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Created: 2024-06-17 06:33

/src/aac/libAACdec/src/usacdec_acelp.h
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/* -----------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
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Forschung e.V. All rights reserved.
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 1.    INTRODUCTION
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The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
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that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
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scheme for digital audio. This FDK AAC Codec software is intended to be used on
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a wide variety of Android devices.
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AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
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general perceptual audio codecs. AAC-ELD is considered the best-performing
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full-bandwidth communications codec by independent studies and is widely
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deployed. AAC has been standardized by ISO and IEC as part of the MPEG
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specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including
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those of Fraunhofer) may be obtained through Via Licensing
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(www.vialicensing.com) or through the respective patent owners individually for
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the purpose of encoding or decoding bit streams in products that are compliant
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with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
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Android devices already license these patent claims through Via Licensing or
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directly from the patent owners, and therefore FDK AAC Codec software may
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already be covered under those patent licenses when it is used for those
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licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions
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with enhanced sound quality, are also available from Fraunhofer. Users are
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encouraged to check the Fraunhofer website for additional applications
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information and documentation.
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2.    COPYRIGHT LICENSE
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Redistribution and use in source and binary forms, with or without modification,
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are permitted without payment of copyright license fees provided that you
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satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of
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the FDK AAC Codec or your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation
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and/or other materials provided with redistributions of the FDK AAC Codec or
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your modifications thereto in binary form. You must make available free of
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charge copies of the complete source code of the FDK AAC Codec and your
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modifications thereto to recipients of copies in binary form.
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The name of Fraunhofer may not be used to endorse or promote products derived
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from this library without prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute
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the FDK AAC Codec software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating
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that you changed the software and the date of any change. For modified versions
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of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
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must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
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AAC Codec Library for Android."
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3.    NO PATENT LICENSE
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NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
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limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
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Fraunhofer provides no warranty of patent non-infringement with respect to this
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software.
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You may use this FDK AAC Codec software or modifications thereto only for
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purposes that are authorized by appropriate patent licenses.
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4.    DISCLAIMER
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This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
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holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
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including but not limited to the implied warranties of merchantability and
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fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
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CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
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or consequential damages, including but not limited to procurement of substitute
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goods or services; loss of use, data, or profits, or business interruption,
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however caused and on any theory of liability, whether in contract, strict
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liability, or tort (including negligence), arising in any way out of the use of
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this software, even if advised of the possibility of such damage.
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5.    CONTACT INFORMATION
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Fraunhofer Institute for Integrated Circuits IIS
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Attention: Audio and Multimedia Departments - FDK AAC LL
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Am Wolfsmantel 33
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91058 Erlangen, Germany
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www.iis.fraunhofer.de/amm
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amm-info@iis.fraunhofer.de
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----------------------------------------------------------------------------- */
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/**************************** AAC decoder library ******************************
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   Author(s):   Matthias Hildenbrand
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   Description: USAC ACELP frame decoder
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*******************************************************************************/
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#ifndef USACDEC_ACELP_H
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#define USACDEC_ACELP_H
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#include "common_fix.h"
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#include "FDK_bitstream.h"
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#include "usacdec_const.h"
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#include "usacdec_rom.h"
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//#define ENHANCED_TCX_TD_CONCEAL_ENABLE
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/** Structure which holds the ACELP internal persistent memory */
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typedef struct {
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  FIXP_DBL old_exc_mem[PIT_MAX_MAX + L_INTERPOL];
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  FIXP_DBL old_syn_mem[M_LP_FILTER_ORDER]; /* synthesis filter states */
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  FIXP_SGL A[M_LP_FILTER_ORDER];
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  INT A_exp;
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  FIXP_DBL gc_threshold;
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  FIXP_DBL de_emph_mem;
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  FIXP_SGL past_gpit;
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  FIXP_DBL past_gcode;
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  USHORT old_T0;
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  UCHAR old_T0_frac;
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  FIXP_DBL deemph_mem_wsyn;
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  FIXP_DBL wsyn_rms;
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  SHORT seed_ace;
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} CAcelpStaticMem;
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/** Structure which holds the parameter data needed to decode one ACELP frame.
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 */
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typedef struct {
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  UCHAR
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  acelp_core_mode;   /**< mean excitation energy index for whole ACELP frame
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                      */
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  UCHAR mean_energy; /**< acelp core mode for whole ACELP frame */
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  USHORT T0[NB_SUBFR];
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  UCHAR T0_frac[NB_SUBFR];
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  UCHAR ltp_filtering_flag[NB_SUBFR]; /**< controlls whether LTP postfilter is
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                                         active for each ACELP subframe */
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  SHORT icb_index[NB_SUBFR]
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                 [8]; /**< innovative codebook index for each ACELP subframe */
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  UCHAR gains[NB_SUBFR]; /**< gain index for each ACELP subframe */
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} CAcelpChannelData;
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/**
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 * \brief Read the acelp_coding() bitstream part.
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 * \param[in] hBs bitstream handle to read data from.
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 * \param[out] acelpData pointer to structure to store the parsed data of one
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 * ACELP frame.
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 * \param[in] acelp_core_mode the ACELP core mode index.
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 * \param[in] coreCoderFrameLength length of core coder frame (1024|768)
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 */
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INT CLpd_AcelpRead(HANDLE_FDK_BITSTREAM hBs, CAcelpChannelData *acelpData,
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                   INT acelp_core_mode, INT i_offset, INT coreCoderFrameLength);
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/**
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 * \brief Initialization of memory before one LPD frame is decoded
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 * \param[out] synth_buf synthesis buffer to be initialized, exponent = SF_SYNTH
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 * \param[in] old_synth past synthesis of previous LPD frame, exponent =
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 * SF_SYNTH
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 * \param[out] synth_buf_fb fullband synthesis buffer to be initialized,
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 * exponent = SF_SYNTH
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 * \param[in] old_synth_fb past fullband synthesis of previous LPD frame,
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 * exponent = SF_SYNTH
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 * \param[out] pitch vector where decoded pitch lag values are stored
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 * \param[in] old_T_pf past pitch lag values of previous LPD frame
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 * \param[in] samplingRate sampling rate for pitch lag offset calculation
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 * \param[out] i_offset pitch lag offset for the decoding of the pitch lag
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 * \param[in] coreCoderFrameLength length of core coder frame (1024|768)
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 */
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void Acelp_PreProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch,
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                         INT *old_T_pf, FIXP_DBL *pit_gain,
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                         FIXP_DBL *old_gain_pf, INT samplingRate, INT *i_offset,
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                         INT coreCoderFrameLength, INT synSfd,
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                         INT nbSubfrSuperfr);
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/**
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 * \brief Save tail of buffers for the initialization of the next LPD frame
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 * \param[in] synth_buf synthesis of current LPD frame, exponent = SF_SYNTH
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 * \param[out] old_synth memory where tail of fullband synth_buf is stored,
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 * exponent = SF_SYNTH
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 * \param[in] synth_buf_fb fullband synthesis of current LPD frame, exponent =
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 * SF_SYNTH
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 * \param[out] old_synth_fb memory where tail of fullband synth_buf is stored,
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 * exponent = SF_SYNTH
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 * \param[in] pitch decoded pitch lag values of current LPD frame
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 * \param[out] old_T_pf memory where last SYN_SFD pitch lag values are stored
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 */
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void Acelp_PostProcessing(FIXP_DBL *synth_buf, FIXP_DBL *old_synth, INT *pitch,
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                          INT *old_T_pf, INT coreCoderFrameLength, INT synSfd,
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                          INT nbSubfrSuperfr);
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/**
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 * \brief Decode one ACELP frame (three or four ACELP subframes with 64 samples
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 * each)
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 * \param[in,out] acelp_mem pointer to ACELP memory structure
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 * \param[in] i_offset pitch lag offset
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 * \param[in] lsp_old LPC filter in LSP domain corresponding to previous frame
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 * \param[in] lsp_new LPC filter in LSP domain corresponding to current frame
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 * \param[in] stab_fac stability factor constrained by 0<=stab_fac<=1.0,
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 * exponent = SF_STAB
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 * \param[in] acelpData pointer to struct with data which is needed for decoding
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 * one ACELP frame
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 * \param[out] synth ACELP output signal
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 * \param[out] pT four decoded pitch lag values
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 * \param[in] coreCoderFrameLength length of core coder frame (1024|768)
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 */
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void CLpd_AcelpDecode(CAcelpStaticMem *acelp_mem, INT i_offset,
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                      const FIXP_LPC lsp_old[M_LP_FILTER_ORDER],
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                      const FIXP_LPC lsp_new[M_LP_FILTER_ORDER],
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                      FIXP_SGL stab_fac, CAcelpChannelData *acelpData,
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                      INT numLostSubframes, int lastLpcLost, int frameCnt,
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                      FIXP_DBL synth[], int pT[], FIXP_DBL *pit_gain,
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                      INT coreCoderFrameLength);
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/**
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 * \brief Reset ACELP internal memory.
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 * \param[out] acelp_mem pointer to ACELP memory structure
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 */
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void CLpd_AcelpReset(CAcelpStaticMem *acelp_mem);
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/**
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 * \brief Initialize ACELP internal memory in case of FAC before ACELP decoder
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 * is called
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 * \param[in] synth points to end+1 of past valid synthesis signal, exponent =
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 * SF_SYNTH
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 * \param[in] last_lpd_mode last lpd mode
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 * \param[in] last_last_lpd_mode lpd mode before last_lpd_mode
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 * \param[in] A_new LP synthesis filter coeffs corresponding to last frame,
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 * exponent = SF_A_COEFFS
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 * \param[in] A_old LP synthesis filter coeffs corresponding to the frame before
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 * last frame, exponent = SF_A_COEFFS
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 * \param[in,out] acelp_mem pointer to ACELP memory structure
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 * \param[in] coreCoderFrameLength length of core coder frame (1024|768)
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 */
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void CLpd_AcelpPrepareInternalMem(const FIXP_DBL *synth, UCHAR last_lpd_mode,
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                                  UCHAR last_last_lpd_mode,
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                                  const FIXP_LPC *A_new, const INT A_new_exp,
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                                  const FIXP_LPC *A_old, const INT A_old_exp,
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                                  CAcelpStaticMem *acelp_mem,
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                                  INT coreCoderFrameLength, INT clearOldExc,
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                                  UCHAR lpd_mode);
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/**
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 * \brief Calculate zero input response (zir) of the acelp synthesis filter
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 * \param[in] A LP synthesis filter coefficients, exponent = SF_A_COEFFS
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 * \param[in,out] acelp_mem pointer to ACELP memory structure
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 * \param[in] length length of zir
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 * \param[out] zir pointer to zir output buffer, exponent = SF_SYNTH
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 */
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void CLpd_Acelp_Zir(const FIXP_LPC A[], const INT A_exp,
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                    CAcelpStaticMem *acelp_mem, const INT length,
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                    FIXP_DBL zir[], int doDeemph);
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/**
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 * \brief Borrow static excitation memory from ACELP decoder
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 * \param[in] acelp_mem pointer to ACELP memory structure
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 * \param[in] length number of requested FIXP_DBL values
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 * \return pointer to requested memory
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 *
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 * The caller has to take care not to overwrite valid memory areas.
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 * During TCX/FAC calculations and before CLpd_AcelpPrepareInternalMem() is
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 * called, the following memory size is available:
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 * - 256 samples in case of ACELP -> TCX20 -> ACELP transition
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 * - PIT_MAX_MAX+L_INTERPOL samples in all other cases
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 */
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FIXP_DBL *CLpd_ACELP_GetFreeExcMem(CAcelpStaticMem *acelp_mem, INT length);
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void CLpd_TcxTDConceal(CAcelpStaticMem *acelp_mem, SHORT *pitch,
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                       const FIXP_LPC lsp_old[M_LP_FILTER_ORDER],
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                       const FIXP_LPC lsp_new[M_LP_FILTER_ORDER],
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                       const FIXP_SGL stab_fac, INT numLostSubframes,
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                       FIXP_DBL synth[], INT coreCoderFrameLength,
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                       UCHAR last_tcx_noise_factor);
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481k
inline SHORT E_UTIL_random(SHORT *seed) {
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  *seed = (SHORT)((((LONG)*seed * (LONG)31821) >> 1) + (LONG)13849);
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  return (*seed);
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}
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#endif /* USACDEC_ACELP_H */