xref: /aosp_15_r20/external/aac/libAACdec/src/usacdec_ace_ltp.cpp (revision e54365361535b070c2db7374cec45c159c7d0e7a)

/* -----------------------------------------------------------------------------
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
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charge copies of the complete source code of the FDK AAC Codec and your
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limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
Fraunhofer provides no warranty of patent non-infringement with respect to this
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You may use this FDK AAC Codec software or modifications thereto only for
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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,
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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
[email protected]
----------------------------------------------------------------------------- */

/**************************** AAC decoder library ******************************

   Author(s):   Matthias Hildenbrand

   Description: USAC ACELP LTP filter

*******************************************************************************/

#include "usacdec_ace_ltp.h"

#include "genericStds.h"
#include "common_fix.h"

#define UP_SAMP 4
#define L_INTERPOL2 16
#define L_SUBFR 64

#define A2 FL2FX_SGL(2 * 0.18f)
#define B FL2FX_SGL(0.64f)

static const LONG Pred_lt4_inter4_2[UP_SAMP][L_INTERPOL2] = {
    {(LONG)0x0000FFFC, (LONG)0x0008FFFC, (LONG)0xFFEB004C, (LONG)0xFF50014A,
     (LONG)0xFDD90351, (LONG)0xFB2A06CD, (LONG)0xF6920D46, (LONG)0xEBB42B35,
     (LONG)0x6D9EEF39, (LONG)0x0618FE0F, (LONG)0xFFE00131, (LONG)0xFE5501C5,
     (LONG)0xFE5E015D, (LONG)0xFEF700B6, (LONG)0xFF920037, (LONG)0xFFEC0003},
    {(LONG)0x0002FFF2, (LONG)0x0026FFBD, (LONG)0x005DFF98, (LONG)0x0055FFEF,
     (LONG)0xFF89015F, (LONG)0xFD3A04E5, (LONG)0xF7D90DAA, (LONG)0xE67A50EE,
     (LONG)0x50EEE67A, (LONG)0x0DAAF7D9, (LONG)0x04E5FD3A, (LONG)0x015FFF89,
     (LONG)0xFFEF0055, (LONG)0xFF98005D, (LONG)0xFFBD0026, (LONG)0xFFF20002},
    {(LONG)0x0003FFEC, (LONG)0x0037FF92, (LONG)0x00B6FEF7, (LONG)0x015DFE5E,
     (LONG)0x01C5FE55, (LONG)0x0131FFE0, (LONG)0xFE0F0618, (LONG)0xEF396D9E,
     (LONG)0x2B35EBB4, (LONG)0x0D46F692, (LONG)0x06CDFB2A, (LONG)0x0351FDD9,
     (LONG)0x014AFF50, (LONG)0x004CFFEB, (LONG)0xFFFC0008, (LONG)0xFFFC0000},
    {(LONG)0x0002FFF2, (LONG)0x002BFF9E, (LONG)0x00B9FECE, (LONG)0x01CFFD75,
     (LONG)0x035EFBC1, (LONG)0x0521FA0C, (LONG)0x06AAF8C9, (LONG)0x07907852,
     (LONG)0x0790F8C9, (LONG)0x06AAFA0C, (LONG)0x0521FBC1, (LONG)0x035EFD75,
     (LONG)0x01CFFECE, (LONG)0x00B9FF9E, (LONG)0x002BFFF2, (LONG)0x00020000}};

void Pred_lt4(FIXP_DBL exc[], /* in/out: excitation buffer              */
              int T0,         /* input : integer pitch lag              */
              int frac        /* input : fraction of lag in range 0..3  */
) {
  int j;
  FIXP_DBL *x;
  const LONG *interpol;
  FIXP_DBL L_sumb, L_sumt;

  x = &exc[-T0 - L_INTERPOL2 + 1];

  /* remap frac and x:
           0 -> 3   x (unchanged)
           1 -> 0   x--
           2 -> 1   x--
           3 -> 2   x--
  */

  if (--frac < 0)
    frac += UP_SAMP;
  else
    x--;

  j = L_SUBFR + 1;
  do {
    LONG filt;
    FIXP_DBL x0, x1;
    FIXP_DBL *xi = x++;
    interpol = Pred_lt4_inter4_2[frac];
    int i = 3;

    filt = *interpol++;
    x0 = *xi++;
    x1 = *xi++;
    L_sumt = fMultDiv2(x0, (FIXP_SGL)((SHORT)(filt >> 16)));
    L_sumb = fMultDiv2(x1, (FIXP_SGL)((SHORT)filt));
    do {
      filt = *interpol++;
      x0 = *xi++;
      x1 = *xi++;
      L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16)));
      L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt));

      filt = *interpol++;
      x0 = *xi++;
      x1 = *xi++;
      L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16)));
      L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt));

      filt = *interpol++;
      x0 = *xi++;
      x1 = *xi++;
      L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16)));
      L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt));

      filt = *interpol++;
      x0 = *xi++;
      x1 = *xi++;
      L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16)));
      L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt));

      filt = *interpol++;
      x0 = *xi++;
      x1 = *xi++;
      L_sumt = fMultAddDiv2(L_sumt, x0, (FIXP_SGL)((SHORT)(filt >> 16)));
      L_sumb = fMultAddDiv2(L_sumb, x1, (FIXP_SGL)((SHORT)filt));
    } while (--i != 0);

    L_sumb <<= 1;
    L_sumb = fAddSaturate(L_sumt << 1, L_sumb);
    *exc++ = L_sumb;
  } while (--j != 0);
  return;
}

void Pred_lt4_postfilter(FIXP_DBL exc[] /* in/out: excitation buffer */
) {
  /*
  exc[i]   = A*exc[i-1] + B*exc[i] + A*exc[i+1]
  exc[i+1] =              A*exc[i] + B*exc[i+1] + A*exc[i+2] ; i = 0:2:62
  */
  int i;
  FIXP_DBL sum0, sum1, a_exc0, a_exc1;
  a_exc0 = fMultDiv2(A2, exc[-1]);
  a_exc1 = fMultDiv2(A2, exc[0]);

  /* ARM926: 22 cycles/iteration */
  for (i = 0; i < L_SUBFR; i += 2) {
    sum0 = a_exc0 + fMult(B, exc[i]);
    sum1 = a_exc1 + fMult(B, exc[i + 1]);
    a_exc0 = fMultDiv2(A2, exc[i + 1]);
    a_exc1 = fMultDiv2(A2, exc[i + 2]);
    exc[i] = sum0 + a_exc0;
    exc[i + 1] = sum1 + a_exc1;
  }
  return;
}