libSBRenc/src/resampler.cpp

/* -----------------------------------------------------------------------------
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
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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.

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Redistribution and use in source and binary forms, with or without modification,
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You must retain the complete text of this software license in redistributions of
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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
amm-info@iis.fraunhofer.de
----------------------------------------------------------------------------- */

/**************************** SBR encoder library ******************************

   Author(s):

   Description:

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

/*!
  \file
  \brief  FDK resampler tool box:$Revision: 91655 $
  \author M. Werner
*/

#include "resampler.h"

#include "genericStds.h"

/**************************************************************************/
/*                   BIQUAD Filter Specifications                         */
/**************************************************************************/

#define B1 0
#define B2 1
#define A1 2
#define A2 3

#define BQC(x) FL2FXCONST_SGL(x / 2)

struct FILTER_PARAM {
  const FIXP_SGL *coeffa; /*! SOS matrix One row/section. Scaled using BQC().
                             Order of coefficients: B1,B2,A1,A2. B0=A0=1.0 */
  FIXP_DBL g;             /*! overall gain */
  int Wc;       /*! normalized passband bandwidth at input samplerate * 1000 */
  int noCoeffs; /*! number of filter coeffs */
  int delay;    /*! delay in samples at input samplerate */
};

#define BIQUAD_COEFSTEP 4

/**
 *\brief Low Pass
 Wc = 0,5, order 30, Stop Band -96dB. Wc criteria is "almost 0dB passband", not
 the usual -3db gain point. [b,a]=cheby2(30,96,0.505) [sos,g]=tf2sos(b,a)
 bandwidth 0.48
 */
static const FIXP_SGL sos48[] = {
    BQC(1.98941075681938),      BQC(0.999999996890811),
    BQC(0.863264527201963),     BQC(0.189553799960663),
    BQC(1.90733804822445),      BQC(1.00000001736189),
    BQC(0.836321575841691),     BQC(0.203505809266564),
    BQC(1.75616665495325),      BQC(0.999999946079721),
    BQC(0.784699225121588),     BQC(0.230471265506986),
    BQC(1.55727745512726),      BQC(1.00000011737815),
    BQC(0.712515423588351),     BQC(0.268752723900498),
    BQC(1.33407591943643),      BQC(0.999999795953228),
    BQC(0.625059117330989),     BQC(0.316194685288965),
    BQC(1.10689898412458),      BQC(1.00000035057114),
    BQC(0.52803514366398),      BQC(0.370517843224669),
    BQC(0.89060371078454),      BQC(0.999999343962822),
    BQC(0.426920462165257),     BQC(0.429608200207746),
    BQC(0.694438261209433),     BQC(1.0000008629792),
    BQC(0.326530699561716),     BQC(0.491714450654174),
    BQC(0.523237800935322),     BQC(1.00000101349782),
    BQC(0.230829556274851),     BQC(0.555559034843281),
    BQC(0.378631165929563),     BQC(0.99998986482665),
    BQC(0.142906422036095),     BQC(0.620338874442411),
    BQC(0.260786911308437),     BQC(1.00003261460178),
    BQC(0.0651008576256505),    BQC(0.685759923926262),
    BQC(0.168409429188098),     BQC(0.999933049695828),
    BQC(-0.000790067789975562), BQC(0.751905896602325),
    BQC(0.100724533818628),     BQC(1.00009472669872),
    BQC(-0.0533772830257041),   BQC(0.81930744384525),
    BQC(0.0561434357867363),    BQC(0.999911636304276),
    BQC(-0.0913550299236405),   BQC(0.88883625875915),
    BQC(0.0341680678662057),    BQC(1.00003667508676),
    BQC(-0.113405185536697),    BQC(0.961756638268446)};

static const FIXP_DBL g48 =
    FL2FXCONST_DBL(0.002712866530047) - (FIXP_DBL)0x8000;

static const struct FILTER_PARAM param_set48 = {
    sos48, g48, 480, 15, 4 /* LF 2 */
};

/**
 *\brief Low Pass
 Wc = 0,5, order 24, Stop Band -96dB. Wc criteria is "almost 0dB passband", not
 the usual -3db gain point. [b,a]=cheby2(24,96,0.5) [sos,g]=tf2sos(b,a)
 bandwidth 0.45
 */
static const FIXP_SGL sos45[] = {
    BQC(1.982962601444),     BQC(1.00000000007504),    BQC(0.646113303737836),
    BQC(0.10851149979981),   BQC(1.85334094281111),    BQC(0.999999999677192),
    BQC(0.612073220102006),  BQC(0.130022141698044),   BQC(1.62541051415425),
    BQC(1.00000000080398),   BQC(0.547879702855959),   BQC(0.171165825133192),
    BQC(1.34554656923247),   BQC(0.9999999980169),     BQC(0.460373914508491),
    BQC(0.228677463376354),  BQC(1.05656568503116),    BQC(1.00000000569363),
    BQC(0.357891894038287),  BQC(0.298676843912185),   BQC(0.787967587877312),
    BQC(0.999999984415017),  BQC(0.248826893211877),   BQC(0.377441803512978),
    BQC(0.555480971120497),  BQC(1.00000003583307),    BQC(0.140614263345315),
    BQC(0.461979302213679),  BQC(0.364986207070964),   BQC(0.999999932084303),
    BQC(0.0392669446074516), BQC(0.55033451180825),    BQC(0.216827267631558),
    BQC(1.00000010534682),   BQC(-0.0506232228865103), BQC(0.641691581560946),
    BQC(0.108951672277119),  BQC(0.999999871167516),   BQC(-0.125584840183225),
    BQC(0.736367748771803),  BQC(0.0387988607229035),  BQC(1.00000011205574),
    BQC(-0.182814849097974), BQC(0.835802108714964),   BQC(0.0042866175809225),
    BQC(0.999999954830813),  BQC(-0.21965740617151),   BQC(0.942623047782363)};

static const FIXP_DBL g45 =
    FL2FXCONST_DBL(0.00242743980909524) - (FIXP_DBL)0x8000;

static const struct FILTER_PARAM param_set45 = {
    sos45, g45, 450, 12, 4 /* LF 2 */
};

/*
 Created by Octave 2.1.73, Mon Oct 13 17:31:32 2008 CEST
 Wc = 0,5, order 16, Stop Band -96dB damping.
 [b,a]=cheby2(16,96,0.5)
 [sos,g]=tf2sos(b,a)
 bandwidth = 0.41
 */

static const FIXP_SGL sos41[] = {
    BQC(1.96193625292),      BQC(0.999999999999964),   BQC(0.169266178786789),
    BQC(0.0128823300475907), BQC(1.68913437662092),    BQC(1.00000000000053),
    BQC(0.124751503206552),  BQC(0.0537472273950989),  BQC(1.27274692366017),
    BQC(0.999999999995674),  BQC(0.0433108625178357),  BQC(0.131015753236317),
    BQC(0.85214175088395),   BQC(1.00000000001813),    BQC(-0.0625658152550408),
    BQC(0.237763778993806),  BQC(0.503841579939009),   BQC(0.999999999953223),
    BQC(-0.179176128722865), BQC(0.367475236424474),   BQC(0.249990711986162),
    BQC(1.00000000007952),   BQC(-0.294425165824676),  BQC(0.516594857170212),
    BQC(0.087971668680286),  BQC(0.999999999915528),   BQC(-0.398956566777928),
    BQC(0.686417767801123),  BQC(0.00965373325350294), BQC(1.00000000003744),
    BQC(-0.48579173764817),  BQC(0.884931534239068)};

static const FIXP_DBL g41 = FL2FXCONST_DBL(0.00155956951169248);

static const struct FILTER_PARAM param_set41 = {
    sos41, g41, 410, 8, 5 /* LF 3 */
};

/*
 # Created by Octave 2.1.73, Mon Oct 13 17:55:33 2008 CEST
 Wc = 0,5, order 12, Stop Band -96dB damping.
 [b,a]=cheby2(12,96,0.5);
 [sos,g]=tf2sos(b,a)
*/
static const FIXP_SGL sos35[] = {
    BQC(1.93299325235762),   BQC(0.999999999999985),  BQC(-0.140733187246596),
    BQC(0.0124139497836062), BQC(1.4890416764109),    BQC(1.00000000000011),
    BQC(-0.198215402588504), BQC(0.0746730616584138), BQC(0.918450161309795),
    BQC(0.999999999999619),  BQC(-0.30133912791941),  BQC(0.192276468839529),
    BQC(0.454877024246818),  BQC(1.00000000000086),   BQC(-0.432337328809815),
    BQC(0.356852933642815),  BQC(0.158017147118507),  BQC(0.999999999998876),
    BQC(-0.574817494249777), BQC(0.566380436970833),  BQC(0.0171834649478749),
    BQC(1.00000000000055),   BQC(-0.718581178041165), BQC(0.83367484487889)};

static const FIXP_DBL g35 = FL2FXCONST_DBL(0.00162580994125131);

static const struct FILTER_PARAM param_set35 = {sos35, g35, 350, 6, 4};

/*
 # Created by Octave 2.1.73, Mon Oct 13 18:15:38 2008 CEST
 Wc = 0,5, order 8, Stop Band -96dB damping.
 [b,a]=cheby2(8,96,0.5);
 [sos,g]=tf2sos(b,a)
*/
static const FIXP_SGL sos25[] = {
    BQC(1.85334094301225),   BQC(1.0),
    BQC(-0.702127214212663), BQC(0.132452403998767),
    BQC(1.056565682167),     BQC(0.999999999999997),
    BQC(-0.789503667880785), BQC(0.236328693569128),
    BQC(0.364986307455489),  BQC(0.999999999999996),
    BQC(-0.955191189843375), BQC(0.442966457936379),
    BQC(0.0387985751642125), BQC(1.0),
    BQC(-1.19817786088084),  BQC(0.770493895456328)};

static const FIXP_DBL g25 = FL2FXCONST_DBL(0.000945182835294559);

static const struct FILTER_PARAM param_set25 = {sos25, g25, 250, 4, 5};

/* Must be sorted in descending order */
static const struct FILTER_PARAM *const filter_paramSet[] = {
    &param_set48, &param_set45, &param_set41, &param_set35, &param_set25};

/**************************************************************************/
/*                         Resampler Functions                            */
/**************************************************************************/

/*!
  \brief   Reset downsampler instance and clear delay lines

  \return  success of operation
*/

INT FDKaacEnc_InitDownsampler(
    DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */
    int Wc,                   /*!< normalized cutoff freq * 1000*  */
    int ratio)                /*!< downsampler ratio */

{
  UINT i;
  const struct FILTER_PARAM *currentSet = NULL;

  FDKmemclear(DownSampler->downFilter.states,
              sizeof(DownSampler->downFilter.states));
  DownSampler->downFilter.ptr = 0;

  /*
    find applicable parameter set
  */
  currentSet = filter_paramSet[0];
  for (i = 1; i < sizeof(filter_paramSet) / sizeof(struct FILTER_PARAM *);
       i++) {
    if (filter_paramSet[i]->Wc <= Wc) {
      break;
    }
    currentSet = filter_paramSet[i];
  }

  DownSampler->downFilter.coeffa = currentSet->coeffa;

  DownSampler->downFilter.gain = currentSet->g;
  FDK_ASSERT(currentSet->noCoeffs <= MAXNR_SECTIONS * 2);

  DownSampler->downFilter.noCoeffs = currentSet->noCoeffs;
  DownSampler->delay = currentSet->delay;
  DownSampler->downFilter.Wc = currentSet->Wc;

  DownSampler->ratio = ratio;
  DownSampler->pending = ratio - 1;
  return (1);
}

/*!
  \brief   faster simple folding operation
           Filter:
           H(z) = A(z)/B(z)
           with
           A(z) = a[0]*z^0 + a[1]*z^1 + a[2]*z^2 ... a[n]*z^n

  \return  filtered value
*/

static inline INT_PCM AdvanceFilter(
    LP_FILTER *downFilter, /*!< pointer to iir filter instance */
    INT_PCM *pInput,       /*!< input of filter                */
    int downRatio) {
  INT_PCM output;
  int i, n;

#define BIQUAD_SCALE 12

  FIXP_DBL y = FL2FXCONST_DBL(0.0f);
  FIXP_DBL input;

  for (n = 0; n < downRatio; n++) {
    FIXP_BQS(*states)[2] = downFilter->states;
    const FIXP_SGL *coeff = downFilter->coeffa;
    int s1, s2;

    s1 = downFilter->ptr;
    s2 = s1 ^ 1;

#if (SAMPLE_BITS == 16)
    input = ((FIXP_DBL)pInput[n]) << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE);
#elif (SAMPLE_BITS == 32)
    input = pInput[n] >> BIQUAD_SCALE;
#else
#error NOT IMPLEMENTED
#endif

    FIXP_BQS state1, state2, state1b, state2b;

    state1 = states[0][s1];
    state2 = states[0][s2];

    /* Loop over sections */
    for (i = 0; i < downFilter->noCoeffs; i++) {
      FIXP_DBL state0;

      /* Load merged states (from next section) */
      state1b = states[i + 1][s1];
      state2b = states[i + 1][s2];

      state0 = input + fMult(state1, coeff[B1]) + fMult(state2, coeff[B2]);
      y = state0 - fMult(state1b, coeff[A1]) - fMult(state2b, coeff[A2]);

      /* Store new feed forward merge state */
      states[i + 1][s2] = y << 1;
      /* Store new feed backward state */
      states[i][s2] = input << 1;

      /* Feedback output to next section. */
      input = y;

      /* Transfer merged states */
      state1 = state1b;
      state2 = state2b;

      /* Step to next coef set */
      coeff += BIQUAD_COEFSTEP;
    }
    downFilter->ptr ^= 1;
  }
  /* Apply global gain */
  y = fMult(y, downFilter->gain);

  /* Apply final gain/scaling to output */
#if (SAMPLE_BITS == 16)
  output = (INT_PCM)SATURATE_RIGHT_SHIFT(
      y + (FIXP_DBL)(1 << (DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE - 1)),
      DFRACT_BITS - SAMPLE_BITS - BIQUAD_SCALE, SAMPLE_BITS);
  // output = (INT_PCM) SATURATE_RIGHT_SHIFT(y,
  // DFRACT_BITS-SAMPLE_BITS-BIQUAD_SCALE, SAMPLE_BITS);
#else
  output = SATURATE_LEFT_SHIFT(y, BIQUAD_SCALE, SAMPLE_BITS);
#endif

  return output;
}

/*!
  \brief   FDKaacEnc_Downsample numInSamples of type INT_PCM
           Returns number of output samples in numOutSamples

  \return  success of operation
*/

INT FDKaacEnc_Downsample(
    DOWNSAMPLER *DownSampler, /*!< pointer to downsampler instance */
    INT_PCM *inSamples,       /*!< pointer to input samples */
    INT numInSamples,         /*!< number  of input samples  */
    INT_PCM *outSamples,      /*!< pointer to output samples */
    INT *numOutSamples        /*!< pointer tp number of output samples */
) {
  INT i;
  *numOutSamples = 0;

  for (i = 0; i < numInSamples; i += DownSampler->ratio) {
    *outSamples = AdvanceFilter(&(DownSampler->downFilter), &inSamples[i],
                                DownSampler->ratio);
    outSamples++;
  }
  *numOutSamples = numInSamples / DownSampler->ratio;

  return 0;
}