Imported Debian version 0.1.3.1

[deb_fdk-aac.git] / libSBRenc / src / nf_est.cpp

/* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android

© Copyright  1995 - 2013 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
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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
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"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."

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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
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or business interruption, however caused and on any theory of liability, whether in contract, strict
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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
----------------------------------------------------------------------------------------------------------- */

#include "nf_est.h"

#include "sbr_misc.h"

#include "genericStds.h"

/* smoothFilter[4]  = {0.05857864376269f, 0.2f, 0.34142135623731f, 0.4f}; */
static const FIXP_DBL smoothFilter[4]  = { 0x077f813d, 0x19999995, 0x2bb3b1f5, 0x33333335 };

/* static const INT smoothFilterLength = 4; */

static const FIXP_DBL QuantOffset = (INT)0xfc000000;  /* ld64(0.25) */

#ifndef min
#define min(a,b) ( a < b ? a:b)
#endif

#ifndef max
#define max(a,b) ( a > b ? a:b)
#endif

#define NOISE_FLOOR_OFFSET_SCALING  (4)



/**************************************************************************/
/*!
  \brief     The function applies smoothing to the noise levels.



  \return    none

*/
/**************************************************************************/
static void
smoothingOfNoiseLevels(FIXP_DBL *NoiseLevels,        /*!< pointer to noise-floor levels.*/
                       INT nEnvelopes,               /*!< Number of noise floor envelopes.*/
                       INT noNoiseBands,             /*!< Number of noise bands for every noise floor envelope. */
                       FIXP_DBL prevNoiseLevels[NF_SMOOTHING_LENGTH][MAX_NUM_NOISE_VALUES],/*!< Previous noise floor envelopes. */
                       const FIXP_DBL *smoothFilter, /*!< filter used for smoothing the noise floor levels. */
                       INT transientFlag)            /*!< flag indicating if a transient is present*/

{
  INT i,band,env;
  FIXP_DBL accu;

  for(env = 0; env < nEnvelopes; env++){
    if(transientFlag){
      for (i = 0; i < NF_SMOOTHING_LENGTH; i++){
        FDKmemcpy(prevNoiseLevels[i],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL));
      }
    }
    else {
      for (i = 1; i < NF_SMOOTHING_LENGTH; i++){
        FDKmemcpy(prevNoiseLevels[i - 1],prevNoiseLevels[i],noNoiseBands*sizeof(FIXP_DBL));
      }
      FDKmemcpy(prevNoiseLevels[NF_SMOOTHING_LENGTH - 1],NoiseLevels+env*noNoiseBands,noNoiseBands*sizeof(FIXP_DBL));
    }

    for (band = 0; band < noNoiseBands; band++){
      accu = FL2FXCONST_DBL(0.0f);
      for (i = 0; i < NF_SMOOTHING_LENGTH; i++){
        accu += fMultDiv2(smoothFilter[i], prevNoiseLevels[i][band]);
      }
      FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES);
      NoiseLevels[band+ env*noNoiseBands] = accu<<1;
    }
  }
}

/**************************************************************************/
/*!
  \brief     Does the noise floor level estiamtion.

  The noiseLevel samples are scaled by the factor 0.25

  \return    none

*/
/**************************************************************************/
static void
qmfBasedNoiseFloorDetection(FIXP_DBL *noiseLevel,              /*!< Pointer to vector to store the noise levels in.*/
                            FIXP_DBL ** quotaMatrixOrig,       /*!< Matrix holding the quota values of the original. */
                            SCHAR *indexVector,                /*!< Index vector to obtain the patched data. */
                            INT startIndex,                    /*!< Start index. */
                            INT stopIndex,                     /*!< Stop index. */
                            INT startChannel,                  /*!< Start channel of the current noise floor band.*/
                            INT stopChannel,                   /*!< Stop channel of the current noise floor band. */
                            FIXP_DBL ana_max_level,            /*!< Maximum level of the adaptive noise.*/
                            FIXP_DBL noiseFloorOffset,         /*!< Noise floor offset. */
                            INT missingHarmonicFlag,           /*!< Flag indicating if a strong tonal component is missing.*/
                            FIXP_DBL weightFac,                /*!< Weightening factor for the difference between orig and sbr. */
                            INVF_MODE diffThres,               /*!< Threshold value to control the inverse filtering decision.*/
                            INVF_MODE inverseFilteringLevel)   /*!< Inverse filtering level of the current band.*/
{
  INT scale, l, k;
  FIXP_DBL meanOrig=FL2FXCONST_DBL(0.0f), meanSbr=FL2FXCONST_DBL(0.0f), diff;
  FIXP_DBL invIndex = GetInvInt(stopIndex-startIndex);
  FIXP_DBL invChannel = GetInvInt(stopChannel-startChannel);
  FIXP_DBL accu;

   /*
   Calculate the mean value, over the current time segment, for the original, the HFR
   and the difference, over all channels in the current frequency range.
   */

  if(missingHarmonicFlag == 1){
    for(l = startChannel; l < stopChannel;l++){
      /* tonalityOrig */
      accu = FL2FXCONST_DBL(0.0f);
      for(k = startIndex ; k < stopIndex; k++){
        accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex);
      }
      meanOrig = fixMax(meanOrig,(accu<<1));

      /* tonalitySbr */
      accu = FL2FXCONST_DBL(0.0f);
      for(k = startIndex ; k < stopIndex; k++){
        accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex);
      }
      meanSbr  = fixMax(meanSbr,(accu<<1));

    }
  }
  else{
    for(l = startChannel; l < stopChannel;l++){
      /* tonalityOrig */
      accu = FL2FXCONST_DBL(0.0f);
      for(k = startIndex ; k < stopIndex; k++){
        accu += fMultDiv2(quotaMatrixOrig[k][l], invIndex);
      }
      meanOrig += fMult((accu<<1), invChannel);

      /* tonalitySbr */
      accu = FL2FXCONST_DBL(0.0f);
      for(k = startIndex ; k < stopIndex; k++){
        accu += fMultDiv2(quotaMatrixOrig[k][indexVector[l]], invIndex);
      }
      meanSbr  += fMult((accu<<1), invChannel);
    }
  }

  /* Small fix to avoid noise during silent passages.*/
  if( meanOrig <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) &&
      meanSbr <= FL2FXCONST_DBL(0.000976562f*RELAXATION_FLOAT) )
  {
    meanOrig = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT);
    meanSbr  = FL2FXCONST_DBL(101.5936673f*RELAXATION_FLOAT);
  }

  meanOrig = fixMax(meanOrig,RELAXATION);
  meanSbr  = fixMax(meanSbr,RELAXATION);

  if (missingHarmonicFlag == 1 ||
      inverseFilteringLevel == INVF_MID_LEVEL ||
      inverseFilteringLevel == INVF_LOW_LEVEL ||
      inverseFilteringLevel == INVF_OFF ||
      inverseFilteringLevel <= diffThres)
  {
    diff = RELAXATION;
  }
  else {
    accu = fDivNorm(meanSbr, meanOrig, &scale);

    diff = fixMax( RELAXATION,
                   fMult(RELAXATION_FRACT,fMult(weightFac,accu)) >>( RELAXATION_SHIFT-scale ) ) ;
  }

  /*
   * noise Level is now a positive value, i.e.
   * the more harmonic the signal is the higher noise level,
   * this makes no sense so we change the sign.
   *********************************************************/
  accu = fDivNorm(diff, meanOrig, &scale);
  scale -= 2;

  if ( (scale>0) && (accu > ((FIXP_DBL)MAXVAL_DBL)>>scale) ) {
    *noiseLevel = (FIXP_DBL)MAXVAL_DBL;
  }
  else {
    *noiseLevel = scaleValue(accu, scale);
  }

  /*
   * Add a noise floor offset to compensate for bias in the detector
   *****************************************************************/
  if(!missingHarmonicFlag)
    *noiseLevel = fMult(*noiseLevel, noiseFloorOffset)<<(NOISE_FLOOR_OFFSET_SCALING);

  /*
   * check to see that we don't exceed the maximum allowed level
   **************************************************************/
  *noiseLevel = fixMin(*noiseLevel, ana_max_level);     /* ana_max_level is scaled with factor 0.25 */
}

/**************************************************************************/
/*!
  \brief     Does the noise floor level estiamtion.
  The function calls the Noisefloor estimation function
  for the time segments decided based upon the transient
  information. The block is always divided into one or two segments.


  \return    none

*/
/**************************************************************************/
void
FDKsbrEnc_sbrNoiseFloorEstimateQmf(HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */
                         const SBR_FRAME_INFO *frame_info,   /*!< Time frequency grid of the current frame. */
                         FIXP_DBL *noiseLevels,              /*!< Pointer to vector to store the noise levels in.*/
                         FIXP_DBL **quotaMatrixOrig,         /*!< Matrix holding the quota values of the original. */
                         SCHAR    *indexVector,              /*!< Index vector to obtain the patched data. */
                         INT missingHarmonicsFlag,           /*!< Flag indicating if a strong tonal component will be missing. */
                         INT startIndex,                     /*!< Start index. */
                         int numberOfEstimatesPerFrame,      /*!< The number of tonality estimates per frame. */
                         int transientFrame,                 /*!< A flag indicating if a transient is present. */
                         INVF_MODE* pInvFiltLevels,          /*!< Pointer to the vector holding the inverse filtering levels. */
                         UINT sbrSyntaxFlags
                         )

{

  INT nNoiseEnvelopes, startPos[2], stopPos[2], env, band;

  INT noNoiseBands      = h_sbrNoiseFloorEstimate->noNoiseBands;
  INT *freqBandTable    = h_sbrNoiseFloorEstimate->freqBandTableQmf;

  nNoiseEnvelopes = frame_info->nNoiseEnvelopes;

  if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
    nNoiseEnvelopes = 1;
    startPos[0] = startIndex;
    stopPos[0]  = startIndex + min(numberOfEstimatesPerFrame,2);
  } else
  if(nNoiseEnvelopes == 1){
    startPos[0] = startIndex;
    stopPos[0]  = startIndex + 2;
  }
  else{
    startPos[0] = startIndex;
    stopPos[0]  = startIndex + 1;
    startPos[1] = startIndex + 1;
    stopPos[1]  = startIndex + 2;
  }

  /*
   * Estimate the noise floor.
   **************************************/
  for(env = 0; env < nNoiseEnvelopes; env++){
    for(band = 0; band < noNoiseBands; band++){
      FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES);
      qmfBasedNoiseFloorDetection(&noiseLevels[band + env*noNoiseBands],
                                  quotaMatrixOrig,
                                  indexVector,
                                  startPos[env],
                                  stopPos[env],
                                  freqBandTable[band],
                                  freqBandTable[band+1],
                                  h_sbrNoiseFloorEstimate->ana_max_level,
                                  h_sbrNoiseFloorEstimate->noiseFloorOffset[band],
                                  missingHarmonicsFlag,
                                  h_sbrNoiseFloorEstimate->weightFac,
                                  h_sbrNoiseFloorEstimate->diffThres,
                                  pInvFiltLevels[band]);
    }
  }


  /*
   * Smoothing of the values.
   **************************/
  smoothingOfNoiseLevels(noiseLevels,
                         nNoiseEnvelopes,
                         h_sbrNoiseFloorEstimate->noNoiseBands,
                         h_sbrNoiseFloorEstimate->prevNoiseLevels,
                         h_sbrNoiseFloorEstimate->smoothFilter,
                         transientFrame);


  /* quantisation*/
  for(env = 0; env < nNoiseEnvelopes; env++){
    for(band = 0; band < noNoiseBands; band++){
      FDK_ASSERT( (band + env*noNoiseBands) < MAX_NUM_NOISE_VALUES);
      noiseLevels[band + env*noNoiseBands] =
         (FIXP_DBL)NOISE_FLOOR_OFFSET_64 - (FIXP_DBL)CalcLdData(noiseLevels[band + env*noNoiseBands]+(FIXP_DBL)1) + QuantOffset;
    }
  }
}

/**************************************************************************/
/*!
  \brief


  \return    errorCode, noError if successful

*/
/**************************************************************************/
static INT
downSampleLoRes(INT *v_result,              /*!<    */
                INT num_result,             /*!<    */
                const UCHAR *freqBandTableRef,/*!<    */
                INT num_Ref)                /*!<    */
{
  INT step;
  INT i,j;
  INT org_length,result_length;
  INT v_index[MAX_FREQ_COEFFS/2];

  /* init */
  org_length=num_Ref;
  result_length=num_result;

  v_index[0]=0; /* Always use left border */
  i=0;
  while(org_length > 0) /* Create downsample vector */
    {
      i++;
      step=org_length/result_length; /* floor; */
      org_length=org_length - step;
      result_length--;
      v_index[i]=v_index[i-1]+step;
    }

  if(i != num_result )  /* Should never happen */
    return (1);/* error downsampling */

  for(j=0;j<=i;j++)     /* Use downsample vector to index LoResolution vector. */
    {
      v_result[j]=freqBandTableRef[v_index[j]];
    }

  return (0);
}

/**************************************************************************/
/*!
  \brief    Initialize an instance of the noise floor level estimation module.


  \return    errorCode, noError if successful

*/
/**************************************************************************/
INT
FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE  h_sbrNoiseFloorEstimate,   /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */
                             INT ana_max_level,                       /*!< Maximum level of the adaptive noise. */
                             const UCHAR *freqBandTable,      /*!< Frequany band table. */
                             INT nSfb,                                /*!< Number of frequency bands. */
                             INT noiseBands,                          /*!< Number of noise bands per octave. */
                             INT noiseFloorOffset,                    /*!< Noise floor offset. */
                             INT timeSlots,                           /*!< Number of time slots in a frame. */
                             UINT useSpeechConfig             /*!< Flag: adapt tuning parameters according to speech */
                            )
{
  INT i, qexp, qtmp;
  FIXP_DBL tmp, exp;

  FDKmemclear(h_sbrNoiseFloorEstimate,sizeof(SBR_NOISE_FLOOR_ESTIMATE));

  h_sbrNoiseFloorEstimate->smoothFilter = smoothFilter;
  if (useSpeechConfig) {
    h_sbrNoiseFloorEstimate->weightFac = (FIXP_DBL)MAXVAL_DBL;
    h_sbrNoiseFloorEstimate->diffThres = INVF_LOW_LEVEL;
  }
  else {
    h_sbrNoiseFloorEstimate->weightFac = FL2FXCONST_DBL(0.25f);
    h_sbrNoiseFloorEstimate->diffThres = INVF_MID_LEVEL;
  }

  h_sbrNoiseFloorEstimate->timeSlots     = timeSlots;
  h_sbrNoiseFloorEstimate->noiseBands    = noiseBands;

  /* h_sbrNoiseFloorEstimate->ana_max_level is scaled by 0.25  */
  switch(ana_max_level)
  {
  case 6:
      h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL;
      break;
  case 3:
      h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.5);
      break;
  case -3:
      h_sbrNoiseFloorEstimate->ana_max_level = FL2FXCONST_DBL(0.125);
      break;
  default:
      /* Should not enter here */
      h_sbrNoiseFloorEstimate->ana_max_level = (FIXP_DBL)MAXVAL_DBL;
      break;
  }

  /*
    calculate number of noise bands and allocate
  */
  if(FDKsbrEnc_resetSbrNoiseFloorEstimate(h_sbrNoiseFloorEstimate,freqBandTable,nSfb))
    return(1);

  if(noiseFloorOffset == 0) {
    tmp = ((FIXP_DBL)MAXVAL_DBL)>>NOISE_FLOOR_OFFSET_SCALING;
  }
  else {
    /* noiseFloorOffset has to be smaller than 12, because
       the result of the calculation below must be smaller than 1:
       (2^(noiseFloorOffset/3))*2^4<1                                        */
    FDK_ASSERT(noiseFloorOffset<12);

    /* Assumes the noise floor offset in tuning table are in q31    */
    /* Change the qformat here when non-zero values would be filled */
    exp = fDivNorm((FIXP_DBL)noiseFloorOffset, 3, &qexp);
    tmp = fPow(2, DFRACT_BITS-1, exp, qexp, &qtmp);
    tmp = scaleValue(tmp, qtmp-NOISE_FLOOR_OFFSET_SCALING);
  }

  for(i=0;i<h_sbrNoiseFloorEstimate->noNoiseBands;i++) {
    h_sbrNoiseFloorEstimate->noiseFloorOffset[i] = tmp;
  }

  return (0);
}

/**************************************************************************/
/*!
  \brief     Resets the current instance of the noise floor estiamtion
          module.


  \return    errorCode, noError if successful

*/
/**************************************************************************/
INT
FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate, /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */
                            const UCHAR *freqBandTable,             /*!< Frequany band table. */
                            INT nSfb)                             /*!< Number of bands in the frequency band table. */
{
    INT k2,kx;

    /*
    * Calculate number of noise bands
    ***********************************/
    k2=freqBandTable[nSfb];
    kx=freqBandTable[0];
    if(h_sbrNoiseFloorEstimate->noiseBands == 0){
        h_sbrNoiseFloorEstimate->noNoiseBands = 1;
    }
    else{
        /*
        * Calculate number of noise bands 1,2 or 3 bands/octave
        ********************************************************/
        FIXP_DBL tmp, ratio, lg2;
        INT ratio_e, qlg2, nNoiseBands;

        ratio = fDivNorm(k2, kx, &ratio_e);
        lg2 = fLog2(ratio, ratio_e, &qlg2);
        tmp = fMult((FIXP_DBL)(h_sbrNoiseFloorEstimate->noiseBands<<24), lg2);
        tmp = scaleValue(tmp, qlg2-23);

        nNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1);


        if (nNoiseBands > MAX_NUM_NOISE_COEFFS ) {
          nNoiseBands = MAX_NUM_NOISE_COEFFS;
        }

        if( nNoiseBands == 0 ) {
          nNoiseBands = 1;
        }

        h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands;

    }


    return(downSampleLoRes(h_sbrNoiseFloorEstimate->freqBandTableQmf,
        h_sbrNoiseFloorEstimate->noNoiseBands,
        freqBandTable,nSfb));
}

/**************************************************************************/
/*!
  \brief     Deletes the current instancce of the noise floor level
  estimation module.


  \return    none

*/
/**************************************************************************/
void
FDKsbrEnc_deleteSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoiseFloorEstimate)  /*!< Handle to SBR_NOISE_FLOOR_ESTIMATE struct */
{

  if (h_sbrNoiseFloorEstimate) {
    /*
      nothing to do
    */
  }
}