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

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

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

/************************* MPEG-D DRC decoder library **************************

   Author(s):

   Description:

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

#include "drcDec_types.h"

#include "drcDec_gainDecoder.h"

#include "drcGainDec_preprocess.h"

#include "drcGainDec_init.h"

#include "drcGainDec_process.h"

#include "drcDec_tools.h"

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

/* static functions                        */

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

static int _fitsLocation(DRC_INSTRUCTIONS_UNI_DRC* pInst,

                         const GAIN_DEC_LOCATION drcLocation) {

  int downmixId = pInst->drcApplyToDownmix ? pInst->downmixId[0] : 0;

  switch (drcLocation) {

    case GAIN_DEC_DRC1:

      return (downmixId == 0);

    case GAIN_DEC_DRC1_DRC2:

      return ((downmixId == 0) || (downmixId == DOWNMIX_ID_ANY_DOWNMIX));

    case GAIN_DEC_DRC2:

      return (downmixId == DOWNMIX_ID_ANY_DOWNMIX);

    case GAIN_DEC_DRC3:

      return ((downmixId != 0) && (downmixId != DOWNMIX_ID_ANY_DOWNMIX));

    case GAIN_DEC_DRC2_DRC3:

      return (downmixId != 0);

  }

  return 0;

}

static void _setChannelGains(HANDLE_DRC_GAIN_DECODER hGainDec,

                             const int numChannelGains,

                             const FIXP_DBL* channelGainDb) {

  int i, channelGain_e;

  FIXP_DBL channelGain;

  FDK_ASSERT(numChannelGains <= 8);

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

    if (channelGainDb[i] == (FIXP_DBL)MINVAL_DBL) {

      hGainDec->channelGain[i] = (FIXP_DBL)0;

    } else {

      /* add loudness normalisation gain (dB) to channel gain (dB) */

      FIXP_DBL tmp_channelGainDb = (channelGainDb[i] >> 1) +

                                   (hGainDec->loudnessNormalisationGainDb >> 2);

      tmp_channelGainDb =

          SATURATE_LEFT_SHIFT(tmp_channelGainDb, 1, DFRACT_BITS);

      channelGain = dB2lin(tmp_channelGainDb, 8, &channelGain_e);

      hGainDec->channelGain[i] = scaleValue(channelGain, channelGain_e - 8);

    }

  }

}

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

/* public functions                        */

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

DRC_ERROR

drcDec_GainDecoder_Open(HANDLE_DRC_GAIN_DECODER* phGainDec) {

  DRC_GAIN_DECODER* hGainDec = NULL;

  hGainDec = (DRC_GAIN_DECODER*)FDKcalloc(1, sizeof(DRC_GAIN_DECODER));

  if (hGainDec == NULL) return DE_MEMORY_ERROR;

  hGainDec->multiBandActiveDrcIndex = -1;

  hGainDec->channelGainActiveDrcIndex = -1;

  *phGainDec = hGainDec;

  return DE_OK;

}

DRC_ERROR

drcDec_GainDecoder_Init(HANDLE_DRC_GAIN_DECODER hGainDec) {

  DRC_ERROR err = DE_OK;

  err = initGainDec(hGainDec);

  if (err) return err;

  initDrcGainBuffers(hGainDec->frameSize, &hGainDec->drcGainBuffers);

  return err;

}

DRC_ERROR

drcDec_GainDecoder_SetParam(HANDLE_DRC_GAIN_DECODER hGainDec,

                            const GAIN_DEC_PARAM paramType,

                            const int paramValue) {

  switch (paramType) {

    case GAIN_DEC_FRAME_SIZE:

      if (paramValue < 0) return DE_PARAM_OUT_OF_RANGE;

      hGainDec->frameSize = paramValue;

      break;

    case GAIN_DEC_SAMPLE_RATE:

      if (paramValue < 0) return DE_PARAM_OUT_OF_RANGE;

      hGainDec->deltaTminDefault = getDeltaTmin(paramValue);

      break;

    default:

      return DE_PARAM_INVALID;

  }

  return DE_OK;

}

DRC_ERROR

drcDec_GainDecoder_SetCodecDependentParameters(

    HANDLE_DRC_GAIN_DECODER hGainDec, const DELAY_MODE delayMode,

    const int timeDomainSupported,

    const SUBBAND_DOMAIN_MODE subbandDomainSupported) {

  if ((delayMode != DM_REGULAR_DELAY) && (delayMode != DM_LOW_DELAY)) {

    return DE_NOT_OK;

  }

  hGainDec->delayMode = delayMode;

  hGainDec->timeDomainSupported = timeDomainSupported;

  hGainDec->subbandDomainSupported = subbandDomainSupported;

  return DE_OK;

}

DRC_ERROR

drcDec_GainDecoder_Config(HANDLE_DRC_GAIN_DECODER hGainDec,

                          HANDLE_UNI_DRC_CONFIG hUniDrcConfig,

                          const UCHAR numSelectedDrcSets,

                          const SCHAR* selectedDrcSetIds,

                          const UCHAR* selectedDownmixIds) {

  DRC_ERROR err = DE_OK;

  int a;

  hGainDec->nActiveDrcs = 0;

  hGainDec->multiBandActiveDrcIndex = -1;

  hGainDec->channelGainActiveDrcIndex = -1;

  for (a = 0; a < numSelectedDrcSets; a++) {

    err = initActiveDrc(hGainDec, hUniDrcConfig, selectedDrcSetIds[a],

                        selectedDownmixIds[a]);

    if (err) return err;

  }

  err = initActiveDrcOffset(hGainDec);

  if (err) return err;

  return err;

}

DRC_ERROR

drcDec_GainDecoder_Close(HANDLE_DRC_GAIN_DECODER* phGainDec) {

  if (*phGainDec != NULL) {

    FDKfree(*phGainDec);

    *phGainDec = NULL;

  }

  return DE_OK;

}

DRC_ERROR

drcDec_GainDecoder_Preprocess(HANDLE_DRC_GAIN_DECODER hGainDec,

                              HANDLE_UNI_DRC_GAIN hUniDrcGain,

                              const FIXP_DBL loudnessNormalizationGainDb,

                              const FIXP_SGL boost, const FIXP_SGL compress) {

  DRC_ERROR err = DE_OK;

  int a, c;

  /* lnbPointer is the index on the most recent node buffer */

  hGainDec->drcGainBuffers.lnbPointer++;

  if (hGainDec->drcGainBuffers.lnbPointer >= NUM_LNB_FRAMES)

    hGainDec->drcGainBuffers.lnbPointer = 0;

  for (a = 0; a < hGainDec->nActiveDrcs; a++) {

    /* prepare gain interpolation of sequences used by copying and modifying

     * nodes in node buffers */

    err = prepareDrcGain(hGainDec, hUniDrcGain, compress, boost,

                         loudnessNormalizationGainDb, a);

    if (err) return err;

  }

  for (a = 0; a < MAX_ACTIVE_DRCS; a++) {

    for (c = 0; c < 8; c++) {

      hGainDec->activeDrc[a]

          .lnbIndexForChannel[c][hGainDec->drcGainBuffers.lnbPointer] =

          -1; /* "no DRC processing" */

    }

    hGainDec->activeDrc[a].subbandGainsReady = 0;

  }

  for (c = 0; c < 8; c++) {

    hGainDec->drcGainBuffers

        .channelGain[c][hGainDec->drcGainBuffers.lnbPointer] =

        FL2FXCONST_DBL(1.0f / (float)(1 << 8));

  }

  return err;

}

/* create gain sequence out of gain sequences of last frame for concealment and

 * flushing */

DRC_ERROR

drcDec_GainDecoder_Conceal(HANDLE_DRC_GAIN_DECODER hGainDec,

                           HANDLE_UNI_DRC_CONFIG hUniDrcConfig,

                           HANDLE_UNI_DRC_GAIN hUniDrcGain) {

  int seq, gainSequenceCount;

  DRC_COEFFICIENTS_UNI_DRC* pCoef =

      selectDrcCoefficients(hUniDrcConfig, LOCATION_SELECTED);

  if (pCoef && pCoef->gainSequenceCount) {

    gainSequenceCount = fMin(pCoef->gainSequenceCount, (UCHAR)12);

  } else {

    gainSequenceCount = 1;

  }

  for (seq = 0; seq < gainSequenceCount; seq++) {

    int lastNodeIndex = 0;

    FIXP_SGL lastGainDb = (FIXP_SGL)0;

    lastNodeIndex = hUniDrcGain->nNodes[seq] - 1;

    if ((lastNodeIndex >= 0) && (lastNodeIndex < 16)) {

      lastGainDb = hUniDrcGain->gainNode[seq][lastNodeIndex].gainDb;

    }

    hUniDrcGain->nNodes[seq] = 1;

    if (lastGainDb > (FIXP_SGL)0) {

      hUniDrcGain->gainNode[seq][0].gainDb =

          FX_DBL2FX_SGL(fMult(FL2FXCONST_SGL(0.9f), lastGainDb));

    } else {

      hUniDrcGain->gainNode[seq][0].gainDb =

          FX_DBL2FX_SGL(fMult(FL2FXCONST_SGL(0.98f), lastGainDb));

    }

    hUniDrcGain->gainNode[seq][0].time = hGainDec->frameSize - 1;

  }

  return DE_OK;

}

void drcDec_GainDecoder_SetChannelGains(HANDLE_DRC_GAIN_DECODER hGainDec,

                                        const int numChannels,

                                        const int frameSize,

                                        const FIXP_DBL* channelGainDb,

                                        const int audioBufferChannelOffset,

                                        FIXP_DBL* audioBuffer) {

  int c, i;

  if (hGainDec->channelGainActiveDrcIndex >= 0) {

    /* channel gains will be applied in drcDec_GainDecoder_ProcessTimeDomain or

     * drcDec_GainDecoder_ProcessSubbandDomain, respectively. */

    _setChannelGains(hGainDec, numChannels, channelGainDb);

    if (!hGainDec->status) { /* overwrite previous channel gains at startup */

      DRC_GAIN_BUFFERS* pDrcGainBuffers = &hGainDec->drcGainBuffers;

      for (c = 0; c < numChannels; c++) {

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

          pDrcGainBuffers->channelGain[c][i] = hGainDec->channelGain[c];

        }

      }

      hGainDec->status = 1;

    }

  } else {

    /* smooth and apply channel gains */

    FIXP_DBL prevChannelGain[8];

    for (c = 0; c < numChannels; c++) {

      prevChannelGain[c] = hGainDec->channelGain[c];

    }

    _setChannelGains(hGainDec, numChannels, channelGainDb);

    if (!hGainDec->status) { /* overwrite previous channel gains at startup */

      for (c = 0; c < numChannels; c++)

        prevChannelGain[c] = hGainDec->channelGain[c];

      hGainDec->status = 1;

    }

    for (c = 0; c < numChannels; c++) {

      INT n_min = fMin(fMin(CntLeadingZeros(prevChannelGain[c]),

                            CntLeadingZeros(hGainDec->channelGain[c])) -

                           1,

                       9);

      FIXP_DBL gain = prevChannelGain[c] << n_min;

      FIXP_DBL stepsize = ((hGainDec->channelGain[c] << n_min) - gain);

      if (stepsize != (FIXP_DBL)0) {

        if (frameSize == 1024)

          stepsize = stepsize >> 10;

        else

          stepsize = (LONG)stepsize / frameSize;

      }

      n_min = 9 - n_min;

#ifdef FUNCTION_drcDec_GainDecoder_SetChannelGains_func1

      drcDec_GainDecoder_SetChannelGains_func1(audioBuffer, gain, stepsize,

                                               n_min, frameSize);

#else

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

        audioBuffer[i] = fMultDiv2(audioBuffer[i], gain) << n_min;

        gain += stepsize;

      }

#endif

      audioBuffer += audioBufferChannelOffset;

    }

  }

}

DRC_ERROR

drcDec_GainDecoder_ProcessTimeDomain(

    HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples,

    const GAIN_DEC_LOCATION drcLocation, const int channelOffset,

    const int drcChannelOffset, const int numChannelsProcessed,

    const int timeDataChannelOffset, FIXP_DBL* audioIOBuffer) {

  DRC_ERROR err = DE_OK;

  int a;

  if (!hGainDec->timeDomainSupported) {

    return DE_NOT_OK;

  }

  for (a = 0; a < hGainDec->nActiveDrcs; a++) {

    if (!_fitsLocation(hGainDec->activeDrc[a].pInst, drcLocation)) continue;

    /* Apply DRC */

    err = processDrcTime(hGainDec, a, delaySamples, channelOffset,

                         drcChannelOffset, numChannelsProcessed,

                         timeDataChannelOffset, audioIOBuffer);

    if (err) return err;

  }

  return err;

}

DRC_ERROR

drcDec_GainDecoder_ProcessSubbandDomain(

    HANDLE_DRC_GAIN_DECODER hGainDec, const int delaySamples,

    const GAIN_DEC_LOCATION drcLocation, const int channelOffset,

    const int drcChannelOffset, const int numChannelsProcessed,

    const int processSingleTimeslot, FIXP_DBL* audioIOBufferReal[],

    FIXP_DBL* audioIOBufferImag[]) {

  DRC_ERROR err = DE_OK;

  int a;

  if (hGainDec->subbandDomainSupported == SDM_OFF) {

    return DE_NOT_OK;

  }

  for (a = 0; a < hGainDec->nActiveDrcs; a++) {

    if (!_fitsLocation(hGainDec->activeDrc[a].pInst, drcLocation)) continue;

    /* Apply DRC */

    err = processDrcSubband(hGainDec, a, delaySamples, channelOffset,

                            drcChannelOffset, numChannelsProcessed,

                            processSingleTimeslot, audioIOBufferReal,

                            audioIOBufferImag);

    if (err) return err;

  }

  return err;

}

DRC_ERROR

drcDec_GainDecoder_SetLoudnessNormalizationGainDb(

    HANDLE_DRC_GAIN_DECODER hGainDec, FIXP_DBL loudnessNormalizationGainDb) {

  hGainDec->loudnessNormalisationGainDb = loudnessNormalizationGainDb;

  return DE_OK;

}

int drcDec_GainDecoder_GetFrameSize(HANDLE_DRC_GAIN_DECODER hGainDec) {

  if (hGainDec == NULL) return -1;

  return hGainDec->frameSize;

}

int drcDec_GainDecoder_GetDeltaTminDefault(HANDLE_DRC_GAIN_DECODER hGainDec) {

  if (hGainDec == NULL) return -1;

  return hGainDec->deltaTminDefault;

}