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

Software License for The Fraunhofer FDK AAC Codec Library for Android

© Copyright  1995 - 2020 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

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

/**************************** AAC encoder library ******************************

   Author(s):   M. Schug / A. Groeschel

   Description: fast aac coder functions

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

#include "aacenc.h"

#include "bitenc.h"

#include "interface.h"

#include "psy_configuration.h"

#include "psy_main.h"

#include "qc_main.h"

#include "bandwidth.h"

#include "channel_map.h"

#include "tns_func.h"

#include "aacEnc_ram.h"

#include "genericStds.h"

#define BITRES_MIN \

  300 /* default threshold for using reduced/disabled bitres mode */

#define BITRES_MAX_LD 4000

#define BITRES_MIN_LD 500

#define BITRATE_MAX_LD 70000 /* Max assumed bitrate for bitres calculation */

#define BITRATE_MIN_LD 12000 /* Min assumed bitrate for bitres calculation */

INT FDKaacEnc_CalcBitsPerFrame(const INT bitRate, const INT frameLength,

                               const INT samplingRate) {

  int shift = 0;

  while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength &&

         (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate) {

    shift++;

  }

  return (bitRate * (frameLength >> shift)) / (samplingRate >> shift);

}

INT FDKaacEnc_CalcBitrate(const INT bitsPerFrame, const INT frameLength,

                          const INT samplingRate) {

  int shift = 0;

  while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength &&

         (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate) {

    shift++;

  }

  return (bitsPerFrame * (samplingRate >> shift)) / (frameLength >> shift);

}

static AAC_ENCODER_ERROR FDKaacEnc_InitCheckAncillary(

    INT bitRate, INT framelength, INT ancillaryRate, INT *ancillaryBitsPerFrame,

    INT sampleRate);

INT FDKaacEnc_LimitBitrate(HANDLE_TRANSPORTENC hTpEnc, AUDIO_OBJECT_TYPE aot,

                           INT coreSamplingRate, INT frameLength, INT nChannels,

                           INT nChannelsEff, INT bitRate, INT averageBits,

                           INT *pAverageBitsPerFrame,

                           AACENC_BITRATE_MODE bitrateMode, INT nSubFrames) {

  INT transportBits, prevBitRate, averageBitsPerFrame, minBitrate = 0, iter = 0;

  INT minBitsPerFrame = 40 * nChannels;

  if (isLowDelay(aot)) {

    minBitrate = 8000 * nChannelsEff;

  }

  do {

    prevBitRate = bitRate;

    averageBitsPerFrame =

        FDKaacEnc_CalcBitsPerFrame(bitRate, frameLength, coreSamplingRate) /

        nSubFrames;

    if (pAverageBitsPerFrame != NULL) {

      *pAverageBitsPerFrame = averageBitsPerFrame;

    }

    if (hTpEnc != NULL) {

      transportBits = transportEnc_GetStaticBits(hTpEnc, averageBitsPerFrame);

    } else {

      /* Assume some worst case */

      transportBits = 208;

    }

    bitRate = fMax(bitRate,

                   fMax(minBitrate,

                        FDKaacEnc_CalcBitrate((minBitsPerFrame + transportBits),

                                              frameLength, coreSamplingRate)));

    FDK_ASSERT(bitRate >= 0);

    bitRate = fMin(bitRate, FDKaacEnc_CalcBitrate(

                                (nChannelsEff * MIN_BUFSIZE_PER_EFF_CHAN),

                                frameLength, coreSamplingRate));

    FDK_ASSERT(bitRate >= 0);

  } while (prevBitRate != bitRate && iter++ < 3);

  return bitRate;

}

typedef struct {

  AACENC_BITRATE_MODE bitrateMode;

  int chanBitrate[2]; /* mono/stereo settings */

} CONFIG_TAB_ENTRY_VBR;

static const CONFIG_TAB_ENTRY_VBR configTabVBR[] = {

    {AACENC_BR_MODE_CBR, {0, 0}},

    {AACENC_BR_MODE_VBR_1, {32000, 20000}},

    {AACENC_BR_MODE_VBR_2, {40000, 32000}},

    {AACENC_BR_MODE_VBR_3, {56000, 48000}},

    {AACENC_BR_MODE_VBR_4, {72000, 64000}},

    {AACENC_BR_MODE_VBR_5, {112000, 96000}}};

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

     functionname: FDKaacEnc_GetVBRBitrate

     description:  Get VBR bitrate from vbr quality

     input params: int vbrQuality (VBR0, VBR1, VBR2)

                   channelMode

     returns:      vbr bitrate

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

INT FDKaacEnc_GetVBRBitrate(AACENC_BITRATE_MODE bitrateMode,

                            CHANNEL_MODE channelMode) {

  INT bitrate = 0;

  INT monoStereoMode = 0; /* default mono */

  if (FDKaacEnc_GetMonoStereoMode(channelMode) == EL_MODE_STEREO) {

    monoStereoMode = 1;

  }

  switch (bitrateMode) {

    case AACENC_BR_MODE_VBR_1:

    case AACENC_BR_MODE_VBR_2:

    case AACENC_BR_MODE_VBR_3:

    case AACENC_BR_MODE_VBR_4:

    case AACENC_BR_MODE_VBR_5:

      bitrate = configTabVBR[bitrateMode].chanBitrate[monoStereoMode];

      break;

    case AACENC_BR_MODE_INVALID:

    case AACENC_BR_MODE_CBR:

    case AACENC_BR_MODE_SFR:

    case AACENC_BR_MODE_FF:

    default:

      bitrate = 0;

      break;

  }

  /* convert channel bitrate to overall bitrate*/

  bitrate *= FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff;

  return bitrate;

}

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

    functionname: FDKaacEnc_AdjustVBRBitrateMode

    description:  Adjust bitrate mode to given bitrate parameter

    input params: int vbrQuality (VBR0, VBR1, VBR2)

                  bitrate

                  channelMode

    returns:      vbr bitrate mode

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

AACENC_BITRATE_MODE FDKaacEnc_AdjustVBRBitrateMode(

    AACENC_BITRATE_MODE bitrateMode, INT bitrate, CHANNEL_MODE channelMode) {

  AACENC_BITRATE_MODE newBitrateMode = bitrateMode;

  if (bitrate != -1) {

    const INT monoStereoMode =

        (FDKaacEnc_GetMonoStereoMode(channelMode) == EL_MODE_STEREO) ? 1 : 0;

    const INT nChannelsEff =

        FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff;

    newBitrateMode = AACENC_BR_MODE_INVALID;

    for (int idx = (int)(sizeof(configTabVBR) / sizeof(*configTabVBR)) - 1;

         idx >= 0; idx--) {

      if (bitrate >=

          configTabVBR[idx].chanBitrate[monoStereoMode] * nChannelsEff) {

        if (configTabVBR[idx].chanBitrate[monoStereoMode] * nChannelsEff <

            FDKaacEnc_GetVBRBitrate(bitrateMode, channelMode)) {

          newBitrateMode = configTabVBR[idx].bitrateMode;

        } else {

          newBitrateMode = bitrateMode;

        }

        break;

      }

    }

  }

  return AACENC_BR_MODE_IS_VBR(newBitrateMode) ? newBitrateMode

                                               : AACENC_BR_MODE_INVALID;

}

/**

 * \brief  Convert encoder bitreservoir value for transport library.

 *

 * \param hAacEnc               Encoder handle

 *

 * \return  Corrected bitreservoir level used in transport library.

 */

static INT FDKaacEnc_EncBitresToTpBitres(const HANDLE_AAC_ENC hAacEnc) {

  INT transportBitreservoir = 0;

  switch (hAacEnc->bitrateMode) {

    case AACENC_BR_MODE_CBR:

      transportBitreservoir =

          hAacEnc->qcKernel->bitResTot; /* encoder bitreservoir level */

      break;

    case AACENC_BR_MODE_VBR_1:

    case AACENC_BR_MODE_VBR_2:

    case AACENC_BR_MODE_VBR_3:

    case AACENC_BR_MODE_VBR_4:

    case AACENC_BR_MODE_VBR_5:

      transportBitreservoir = FDK_INT_MAX; /* signal variable bitrate */

      break;

    case AACENC_BR_MODE_SFR:

      transportBitreservoir = 0; /* super framing and fixed framing */

      break;                     /* without bitreservoir signaling */

    default:

    case AACENC_BR_MODE_INVALID:

      transportBitreservoir = 0; /* invalid configuration*/

  }

  if (hAacEnc->config->audioMuxVersion == 2) {

    transportBitreservoir =

        MIN_BUFSIZE_PER_EFF_CHAN * hAacEnc->channelMapping.nChannelsEff;

  }

  return transportBitreservoir;

}

INT FDKaacEnc_GetBitReservoirState(const HANDLE_AAC_ENC hAacEncoder) {

  return FDKaacEnc_EncBitresToTpBitres(hAacEncoder);

}

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

     functionname: FDKaacEnc_AacInitDefaultConfig

     description:  gives reasonable default configuration

     returns:      ---

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

void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config) {

  /* make the preinitialization of the structs flexible */

  FDKmemclear(config, sizeof(AACENC_CONFIG));

  /* default ancillary */

  config->anc_Rate = 0;       /* no ancillary data */

  config->ancDataBitRate = 0; /* no additional consumed bitrate */

  /* default configurations */

  config->bitRate = -1; /* bitrate must be set*/

  config->averageBits =

      -1; /* instead of bitrate/s we can configure bits/superframe */

  config->bitrateMode =

      AACENC_BR_MODE_CBR;           /* set bitrate mode to constant bitrate */

  config->bandWidth = 0;            /* get bandwidth from table */

  config->useTns = TNS_ENABLE_MASK; /* tns enabled completly */

  config->usePns =

      1; /* depending on channelBitrate this might be set to 0 later */

  config->useIS = 1;        /* Intensity Stereo Configuration */

  config->useMS = 1;        /* MS Stereo tool */

  config->framelength = -1; /* Framesize not configured */

  config->syntaxFlags = 0;  /* default syntax with no specialities */

  config->epConfig = -1;    /* no ER syntax -> no additional error protection */

  config->nSubFrames = 1;   /* default, no sub frames */

  config->channelOrder = CH_ORDER_MPEG; /* Use MPEG channel ordering. */

  config->channelMode = MODE_UNKNOWN;

  config->minBitsPerFrame = -1; /* minum number of bits in each AU */

  config->maxBitsPerFrame = -1; /* minum number of bits in each AU */

  config->audioMuxVersion = -1; /* audio mux version not configured */

  config->downscaleFactor =

      1; /* downscale factor for ELD reduced delay mode, 1 is normal ELD */

}

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

    functionname: FDKaacEnc_Open

    description:  allocate and initialize a new encoder instance

    returns:      error code

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

AAC_ENCODER_ERROR FDKaacEnc_Open(HANDLE_AAC_ENC *phAacEnc, const INT nElements,

                                 const INT nChannels, const INT nSubFrames) {

  AAC_ENCODER_ERROR ErrorStatus;

  AAC_ENC *hAacEnc = NULL;

  UCHAR *dynamicRAM = NULL;

  if (phAacEnc == NULL) {

    return AAC_ENC_INVALID_HANDLE;

  }

  /* allocate encoder structure */

  hAacEnc = GetRam_aacEnc_AacEncoder();

  if (hAacEnc == NULL) {

    ErrorStatus = AAC_ENC_NO_MEMORY;

    goto bail;

  }

  FDKmemclear(hAacEnc, sizeof(AAC_ENC));

  if (NULL == (hAacEnc->dynamic_RAM = GetAACdynamic_RAM())) {

    ErrorStatus = AAC_ENC_NO_MEMORY;

    goto bail;

  }

  dynamicRAM = (UCHAR *)hAacEnc->dynamic_RAM;

  /* allocate the Psy aud Psy Out structure */

  ErrorStatus =

      FDKaacEnc_PsyNew(&hAacEnc->psyKernel, nElements, nChannels, dynamicRAM);

  if (ErrorStatus != AAC_ENC_OK) goto bail;

  ErrorStatus = FDKaacEnc_PsyOutNew(hAacEnc->psyOut, nElements, nChannels,

                                    nSubFrames, dynamicRAM);

  if (ErrorStatus != AAC_ENC_OK) goto bail;

  /* allocate the Q&C Out structure */

  ErrorStatus = FDKaacEnc_QCOutNew(hAacEnc->qcOut, nElements, nChannels,

                                   nSubFrames, dynamicRAM);

  if (ErrorStatus != AAC_ENC_OK) goto bail;

  /* allocate the Q&C kernel */

  ErrorStatus = FDKaacEnc_QCNew(&hAacEnc->qcKernel, nElements, dynamicRAM);

  if (ErrorStatus != AAC_ENC_OK) goto bail;

  hAacEnc->maxChannels = nChannels;

  hAacEnc->maxElements = nElements;

  hAacEnc->maxFrames = nSubFrames;

bail:

  *phAacEnc = hAacEnc;

  return ErrorStatus;

}

AAC_ENCODER_ERROR FDKaacEnc_Initialize(

    HANDLE_AAC_ENC hAacEnc,

    AACENC_CONFIG *config, /* pre-initialized config struct */

    HANDLE_TRANSPORTENC hTpEnc, ULONG initFlags) {

  AAC_ENCODER_ERROR ErrorStatus;

  INT psyBitrate, tnsMask;  // INT profile = 1;

  CHANNEL_MAPPING *cm = NULL;

  INT mbfac_e, qbw;

  FIXP_DBL mbfac, bw_ratio;

  QC_INIT qcInit;

  INT averageBitsPerFrame = 0;

  const CHANNEL_MODE prevChannelMode = hAacEnc->encoderMode;

  if (config == NULL) return AAC_ENC_INVALID_HANDLE;

  /******************* sanity checks *******************/

  /* check config structure */

  if (config->nChannels < 1 || config->nChannels > (8)) {

    return AAC_ENC_UNSUPPORTED_CHANNELCONFIG;

  }

  /* check sample rate */

  switch (config->sampleRate) {

    case 8000:

    case 11025:

    case 12000:

    case 16000:

    case 22050:

    case 24000:

    case 32000:

    case 44100:

    case 48000:

    case 64000:

    case 88200:

    case 96000:

      break;

    default:

      return AAC_ENC_UNSUPPORTED_SAMPLINGRATE;

  }

  /* bitrate has to be set */

  if (config->bitRate == -1) {

    return AAC_ENC_UNSUPPORTED_BITRATE;

  }

  /* check bit rate */

  if (FDKaacEnc_LimitBitrate(

          hTpEnc, config->audioObjectType, config->sampleRate,

          config->framelength, config->nChannels,

          FDKaacEnc_GetChannelModeConfiguration(config->channelMode)

              ->nChannelsEff,

          config->bitRate, config->averageBits, &averageBitsPerFrame,

          config->bitrateMode, config->nSubFrames) != config->bitRate &&

      !(AACENC_BR_MODE_IS_VBR(config->bitrateMode))) {

    return AAC_ENC_UNSUPPORTED_BITRATE;

  }

  if (config->syntaxFlags & AC_ER_VCB11) {

    return AAC_ENC_UNSUPPORTED_ER_FORMAT;

  }

  if (config->syntaxFlags & AC_ER_HCR) {

    return AAC_ENC_UNSUPPORTED_ER_FORMAT;

  }

  /* check frame length */

  switch (config->framelength) {

    case 1024:

      if (isLowDelay(config->audioObjectType)) {

        return AAC_ENC_INVALID_FRAME_LENGTH;

      }

      break;

    case 128:

    case 256:

    case 512:

    case 120:

    case 240:

    case 480:

      if (!isLowDelay(config->audioObjectType)) {

        return AAC_ENC_INVALID_FRAME_LENGTH;

      }

      break;

    default:

      return AAC_ENC_INVALID_FRAME_LENGTH;

  }

  if (config->anc_Rate != 0) {

    ErrorStatus = FDKaacEnc_InitCheckAncillary(

        config->bitRate, config->framelength, config->anc_Rate,

        &hAacEnc->ancillaryBitsPerFrame, config->sampleRate);

    if (ErrorStatus != AAC_ENC_OK) goto bail;

    /* update estimated consumed bitrate */

    config->ancDataBitRate +=

        FDKaacEnc_CalcBitrate(hAacEnc->ancillaryBitsPerFrame,

                              config->framelength, config->sampleRate);

  }

  /* maximal allowed DSE bytes in frame */

  config->maxAncBytesPerAU =

      fMin((256), fMax(0, FDKaacEnc_CalcBitsPerFrame(

                              (config->bitRate - (config->nChannels * 8000)),

                              config->framelength, config->sampleRate) >>

                              3));

  /* bind config to hAacEnc->config */

  hAacEnc->config = config;

  /* set hAacEnc->bitrateMode */

  hAacEnc->bitrateMode = config->bitrateMode;

  hAacEnc->encoderMode = config->channelMode;

  ErrorStatus = FDKaacEnc_InitChannelMapping(

      hAacEnc->encoderMode, config->channelOrder, &hAacEnc->channelMapping);

  if (ErrorStatus != AAC_ENC_OK) goto bail;

  cm = &hAacEnc->channelMapping;

  ErrorStatus = FDKaacEnc_DetermineBandWidth(

      config->bandWidth, config->bitRate - config->ancDataBitRate,

      hAacEnc->bitrateMode, config->sampleRate, config->framelength, cm,

      hAacEnc->encoderMode, &hAacEnc->config->bandWidth);

  if (ErrorStatus != AAC_ENC_OK) goto bail;

  hAacEnc->bandwidth90dB = (INT)hAacEnc->config->bandWidth;

  tnsMask = config->useTns ? TNS_ENABLE_MASK : 0x0;

  psyBitrate = config->bitRate - config->ancDataBitRate;

  if ((hAacEnc->encoderMode != prevChannelMode) || (initFlags != 0)) {

    /* Reinitialize psych states in case of channel configuration change ore if

     * full reset requested. */

    ErrorStatus = FDKaacEnc_psyInit(hAacEnc->psyKernel, hAacEnc->psyOut,

                                    hAacEnc->maxFrames, hAacEnc->maxChannels,

                                    config->audioObjectType, cm);

    if (ErrorStatus != AAC_ENC_OK) goto bail;

  }

  ErrorStatus = FDKaacEnc_psyMainInit(

      hAacEnc->psyKernel, config->audioObjectType, cm, config->sampleRate,

      config->framelength, psyBitrate, tnsMask, hAacEnc->bandwidth90dB,

      config->usePns, config->useIS, config->useMS, config->syntaxFlags,

      initFlags);

  if (ErrorStatus != AAC_ENC_OK) goto bail;

  ErrorStatus = FDKaacEnc_QCOutInit(hAacEnc->qcOut, hAacEnc->maxFrames, cm);

  if (ErrorStatus != AAC_ENC_OK) goto bail;

  qcInit.channelMapping = &hAacEnc->channelMapping;

  qcInit.sceCpe = 0;

  if (AACENC_BR_MODE_IS_VBR(config->bitrateMode)) {

    qcInit.averageBits = (averageBitsPerFrame + 7) & ~7;

    qcInit.bitRes = MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff;

    qcInit.maxBits = MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff;

    qcInit.maxBits = (config->maxBitsPerFrame != -1)

                         ? fixMin(qcInit.maxBits, config->maxBitsPerFrame)

                         : qcInit.maxBits;

    qcInit.maxBits = fixMax(qcInit.maxBits, (averageBitsPerFrame + 7) & ~7);

    qcInit.minBits =

        (config->minBitsPerFrame != -1) ? config->minBitsPerFrame : 0;

    qcInit.minBits = fixMin(qcInit.minBits, averageBitsPerFrame & ~7);

  } else {

    INT bitreservoir = -1; /* default bitreservoir size*/

    if (isLowDelay(config->audioObjectType)) {

      INT brPerChannel = config->bitRate / config->nChannels;

      brPerChannel = fMin(BITRATE_MAX_LD, fMax(BITRATE_MIN_LD, brPerChannel));

      /* bitreservoir  =

       * (maxBitRes-minBitRes)/(maxBitRate-minBitrate)*(bitRate-minBitrate)+minBitRes;

       */

      FIXP_DBL slope = fDivNorm(

          (brPerChannel - BITRATE_MIN_LD),

          BITRATE_MAX_LD - BITRATE_MIN_LD); /* calc slope for interpolation */

      bitreservoir = fMultI(slope, (INT)(BITRES_MAX_LD - BITRES_MIN_LD)) +

                     BITRES_MIN_LD;     /* interpolate */

      bitreservoir = bitreservoir & ~7; /* align to bytes */

    }

    int maxBitres;

    qcInit.averageBits = (averageBitsPerFrame + 7) & ~7;

    maxBitres =

        (MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff) - qcInit.averageBits;

    qcInit.bitRes =

        (bitreservoir != -1) ? fMin(bitreservoir, maxBitres) : maxBitres;

    qcInit.maxBits = fixMin(MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff,

                            ((averageBitsPerFrame + 7) & ~7) + qcInit.bitRes);

    qcInit.maxBits = (config->maxBitsPerFrame != -1)

                         ? fixMin(qcInit.maxBits, config->maxBitsPerFrame)

                         : qcInit.maxBits;

    qcInit.maxBits =

        fixMin(MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff,

               fixMax(qcInit.maxBits, (averageBitsPerFrame + 7 + 8) & ~7));

    qcInit.minBits = fixMax(

        0, ((averageBitsPerFrame - 1) & ~7) - qcInit.bitRes -

               transportEnc_GetStaticBits(

                   hTpEnc, ((averageBitsPerFrame + 7) & ~7) + qcInit.bitRes));

    qcInit.minBits = (config->minBitsPerFrame != -1)

                         ? fixMax(qcInit.minBits, config->minBitsPerFrame)

                         : qcInit.minBits;

    qcInit.minBits = fixMin(

        qcInit.minBits, (averageBitsPerFrame -

                         transportEnc_GetStaticBits(hTpEnc, qcInit.maxBits)) &

                            ~7);

  }

  qcInit.sampleRate = config->sampleRate;

  qcInit.isLowDelay = isLowDelay(config->audioObjectType) ? 1 : 0;

  qcInit.nSubFrames = config->nSubFrames;

  qcInit.padding.paddingRest = config->sampleRate;

  if (qcInit.maxBits - qcInit.averageBits >=

      ((qcInit.isLowDelay) ? BITRES_MIN_LD : BITRES_MIN) * config->nChannels) {

    qcInit.bitResMode = AACENC_BR_MODE_FULL; /* full bitreservoir */

  } else if (qcInit.maxBits > qcInit.averageBits) {

    qcInit.bitResMode = AACENC_BR_MODE_REDUCED; /* reduced bitreservoir */

  } else {

    qcInit.bitResMode = AACENC_BR_MODE_DISABLED; /* disabled bitreservoir */

  }

  /* Configure bitrate distribution strategy. */

  switch (config->channelMode) {

    case MODE_1_2:

    case MODE_1_2_1:

    case MODE_1_2_2:

    case MODE_1_2_2_1:

    case MODE_6_1:

    case MODE_1_2_2_2_1:

    case MODE_7_1_BACK:

    case MODE_7_1_TOP_FRONT:

    case MODE_7_1_REAR_SURROUND:

    case MODE_7_1_FRONT_CENTER:

      qcInit.bitDistributionMode = 0; /* over all elements bitrate estimation */

      break;

    case MODE_1:

    case MODE_2:

    default:                          /* all non mpeg defined channel modes */

      qcInit.bitDistributionMode = 1; /* element-wise bit bitrate estimation */

  }                                   /* config->channelMode */

  /* Calc meanPe: qcInit.meanPe = 10.0f * FRAME_LEN_LONG *

   * hAacEnc->bandwidth90dB/(config->sampleRate/2.0f); */

  bw_ratio =

      fDivNorm((FIXP_DBL)(10 * config->framelength * hAacEnc->bandwidth90dB),

               (FIXP_DBL)(config->sampleRate), &qbw);

  qcInit.meanPe =

      fMax((INT)scaleValue(bw_ratio, qbw + 1 - (DFRACT_BITS - 1)), 1);

  /* Calc maxBitFac, scale it to 24 bit accuracy */

  mbfac = fDivNorm(qcInit.maxBits, qcInit.averageBits / qcInit.nSubFrames,

                   &mbfac_e);

  qcInit.maxBitFac = scaleValue(mbfac, -(DFRACT_BITS - 1 - 24 - mbfac_e));

  switch (config->bitrateMode) {

    case AACENC_BR_MODE_CBR:

      qcInit.bitrateMode = QCDATA_BR_MODE_CBR;

      break;

    case AACENC_BR_MODE_VBR_1:

      qcInit.bitrateMode = QCDATA_BR_MODE_VBR_1;

      break;

    case AACENC_BR_MODE_VBR_2:

      qcInit.bitrateMode = QCDATA_BR_MODE_VBR_2;

      break;

    case AACENC_BR_MODE_VBR_3:

      qcInit.bitrateMode = QCDATA_BR_MODE_VBR_3;

      break;

    case AACENC_BR_MODE_VBR_4:

      qcInit.bitrateMode = QCDATA_BR_MODE_VBR_4;

      break;

    case AACENC_BR_MODE_VBR_5:

      qcInit.bitrateMode = QCDATA_BR_MODE_VBR_5;

      break;

    case AACENC_BR_MODE_SFR:

      qcInit.bitrateMode = QCDATA_BR_MODE_SFR;

      break;

    case AACENC_BR_MODE_FF:

      qcInit.bitrateMode = QCDATA_BR_MODE_FF;

      break;

    default:

      ErrorStatus = AAC_ENC_UNSUPPORTED_BITRATE_MODE;

      goto bail;

  }

  qcInit.invQuant = (config->useRequant) ? 2 : 0;

  /* maxIterations should be set to the maximum number of requantization

   * iterations that are allowed before the crash recovery functionality is

   * activated. This setting should be adjusted to the processing power

   * available, i.e. to the processing power headroom in one frame that is still

   * left after normal encoding without requantization. Please note that if

   * activated this functionality is used most likely only in cases where the

   * encoder is operating beyond recommended settings, i.e. the audio quality is

   * suboptimal anyway. Activating the crash recovery does not further reduce

   * audio quality significantly in these cases. */

  if (isLowDelay(config->audioObjectType)) {

    qcInit.maxIterations = 2;

  } else {

    qcInit.maxIterations = 5;

  }

  qcInit.bitrate = config->bitRate - config->ancDataBitRate;

  qcInit.staticBits = transportEnc_GetStaticBits(

      hTpEnc, qcInit.averageBits / qcInit.nSubFrames);

  ErrorStatus = FDKaacEnc_QCInit(hAacEnc->qcKernel, &qcInit, initFlags);

  if (ErrorStatus != AAC_ENC_OK) goto bail;

  /* Map virtual aot's to intern aot used in bitstream writer. */

  switch (hAacEnc->config->audioObjectType) {

    case AOT_MP2_AAC_LC:

      hAacEnc->aot = AOT_AAC_LC;

      break;

    case AOT_MP2_SBR:

      hAacEnc->aot = AOT_SBR;

      break;

    default:

      hAacEnc->aot = hAacEnc->config->audioObjectType;

  }

  /* common things */

  return AAC_ENC_OK;

bail:

  return ErrorStatus;

}

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

    functionname: FDKaacEnc_EncodeFrame

    description:  encodes one frame

    returns:      error code

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

AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame(

    HANDLE_AAC_ENC hAacEnc, /* encoder handle */

    HANDLE_TRANSPORTENC hTpEnc, INT_PCM *RESTRICT inputBuffer,

    const UINT inputBufferBufSize, INT *nOutBytes,

    AACENC_EXT_PAYLOAD extPayload[MAX_TOTAL_EXT_PAYLOADS]) {

  AAC_ENCODER_ERROR ErrorStatus;

  int el, n, c = 0;

  UCHAR extPayloadUsed[MAX_TOTAL_EXT_PAYLOADS];

  CHANNEL_MAPPING *cm = &hAacEnc->channelMapping;

  PSY_OUT *psyOut = hAacEnc->psyOut[c];

  QC_OUT *qcOut = hAacEnc->qcOut[c];

  FDKmemclear(extPayloadUsed, MAX_TOTAL_EXT_PAYLOADS * sizeof(UCHAR));

  qcOut->elementExtBits = 0; /* sum up all extended bit of each element */

  qcOut->staticBits = 0;     /* sum up side info bits of each element */

  qcOut->totalNoRedPe = 0;   /* sum up PE */

  /* advance psychoacoustics */

  for (el = 0; el < cm->nElements; el++) {

    ELEMENT_INFO elInfo = cm->elInfo[el];

    if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) ||

        (elInfo.elType == ID_LFE)) {

      int ch;

      /* update pointer!*/

      for (ch = 0; ch < elInfo.nChannelsInEl; ch++) {

        PSY_OUT_CHANNEL *psyOutChan =

            psyOut->psyOutElement[el]->psyOutChannel[ch];

        QC_OUT_CHANNEL *qcOutChan = qcOut->qcElement[el]->qcOutChannel[ch];

        psyOutChan->mdctSpectrum = qcOutChan->mdctSpectrum;

        psyOutChan->sfbSpreadEnergy = qcOutChan->sfbSpreadEnergy;

        psyOutChan->sfbEnergy = qcOutChan->sfbEnergy;

        psyOutChan->sfbEnergyLdData = qcOutChan->sfbEnergyLdData;

        psyOutChan->sfbMinSnrLdData = qcOutChan->sfbMinSnrLdData;

        psyOutChan->sfbThresholdLdData = qcOutChan->sfbThresholdLdData;

      }

      ErrorStatus = FDKaacEnc_psyMain(

          elInfo.nChannelsInEl, hAacEnc->psyKernel->psyElement[el],

          hAacEnc->psyKernel->psyDynamic, hAacEnc->psyKernel->psyConf,

          psyOut->psyOutElement[el], inputBuffer, inputBufferBufSize,

          cm->elInfo[el].ChannelIndex, cm->nChannels);

      if (ErrorStatus != AAC_ENC_OK) return ErrorStatus;

      /* FormFactor, Pe and staticBitDemand calculation */

      ErrorStatus = FDKaacEnc_QCMainPrepare(

          &elInfo, hAacEnc->qcKernel->hAdjThr->adjThrStateElem[el],

          psyOut->psyOutElement[el], qcOut->qcElement[el], hAacEnc->aot,

          hAacEnc->config->syntaxFlags, hAacEnc->config->epConfig);

      if (ErrorStatus != AAC_ENC_OK) return ErrorStatus;

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

      qcOut->qcElement[el]->extBitsUsed = 0;

      qcOut->qcElement[el]->nExtensions = 0;

      /* reset extension payload */

      FDKmemclear(&qcOut->qcElement[el]->extension,

                  (1) * sizeof(QC_OUT_EXTENSION));

      for (n = 0; n < MAX_TOTAL_EXT_PAYLOADS; n++) {

        if (!extPayloadUsed[n] && (extPayload[n].associatedChElement == el) &&

            (extPayload[n].dataSize > 0) && (extPayload[n].pData != NULL)) {

          int idx = qcOut->qcElement[el]->nExtensions++;

          qcOut->qcElement[el]->extension[idx].type =

              extPayload[n].dataType; /* Perform a sanity check on the type? */

          qcOut->qcElement[el]->extension[idx].nPayloadBits =

              extPayload[n].dataSize;

          qcOut->qcElement[el]->extension[idx].pPayload = extPayload[n].pData;

          /* Now ask the bitstream encoder how many bits we need to encode the

           * data with the current bitstream syntax: */

          qcOut->qcElement[el]->extBitsUsed += FDKaacEnc_writeExtensionData(

              NULL, &qcOut->qcElement[el]->extension[idx], 0, 0,

              hAacEnc->config->syntaxFlags, hAacEnc->aot,

              hAacEnc->config->epConfig);

          extPayloadUsed[n] = 1;

        }

      }

      /* sum up extension and static bits for all channel elements */

      qcOut->elementExtBits += qcOut->qcElement[el]->extBitsUsed;

      qcOut->staticBits += qcOut->qcElement[el]->staticBitsUsed;

      /* sum up pe */

      qcOut->totalNoRedPe += qcOut->qcElement[el]->peData.pe;

    }

  }

  qcOut->nExtensions = 0;

  qcOut->globalExtBits = 0;

  /* reset extension payload */

  FDKmemclear(&qcOut->extension, (2 + 2) * sizeof(QC_OUT_EXTENSION));

  /* Add extension payload not assigned to an channel element

    (Ancillary data is the only supported type up to now) */

  for (n = 0; n < MAX_TOTAL_EXT_PAYLOADS; n++) {

    if (!extPayloadUsed[n] && (extPayload[n].associatedChElement == -1) &&

        (extPayload[n].pData != NULL)) {

      UINT payloadBits = 0;

      if (extPayload[n].dataType == EXT_DATA_ELEMENT) {

        if (hAacEnc->ancillaryBitsPerFrame) {

          /* granted frame dse bitrate */

          payloadBits = hAacEnc->ancillaryBitsPerFrame;

        } else {

          /* write anc data if bitrate constraint fulfilled */

          if ((extPayload[n].dataSize >> 3) <=

              hAacEnc->config->maxAncBytesPerAU) {

            payloadBits = extPayload[n].dataSize;

          }

        }

        payloadBits = fixMin(extPayload[n].dataSize, payloadBits);

      } else {

        payloadBits = extPayload[n].dataSize;

      }

      if (payloadBits > 0) {

        int idx = qcOut->nExtensions++;

        qcOut->extension[idx].type =

            extPayload[n].dataType; /* Perform a sanity check on the type? */

        qcOut->extension[idx].nPayloadBits = payloadBits;

        qcOut->extension[idx].pPayload = extPayload[n].pData;

        /* Now ask the bitstream encoder how many bits we need to encode the

         * data with the current bitstream syntax: */

        qcOut->globalExtBits += FDKaacEnc_writeExtensionData(

            NULL, &qcOut->extension[idx], 0, 0, hAacEnc->config->syntaxFlags,

            hAacEnc->aot, hAacEnc->config->epConfig);

        if (extPayload[n].dataType == EXT_DATA_ELEMENT) {

          /* substract the processed bits */

          extPayload[n].dataSize -= payloadBits;

        }