fdk-aac/libFDK/include/qmf_pcm.h

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

/******************* Library for basic calculation routines ********************

   Author(s):   Markus Lohwasser, Josef Hoepfl, Manuel Jander

   Description: QMF filterbank

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

#ifndef QMF_PCM_H
#define QMF_PCM_H

/*
   All Synthesis functions dependent on datatype INT_PCM_QMFOUT
   Should only be included by qmf.cpp, but not compiled separately, please
   exclude compilation from project, if done otherwise. Is optional included
   twice to duplicate all functions with two different pre-definitions, as:
        #define INT_PCM_QMFOUT LONG
    and ...
        #define INT_PCM_QMFOUT SHORT
    needed to run QMF synthesis in both 16bit and 32bit sample output format.
*/

#define QSSCALE (0)
#define FX_DBL2FX_QSS(x) (x)
#define FX_QSS2FX_DBL(x) (x)

/*!
  \brief Perform Synthesis Prototype Filtering on a single slot of input data.

  The filter takes 2 * qmf->no_channels of input data and
  generates qmf->no_channels time domain output samples.
*/
/* static */
#ifndef FUNCTION_qmfSynPrototypeFirSlot
void qmfSynPrototypeFirSlot(
#else
void qmfSynPrototypeFirSlot_fallback(
#endif
    HANDLE_QMF_FILTER_BANK qmf,
    FIXP_DBL *RESTRICT realSlot,      /*!< Input: Pointer to real Slot */
    FIXP_DBL *RESTRICT imagSlot,      /*!< Input: Pointer to imag Slot */
    INT_PCM_QMFOUT *RESTRICT timeOut, /*!< Time domain data */
    int stride) {
  FIXP_QSS *FilterStates = (FIXP_QSS *)qmf->FilterStates;
  int no_channels = qmf->no_channels;
  const FIXP_PFT *p_Filter = qmf->p_filter;
  int p_stride = qmf->p_stride;
  int j;
  FIXP_QSS *RESTRICT sta = FilterStates;
  const FIXP_PFT *RESTRICT p_flt, *RESTRICT p_fltm;
  int scale = (DFRACT_BITS - SAMPLE_BITS_QMFOUT) - 1 - qmf->outScalefactor -
              qmf->outGain_e;

  p_flt =
      p_Filter + p_stride * QMF_NO_POLY; /*                     5th of 330 */
  p_fltm = p_Filter + (qmf->FilterSize / 2) -
           p_stride * QMF_NO_POLY; /* 5 + (320 - 2*5) = 315th of 330 */

  FIXP_SGL gain = FX_DBL2FX_SGL(qmf->outGain_m);

  FIXP_DBL rnd_val = 0;

  if (scale > 0) {
    if (scale < (DFRACT_BITS - 1))
      rnd_val = FIXP_DBL(1 << (scale - 1));
    else
      scale = (DFRACT_BITS - 1);
  } else {
    scale = fMax(scale, -(DFRACT_BITS - 1));
  }

  for (j = no_channels - 1; j >= 0; j--) {
    FIXP_DBL imag = imagSlot[j]; /* no_channels-1 .. 0 */
    FIXP_DBL real = realSlot[j]; /* no_channels-1 .. 0 */
    {
      INT_PCM_QMFOUT tmp;
      FIXP_DBL Are = fMultAddDiv2(FX_QSS2FX_DBL(sta[0]), p_fltm[0], real);

      /* This PCM formatting performs:
         - multiplication with 16-bit gain, if not -1.0f
         - rounding, if shift right is applied
         - apply shift left (or right) with saturation to 32 (or 16) bits
         - store output with --stride in 32 (or 16) bit format
      */
      if (gain != (FIXP_SGL)(-32768)) /* -1.0f */
      {
        Are = fMult(Are, gain);
      }
      if (scale >= 0) {
        FDK_ASSERT(
            Are <=
            (Are + rnd_val)); /* Round-addition must not overflow, might be
                                 equal for rnd_val=0 */
        tmp = (INT_PCM_QMFOUT)(
            SATURATE_RIGHT_SHIFT(Are + rnd_val, scale, SAMPLE_BITS_QMFOUT));
      } else {
        tmp = (INT_PCM_QMFOUT)(
            SATURATE_LEFT_SHIFT(Are, -scale, SAMPLE_BITS_QMFOUT));
      }

      { timeOut[(j)*stride] = tmp; }
    }

    sta[0] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[1]), p_flt[4], imag));
    sta[1] =
        FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[2]), p_fltm[1], real));
    sta[2] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[3]), p_flt[3], imag));
    sta[3] =
        FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[4]), p_fltm[2], real));
    sta[4] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[5]), p_flt[2], imag));
    sta[5] =
        FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[6]), p_fltm[3], real));
    sta[6] = FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[7]), p_flt[1], imag));
    sta[7] =
        FX_DBL2FX_QSS(fMultAddDiv2(FX_QSS2FX_DBL(sta[8]), p_fltm[4], real));
    sta[8] = FX_DBL2FX_QSS(fMultDiv2(p_flt[0], imag));
    p_flt += (p_stride * QMF_NO_POLY);
    p_fltm -= (p_stride * QMF_NO_POLY);
    sta += 9;  // = (2*QMF_NO_POLY-1);
  }
}

#ifndef FUNCTION_qmfSynPrototypeFirSlot_NonSymmetric
/*!
  \brief Perform Synthesis Prototype Filtering on a single slot of input data.

  The filter takes 2 * qmf->no_channels of input data and
  generates qmf->no_channels time domain output samples.
*/
static void qmfSynPrototypeFirSlot_NonSymmetric(
    HANDLE_QMF_FILTER_BANK qmf,
    FIXP_DBL *RESTRICT realSlot,      /*!< Input: Pointer to real Slot */
    FIXP_DBL *RESTRICT imagSlot,      /*!< Input: Pointer to imag Slot */
    INT_PCM_QMFOUT *RESTRICT timeOut, /*!< Time domain data */
    int stride) {
  FIXP_QSS *FilterStates = (FIXP_QSS *)qmf->FilterStates;
  int no_channels = qmf->no_channels;
  const FIXP_PFT *p_Filter = qmf->p_filter;
  int p_stride = qmf->p_stride;
  int j;
  FIXP_QSS *RESTRICT sta = FilterStates;
  const FIXP_PFT *RESTRICT p_flt, *RESTRICT p_fltm;
  int scale = (DFRACT_BITS - SAMPLE_BITS_QMFOUT) - 1 - qmf->outScalefactor -
              qmf->outGain_e;

  p_flt = p_Filter; /*!< Pointer to first half of filter coefficients */
  p_fltm =
      &p_flt[qmf->FilterSize / 2]; /* at index 320, overall 640 coefficients */

  FIXP_SGL gain = FX_DBL2FX_SGL(qmf->outGain_m);

  FIXP_DBL rnd_val = (FIXP_DBL)0;

  if (scale > 0) {
    if (scale < (DFRACT_BITS - 1))
      rnd_val = FIXP_DBL(1 << (scale - 1));
    else
      scale = (DFRACT_BITS - 1);
  } else {
    scale = fMax(scale, -(DFRACT_BITS - 1));
  }

  for (j = no_channels - 1; j >= 0; j--) {
    FIXP_DBL imag = imagSlot[j]; /* no_channels-1 .. 0 */
    FIXP_DBL real = realSlot[j]; /* no_channels-1 .. 0 */
    {
      INT_PCM_QMFOUT tmp;
      FIXP_DBL Are = sta[0] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[4], real));

      /* This PCM formatting performs:
         - multiplication with 16-bit gain, if not -1.0f
         - rounding, if shift right is applied
         - apply shift left (or right) with saturation to 32 (or 16) bits
         - store output with --stride in 32 (or 16) bit format
      */
      if (gain != (FIXP_SGL)(-32768)) /* -1.0f */
      {
        Are = fMult(Are, gain);
      }
      if (scale > 0) {
        FDK_ASSERT(Are <
                   (Are + rnd_val)); /* Round-addition must not overflow */
        tmp = (INT_PCM_QMFOUT)(
            SATURATE_RIGHT_SHIFT(Are + rnd_val, scale, SAMPLE_BITS_QMFOUT));
      } else {
        tmp = (INT_PCM_QMFOUT)(
            SATURATE_LEFT_SHIFT(Are, -scale, SAMPLE_BITS_QMFOUT));
      }
      timeOut[j * stride] = tmp;
    }

    sta[0] = sta[1] + FX_DBL2FX_QSS(fMultDiv2(p_flt[4], imag));
    sta[1] = sta[2] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[3], real));
    sta[2] = sta[3] + FX_DBL2FX_QSS(fMultDiv2(p_flt[3], imag));

    sta[3] = sta[4] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[2], real));
    sta[4] = sta[5] + FX_DBL2FX_QSS(fMultDiv2(p_flt[2], imag));
    sta[5] = sta[6] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[1], real));
    sta[6] = sta[7] + FX_DBL2FX_QSS(fMultDiv2(p_flt[1], imag));

    sta[7] = sta[8] + FX_DBL2FX_QSS(fMultDiv2(p_fltm[0], real));
    sta[8] = FX_DBL2FX_QSS(fMultDiv2(p_flt[0], imag));

    p_flt += (p_stride * QMF_NO_POLY);
    p_fltm += (p_stride * QMF_NO_POLY);
    sta += 9;  // = (2*QMF_NO_POLY-1);
  }
}
#endif /* FUNCTION_qmfSynPrototypeFirSlot_NonSymmetric */

void qmfSynthesisFilteringSlot(HANDLE_QMF_FILTER_BANK synQmf,
                               const FIXP_DBL *realSlot,
                               const FIXP_DBL *imagSlot,
                               const int scaleFactorLowBand,
                               const int scaleFactorHighBand,
                               INT_PCM_QMFOUT *timeOut, const int stride,
                               FIXP_DBL *pWorkBuffer) {
  if (!(synQmf->flags & QMF_FLAG_LP))
    qmfInverseModulationHQ(synQmf, realSlot, imagSlot, scaleFactorLowBand,
                           scaleFactorHighBand, pWorkBuffer);
  else {
    if (synQmf->flags & QMF_FLAG_CLDFB) {
      qmfInverseModulationLP_odd(synQmf, realSlot, scaleFactorLowBand,
                                 scaleFactorHighBand, pWorkBuffer);
    } else {
      qmfInverseModulationLP_even(synQmf, realSlot, scaleFactorLowBand,
                                  scaleFactorHighBand, pWorkBuffer);
    }
  }

  if (synQmf->flags & QMF_FLAG_NONSYMMETRIC) {
    qmfSynPrototypeFirSlot_NonSymmetric(synQmf, pWorkBuffer,
                                        pWorkBuffer + synQmf->no_channels,
                                        timeOut, stride);
  } else {
    qmfSynPrototypeFirSlot(synQmf, pWorkBuffer,
                           pWorkBuffer + synQmf->no_channels, timeOut, stride);
  }
}

/*!
 *
 * \brief Perform complex-valued subband synthesis of the
 *        low band and the high band and store the
 *        time domain data in timeOut
 *
 * First step: Calculate the proper scaling factor of current
 * spectral data in qmfReal/qmfImag, old spectral data in the overlap
 * range and filter states.
 *
 * Second step: Perform Frequency-to-Time mapping with inverse
 * Modulation slot-wise.
 *
 * Third step: Perform FIR-filter slot-wise. To save space for filter
 * states, the MAC operations are executed directly on the filter states
 * instead of accumulating several products in the accumulator. The
 * buffer shift at the end of the function should be replaced by a
 * modulo operation, which is available on some DSPs.
 *
 * Last step: Copy the upper part of the spectral data to the overlap buffer.
 *
 * The qmf coefficient table is symmetric. The symmetry is exploited by
 * shrinking the coefficient table to half the size. The addressing mode
 * takes care of the symmetries.  If the #define #QMFTABLE_FULL is set,
 * coefficient addressing works on the full table size. The code will be
 * slightly faster and slightly more compact.
 *
 * Workbuffer requirement: 2 x sizeof(**QmfBufferReal) * synQmf->no_channels
 * The workbuffer must be aligned
 */
void qmfSynthesisFiltering(
    HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank  */
    FIXP_DBL **QmfBufferReal,      /*!< Low and High band, real */
    FIXP_DBL **QmfBufferImag,      /*!< Low and High band, imag */
    const QMF_SCALE_FACTOR *scaleFactor,
    const INT ov_len,        /*!< split Slot of overlap and actual slots */
    INT_PCM_QMFOUT *timeOut, /*!< Pointer to output */
    const INT stride,        /*!< stride factor of output */
    FIXP_DBL *pWorkBuffer    /*!< pointer to temporal working buffer */
) {
  int i;
  int L = synQmf->no_channels;
  int scaleFactorHighBand;
  int scaleFactorLowBand_ov, scaleFactorLowBand_no_ov;

  FDK_ASSERT(synQmf->no_channels >= synQmf->lsb);
  FDK_ASSERT(synQmf->no_channels >= synQmf->usb);

  /* adapt scaling */
  scaleFactorHighBand = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK -
                        scaleFactor->hb_scale - synQmf->filterScale;
  scaleFactorLowBand_ov = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK -
                          scaleFactor->ov_lb_scale - synQmf->filterScale;
  scaleFactorLowBand_no_ov = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK -
                             scaleFactor->lb_scale - synQmf->filterScale;

  for (i = 0; i < synQmf->no_col; i++) /* ----- no_col loop ----- */
  {
    const FIXP_DBL *QmfBufferImagSlot = NULL;

    int scaleFactorLowBand =
        (i < ov_len) ? scaleFactorLowBand_ov : scaleFactorLowBand_no_ov;

    if (!(synQmf->flags & QMF_FLAG_LP)) QmfBufferImagSlot = QmfBufferImag[i];

    qmfSynthesisFilteringSlot(synQmf, QmfBufferReal[i], QmfBufferImagSlot,
                              scaleFactorLowBand, scaleFactorHighBand,
                              timeOut + (i * L * stride), stride, pWorkBuffer);
  } /* no_col loop  i  */
}
#endif /* QMF_PCM_H */