX = PSENC_NENV_4
    128 
    129 } PSENC_NENV_CONFIG;
    130 
    131 typedef struct {
    132   UINT bitrateFrom; /* inclusive */
    133   UINT bitrateTo;   /* exclusive */
    134   PSENC_STEREO_BANDS_CONFIG nStereoBands;
    135   PSENC_NENV_CONFIG nEnvelopes;
    136   LONG iidQuantErrorThreshold; /* quantization threshold to switch between
    137                                   coarse and fine iid quantization */
    138 
    139 } psTuningTable_t;
    140 
    141 /* Function / Class Declarations *********************************************/
    142 
    143 typedef struct T_PARAMETRIC_STEREO {
    144   HANDLE_PS_ENCODE hPsEncode;
    145   PS_OUT psOut[2];
    146 
    147   FIXP_DBL __staticHybridData[HYBRID_READ_OFFSET][MAX_PS_CHANNELS][2]
    148                              [MAX_HYBRID_BANDS];
    149   FIXP_DBL
    150   *pHybridData[HYBRID_READ_OFFSET + HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2];
    151 
    152   FIXP_DBL qmfDelayLines[2][32 >> 1][64];
    153   int qmfDelayScale;
    154 
    155   INT psDelay;
    156   UINT maxEnvelopes;
    157   UCHAR dynBandScale[PS_MAX_BANDS];
    158   FIXP_DBL maxBandValue[PS_MAX_BANDS];
    159   SCHAR dmxScale;
    160   INT initPS;
    161   INT noQmfSlots;
    162   INT noQmfBands;
    163 
    164   FIXP_DBL __staticHybAnaStatesLF[MAX_PS_CHANNELS][2 * HYBRID_FILTER_LENGTH *
    165                                                    HYBRID_MAX_QMF_BANDS];
    166   FIXP_DBL __staticHybAnaStatesHF[MAX_PS_CHANNELS][2 * HYBRID_FILTER_DELAY *
    167                                                    (64 - HYBRID_MAX_QMF_BANDS)];
    168   FDK_ANA_HYB_FILTER fdkHybAnaFilter[MAX_PS_CHANNELS];
    169   FDK_SYN_HYB_FILTER fdkHybSynFilter;
    170 
    171 } PARAMETRIC_STEREO;
    172 
    173 typedef struct T_PSENC_CONFIG {
    174   INT frameSize;
    175   INT qmfFilterMode;
    176   INT sbrPsDelay;
    177   PSENC_STEREO_BANDS_CONFIG nStereoBands;
    178   PSENC_NENV_CONFIG maxEnvelopes;
    179   FIXP_DBL iidQuantErrorThreshold;
    180 
    181 } PSENC_CONFIG, *HANDLE_PSENC_CONFIG;
    182 
    183 typedef struct T_PARAMETRIC_STEREO *HANDLE_PARAMETRIC_STEREO;
    184 
    185 /**
    186  * \brief  Create a parametric stereo encoder instance.
    187  *
    188  * \param phParametricStereo    A pointer to a parametric stereo handle to be
    189  * allocated. Initialized on return.
    190  *
    191  * \return
    192  *          - PSENC_OK, on succes.
    193  *          - PSENC_INVALID_HANDLE, PSENC_MEMORY_ERROR, on failure.
    194  */
    195 FDK_PSENC_ERROR PSEnc_Create(HANDLE_PARAMETRIC_STEREO *phParametricStereo);
    196 
    197 /**
    198  * \brief  Initialize a parametric stereo encoder instance.
    199  *
    200  * \param hParametricStereo     Meta Data handle.
    201  * \param hPsEncConfig          Filled parametric stereo configuration
    202  * structure.
    203  * \param noQmfSlots            Number of slots within one audio frame.
    204  * \param noQmfBands            Number of QMF bands.
    205  * \param dynamic_RAM           Pointer to preallocated workbuffer.
    206  *
    207  * \return
    208  *          - PSENC_OK, on succes.
    209  *          - PSENC_INVALID_HANDLE, PSENC_INIT_ERROR, on failure.
    210  */
    211 FDK_PSENC_ERROR PSEnc_Init(HANDLE_PARAMETRIC_STEREO hParametricStereo,
    212                            const HANDLE_PSENC_CONFIG hPsEncConfig,
    213                            INT noQmfSlots, INT noQmfBands, UCHAR *dynamic_RAM);
    214 
    215 /**
    216  * \brief  Destroy parametric stereo encoder instance.
    217  *
    218  * Deallocate instance and free whole memory.
    219  *
    220  * \param phParametricStereo    Pointer to the parametric stereo handle to be
    221  * deallocated.
    222  *
    223  * \return
    224  *          - PSENC_OK, on succes.
    225  *          - PSENC_INVALID_HANDLE, on failure.
    226  */
    227 FDK_PSENC_ERROR PSEnc_Destroy(HANDLE_PARAMETRIC_STEREO *phParametricStereo);
    228 
    229 /**
    230  * \brief  Apply parametric stereo processing.
    231  *
    232  * \param hParametricStereo     Meta Data handle.
    233  * \param samples               Pointer to 2 channel audio input signal.
    234  * \param timeInStride,         Stride factor of input buffer.
    235  * \param hQmfAnalysis,         Pointer to QMF analysis filterbanks.
    236  * \param downmixedRealQmfData  Pointer to real QMF buffer to be written to.
    237  * \param downmixedImagQmfData  Pointer to imag QMF buffer to be written to.
    238  * \param downsampledOutSignal  Pointer to buffer where to write downmixed
    239  * timesignal.
    240  * \param sbrSynthQmf           Pointer to QMF synthesis filterbank.
    241  * \param qmfScale              Return scaling factor of the qmf data.
    242  * \param sendHeader            Signal whether to write header data.
    243  *
    244  * \return
    245  *          - PSENC_OK, on succes.
    246  *          - PSENC_INVALID_HANDLE, PSENC_ENCODE_ERROR, on failure.
    247  */
    248 FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing(
    249     HANDLE_PARAMETRIC_STEREO hParametricStereo, INT_PCM *samples[2],
    250     UINT timeInStride, QMF_FILTER_BANK **hQmfAnalysis,
    251     FIXP_DBL **RESTRICT downmixedRealQmfData,
    252     FIXP_DBL **RESTRICT downmixedImagQmfData, INT_PCM *downsampledOutSignal,
    253     HANDLE_QMF_FILTER_BANK sbrSynthQmf, SCHAR *qmfScale, const int sendHeader);
    254 
    255 /**
    256  * \brief  Write parametric stereo bitstream.
    257  *
    258  * Write ps_data() element to bitstream and return number of written bits.
    259  * Returns number of written bits only, if hBitstream == NULL.
    260  *
    261  * \param hParametricStereo     Meta Data handle.
    262  * \param hBitstream            Bitstream buffer handle.
    263  *
    264  * \return
    265  *          - number of written bits.
    266  */
    267 INT FDKsbrEnc_PSEnc_WritePSData(HANDLE_PARAMETRIC_STEREO hParametricStereo,
    268                                 HANDLE_FDK_BITSTREAM hBitstream);
    269 
    270 #endif /* PS_MAIN_H */
    271