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

/**************************** PCM utility library ******************************

   Author(s):   Matthias Neusinger

   Description: Hard limiter for clipping prevention

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

#include "limiter.h"
#include "FDK_core.h"

/* library version */
#include "version.h"
/* library title */
#define TDLIMIT_LIB_TITLE "TD Limiter Lib"

/* create limiter */
TDLimiterPtr pcmLimiter_Create(unsigned int maxAttackMs, unsigned int releaseMs,
                               FIXP_DBL threshold, unsigned int maxChannels,
                               UINT maxSampleRate) {
  TDLimiterPtr limiter = NULL;
  unsigned int attack, release;
  FIXP_DBL attackConst, releaseConst, exponent;
  INT e_ans;

  /* calc attack and release time in samples */
  attack = (unsigned int)(maxAttackMs * maxSampleRate / 1000);
  release = (unsigned int)(releaseMs * maxSampleRate / 1000);

  /* alloc limiter struct */
  limiter = (TDLimiterPtr)FDKcalloc(1, sizeof(struct TDLimiter));
  if (!limiter) return NULL;

  /* alloc max and delay buffers */
  limiter->maxBuf = (FIXP_DBL*)FDKcalloc(attack + 1, sizeof(FIXP_DBL));
  limiter->delayBuf =
      (FIXP_DBL*)FDKcalloc(attack * maxChannels, sizeof(FIXP_DBL));

  if (!limiter->maxBuf || !limiter->delayBuf) {
    pcmLimiter_Destroy(limiter);
    return NULL;
  }

  /* attackConst = pow(0.1, 1.0 / (attack + 1)) */
  exponent = invFixp(attack + 1);
  attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
  attackConst = scaleValue(attackConst, e_ans);

  /* releaseConst  = (float)pow(0.1, 1.0 / (release + 1)) */
  exponent = invFixp(release + 1);
  releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
  releaseConst = scaleValue(releaseConst, e_ans);

  /* init parameters */
  limiter->attackMs = maxAttackMs;
  limiter->maxAttackMs = maxAttackMs;
  limiter->releaseMs = releaseMs;
  limiter->attack = attack;
  limiter->attackConst = attackConst;
  limiter->releaseConst = releaseConst;
  limiter->threshold = threshold >> TDL_GAIN_SCALING;
  limiter->channels = maxChannels;
  limiter->maxChannels = maxChannels;
  limiter->sampleRate = maxSampleRate;
  limiter->maxSampleRate = maxSampleRate;

  pcmLimiter_Reset(limiter);

  return limiter;
}

/* apply limiter */
TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn,
                                 INT_PCM* samplesOut, FIXP_DBL* RESTRICT pGain,
                                 const INT* RESTRICT gain_scale,
                                 const UINT gain_size, const UINT gain_delay,
                                 const UINT nSamples) {
  unsigned int i, j;
  FIXP_DBL tmp1;
  FIXP_DBL tmp2;
  FIXP_DBL tmp, old, gain, additionalGain = 0, additionalGainUnfiltered;
  FIXP_DBL minGain = FL2FXCONST_DBL(1.0f / (1 << 1));

  FDK_ASSERT(gain_size == 1);
  FDK_ASSERT(gain_delay <= nSamples);

  if (limiter == NULL) return TDLIMIT_INVALID_HANDLE;

  {
    unsigned int channels = limiter->channels;
    unsigned int attack = limiter->attack;
    FIXP_DBL attackConst = limiter->attackConst;
    FIXP_DBL releaseConst = limiter->releaseConst;
    FIXP_DBL threshold = limiter->threshold;

    FIXP_DBL max = limiter->max;
    FIXP_DBL* maxBuf = limiter->maxBuf;
    unsigned int maxBufIdx = limiter->maxBufIdx;
    FIXP_DBL cor = limiter->cor;
    FIXP_DBL* delayBuf = limiter->delayBuf;
    unsigned int delayBufIdx = limiter->delayBufIdx;

    FIXP_DBL smoothState0 = limiter->smoothState0;
    FIXP_DBL additionalGainSmoothState = limiter->additionalGainFilterState;
    FIXP_DBL additionalGainSmoothState1 = limiter->additionalGainFilterState1;

    if (!gain_delay) {
      additionalGain = pGain[0];
      if (gain_scale[0] > 0) {
        additionalGain <<= gain_scale[0];
      } else {
        additionalGain >>= -gain_scale[0];
      }
    }

    for (i = 0; i < nSamples; i++) {
      if (gain_delay) {
        if (i < gain_delay) {
          additionalGainUnfiltered = limiter->additionalGainPrev;
        } else {
          additionalGainUnfiltered = pGain[0];
        }

        /* Smooth additionalGain */
        /* [b,a] = butter(1, 0.01) */
        static const FIXP_SGL b[] = {FL2FXCONST_SGL(0.015466 * 2.0),
                                     FL2FXCONST_SGL(0.015466 * 2.0)};
        static const FIXP_SGL a[] = {(FIXP_SGL)MAXVAL_SGL,
                                     FL2FXCONST_SGL(-0.96907)};
        additionalGain = -fMult(additionalGainSmoothState, a[1]) +
                         fMultDiv2(additionalGainUnfiltered, b[0]) +
                         fMultDiv2(additionalGainSmoothState1, b[1]);
        additionalGainSmoothState1 = additionalGainUnfiltered;
        additionalGainSmoothState = additionalGain;

        /* Apply the additional scaling that has no delay and no smoothing */
        if (gain_scale[0] > 0) {
          additionalGain <<= gain_scale[0];
        } else {
          additionalGain >>= -gain_scale[0];
        }
      }
      /* get maximum absolute sample value of all channels, including the
       * additional gain. */
      tmp1 = (FIXP_DBL)0;
      for (j = 0; j < channels; j++) {
        tmp2 = PCM_LIM2FIXP_DBL(samplesIn[j]);
        tmp2 = fAbs(tmp2);
        tmp2 = FIXP_DBL(INT(tmp2) ^ INT((tmp2 >> (SAMPLE_BITS_LIM - 1))));
        tmp1 = fMax(tmp1, tmp2);
      }
      tmp = fMult(tmp1, additionalGain);

      /* set threshold as lower border to save calculations in running maximum
       * algorithm */
      tmp = fMax(tmp, threshold);

      /* running maximum */
      old = maxBuf[maxBufIdx];
      maxBuf[maxBufIdx] = tmp;

      if (tmp >= max) {
        /* new sample is greater than old maximum, so it is the new maximum */
        max = tmp;
      } else if (old < max) {
        /* maximum does not change, as the sample, which has left the window was
           not the maximum */
      } else {
        /* the old maximum has left the window, we have to search the complete
           buffer for the new max */
        max = maxBuf[0];
        for (j = 1; j <= attack; j++) {
          max = fMax(max, maxBuf[j]);
        }
      }
      maxBufIdx++;
      if (maxBufIdx >= attack + 1) maxBufIdx = 0;

      /* calc gain */
      /* gain is downscaled by one, so that gain = 1.0 can be represented */
      if (max > threshold) {
        gain = fDivNorm(threshold, max) >> 1;
      } else {
        gain = FL2FXCONST_DBL(1.0f / (1 << 1));
      }

      /* gain smoothing, method: TDL_EXPONENTIAL */
      /* first order IIR filter with attack correction to avoid overshoots */

      /* correct the 'aiming' value of the exponential attack to avoid the
       * remaining overshoot */
      if (gain < smoothState0) {
        cor = fMin(cor,
                   fMultDiv2((gain - fMultDiv2(FL2FXCONST_SGL(0.1f * (1 << 1)),
                                               smoothState0)),
                             FL2FXCONST_SGL(1.11111111f / (1 << 1)))
                       << 2);
      } else {
        cor = gain;
      }

      /* smoothing filter */
      if (cor < smoothState0) {
        smoothState0 =
            fMult(attackConst, (smoothState0 - cor)) + cor; /* attack */
        smoothState0 = fMax(smoothState0, gain); /* avoid overshooting target */
      } else {
        /* sign inversion twice to round towards +infinity,
           so that gain can converge to 1.0 again,
           for bit-identical output when limiter is not active */
        smoothState0 =
            -fMult(releaseConst, -(smoothState0 - cor)) + cor; /* release */
      }

      gain = smoothState0;

      FIXP_DBL* p_delayBuf = &delayBuf[delayBufIdx * channels + 0];
      if (gain < FL2FXCONST_DBL(1.0f / (1 << 1))) {
        gain <<= 1;
        /* lookahead delay, apply gain */
        for (j = 0; j < channels; j++) {
          tmp = p_delayBuf[j];
          p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain);

          /* Apply gain to delayed signal */
          tmp = fMultDiv2(tmp, gain);

          samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT(
              tmp, TDL_GAIN_SCALING + 1, DFRACT_BITS));
        }
        gain >>= 1;
      } else {
        /* lookahead delay, apply gain=1.0f */
        for (j = 0; j < channels; j++) {
          tmp = p_delayBuf[j];
          p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain);
          samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT(
              tmp, TDL_GAIN_SCALING, DFRACT_BITS));
        }
      }

      delayBufIdx++;
      if (delayBufIdx >= attack) {
        delayBufIdx = 0;
      }

      /* save minimum gain factor */
      if (gain < minGain) {
        minGain = gain;
      }

      /* advance sample pointer by <channel> samples */
      samplesIn += channels;
      samplesOut += channels;
    }

    limiter->max = max;
    limiter->maxBufIdx = maxBufIdx;
    limiter->cor = cor;
    limiter->delayBufIdx = delayBufIdx;

    limiter->smoothState0 = smoothState0;
    limiter->additionalGainFilterState = additionalGainSmoothState;
    limiter->additionalGainFilterState1 = additionalGainSmoothState1;

    limiter->minGain = minGain;

    limiter->additionalGainPrev = pGain[0];

    return TDLIMIT_OK;
  }
}

/* set limiter threshold */
TDLIMITER_ERROR pcmLimiter_SetThreshold(TDLimiterPtr limiter,
                                        FIXP_DBL threshold) {
  if (limiter == NULL) return TDLIMIT_INVALID_HANDLE;

  limiter->threshold = threshold >> TDL_GAIN_SCALING;

  return TDLIMIT_OK;
}

/* reset limiter */
TDLIMITER_ERROR pcmLimiter_Reset(TDLimiterPtr limiter) {
  if (limiter != NULL) {
    limiter->maxBufIdx = 0;
    limiter->delayBufIdx = 0;
    limiter->max = (FIXP_DBL)0;
    limiter->cor = FL2FXCONST_DBL(1.0f / (1 << 1));
    limiter->smoothState0 = FL2FXCONST_DBL(1.0f / (1 << 1));
    limiter->minGain = FL2FXCONST_DBL(1.0f / (1 << 1));

    limiter->additionalGainPrev =
        FL2FXCONST_DBL(1.0f / (1 << TDL_GAIN_SCALING));
    limiter->additionalGainFilterState =
        FL2FXCONST_DBL(1.0f / (1 << TDL_GAIN_SCALING));
    limiter->additionalGainFilterState1 =
        FL2FXCONST_DBL(1.0f / (1 << TDL_GAIN_SCALING));

    FDKmemset(limiter->maxBuf, 0, (limiter->attack + 1) * sizeof(FIXP_DBL));
    FDKmemset(limiter->delayBuf, 0,
              limiter->attack * limiter->channels * sizeof(FIXP_DBL));
  } else {
    return TDLIMIT_INVALID_HANDLE;
  }

  return TDLIMIT_OK;
}

/* destroy limiter */
TDLIMITER_ERROR pcmLimiter_Destroy(TDLimiterPtr limiter) {
  if (limiter != NULL) {
    FDKfree(limiter->maxBuf);
    FDKfree(limiter->delayBuf);

    FDKfree(limiter);
  } else {
    return TDLIMIT_INVALID_HANDLE;
  }
  return TDLIMIT_OK;
}

/* get delay in samples */
unsigned int pcmLimiter_GetDelay(TDLimiterPtr limiter) {
  FDK_ASSERT(limiter != NULL);
  return limiter->attack;
}

/* get maximum gain reduction of last processed block */
INT pcmLimiter_GetMaxGainReduction(TDLimiterPtr limiter) {
  /* maximum gain reduction in dB = -20 * log10(limiter->minGain)
     = -20 * log2(limiter->minGain)/log2(10) = -6.0206*log2(limiter->minGain) */
  int e_ans;
  FIXP_DBL loggain, maxGainReduction;

  FDK_ASSERT(limiter != NULL);

  loggain = fLog2(limiter->minGain, 1, &e_ans);

  maxGainReduction = fMult(loggain, FL2FXCONST_DBL(-6.0206f / (1 << 3)));

  return fixp_roundToInt(maxGainReduction, (e_ans + 3));
}

/* set number of channels */
TDLIMITER_ERROR pcmLimiter_SetNChannels(TDLimiterPtr limiter,
                                        unsigned int nChannels) {
  if (limiter == NULL) return TDLIMIT_INVALID_HANDLE;

  if (nChannels > limiter->maxChannels) return TDLIMIT_INVALID_PARAMETER;

  limiter->channels = nChannels;
  // pcmLimiter_Reset(limiter);

  return TDLIMIT_OK;
}

/* set sampling rate */
TDLIMITER_ERROR pcmLimiter_SetSampleRate(TDLimiterPtr limiter,
                                         UINT sampleRate) {
  unsigned int attack, release;
  FIXP_DBL attackConst, releaseConst, exponent;
  INT e_ans;

  if (limiter == NULL) return TDLIMIT_INVALID_HANDLE;

  if (sampleRate > limiter->maxSampleRate) return TDLIMIT_INVALID_PARAMETER;

  /* update attack and release time in samples */
  attack = (unsigned int)(limiter->attackMs * sampleRate / 1000);
  release = (unsigned int)(limiter->releaseMs * sampleRate / 1000);

  /* attackConst = pow(0.1, 1.0 / (attack + 1)) */
  exponent = invFixp(attack + 1);
  attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
  attackConst = scaleValue(attackConst, e_ans);

  /* releaseConst  = (float)pow(0.1, 1.0 / (release + 1)) */
  exponent = invFixp(release + 1);
  releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
  releaseConst = scaleValue(releaseConst, e_ans);

  limiter->attack = attack;
  limiter->attackConst = attackConst;
  limiter->releaseConst = releaseConst;
  limiter->sampleRate = sampleRate;

  /* reset */
  // pcmLimiter_Reset(limiter);

  return TDLIMIT_OK;
}

/* set attack time */
TDLIMITER_ERROR pcmLimiter_SetAttack(TDLimiterPtr limiter,
                                     unsigned int attackMs) {
  unsigned int attack;
  FIXP_DBL attackConst, exponent;
  INT e_ans;

  if (limiter == NULL) return TDLIMIT_INVALID_HANDLE;

  if (attackMs > limiter->maxAttackMs) return TDLIMIT_INVALID_PARAMETER;

  /* calculate attack time in samples */
  attack = (unsigned int)(attackMs * limiter->sampleRate / 1000);

  /* attackConst = pow(0.1, 1.0 / (attack + 1)) */
  exponent = invFixp(attack + 1);
  attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
  attackConst = scaleValue(attackConst, e_ans);

  limiter->attack = attack;
  limiter->attackConst = attackConst;
  limiter->attackMs = attackMs;

  return TDLIMIT_OK;
}

/* set release time */
TDLIMITER_ERROR pcmLimiter_SetRelease(TDLimiterPtr limiter,
                                      unsigned int releaseMs) {
  unsigned int release;
  FIXP_DBL releaseConst, exponent;
  INT e_ans;

  if (limiter == NULL) return TDLIMIT_INVALID_HANDLE;

  /* calculate  release time in samples */
  release = (unsigned int)(releaseMs * limiter->sampleRate / 1000);

  /* releaseConst  = (float)pow(0.1, 1.0 / (release + 1)) */
  exponent = invFixp(release + 1);
  releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
  releaseConst = scaleValue(releaseConst, e_ans);

  limiter->releaseConst = releaseConst;
  limiter->releaseMs = releaseMs;

  return TDLIMIT_OK;
}

/* Get library info for this module. */
TDLIMITER_ERROR pcmLimiter_GetLibInfo(LIB_INFO* info) {
  int i;

  if (info == NULL) {
    return TDLIMIT_INVALID_PARAMETER;
  }

  /* Search for next free tab */
  for (i = 0; i < FDK_MODULE_LAST; i++) {
    if (info[i].module_id == FDK_NONE) break;
  }
  if (i == FDK_MODULE_LAST) {
    return TDLIMIT_UNKNOWN;
  }

  /* Add the library info */
  info[i].module_id = FDK_TDLIMIT;
  info[i].version =
      LIB_VERSION(PCMUTIL_LIB_VL0, PCMUTIL_LIB_VL1, PCMUTIL_LIB_VL2);
  LIB_VERSION_STRING(info + i);
  info[i].build_date = PCMUTIL_LIB_BUILD_DATE;
  info[i].build_time = PCMUTIL_LIB_BUILD_TIME;
  info[i].title = TDLIMIT_LIB_TITLE;

  /* Set flags */
  info[i].flags = CAPF_LIMITER;

  /* Add lib info for FDK tools (if not yet done). */
  FDK_toolsGetLibInfo(info);

  return TDLIMIT_OK;
}