/* -----------------------------------------------------------------------------

Software License for The Fraunhofer FDK AAC Codec Library for Android

© Copyright  1995 - 2019 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):   Manuel Jander

   Description: USAC Linear Prediction Domain coding

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

#include "usacdec_lpd.h"

#include "usacdec_rom.h"

#include "usacdec_fac.h"

#include "usacdec_lpc.h"

#include "FDK_tools_rom.h"

#include "fft.h"

#include "mdct.h"

#include "usacdec_acelp.h"

#include "overlapadd.h"

#include "conceal.h"

#include "block.h"

#define SF_PITCH_TRACK 6

#define SF_GAIN 3

#define MIN_VAL FL2FXCONST_DBL(0.0f)

#define MAX_VAL (FIXP_DBL) MAXVAL_DBL

#include "ac_arith_coder.h"

void filtLP(const FIXP_DBL *syn, PCM_DEC *syn_out, FIXP_DBL *noise,

            const FIXP_SGL *filt, const INT aacOutDataHeadroom, INT stop,

            int len) {

  INT i, j;

  FIXP_DBL tmp;

  FDK_ASSERT((aacOutDataHeadroom - 1) >= -(MDCT_OUTPUT_SCALE));

  for (i = 0; i < stop; i++) {

    tmp = fMultDiv2(noise[i], filt[0]);  // Filt in Q-1.16

    for (j = 1; j <= len; j++) {

      tmp += fMult((noise[i - j] >> 1) + (noise[i + j] >> 1), filt[j]);

    }

    syn_out[i] = (PCM_DEC)(

        IMDCT_SCALE((syn[i] >> 1) - (tmp >> 1), aacOutDataHeadroom - 1));

  }

}

void bass_pf_1sf_delay(

    FIXP_DBL *syn,   /* (i) : 12.8kHz synthesis to postfilter              */

    const INT *T_sf, /* (i) : Pitch period for all subframes (T_sf[16])    */

    FIXP_DBL *pit_gain,

    const int frame_length, /* (i) : frame length (should be 768|1024) */

    const INT l_frame,

    const INT l_next,   /* (i) : look ahead for symmetric filtering           */

    PCM_DEC *synth_out, /* (o) : filtered synthesis (with delay of 1 subfr)   */

    const INT aacOutDataHeadroom, /* (i) : headroom of the output time signal to

                                     prevent clipping */

    FIXP_DBL mem_bpf[]) /* i/o : memory state [L_FILT+L_SUBFR]                */

{

  INT i, sf, i_subfr, T, T2, lg;

  FIXP_DBL tmp, ener, corr, gain;

  FIXP_DBL *noise, *noise_in;

  FIXP_DBL

  noise_buf[L_FILT + (2 * L_SUBFR)];  // L_FILT = 12, L_SUBFR = 64 => 140

  const FIXP_DBL *x, *y;

  {

    noise = noise_buf + L_FILT;  // L_FILT = 12 delay of upsampling filter

    noise_in = noise_buf + L_FILT + L_SUBFR;

    /* Input scaling of the BPF memory */

    scaleValues(mem_bpf, (L_FILT + L_SUBFR), 1);

  }

  int gain_exp = 17;

  sf = 0;

  for (i_subfr = 0; i_subfr < l_frame; i_subfr += L_SUBFR, sf++) {

    T = T_sf[sf];

    gain = pit_gain[sf];

    /* Gain is in Q17.14 */

    /* If gain > 1 set to 1 */

    if (gain > (FIXP_DBL)(1 << 14)) gain = (FIXP_DBL)(1 << 14);

    /* If gain < 0 set to 0 */

    if (gain < (FIXP_DBL)0) gain = (FIXP_DBL)0;

    if (gain > (FIXP_DBL)0) {

      /* pitch tracker: test pitch/2 to avoid continuous pitch doubling */

      /* Note: pitch is limited to PIT_MIN (34 = 376Hz) at the encoder  */

      T2 = T >> 1;

      x = &syn[i_subfr - L_EXTRA];

      y = &syn[i_subfr - T2 - L_EXTRA];

      ener = (FIXP_DBL)0;

      corr = (FIXP_DBL)0;

      tmp = (FIXP_DBL)0;

      int headroom_x = getScalefactor(x, L_SUBFR + L_EXTRA);

      int headroom_y = getScalefactor(y, L_SUBFR + L_EXTRA);

      int width_shift = 7;

      for (i = 0; i < (L_SUBFR + L_EXTRA); i++) {

        ener += fPow2Div2((x[i] << headroom_x)) >> width_shift;

        corr += fMultDiv2((x[i] << headroom_x), (y[i] << headroom_y)) >>

                width_shift;

        tmp += fPow2Div2((y[i] << headroom_y)) >> width_shift;

      }

      int exp_ener = ((17 - headroom_x) << 1) + width_shift + 1;

      int exp_corr = (17 - headroom_x) + (17 - headroom_y) + width_shift + 1;

      int exp_tmp = ((17 - headroom_y) << 1) + width_shift + 1;

      /* Add 0.01 to "ener". Adjust exponents */

      FIXP_DBL point_zero_one = (FIXP_DBL)0x51eb851f; /* In Q-6.37 */

      int diff;

      ener = fAddNorm(ener, exp_ener, point_zero_one, -6, &exp_ener);

      corr = fAddNorm(corr, exp_corr, point_zero_one, -6, &exp_corr);

      tmp = fAddNorm(tmp, exp_tmp, point_zero_one, -6, &exp_tmp);

      /* use T2 if normalized correlation > 0.95 */

      INT s1, s2;

      s1 = CntLeadingZeros(ener) - 1;

      s2 = CntLeadingZeros(tmp) - 1;

      FIXP_DBL ener_by_tmp = fMultDiv2(ener << s1, tmp << s2);

      int ener_by_tmp_exp = (exp_ener - s1) + (exp_tmp - s2) + 1;

      if (ener_by_tmp_exp & 1) {

        ener_by_tmp <<= 1;

        ener_by_tmp_exp -= 1;

      }

      int temp_exp = 0;

      FIXP_DBL temp1 = invSqrtNorm2(ener_by_tmp, &temp_exp);

      int temp1_exp = temp_exp - (ener_by_tmp_exp >> 1);

      FIXP_DBL tmp_result = fMult(corr, temp1);

      int tmp_result_exp = exp_corr + temp1_exp;

      diff = tmp_result_exp - 0;

      FIXP_DBL point95 = FL2FXCONST_DBL(0.95f);

      if (diff >= 0) {

        diff = fMin(diff, 31);

        point95 = FL2FXCONST_DBL(0.95f) >> diff;

      } else {

        diff = fMax(diff, -31);

        tmp_result >>= (-diff);

      }

      if (tmp_result > point95) T = T2;

      /* prevent that noise calculation below reaches into not defined signal

         parts at the end of the synth_buf or in other words restrict the below

         used index (i+i_subfr+T) < l_frame + l_next

      */

      lg = l_frame + l_next - T - i_subfr;

      if (lg > L_SUBFR)

        lg = L_SUBFR;

      else if (lg < 0)

        lg = 0;

      /* limit gain to avoid problem on burst */

      if (lg > 0) {

        FIXP_DBL tmp1;

        /* max(lg) = 64 => scale with 6 bits minus 1 (fPow2Div2) */

        s1 = getScalefactor(&syn[i_subfr], lg);

        s2 = getScalefactor(&syn[i_subfr + T], lg);

        INT s = fixMin(s1, s2);

        tmp = (FIXP_DBL)0;

        ener = (FIXP_DBL)0;

        for (i = 0; i < lg; i++) {

          tmp += fPow2Div2(syn[i + i_subfr] << s1) >> (SF_PITCH_TRACK);

          ener += fPow2Div2(syn[i + i_subfr + T] << s2) >> (SF_PITCH_TRACK);

        }

        tmp = tmp >> fMin(DFRACT_BITS - 1, (2 * (s1 - s)));

        ener = ener >> fMin(DFRACT_BITS - 1, (2 * (s2 - s)));

        /* error robustness: for the specific case syn[...] == -1.0f for all 64

           samples ener or tmp might overflow and become negative. For all sane

           cases we have enough headroom.

        */

        if (ener <= (FIXP_DBL)0) {

          ener = (FIXP_DBL)1;

        }

        if (tmp <= (FIXP_DBL)0) {

          tmp = (FIXP_DBL)1;

        }

        FDK_ASSERT(ener > (FIXP_DBL)0);

        /* tmp = sqrt(tmp/ener) */

        int result_e = 0;

        tmp1 = fDivNorm(tmp, ener, &result_e);

        if (result_e & 1) {

          tmp1 >>= 1;

          result_e += 1;

        }

        tmp = sqrtFixp(tmp1);

        result_e >>= 1;

        gain_exp = 17;

        diff = result_e - gain_exp;

        FIXP_DBL gain1 = gain;

        if (diff >= 0) {

          diff = fMin(diff, 31);

          gain1 >>= diff;

        } else {

          result_e += (-diff);

          diff = fMax(diff, -31);

          tmp >>= (-diff);

        }

        if (tmp < gain1) {

          gain = tmp;

          gain_exp = result_e;

        }

      }

      /* calculate noise based on voiced pitch */

      /* fMultDiv2() replaces weighting of gain with 0.5 */

      diff = gain_exp - 17;

      if (diff >= 0) {

        gain <<= diff;

      } else {

        gain >>= (-diff);

      }

      s1 = CntLeadingZeros(gain) - 1;

      s1 -= 16; /* Leading bits for SGL */

      FIXP_SGL gainSGL = FX_DBL2FX_SGL(gain << 16);

      gainSGL = gainSGL << s1;

      {

        for (i = 0; i < lg; i++) {

          /* scaled with SF_SYNTH + gain_sf + 1; composition of scalefactor 2:

           * one additional shift of syn values + fMult => fMultDiv2 */

          noise_in[i] =

              scaleValue(fMultDiv2(gainSGL, (syn[i + i_subfr] >> 1) -

                                                (syn[i + i_subfr - T] >> 2) -

                                                (syn[i + i_subfr + T] >> 2)),

                         2 - s1);

        }

        for (i = lg; i < L_SUBFR; i++) {

          /* scaled with SF_SYNTH + gain_sf + 1; composition of scalefactor 2:

           * one additional shift of syn values + fMult => fMultDiv2 */

          noise_in[i] =

              scaleValue(fMultDiv2(gainSGL, (syn[i + i_subfr] >> 1) -

                                                (syn[i + i_subfr - T] >> 1)),

                         2 - s1);

        }

      }

    } else {

      FDKmemset(noise_in, (FIXP_DBL)0, L_SUBFR * sizeof(FIXP_DBL));

    }

    {

      FDKmemcpy(noise_buf, mem_bpf, (L_FILT + L_SUBFR) * sizeof(FIXP_DBL));

      FDKmemcpy(mem_bpf, noise_buf + L_SUBFR,

                (L_FILT + L_SUBFR) * sizeof(FIXP_DBL));

    }

    /* substract from voiced speech low-pass filtered noise */

    /* filter coefficients are scaled with factor SF_FILT_LP (1) */

    {

      filtLP(&syn[i_subfr - L_SUBFR], &synth_out[i_subfr], noise,

             fdk_dec_filt_lp, aacOutDataHeadroom, L_SUBFR, L_FILT);

    }

  }

  {

  }

  // To be determined (info from Ben)

  {

    /* Output scaling of the BPF memory */

    scaleValues(mem_bpf, (L_FILT + L_SUBFR), -1);

    /* Copy the rest of the signal (after the fac) */

    scaleValuesSaturate(

        (PCM_DEC *)&synth_out[l_frame], (FIXP_DBL *)&syn[l_frame - L_SUBFR],

        (frame_length - l_frame), MDCT_OUT_HEADROOM - aacOutDataHeadroom);

  }

  return;

}

/*

 * Frequency Domain Noise Shaping

 */

/**

 * \brief Adaptive Low Frequencies Deemphasis of spectral coefficients.

 *

 * Ensure quantization of low frequencies in case where the

 * signal dynamic is higher than the LPC noise shaping.

 * This is the inverse operation of adap_low_freq_emph().

 * Output gain of all blocks.

 *

 * \param x pointer to the spectral coefficients, requires 1 bit headroom.

 * \param lg length of x.

 * \param bUseNewAlfe if set, apply ALFD for fullband lpd.

 * \param gainLpc1 pointer to gain based on old input LPC coefficients.

 * \param gainLpc2 pointer to gain based on new input LPC coefficients.

 * \param alfd_gains pointer to output gains.

 * \param s current scale shift factor of x.

 */

#define ALFDPOW2_SCALE 3

/*static*/

void CLpd_AdaptLowFreqDeemph(FIXP_DBL x[], int lg, FIXP_DBL alfd_gains[],

                             INT s) {

  {

    int i, j, k, i_max;

    FIXP_DBL max, fac;

    /* Note: This stack array saves temporary accumulation results to be used in

     * a second run */

    /*       The size should be limited to (1024/4)/8=32 */

    FIXP_DBL tmp_pow2[32];

    s = s * 2 + ALFDPOW2_SCALE;

    s = fMin(31, s);

    k = 8;

    i_max = lg / 4; /* ALFD range = 1600Hz (lg = 6400Hz) */

    /* find spectral peak */

    max = FL2FX_DBL(0.01f) >> s;

    for (i = 0; i < i_max; i += k) {

      FIXP_DBL tmp;

      tmp = FIXP_DBL(0);

      FIXP_DBL *pX = &x[i];

      j = 8;

      do {

        FIXP_DBL x0 = *pX++;

        FIXP_DBL x1 = *pX++;

        x0 = fPow2Div2(x0);

        x1 = fPow2Div2(x1);

        tmp = tmp + (x0 >> (ALFDPOW2_SCALE - 1));

        tmp = tmp + (x1 >> (ALFDPOW2_SCALE - 1));

      } while ((j = j - 2) != 0);

      tmp = fMax(tmp, (FL2FX_DBL(0.01f) >> s));

      tmp_pow2[i >> 3] = tmp;

      if (tmp > max) {

        max = tmp;

      }

    }

    /* deemphasis of all blocks below the peak */

    fac = FL2FX_DBL(0.1f) >> 1;

    for (i = 0; i < i_max; i += k) {

      FIXP_DBL tmp;

      INT shifti;

      tmp = tmp_pow2[i >> 3];

      /* tmp = (float)sqrt(tmp/max); */

      /* value of tmp is between 8/2*max^2 and max^2 / 2. */

      /* required shift factor of division can grow up to 27

         (grows exponentially for values toward zero)

         thus using normalized division to assure valid result. */

      {

        INT sd;

        if (tmp != (FIXP_DBL)0) {

          tmp = fDivNorm(max, tmp, &sd);

          if (sd & 1) {

            sd++;

            tmp >>= 1;

          }

        } else {

          tmp = (FIXP_DBL)MAXVAL_DBL;

          sd = 0;

        }

        tmp = invSqrtNorm2(tmp, &shifti);

        tmp = scaleValue(tmp, shifti - 1 - (sd / 2));

      }

      if (tmp > fac) {

        fac = tmp;

      }

      FIXP_DBL *pX = &x[i];

      j = 8;

      do {

        FIXP_DBL x0 = pX[0];

        FIXP_DBL x1 = pX[1];

        x0 = fMultDiv2(x0, fac);

        x1 = fMultDiv2(x1, fac);

        x0 = x0 << 2;

        x1 = x1 << 2;

        *pX++ = x0;

        *pX++ = x1;

      } while ((j = j - 2) != 0);

      /* Store gains for FAC */

      *alfd_gains++ = fac;

    }

  }

}

/**

 * \brief Interpolated Noise Shaping for mdct coefficients.

 * This algorithm shapes temporally the spectral noise between

 * the two spectral noise represention (FDNS_NPTS of resolution).

 * The noise is shaped monotonically between the two points

 * using a curved shape to favor the lower gain in mid-frame.

 * ODFT and amplitud calculation are applied to the 2 LPC coefficients first.

 *

 * \param r pointer to spectrum data.

 * \param rms RMS of output spectrum.

 * \param lg length of r.

 * \param A1 pointer to old input LPC coefficients of length M_LP_FILTER_ORDER

 * scaled by SF_A_COEFFS.

 * \param A2 pointer to new input LPC coefficients of length M_LP_FILTER_ORDER

 * scaled by SF_A_COEFFS.

 * \param bLpc2Mdct flags control lpc2mdct conversion and noise shaping.

 * \param gainLpc1 pointer to gain based on old input LPC coefficients.

 * \param gainLpc2 pointer to gain based on new input LPC coefficients.

 * \param gLpc_e pointer to exponent of gainLpc1 and gainLpc2.

 */

/* static */

#define NSHAPE_SCALE (4)

#define LPC2MDCT_CALC (1)

#define LPC2MDCT_GAIN_LOAD (2)

#define LPC2MDCT_GAIN_SAVE (4)

#define LPC2MDCT_APPLY_NSHAPE (8)

void lpc2mdctAndNoiseShaping(FIXP_DBL *r, SHORT *pScale, const INT lg,

                             const INT fdns_npts, const FIXP_LPC *A1,

                             const INT A1_exp, const FIXP_LPC *A2,

                             const INT A2_exp) {

  FIXP_DBL *tmp2 = NULL;

  FIXP_DBL rr_minus_one;

  int i, k, s, step;

  C_AALLOC_SCRATCH_START(tmp1, FIXP_DBL, FDNS_NPTS * 8)

  {

    tmp2 = tmp1 + fdns_npts * 4;

    /* lpc2mdct() */

    /* ODFT. E_LPC_a_weight() for A1 and A2 vectors is included into the loop

     * below. */

    FIXP_DBL f = FL2FXCONST_DBL(0.92f);

    const FIXP_STP *SinTab;

    int k_step;

    /* needed values: sin(phi), cos(phi); phi = i*PI/(2*fdns_npts), i = 0 ...

     * M_LP_FILTER_ORDER */

    switch (fdns_npts) {

      case 64:

        SinTab = SineTable512;

        k_step = (512 / 64);

        FDK_ASSERT(512 >= 64);

        break;

      case 48:

        SinTab = SineTable384;

        k_step = 384 / 48;

        FDK_ASSERT(384 >= 48);

        break;

      default:

        FDK_ASSERT(0);

        return;

    }

    for (i = 0, k = k_step; i < M_LP_FILTER_ORDER; i++, k += k_step) {

      FIXP_STP cs = SinTab[k];

      FIXP_DBL wA1, wA2;

      wA1 = fMult(A1[i], f);

      wA2 = fMult(A2[i], f);

      /* r[i] = A[i]*cos() */

      tmp1[2 + i * 2] = fMult(wA1, cs.v.re);

      tmp2[2 + i * 2] = fMult(wA2, cs.v.re);

      /* i[i] = A[i]*sin() */

      tmp1[3 + i * 2] = -fMult(wA1, cs.v.im);

      tmp2[3 + i * 2] = -fMult(wA2, cs.v.im);

      f = fMult(f, FL2FXCONST_DBL(0.92f));

    }

    /* Guarantee at least 2 bits of headroom for the FFT */

    /* "3" stands for 1.0 with 2 bits of headroom; (A1_exp + 2) guarantess 2

     * bits of headroom if A1_exp > 1 */

    int A1_exp_fix = fMax(3, A1_exp + 2);

    int A2_exp_fix = fMax(3, A2_exp + 2);

    /* Set 1.0 in the proper format */

    tmp1[0] = (FIXP_DBL)(INT)((ULONG)0x80000000 >> A1_exp_fix);

    tmp2[0] = (FIXP_DBL)(INT)((ULONG)0x80000000 >> A2_exp_fix);

    tmp1[1] = tmp2[1] = (FIXP_DBL)0;

    /* Clear the resto of the array */

    FDKmemclear(

        tmp1 + 2 * (M_LP_FILTER_ORDER + 1),

        2 * (fdns_npts * 2 - (M_LP_FILTER_ORDER + 1)) * sizeof(FIXP_DBL));

    FDKmemclear(

        tmp2 + 2 * (M_LP_FILTER_ORDER + 1),

        2 * (fdns_npts * 2 - (M_LP_FILTER_ORDER + 1)) * sizeof(FIXP_DBL));

    /* Guarantee 2 bits of headroom for FFT */

    scaleValues(&tmp1[2], (2 * M_LP_FILTER_ORDER), (A1_exp - A1_exp_fix));

    scaleValues(&tmp2[2], (2 * M_LP_FILTER_ORDER), (A2_exp - A2_exp_fix));

    INT s2;

    s = A1_exp_fix;

    s2 = A2_exp_fix;

    fft(2 * fdns_npts, tmp1, &s);

    fft(2 * fdns_npts, tmp2, &s2);

    /* Adjust the exponents of both fft outputs if necessary*/

    if (s > s2) {

      scaleValues(tmp2, 2 * fdns_npts, s2 - s);

      s2 = s;

    } else if (s < s2) {

      scaleValues(tmp1, 2 * fdns_npts, s - s2);

      s = s2;

    }

    FDK_ASSERT(s == s2);

  }

  /* Get amplitude and apply gains */

  step = lg / fdns_npts;

  rr_minus_one = (FIXP_DBL)0;

  for (k = 0; k < fdns_npts; k++) {

    FIXP_DBL g1, g2, inv_g1_g2, a, b;

    INT inv_g1_g2_e;

    int g_e, shift;

    {

      FIXP_DBL real, imag;

      int si1, si2, sInput;

      real = tmp1[k * 2];

      imag = tmp1[k * 2 + 1];

      sInput = fMax(fMin(fNorm(real), fNorm(imag)) - 1, 0);

      real <<= sInput;

      imag <<= sInput;

      /* g1_e = si1 - 2*s/2 */

      g1 = invSqrtNorm2(fPow2(real) + fPow2(imag), &si1);

      si1 += sInput;

      real = tmp2[k * 2];

      imag = tmp2[k * 2 + 1];

      sInput = fMax(fMin(fNorm(real), fNorm(imag)) - 1, 0);

      real <<= sInput;

      imag <<= sInput;

      /* g2_e = si2 - 2*s/2 */

      g2 = invSqrtNorm2(fPow2(real) + fPow2(imag), &si2);

      si2 += sInput;

      /* Pick a common scale factor for g1 and g2 */

      if (si1 > si2) {

        g2 >>= si1 - si2;

        g_e = si1 - s;

      } else {

        g1 >>= si2 - si1;

        g_e = si2 - s;

      }

    }

    /* end of lpc2mdct() */

    FDK_ASSERT(g1 >= (FIXP_DBL)0);

    FDK_ASSERT(g2 >= (FIXP_DBL)0);

    /* mdct_IntNoiseShaping() */

    {

      /* inv_g1_g2 * 2^inv_g1_g2_e = 1/(g1+g2) */

      inv_g1_g2 = (g1 >> 1) + (g2 >> 1);

      if (inv_g1_g2 != (FIXP_DBL)0) {

        inv_g1_g2 = fDivNorm(FL2FXCONST_DBL(0.5f), inv_g1_g2, &inv_g1_g2_e);

        inv_g1_g2_e = inv_g1_g2_e - g_e;

      } else {

        inv_g1_g2 = (FIXP_DBL)MAXVAL_DBL;

        inv_g1_g2_e = 0;

      }

      if (g_e < 0) {

        /* a_e = g_e + inv_g1_g2_e + 1 */

        a = scaleValue(fMult(fMult(g1, g2), inv_g1_g2), g_e);

        /* b_e = g_e + inv_g1_g2_e */

        b = fMult(g2 - g1, inv_g1_g2);

        shift = g_e + inv_g1_g2_e + 1 - NSHAPE_SCALE;

      } else {

        /* a_e = (g_e+g_e) + inv_g1_g2_e + 1 */

        a = fMult(fMult(g1, g2), inv_g1_g2);

        /* b_e = (g_e+g_e) + inv_g1_g2_e */

        b = scaleValue(fMult(g2 - g1, inv_g1_g2), -g_e);

        shift = (g_e + g_e) + inv_g1_g2_e + 1 - NSHAPE_SCALE;

      }

      for (i = k * step; i < (k + 1) * step; i++) {

        FIXP_DBL tmp;

        /* rr[i] = 2*a*r[i] + b*rr[i-1] */

        tmp = fMult(a, r[i]);

        tmp += scaleValue(fMultDiv2(b, rr_minus_one), NSHAPE_SCALE);

        tmp = scaleValueSaturate(tmp, shift);

        rr_minus_one = tmp;

        r[i] = tmp;

      }

    }

  }

  /* end of mdct_IntNoiseShaping() */

  { *pScale += NSHAPE_SCALE; }

  C_AALLOC_SCRATCH_END(tmp1, FIXP_DBL, FDNS_NPTS * 8)

}

/**

 * \brief Calculates the energy.

 * \param r pointer to spectrum.

 * \param rs scale factor of spectrum r.

 * \param lg frame length in audio samples.

 * \param rms_e pointer to exponent of energy value.

 * \return mantissa of energy value.

 */

static FIXP_DBL calcEnergy(const FIXP_DBL *r, const SHORT rs, const INT lg,

                           INT *rms_e) {

  int headroom = getScalefactor(r, lg);

  FIXP_DBL rms_m = 0;

  /* Calculate number of growth bits due to addition */

  INT shift = (INT)(fNormz((FIXP_DBL)lg));

  shift = 31 - shift;

  /* Generate 1e-2 in Q-6.37 */

  const FIXP_DBL value0_01 = 0x51eb851e;

  const INT value0_01_exp = -6;

  /* Find the exponent of the resulting energy value */

  *rms_e = ((rs - headroom) << 1) + shift + 1;

  INT delta = *rms_e - value0_01_exp;

  if (delta > 0) {

    /* Limit shift_to 31*/

    delta = fMin(31, delta);

    rms_m = value0_01 >> delta;

  } else {

    rms_m = value0_01;

    *rms_e = value0_01_exp;

    shift = shift - delta;

    /* Limit shift_to 31*/

    shift = fMin(31, shift);

  }

  for (int i = 0; i < lg; i++) {

    rms_m += fPow2Div2(r[i] << headroom) >> shift;

  }

  return rms_m;

}

/**

 * \brief TCX gain calculation.

 * \param pAacDecoderChannelInfo channel context data.

 * \param r output spectrum.

 * \param rms_e pointer to mantissa of energy value.

 * \param rms_e pointer to exponent of energy value.

 * \param frame the frame index of the LPD super frame.

 * \param lg the frame length in audio samples.

 * \param gain_m pointer to mantissa of TCX gain.

 * \param gain_e pointer to exponent of TCX gain.

 * \param elFlags element specific parser guidance flags.

 * \param lg_fb the fullband frame length in audio samples.

 * \param IGF_bgn the IGF start index.

 */

static void calcTCXGain(CAacDecoderChannelInfo *pAacDecoderChannelInfo,

                        FIXP_DBL *r, FIXP_DBL rms_m, INT rms_e, const INT frame,

                        const INT lg) {

  if ((rms_m != (FIXP_DBL)0)) {

    FIXP_DBL tcx_gain_m;

    INT tcx_gain_e;

    CLpd_DecodeGain(&tcx_gain_m, &tcx_gain_e,

                    pAacDecoderChannelInfo->pDynData->specificTo.usac

                        .tcx_global_gain[frame]);

    /* rms * 2^rms_e = lg/sqrt(sum(spec^2)) */

    if (rms_e & 1) {

      rms_m >>= 1;

      rms_e++;

    }

    {

      FIXP_DBL fx_lg;

      INT fx_lg_e, s;

      INT inv_e;

      /* lg = fx_lg * 2^fx_lg_e */

      s = fNorm((FIXP_DBL)lg);

      fx_lg = (FIXP_DBL)lg << s;

      fx_lg_e = DFRACT_BITS - 1 - s;

      /* 1/sqrt(rms) */

      rms_m = invSqrtNorm2(rms_m, &inv_e);

      rms_m = fMult(rms_m, fx_lg);

      rms_e = inv_e - (rms_e >> 1) + fx_lg_e;

    }

    {

      int s = fNorm(tcx_gain_m);

      tcx_gain_m = tcx_gain_m << s;

      tcx_gain_e -= s;

    }

    tcx_gain_m = fMultDiv2(tcx_gain_m, rms_m);

    tcx_gain_e = tcx_gain_e + rms_e;

    /* global_gain * 2^(global_gain_e+rms_e) = (10^(global_gain/28)) * rms *

     * 2^rms_e */

    {

      { tcx_gain_e += 1; }

    }

    pAacDecoderChannelInfo->data.usac.tcx_gain[frame] = tcx_gain_m;

    pAacDecoderChannelInfo->data.usac.tcx_gain_e[frame] = tcx_gain_e;

    pAacDecoderChannelInfo->specScale[frame] += tcx_gain_e;

  }

}

/**

 * \brief FDNS decoding.

 * \param pAacDecoderChannelInfo channel context data.

 * \param pAacDecoderStaticChannelInfo channel context static data.

 * \param r output spectrum.

 * \param lg the frame length in audio samples.

 * \param frame the frame index of the LPD super frame.

 * \param pScale pointer to current scale shift factor of r[].

 * \param A1 old input LPC coefficients of length M_LP_FILTER_ORDER.

 * \param A2 new input LPC coefficients of length M_LP_FILTER_ORDER.

 * \param pAlfdGains pointer for ALFD gains output scaled by 1.

 * \param fdns_npts number of lines (FDNS_NPTS).

 * \param inf_mask pointer to noise mask.

 * \param IGF_win_mode IGF window mode (LONG, SHORT, TCX10, TCX20).

 * \param frameType (IGF_FRAME_DIVISION_AAC_OR_TCX_LONG or

 * IGF_FRAME_DIVISION_TCX_SHORT_1).

 * \param elFlags element specific parser guidance flags.

 * \param lg_fb the fullband frame length in audio samples.

 * \param IGF_bgn the IGF start index.

 * \param rms_m mantisse of energy.

 * \param rms_e exponent of energy.

 */

/* static */

void CLpd_FdnsDecode(CAacDecoderChannelInfo *pAacDecoderChannelInfo,

                     CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,

                     FIXP_DBL r[], const INT lg, const INT frame, SHORT *pScale,

                     const FIXP_LPC A1[M_LP_FILTER_ORDER], const INT A1_exp,

                     const FIXP_LPC A2[M_LP_FILTER_ORDER], const INT A2_exp,

                     FIXP_DBL pAlfdGains[LFAC / 4], const INT fdns_npts) {

  /* Weight LPC coefficients using Rm values */

  CLpd_AdaptLowFreqDeemph(r, lg, pAlfdGains, *pScale);

  FIXP_DBL rms_m = (FIXP_DBL)0;

  INT rms_e = 0;

  {

    /* Calculate Energy */

    rms_m = calcEnergy(r, *pScale, lg, &rms_e);

  }

  calcTCXGain(pAacDecoderChannelInfo, r, rms_m, rms_e, frame, lg);

  /* Apply ODFT and Noise Shaping. LP coefficient (A1, A2) weighting is done

   * inside on the fly. */

  lpc2mdctAndNoiseShaping(r, pScale, lg, fdns_npts, A1, A1_exp, A2, A2_exp);

}

/**

 * find pitch for TCX20 (time domain) concealment.

 */

static int find_mpitch(FIXP_DBL xri[], int lg) {

  FIXP_DBL max, pitch;

  INT pitch_e;

  int i, n;

  max = (FIXP_DBL)0;

  n = 2;

  /* find maximum below 400Hz */

  for (i = 2; i < (lg >> 4); i += 2) {

    FIXP_DBL tmp = fPow2Div2(xri[i]) + fPow2Div2(xri[i + 1]);

    if (tmp > max) {

      max = tmp;

      n = i;

    }

  }

  // pitch = ((float)lg<<1)/(float)n;

  pitch = fDivNorm((FIXP_DBL)lg << 1, (FIXP_DBL)n, &pitch_e);

  pitch >>= fixMax(0, DFRACT_BITS - 1 - pitch_e - 16);

  /* find pitch multiple under 20ms */

  if (pitch >= (FIXP_DBL)((256 << 16) - 1)) { /*231.0f*/

    n = 256;

  } else {

    FIXP_DBL mpitch = pitch;

    while (mpitch < (FIXP_DBL)(255 << 16)) {

      mpitch += pitch;

    }

    n = (int)(mpitch - pitch) >> 16;

  }

  return (n);

}

/**

 * number of spectral coefficients / time domain samples using frame mode as

 * index.

 */

static const int lg_table_ccfl[2][4] = {

    {256, 256, 512, 1024}, /* coreCoderFrameLength = 1024 */