/*
  * QCELP decoder
  * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */

 /**

  * @file

  * QCELP decoder
  * @author Reynaldo H. Verdejo Pinochet

  * @remark FFmpeg merging spearheaded by Kenan Gillet

  * @remark Development mentored by Benjamin Larson

  */
 
 #include <stddef.h>
 
 #include "avcodec.h"

 #include "internal.h"

 #include "get_bits.h"

 
 #include "qcelpdata.h"
 
 #include "celp_math.h"
 #include "celp_filters.h"

 #include "acelp_filters.h"

 #include "acelp_vectors.h"

 #include "lsp.h"

 
 #undef NDEBUG
 #include <assert.h>
 

 typedef enum
 {
     I_F_Q = -1,    /*!< insufficient frame quality */
     SILENCE,
     RATE_OCTAVE,
     RATE_QUARTER,
     RATE_HALF,
     RATE_FULL
 } qcelp_packet_rate;
 

 typedef struct
 {

     GetBitContext     gb;
     qcelp_packet_rate bitrate;

     QCELPFrame        frame;    /*!< unpacked data frame */
 
     uint8_t  erasure_count;
     uint8_t  octave_count;      /*!< count the consecutive RATE_OCTAVE frames */
     float    prev_lspf[10];

     float    predictor_lspf[10];/*!< LSP predictor for RATE_OCTAVE and I_F_Q */

     float    pitch_synthesis_filter_mem[303];
     float    pitch_pre_filter_mem[303];
     float    rnd_fir_filter_mem[180];
     float    formant_mem[170];
     float    last_codebook_gain;
     int      prev_g1[2];
     int      prev_bitrate;
     float    pitch_gain[4];
     uint8_t  pitch_lag[4];
     uint16_t first16bits;

     uint8_t  warned_buf_mismatch_bitrate;

 
     /* postfilter */
     float    postfilter_synth_mem[10];
     float    postfilter_agc_mem;
     float    postfilter_tilt_mem;

 } QCELPContext;
 

 /**

  * Initialize the speech codec according to the specification.
  *
  * TIA/EIA/IS-733 2.4.9
  */

 static av_cold int qcelp_decode_init(AVCodecContext *avctx)
 {

     QCELPContext *q = avctx->priv_data;
     int i;
 

     avctx->sample_fmt = AV_SAMPLE_FMT_FLT;

     for(i=0; i<10; i++)
         q->prev_lspf[i] = (i+1)/11.;

 
     return 0;
 }
 
 /**

  * Decode the 10 quantized LSP frequencies from the LSPV/LSP
  * transmission codes of any bitrate and check for badly received packets.

  *
  * @param q the context
  * @param lspf line spectral pair frequencies
  *
  * @return 0 on success, -1 if the packet is badly received
  *
  * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
  */

 static int decode_lspf(QCELPContext *q, float *lspf)
 {

     int i;

     float tmp_lspf, smooth, erasure_coeff;
     const float *predictors;

     if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q)
     {

         predictors = (q->prev_bitrate != RATE_OCTAVE &&
                        q->prev_bitrate != I_F_Q ?
                        q->prev_lspf : q->predictor_lspf);

         if(q->bitrate == RATE_OCTAVE)
         {

             q->octave_count++;
 

             for(i=0; i<10; i++)
             {

                 q->predictor_lspf[i] =

                              lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
                                                          : -QCELP_LSP_SPREAD_FACTOR)

                                      + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR
                                      + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
             }
             smooth = (q->octave_count < 10 ? .875 : 0.1);

         }else
         {

             erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;

 
             assert(q->bitrate == I_F_Q);
 

             if(q->erasure_count > 1)

                 erasure_coeff *= (q->erasure_count < 4 ? 0.9 : 0.7);
 

             for(i=0; i<10; i++)
             {

                 q->predictor_lspf[i] =
                              lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
                                      + erasure_coeff * predictors[i];
             }
             smooth = 0.125;
         }
 
         // Check the stability of the LSP frequencies.
         lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);

         for(i=1; i<10; i++)

             lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR));
 
         lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR));

         for(i=9; i>0; i--)

             lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
 
         // Low-pass filter the LSP frequencies.

         ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);

     }else
     {

         q->octave_count = 0;
 
         tmp_lspf = 0.;

         for(i=0; i<5 ; i++)
         {

             lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
             lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;

         }
 
         // Check for badly received packets.

         if(q->bitrate == RATE_QUARTER)
         {
             if(lspf[9] <= .70 || lspf[9] >=  .97)

                 return -1;

             for(i=3; i<10; i++)
                 if(fabs(lspf[i] - lspf[i-2]) < .08)

                     return -1;

         }else
         {
             if(lspf[9] <= .66 || lspf[9] >= .985)

                 return -1;

             for(i=4; i<10; i++)

                 if (fabs(lspf[i] - lspf[i-4]) < .0931)
                     return -1;
         }
     }
     return 0;
 }
 
 /**

  * Convert codebook transmission codes to GAIN and INDEX.

  *
  * @param q the context
  * @param gain array holding the decoded gain
  *
  * TIA/EIA/IS-733 2.4.6.2
  */
 static void decode_gain_and_index(QCELPContext  *q,
                                   float *gain) {
     int   i, subframes_count, g1[16];
     float slope;
 

     if(q->bitrate >= RATE_QUARTER)
     {
         switch(q->bitrate)
         {

             case RATE_FULL: subframes_count = 16; break;
             case RATE_HALF: subframes_count = 4;  break;
             default:        subframes_count = 5;
         }

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

             g1[i] = 4 * q->frame.cbgain[i];

             if(q->bitrate == RATE_FULL && !((i+1) & 3))
             {

                 g1[i] += av_clip((g1[i-1] + g1[i-2] + g1[i-3]) / 3 - 6, 0, 32);
             }
 
             gain[i] = qcelp_g12ga[g1[i]];
 

             if(q->frame.cbsign[i])
             {

                 gain[i] = -gain[i];
                 q->frame.cindex[i] = (q->frame.cindex[i]-89) & 127;
             }
         }
 
         q->prev_g1[0] = g1[i-2];
         q->prev_g1[1] = g1[i-1];
         q->last_codebook_gain = qcelp_g12ga[g1[i-1]];
 

         if(q->bitrate == RATE_QUARTER)
         {

             // Provide smoothing of the unvoiced excitation energy.
             gain[7] =     gain[4];
             gain[6] = 0.4*gain[3] + 0.6*gain[4];
             gain[5] =     gain[3];
             gain[4] = 0.8*gain[2] + 0.2*gain[3];
             gain[3] = 0.2*gain[1] + 0.8*gain[2];
             gain[2] =     gain[1];
             gain[1] = 0.6*gain[0] + 0.4*gain[1];
         }

     }else if (q->bitrate != SILENCE)

     {
         if(q->bitrate == RATE_OCTAVE)
         {

             g1[0] = 2 * q->frame.cbgain[0]
                   + av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
             subframes_count = 8;

         }else
         {

             assert(q->bitrate == I_F_Q);
 
             g1[0] = q->prev_g1[1];

             switch(q->erasure_count)
             {
                 case 1 : break;
                 case 2 : g1[0] -= 1; break;
                 case 3 : g1[0] -= 2; break;
                 default: g1[0] -= 6;

             }

             if(g1[0] < 0)

                 g1[0] = 0;
             subframes_count = 4;
         }
         // This interpolation is done to produce smoother background noise.
         slope = 0.5*(qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;

         for(i=1; i<=subframes_count; i++)

             gain[i-1] = q->last_codebook_gain + slope * i;
 

         q->last_codebook_gain = gain[i-2];

         q->prev_g1[0] = q->prev_g1[1];
         q->prev_g1[1] = g1[0];
     }
 }
 
 /**

  * If the received packet is Rate 1/4 a further sanity check is made of the
  * codebook gain.

  *
  * @param cbgain the unpacked cbgain array
  * @return -1 if the sanity check fails, 0 otherwise
  *
  * TIA/EIA/IS-733 2.4.8.7.3
  */

 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
 {

     int i, diff, prev_diff=0;

     for(i=1; i<5; i++)
     {

         diff = cbgain[i] - cbgain[i-1];

         if(FFABS(diff) > 10)
             return -1;
         else if(FFABS(diff - prev_diff) > 12)
             return -1;
         prev_diff = diff;

     }
     return 0;

 }
 
 /**

  * Compute the scaled codebook vector Cdn From INDEX and GAIN

  * for all rates.
  *
  * The specification lacks some information here.
  *
  * TIA/EIA/IS-733 has an omission on the codebook index determination
  * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
  * you have to subtract the decoded index parameter from the given scaled
  * codebook vector index 'n' to get the desired circular codebook index, but
  * it does not mention that you have to clamp 'n' to [0-9] in order to get
  * RI-compliant results.
  *
  * The reason for this mistake seems to be the fact they forgot to mention you
  * have to do these calculations per codebook subframe and adjust given
  * equation values accordingly.
  *
  * @param q the context
  * @param gain array holding the 4 pitch subframe gain values
  * @param cdn_vector array for the generated scaled codebook vector
  */

 static void compute_svector(QCELPContext *q, const float *gain,

                             float *cdn_vector)
 {

     int      i, j, k;
     uint16_t cbseed, cindex;
     float    *rnd, tmp_gain, fir_filter_value;
 

     switch(q->bitrate)
     {
         case RATE_FULL:
             for(i=0; i<16; i++)
             {
                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;

                 cindex = -q->frame.cindex[i];

                 for(j=0; j<10; j++)
                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
             }

         break;

         case RATE_HALF:
             for(i=0; i<4; i++)
             {
                 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;

                 cindex = -q->frame.cindex[i];

                 for (j = 0; j < 40; j++)

                 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];

             }

         break;

         case RATE_QUARTER:

             cbseed = (0x0003 & q->frame.lspv[4])<<14 |
                      (0x003F & q->frame.lspv[3])<< 8 |
                      (0x0060 & q->frame.lspv[2])<< 1 |
                      (0x0007 & q->frame.lspv[1])<< 3 |
                      (0x0038 & q->frame.lspv[0])>> 3 ;

             rnd = q->rnd_fir_filter_mem + 20;
             for(i=0; i<8; i++)
             {
                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
                 for(k=0; k<20; k++)
                 {
                     cbseed = 521 * cbseed + 259;
                     *rnd = (int16_t)cbseed;
 
                     // FIR filter
                     fir_filter_value = 0.0;
                     for(j=0; j<10; j++)
                         fir_filter_value += qcelp_rnd_fir_coefs[j ]
                                           * (rnd[-j ] + rnd[-20+j]);
 
                     fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
                     *cdn_vector++ = tmp_gain * fir_filter_value;
                     rnd++;
                 }

             }

             memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));

         break;

         case RATE_OCTAVE:
             cbseed = q->first16bits;
             for(i=0; i<8; i++)
             {
                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
                 for(j=0; j<20; j++)
                 {
                     cbseed = 521 * cbseed + 259;
                     *cdn_vector++ = tmp_gain * (int16_t)cbseed;
                 }

             }
         break;

         case I_F_Q:
             cbseed = -44; // random codebook index
             for(i=0; i<4; i++)
             {
                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
                 for(j=0; j<40; j++)
                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
             }

         break;

         case SILENCE:
             memset(cdn_vector, 0, 160 * sizeof(float));
         break;

     }
 }
 
 /**

  * Apply generic gain control.
  *
  * @param v_out output vector
  * @param v_in gain-controlled vector
  * @param v_ref vector to control gain of
  *
  * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
  */

 static void apply_gain_ctrl(float *v_out, const float *v_ref,
                             const float *v_in)
 {

     int i;

     for (i = 0; i < 160; i += 40)
         ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
                                                 ff_dot_productf(v_ref + i,
                                                                 v_ref + i, 40),
                                                 40);

 }
 
 /**

  * Apply filter in pitch-subframe steps.
  *
  * @param memory buffer for the previous state of the filter
  *        - must be able to contain 303 elements
  *        - the 143 first elements are from the previous state
  *        - the next 160 are for output
  * @param v_in input filter vector
  * @param gain per-subframe gain array, each element is between 0.0 and 2.0
  * @param lag per-subframe lag array, each element is
  *        - between 16 and 143 if its corresponding pfrac is 0,
  *        - between 16 and 139 otherwise

  * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
  *        otherwise

  *
  * @return filter output vector
  */

 static const float *do_pitchfilter(float memory[303], const float v_in[160],
                                    const float gain[4], const uint8_t *lag,
                                    const uint8_t pfrac[4])
 {

     int         i, j;
     float       *v_lag, *v_out;
     const float *v_len;
 
     v_out = memory + 143; // Output vector starts at memory[143].
 

     for(i=0; i<4; i++)
     {
         if(gain[i])
         {

             v_lag = memory + 143 + 40 * i - lag[i];

             for(v_len=v_in+40; v_in<v_len; v_in++)
             {
                 if(pfrac[i]) // If it is a fractional lag...
                 {
                     for(j=0, *v_out=0.; j<4; j++)

                         *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);

                 }else

                     *v_out = *v_lag;
 
                 *v_out = *v_in + gain[i] * *v_out;
 
                 v_lag++;
                 v_out++;
             }

         }else
         {

             memcpy(v_out, v_in, 40 * sizeof(float));
             v_in  += 40;
             v_out += 40;
         }

     }

 
     memmove(memory, memory + 160, 143 * sizeof(float));
     return memory + 143;
 }
 

 /**

  * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.

  * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2

  *
  * @param q the context
  * @param cdn_vector the scaled codebook vector
  */

 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
 {

     int         i;
     const float *v_synthesis_filtered, *v_pre_filtered;
 

     if(q->bitrate >= RATE_HALF ||

        q->bitrate == SILENCE ||

        (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF)))
     {

         if(q->bitrate >= RATE_HALF)
         {

 
             // Compute gain & lag for the whole frame.

             for(i=0; i<4; i++)
             {

                 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
 
                 q->pitch_lag[i] = q->frame.plag[i] + 16;
             }

         }else
         {

             float max_pitch_gain;
 

             if (q->bitrate == I_F_Q)
             {

                   if (q->erasure_count < 3)
                       max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
                   else
                       max_pitch_gain = 0.0;

             }else
             {
                 assert(q->bitrate == SILENCE);
                 max_pitch_gain = 1.0;
             }

             for(i=0; i<4; i++)

                 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
 
             memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
         }
 
         // pitch synthesis filter

         v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
                                               cdn_vector, q->pitch_gain,
                                               q->pitch_lag, q->frame.pfrac);

 
         // pitch prefilter update

         for(i=0; i<4; i++)

             q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
 

         v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
                                         v_synthesis_filtered,
                                         q->pitch_gain, q->pitch_lag,
                                         q->frame.pfrac);

 
         apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);

     }else
     {
         memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17,
                143 * sizeof(float));
         memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));

         memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
         memset(q->pitch_lag,  0, sizeof(q->pitch_lag));
     }
 }
 
 /**

  * Reconstruct LPC coefficients from the line spectral pair frequencies
  * and perform bandwidth expansion.

  *
  * @param lspf line spectral pair frequencies
  * @param lpc linear predictive coding coefficients
  *

  * @note: bandwidth_expansion_coeff could be precalculated into a table

  *        but it seems to be slower on x86
  *
  * TIA/EIA/IS-733 2.4.3.3.5
  */

 static void lspf2lpc(const float *lspf, float *lpc)

 {

     double lsp[10];

     double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;

     int   i;
 
     for (i=0; i<10; i++)

         lsp[i] = cos(M_PI * lspf[i]);

     ff_acelp_lspd2lpc(lsp, lpc, 5);

 
     for (i=0; i<10; i++)
     {

         lpc[i] *= bandwidth_expansion_coeff;
         bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;

     }
 }
 
 /**

  * Interpolate LSP frequencies and compute LPC coefficients

  * for a given bitrate & pitch subframe.

  *

  * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2

  *
  * @param q the context
  * @param curr_lspf LSP frequencies vector of the current frame
  * @param lpc float vector for the resulting LPC
  * @param subframe_num frame number in decoded stream
  */

 static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
                             float *lpc, const int subframe_num)

 {

     float interpolated_lspf[10];
     float weight;
 

     if(q->bitrate >= RATE_QUARTER)

         weight = 0.25 * (subframe_num + 1);

     else if(q->bitrate == RATE_OCTAVE && !subframe_num)

         weight = 0.625;

     else

         weight = 1.0;
 

     if(weight != 1.0)
     {

         ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,

                                 weight, 1.0 - weight, 10);

         lspf2lpc(interpolated_lspf, lpc);

     }else if(q->bitrate >= RATE_QUARTER ||
              (q->bitrate == I_F_Q && !subframe_num))

         lspf2lpc(curr_lspf, lpc);

     else if(q->bitrate == SILENCE && !subframe_num)

         lspf2lpc(q->prev_lspf, lpc);

 }
 

 static qcelp_packet_rate buf_size2bitrate(const int buf_size)

 {
     switch(buf_size)
     {

         case 35: return RATE_FULL;
         case 17: return RATE_HALF;
         case  8: return RATE_QUARTER;
         case  4: return RATE_OCTAVE;
         case  1: return SILENCE;

     }

     return I_F_Q;

 }
 

 /**
  * Determine the bitrate from the frame size and/or the first byte of the frame.
  *
  * @param avctx the AV codec context
  * @param buf_size length of the buffer
  * @param buf the bufffer
  *
  * @return the bitrate on success,
  *         I_F_Q  if the bitrate cannot be satisfactorily determined
  *
  * TIA/EIA/IS-733 2.4.8.7.1
  */

 static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx, const int buf_size,

                              const uint8_t **buf)

 {

     qcelp_packet_rate bitrate;
 

     if((bitrate = buf_size2bitrate(buf_size)) >= 0)
     {
         if(bitrate > **buf)
         {

             QCELPContext *q = avctx->priv_data;
             if (!q->warned_buf_mismatch_bitrate)
             {

             av_log(avctx, AV_LOG_WARNING,
                    "Claimed bitrate and buffer size mismatch.\n");

                 q->warned_buf_mismatch_bitrate = 1;
             }

             bitrate = **buf;

         }else if(bitrate < **buf)
         {
             av_log(avctx, AV_LOG_ERROR,
                    "Buffer is too small for the claimed bitrate.\n");

             return I_F_Q;
         }
         (*buf)++;

     }else if((bitrate = buf_size2bitrate(buf_size + 1)) >= 0)
     {

         av_log(avctx, AV_LOG_WARNING,
                "Bitrate byte is missing, guessing the bitrate from packet size.\n");

     }else

         return I_F_Q;
 

     if(bitrate == SILENCE)
     {

         //FIXME: Remove experimental warning when tested with samples.

         av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");

     }
     return bitrate;
 }
 

 static void warn_insufficient_frame_quality(AVCodecContext *avctx,

                                             const char *message)
 {
     av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
            message);

 }

 static void postfilter(QCELPContext *q, float *samples, float *lpc)
 {
     static const float pow_0_775[10] = {
         0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
         0.216676, 0.167924, 0.130141, 0.100859, 0.078166
     }, pow_0_625[10] = {
         0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
         0.059605, 0.037253, 0.023283, 0.014552, 0.009095
     };
     float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
     int n;
 
     for (n = 0; n < 10; n++) {
         lpc_s[n] = lpc[n] * pow_0_625[n];
         lpc_p[n] = lpc[n] * pow_0_775[n];
     }
 
     ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
                                       q->formant_mem + 10, 160, 10);
     memcpy(pole_out, q->postfilter_synth_mem,       sizeof(float) * 10);
     ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
     memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
 
     ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
 
     ff_adaptive_gain_control(samples, pole_out + 10,
         ff_dot_productf(q->formant_mem + 10, q->formant_mem + 10, 160),
         160, 0.9375, &q->postfilter_agc_mem);
 }
 

 static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,

                               AVPacket *avpkt)

 {

     const uint8_t *buf = avpkt->data;
     int buf_size = avpkt->size;

     QCELPContext *q = avctx->priv_data;
     float *outbuffer = data;

     int   i, out_size;

     float quantized_lspf[10], lpc[10];
     float gain[16];
     float *formant_mem;
 

     out_size = 160 * av_get_bytes_per_sample(avctx->sample_fmt);
     if (*data_size < out_size) {
         av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
         return AVERROR(EINVAL);
     }
 

     if((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q)
     {

         warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
         goto erasure;
     }
 

     if(q->bitrate == RATE_OCTAVE &&
        (q->first16bits = AV_RB16(buf)) == 0xFFFF)
     {

         warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
         goto erasure;
     }
 

     if(q->bitrate > SILENCE)
     {

         const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
         const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate]
                                        + qcelp_unpacking_bitmaps_lengths[q->bitrate];
         uint8_t           *unpacked_data = (uint8_t *)&q->frame;
 
         init_get_bits(&q->gb, buf, 8*buf_size);
 
         memset(&q->frame, 0, sizeof(QCELPFrame));
 

         for(; bitmaps < bitmaps_end; bitmaps++)

             unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
 
         // Check for erasures/blanks on rates 1, 1/4 and 1/8.

         if(q->frame.reserved)
         {

             warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
             goto erasure;
         }

         if(q->bitrate == RATE_QUARTER &&
            codebook_sanity_check_for_rate_quarter(q->frame.cbgain))
         {

             warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
             goto erasure;
         }
 

         if(q->bitrate >= RATE_HALF)
         {
             for(i=0; i<4; i++)
             {
                 if(q->frame.pfrac[i] && q->frame.plag[i] >= 124)
                 {

                     warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
                     goto erasure;
                 }
             }
         }
     }
 
     decode_gain_and_index(q, gain);
     compute_svector(q, gain, outbuffer);
 

     if(decode_lspf(q, quantized_lspf) < 0)
     {

         warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
         goto erasure;
     }
 
 
     apply_pitch_filters(q, outbuffer);
 

     if(q->bitrate == I_F_Q)
     {

 erasure:
         q->bitrate = I_F_Q;
         q->erasure_count++;
         decode_gain_and_index(q, gain);
         compute_svector(q, gain, outbuffer);
         decode_lspf(q, quantized_lspf);
         apply_pitch_filters(q, outbuffer);

     }else

         q->erasure_count = 0;
 
     formant_mem = q->formant_mem + 10;

     for(i=0; i<4; i++)
     {

         interpolate_lpc(q, quantized_lspf, lpc, i);

         ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40,
                                      10);

         formant_mem += 40;
     }
 

     // postfilter, as per TIA/EIA/IS-733 2.4.8.6
     postfilter(q, outbuffer, lpc);
 
     memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));

 
     memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
     q->prev_bitrate = q->bitrate;
 

     *data_size = out_size;

     return buf_size;

 }
 

 AVCodec ff_qcelp_decoder =

 {
     .name   = "qcelp",

     .type   = AVMEDIA_TYPE_AUDIO,

     .id     = CODEC_ID_QCELP,
     .init   = qcelp_decode_init,
     .decode = qcelp_decode_frame,
     .priv_data_size = sizeof(QCELPContext),
     .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
 };