/*
  * G.723.1 compatible decoder
  * Copyright (c) 2006 Benjamin Larsson
  * Copyright (c) 2010 Mohamed Naufal Basheer
  *
  * 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
  * G.723.1 compatible decoder
  */
 

 #define BITSTREAM_READER_LE

 #include "libavutil/channel_layout.h"

 #include "libavutil/mem.h"

 #include "libavutil/opt.h"
 #include "avcodec.h"

 #include "get_bits.h"
 #include "acelp_vectors.h"
 #include "celp_filters.h"
 #include "celp_math.h"

 #include "g723_1.h"

 #include "internal.h"

 #define CNG_RANDOM_SEED 12345
 

 static av_cold int g723_1_decode_init(AVCodecContext *avctx)
 {

     G723_1_Context *p = avctx->priv_data;

     avctx->channel_layout = AV_CH_LAYOUT_MONO;
     avctx->sample_fmt     = AV_SAMPLE_FMT_S16;
     avctx->channels       = 1;
     p->pf_gain            = 1 << 12;

     memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(*p->prev_lsp));

     memcpy(p->sid_lsp,  dc_lsp, LPC_ORDER * sizeof(*p->sid_lsp));
 
     p->cng_random_seed = CNG_RANDOM_SEED;
     p->past_frame_type = SID_FRAME;

     return 0;
 }
 
 /**
  * Unpack the frame into parameters.
  *
  * @param p           the context
  * @param buf         pointer to the input buffer
  * @param buf_size    size of the input buffer
  */
 static int unpack_bitstream(G723_1_Context *p, const uint8_t *buf,
                             int buf_size)
 {
     GetBitContext gb;
     int ad_cb_len;
     int temp, info_bits, i;
 
     init_get_bits(&gb, buf, buf_size * 8);
 
     /* Extract frame type and rate info */
     info_bits = get_bits(&gb, 2);
 
     if (info_bits == 3) {

         p->cur_frame_type = UNTRANSMITTED_FRAME;

         return 0;
     }
 
     /* Extract 24 bit lsp indices, 8 bit for each band */
     p->lsp_index[2] = get_bits(&gb, 8);
     p->lsp_index[1] = get_bits(&gb, 8);
     p->lsp_index[0] = get_bits(&gb, 8);
 
     if (info_bits == 2) {

         p->cur_frame_type = SID_FRAME;

         p->subframe[0].amp_index = get_bits(&gb, 6);
         return 0;
     }
 
     /* Extract the info common to both rates */

     p->cur_rate       = info_bits ? RATE_5300 : RATE_6300;
     p->cur_frame_type = ACTIVE_FRAME;

 
     p->pitch_lag[0] = get_bits(&gb, 7);
     if (p->pitch_lag[0] > 123)       /* test if forbidden code */
         return -1;
     p->pitch_lag[0] += PITCH_MIN;
     p->subframe[1].ad_cb_lag = get_bits(&gb, 2);
 
     p->pitch_lag[1] = get_bits(&gb, 7);
     if (p->pitch_lag[1] > 123)
         return -1;
     p->pitch_lag[1] += PITCH_MIN;
     p->subframe[3].ad_cb_lag = get_bits(&gb, 2);
     p->subframe[0].ad_cb_lag = 1;
     p->subframe[2].ad_cb_lag = 1;
 
     for (i = 0; i < SUBFRAMES; i++) {
         /* Extract combined gain */
         temp = get_bits(&gb, 12);
         ad_cb_len = 170;
         p->subframe[i].dirac_train = 0;

         if (p->cur_rate == RATE_6300 && p->pitch_lag[i >> 1] < SUBFRAME_LEN - 2) {

             p->subframe[i].dirac_train = temp >> 11;

             temp &= 0x7FF;

             ad_cb_len = 85;
         }
         p->subframe[i].ad_cb_gain = FASTDIV(temp, GAIN_LEVELS);
         if (p->subframe[i].ad_cb_gain < ad_cb_len) {
             p->subframe[i].amp_index = temp - p->subframe[i].ad_cb_gain *
                                        GAIN_LEVELS;
         } else {
             return -1;
         }
     }
 

     p->subframe[0].grid_index = get_bits1(&gb);
     p->subframe[1].grid_index = get_bits1(&gb);
     p->subframe[2].grid_index = get_bits1(&gb);
     p->subframe[3].grid_index = get_bits1(&gb);

     if (p->cur_rate == RATE_6300) {

         skip_bits1(&gb);  /* skip reserved bit */

 
         /* Compute pulse_pos index using the 13-bit combined position index */
         temp = get_bits(&gb, 13);
         p->subframe[0].pulse_pos = temp / 810;
 
         temp -= p->subframe[0].pulse_pos * 810;
         p->subframe[1].pulse_pos = FASTDIV(temp, 90);
 
         temp -= p->subframe[1].pulse_pos * 90;
         p->subframe[2].pulse_pos = FASTDIV(temp, 9);
         p->subframe[3].pulse_pos = temp - p->subframe[2].pulse_pos * 9;
 
         p->subframe[0].pulse_pos = (p->subframe[0].pulse_pos << 16) +
                                    get_bits(&gb, 16);
         p->subframe[1].pulse_pos = (p->subframe[1].pulse_pos << 14) +
                                    get_bits(&gb, 14);
         p->subframe[2].pulse_pos = (p->subframe[2].pulse_pos << 16) +
                                    get_bits(&gb, 16);
         p->subframe[3].pulse_pos = (p->subframe[3].pulse_pos << 14) +
                                    get_bits(&gb, 14);
 
         p->subframe[0].pulse_sign = get_bits(&gb, 6);
         p->subframe[1].pulse_sign = get_bits(&gb, 5);
         p->subframe[2].pulse_sign = get_bits(&gb, 6);
         p->subframe[3].pulse_sign = get_bits(&gb, 5);

     } else { /* 5300 bps */

         p->subframe[0].pulse_pos  = get_bits(&gb, 12);
         p->subframe[1].pulse_pos  = get_bits(&gb, 12);
         p->subframe[2].pulse_pos  = get_bits(&gb, 12);
         p->subframe[3].pulse_pos  = get_bits(&gb, 12);
 
         p->subframe[0].pulse_sign = get_bits(&gb, 4);
         p->subframe[1].pulse_sign = get_bits(&gb, 4);
         p->subframe[2].pulse_sign = get_bits(&gb, 4);
         p->subframe[3].pulse_sign = get_bits(&gb, 4);
     }
 
     return 0;
 }
 
 /**
  * Bitexact implementation of sqrt(val/2).
  */

 static int16_t square_root(unsigned val)

 {

     av_assert2(!(val & 0x80000000));
 

     return (ff_sqrt(val << 1) >> 1) & (~1);

 }
 
 /**
  * Generate fixed codebook excitation vector.
  *
  * @param vector    decoded excitation vector
  * @param subfrm    current subframe
  * @param cur_rate  current bitrate
  * @param pitch_lag closed loop pitch lag
  * @param index     current subframe index
  */

 static void gen_fcb_excitation(int16_t *vector, G723_1_Subframe *subfrm,

                                enum Rate cur_rate, int pitch_lag, int index)

 {
     int temp, i, j;
 

     memset(vector, 0, SUBFRAME_LEN * sizeof(*vector));

     if (cur_rate == RATE_6300) {

         if (subfrm->pulse_pos >= max_pos[index])

             return;
 
         /* Decode amplitudes and positions */
         j = PULSE_MAX - pulses[index];

         temp = subfrm->pulse_pos;

         for (i = 0; i < SUBFRAME_LEN / GRID_SIZE; i++) {
             temp -= combinatorial_table[j][i];
             if (temp >= 0)
                 continue;
             temp += combinatorial_table[j++][i];

             if (subfrm->pulse_sign & (1 << (PULSE_MAX - j))) {
                 vector[subfrm->grid_index + GRID_SIZE * i] =
                                         -fixed_cb_gain[subfrm->amp_index];

             } else {

                 vector[subfrm->grid_index + GRID_SIZE * i] =
                                          fixed_cb_gain[subfrm->amp_index];

             }
             if (j == PULSE_MAX)
                 break;
         }

         if (subfrm->dirac_train == 1)

             ff_g723_1_gen_dirac_train(vector, pitch_lag);

     } else { /* 5300 bps */

         int cb_gain  = fixed_cb_gain[subfrm->amp_index];
         int cb_shift = subfrm->grid_index;
         int cb_sign  = subfrm->pulse_sign;
         int cb_pos   = subfrm->pulse_pos;

         int offset, beta, lag;
 
         for (i = 0; i < 8; i += 2) {
             offset         = ((cb_pos & 7) << 3) + cb_shift + i;
             vector[offset] = (cb_sign & 1) ? cb_gain : -cb_gain;
             cb_pos  >>= 3;
             cb_sign >>= 1;
         }
 
         /* Enhance harmonic components */

         lag  = pitch_contrib[subfrm->ad_cb_gain << 1] + pitch_lag +
                subfrm->ad_cb_lag - 1;
         beta = pitch_contrib[(subfrm->ad_cb_gain << 1) + 1];

 
         if (lag < SUBFRAME_LEN - 2) {
             for (i = lag; i < SUBFRAME_LEN; i++)
                 vector[i] += beta * vector[i - lag] >> 15;
         }
     }
 }
 
 /**
  * Estimate maximum auto-correlation around pitch lag.
  *

  * @param buf       buffer with offset applied

  * @param offset    offset of the excitation vector
  * @param ccr_max   pointer to the maximum auto-correlation
  * @param pitch_lag decoded pitch lag
  * @param length    length of autocorrelation
  * @param dir       forward lag(1) / backward lag(-1)
  */

 static int autocorr_max(const int16_t *buf, int offset, int *ccr_max,

                         int pitch_lag, int length, int dir)
 {
     int limit, ccr, lag = 0;
     int i;
 
     pitch_lag = FFMIN(PITCH_MAX - 3, pitch_lag);

     if (dir > 0)
         limit = FFMIN(FRAME_LEN + PITCH_MAX - offset - length, pitch_lag + 3);
     else
         limit = pitch_lag + 3;

 
     for (i = pitch_lag - 3; i <= limit; i++) {

         ccr = ff_g723_1_dot_product(buf, buf + dir * i, length);

 
         if (ccr > *ccr_max) {
             *ccr_max = ccr;
             lag = i;
         }
     }
     return lag;
 }
 
 /**
  * Calculate pitch postfilter optimal and scaling gains.
  *
  * @param lag      pitch postfilter forward/backward lag
  * @param ppf      pitch postfilter parameters
  * @param cur_rate current bitrate
  * @param tgt_eng  target energy
  * @param ccr      cross-correlation
  * @param res_eng  residual energy
  */

 static void comp_ppf_gains(int lag, PPFParam *ppf, enum Rate cur_rate,

                            int tgt_eng, int ccr, int res_eng)
 {
     int pf_residual;     /* square of postfiltered residual */

     int temp1, temp2;

 
     ppf->index = lag;
 
     temp1 = tgt_eng * res_eng >> 1;
     temp2 = ccr * ccr << 1;
 
     if (temp2 > temp1) {
         if (ccr >= res_eng) {
             ppf->opt_gain = ppf_gain_weight[cur_rate];
         } else {
             ppf->opt_gain = (ccr << 15) / res_eng *
                             ppf_gain_weight[cur_rate] >> 15;
         }
         /* pf_res^2 = tgt_eng + 2*ccr*gain + res_eng*gain^2 */
         temp1       = (tgt_eng << 15) + (ccr * ppf->opt_gain << 1);
         temp2       = (ppf->opt_gain * ppf->opt_gain >> 15) * res_eng;

         pf_residual = av_sat_add32(temp1, temp2 + (1 << 15)) >> 16;

 
         if (tgt_eng >= pf_residual << 1) {
             temp1 = 0x7fff;
         } else {
             temp1 = (tgt_eng << 14) / pf_residual;
         }
 
         /* scaling_gain = sqrt(tgt_eng/pf_res^2) */
         ppf->sc_gain = square_root(temp1 << 16);
     } else {
         ppf->opt_gain = 0;
         ppf->sc_gain  = 0x7fff;
     }
 
     ppf->opt_gain = av_clip_int16(ppf->opt_gain * ppf->sc_gain >> 15);
 }
 
 /**
  * Calculate pitch postfilter parameters.
  *
  * @param p         the context
  * @param offset    offset of the excitation vector
  * @param pitch_lag decoded pitch lag
  * @param ppf       pitch postfilter parameters
  * @param cur_rate  current bitrate
  */
 static void comp_ppf_coeff(G723_1_Context *p, int offset, int pitch_lag,

                            PPFParam *ppf, enum Rate cur_rate)

 {
 
     int16_t scale;
     int i;

     int temp1, temp2;

 
     /*
      * 0 - target energy
      * 1 - forward cross-correlation
      * 2 - forward residual energy
      * 3 - backward cross-correlation
      * 4 - backward residual energy
      */
     int energy[5] = {0, 0, 0, 0, 0};

     int16_t *buf  = p->audio + LPC_ORDER + offset;

     int fwd_lag   = autocorr_max(buf, offset, &energy[1], pitch_lag,

                                  SUBFRAME_LEN, 1);

     int back_lag  = autocorr_max(buf, offset, &energy[3], pitch_lag,

                                  SUBFRAME_LEN, -1);
 
     ppf->index    = 0;
     ppf->opt_gain = 0;
     ppf->sc_gain  = 0x7fff;
 
     /* Case 0, Section 3.6 */
     if (!back_lag && !fwd_lag)
         return;
 
     /* Compute target energy */

     energy[0] = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN);

 
     /* Compute forward residual energy */
     if (fwd_lag)

         energy[2] = ff_g723_1_dot_product(buf + fwd_lag, buf + fwd_lag,
                                           SUBFRAME_LEN);

 
     /* Compute backward residual energy */
     if (back_lag)

         energy[4] = ff_g723_1_dot_product(buf - back_lag, buf - back_lag,
                                           SUBFRAME_LEN);

 
     /* Normalize and shorten */
     temp1 = 0;
     for (i = 0; i < 5; i++)
         temp1 = FFMAX(energy[i], temp1);
 

     scale = ff_g723_1_normalize_bits(temp1, 31);

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

         energy[i] = (energy[i] << scale) >> 16;

 
     if (fwd_lag && !back_lag) {  /* Case 1 */
         comp_ppf_gains(fwd_lag,  ppf, cur_rate, energy[0], energy[1],
                        energy[2]);
     } else if (!fwd_lag) {       /* Case 2 */
         comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
                        energy[4]);
     } else {                     /* Case 3 */
 
         /*
          * Select the largest of energy[1]^2/energy[2]
          * and energy[3]^2/energy[4]
          */
         temp1 = energy[4] * ((energy[1] * energy[1] + (1 << 14)) >> 15);
         temp2 = energy[2] * ((energy[3] * energy[3] + (1 << 14)) >> 15);
         if (temp1 >= temp2) {
             comp_ppf_gains(fwd_lag, ppf, cur_rate, energy[0], energy[1],
                            energy[2]);
         } else {
             comp_ppf_gains(-back_lag, ppf, cur_rate, energy[0], energy[3],
                            energy[4]);
         }
     }
 }
 
 /**
  * Classify frames as voiced/unvoiced.
  *
  * @param p         the context
  * @param pitch_lag decoded pitch_lag
  * @param exc_eng   excitation energy estimation
  * @param scale     scaling factor of exc_eng
  *
  * @return residual interpolation index if voiced, 0 otherwise
  */
 static int comp_interp_index(G723_1_Context *p, int pitch_lag,
                              int *exc_eng, int *scale)
 {
     int offset = PITCH_MAX + 2 * SUBFRAME_LEN;

     int16_t *buf = p->audio + LPC_ORDER;

 
     int index, ccr, tgt_eng, best_eng, temp;
 

     *scale = ff_g723_1_scale_vector(buf, p->excitation, FRAME_LEN + PITCH_MAX);

     buf   += offset;

 
     /* Compute maximum backward cross-correlation */
     ccr   = 0;

     index = autocorr_max(buf, offset, &ccr, pitch_lag, SUBFRAME_LEN * 2, -1);

     ccr   = av_sat_add32(ccr, 1 << 15) >> 16;

 
     /* Compute target energy */

     tgt_eng  = ff_g723_1_dot_product(buf, buf, SUBFRAME_LEN * 2);

     *exc_eng = av_sat_add32(tgt_eng, 1 << 15) >> 16;

 
     if (ccr <= 0)
         return 0;
 
     /* Compute best energy */

     best_eng = ff_g723_1_dot_product(buf - index, buf - index,
                                      SUBFRAME_LEN * 2);

     best_eng = av_sat_add32(best_eng, 1 << 15) >> 16;

 
     temp = best_eng * *exc_eng >> 3;
 
     if (temp < ccr * ccr) {
         return index;
     } else
         return 0;
 }
 
 /**

  * Perform residual interpolation based on frame classification.

  *
  * @param buf   decoded excitation vector
  * @param out   output vector
  * @param lag   decoded pitch lag
  * @param gain  interpolated gain
  * @param rseed seed for random number generator
  */
 static void residual_interp(int16_t *buf, int16_t *out, int lag,
                             int gain, int *rseed)
 {
     int i;
     if (lag) { /* Voiced */
         int16_t *vector_ptr = buf + PITCH_MAX;
         /* Attenuate */
         for (i = 0; i < lag; i++)

             out[i] = vector_ptr[i - lag] * 3 >> 2;
         av_memcpy_backptr((uint8_t*)(out + lag), lag * sizeof(*out),
                           (FRAME_LEN - lag) * sizeof(*out));

     } else {  /* Unvoiced */
         for (i = 0; i < FRAME_LEN; i++) {
             *rseed = *rseed * 521 + 259;
             out[i] = gain * *rseed >> 15;
         }

         memset(buf, 0, (FRAME_LEN + PITCH_MAX) * sizeof(*buf));

     }
 }
 
 /**
  * Perform IIR filtering.
  *
  * @param fir_coef FIR coefficients
  * @param iir_coef IIR coefficients
  * @param src      source vector
  * @param dest     destination vector
  * @param width    width of the output, 16 bits(0) / 32 bits(1)
  */
 #define iir_filter(fir_coef, iir_coef, src, dest, width)\
 {\
     int m, n;\
     int res_shift = 16 & ~-(width);\
     int in_shift  = 16 - res_shift;\
 \
     for (m = 0; m < SUBFRAME_LEN; m++) {\
         int64_t filter = 0;\
         for (n = 1; n <= LPC_ORDER; n++) {\
             filter -= (fir_coef)[n - 1] * (src)[m - n] -\
                       (iir_coef)[n - 1] * ((dest)[m - n] >> in_shift);\
         }\
 \
         (dest)[m] = av_clipl_int32(((src)[m] << 16) + (filter << 3) +\
                                    (1 << 15)) >> res_shift;\
     }\
 }
 
 /**
  * Adjust gain of postfiltered signal.
  *
  * @param p      the context
  * @param buf    postfiltered output vector
  * @param energy input energy coefficient
  */
 static void gain_scale(G723_1_Context *p, int16_t * buf, int energy)
 {
     int num, denom, gain, bits1, bits2;
     int i;
 
     num   = energy;
     denom = 0;
     for (i = 0; i < SUBFRAME_LEN; i++) {

         int temp = buf[i] >> 2;
         temp *= temp;

         denom = av_sat_dadd32(denom, temp);

     }
 
     if (num && denom) {

         bits1   = ff_g723_1_normalize_bits(num,   31);
         bits2   = ff_g723_1_normalize_bits(denom, 31);

         num     = num << bits1 >> 1;
         denom <<= bits2;
 
         bits2 = 5 + bits1 - bits2;
         bits2 = FFMAX(0, bits2);
 
         gain = (num >> 1) / (denom >> 16);
         gain = square_root(gain << 16 >> bits2);
     } else {
         gain = 1 << 12;
     }
 
     for (i = 0; i < SUBFRAME_LEN; i++) {

         p->pf_gain = (15 * p->pf_gain + gain + (1 << 3)) >> 4;

         buf[i]     = av_clip_int16((buf[i] * (p->pf_gain + (p->pf_gain >> 4)) +
                                    (1 << 10)) >> 11);
     }
 }
 
 /**
  * Perform formant filtering.
  *
  * @param p   the context
  * @param lpc quantized lpc coefficients

  * @param buf input buffer
  * @param dst output buffer

  */

 static void formant_postfilter(G723_1_Context *p, int16_t *lpc,
                                int16_t *buf, int16_t *dst)

 {

     int16_t filter_coef[2][LPC_ORDER];

     int filter_signal[LPC_ORDER + FRAME_LEN], *signal_ptr;
     int i, j, k;
 

     memcpy(buf, p->fir_mem, LPC_ORDER * sizeof(*buf));
     memcpy(filter_signal, p->iir_mem, LPC_ORDER * sizeof(*filter_signal));

 
     for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
         for (k = 0; k < LPC_ORDER; k++) {
             filter_coef[0][k] = (-lpc[k] * postfilter_tbl[0][k] +
                                  (1 << 14)) >> 15;
             filter_coef[1][k] = (-lpc[k] * postfilter_tbl[1][k] +
                                  (1 << 14)) >> 15;
         }

         iir_filter(filter_coef[0], filter_coef[1], buf + i, filter_signal + i, 1);

         lpc += LPC_ORDER;

     }
 
     memcpy(p->fir_mem, buf + FRAME_LEN, LPC_ORDER * sizeof(int16_t));
     memcpy(p->iir_mem, filter_signal + FRAME_LEN, LPC_ORDER * sizeof(int));
 

     buf += LPC_ORDER;

     signal_ptr = filter_signal + LPC_ORDER;
     for (i = 0; i < SUBFRAMES; i++) {

         int temp;

         int auto_corr[2];
         int scale, energy;
 
         /* Normalize */

         scale = ff_g723_1_scale_vector(dst, buf, SUBFRAME_LEN);

 
         /* Compute auto correlation coefficients */

         auto_corr[0] = ff_g723_1_dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
         auto_corr[1] = ff_g723_1_dot_product(dst, dst,     SUBFRAME_LEN);

 
         /* Compute reflection coefficient */
         temp = auto_corr[1] >> 16;
         if (temp) {
             temp = (auto_corr[0] >> 2) / temp;
         }

         p->reflection_coef = (3 * p->reflection_coef + temp + 2) >> 2;

         temp = -p->reflection_coef >> 1 & ~3;

 
         /* Compensation filter */
         for (j = 0; j < SUBFRAME_LEN; j++) {

             dst[j] = av_sat_dadd32(signal_ptr[j],

                                    (signal_ptr[j - 1] >> 16) * temp) >> 16;

         }
 
         /* Compute normalized signal energy */
         temp = 2 * scale + 4;
         if (temp < 0) {
             energy = av_clipl_int32((int64_t)auto_corr[1] << -temp);
         } else
             energy = auto_corr[1] >> temp;
 

         gain_scale(p, dst, energy);

         buf        += SUBFRAME_LEN;

         signal_ptr += SUBFRAME_LEN;

         dst        += SUBFRAME_LEN;

     }
 }
 

 static int sid_gain_to_lsp_index(int gain)
 {
     if (gain < 0x10)
         return gain << 6;
     else if (gain < 0x20)
         return gain - 8 << 7;
     else
         return gain - 20 << 8;
 }
 
 static inline int cng_rand(int *state, int base)
 {
     *state = (*state * 521 + 259) & 0xFFFF;
     return (*state & 0x7FFF) * base >> 15;
 }
 
 static int estimate_sid_gain(G723_1_Context *p)
 {
     int i, shift, seg, seg2, t, val, val_add, x, y;
 
     shift = 16 - p->cur_gain * 2;
     if (shift > 0)
         t = p->sid_gain << shift;
     else
         t = p->sid_gain >> -shift;
     x = t * cng_filt[0] >> 16;
 
     if (x >= cng_bseg[2])
         return 0x3F;
 
     if (x >= cng_bseg[1]) {
         shift = 4;
         seg   = 3;
     } else {
         shift = 3;
         seg   = (x >= cng_bseg[0]);
     }
     seg2 = FFMIN(seg, 3);
 
     val     = 1 << shift;
     val_add = val >> 1;
     for (i = 0; i < shift; i++) {
         t = seg * 32 + (val << seg2);
         t *= t;
         if (x >= t)
             val += val_add;
         else
             val -= val_add;
         val_add >>= 1;
     }
 
     t = seg * 32 + (val << seg2);
     y = t * t - x;
     if (y <= 0) {
         t = seg * 32 + (val + 1 << seg2);
         t = t * t - x;
         val = (seg2 - 1 << 4) + val;
         if (t >= y)
             val++;
     } else {
         t = seg * 32 + (val - 1 << seg2);
         t = t * t - x;
         val = (seg2 - 1 << 4) + val;
         if (t >= y)
             val--;
     }
 
     return val;
 }
 
 static void generate_noise(G723_1_Context *p)
 {
     int i, j, idx, t;
     int off[SUBFRAMES];
     int signs[SUBFRAMES / 2 * 11], pos[SUBFRAMES / 2 * 11];
     int tmp[SUBFRAME_LEN * 2];
     int16_t *vector_ptr;
     int64_t sum;
     int b0, c, delta, x, shift;
 
     p->pitch_lag[0] = cng_rand(&p->cng_random_seed, 21) + 123;
     p->pitch_lag[1] = cng_rand(&p->cng_random_seed, 19) + 123;
 
     for (i = 0; i < SUBFRAMES; i++) {
         p->subframe[i].ad_cb_gain = cng_rand(&p->cng_random_seed, 50) + 1;
         p->subframe[i].ad_cb_lag  = cng_adaptive_cb_lag[i];
     }
 
     for (i = 0; i < SUBFRAMES / 2; i++) {
         t = cng_rand(&p->cng_random_seed, 1 << 13);
         off[i * 2]     =   t       & 1;
         off[i * 2 + 1] = ((t >> 1) & 1) + SUBFRAME_LEN;
         t >>= 2;
         for (j = 0; j < 11; j++) {
             signs[i * 11 + j] = (t & 1) * 2 - 1 << 14;
             t >>= 1;
         }
     }
 
     idx = 0;
     for (i = 0; i < SUBFRAMES; i++) {
         for (j = 0; j < SUBFRAME_LEN / 2; j++)
             tmp[j] = j;
         t = SUBFRAME_LEN / 2;
         for (j = 0; j < pulses[i]; j++, idx++) {
             int idx2 = cng_rand(&p->cng_random_seed, t);
 
             pos[idx]  = tmp[idx2] * 2 + off[i];
             tmp[idx2] = tmp[--t];
         }
     }
 
     vector_ptr = p->audio + LPC_ORDER;
     memcpy(vector_ptr, p->prev_excitation,
            PITCH_MAX * sizeof(*p->excitation));
     for (i = 0; i < SUBFRAMES; i += 2) {

         ff_g723_1_gen_acb_excitation(vector_ptr, vector_ptr,
                                      p->pitch_lag[i >> 1], &p->subframe[i],
                                      p->cur_rate);
         ff_g723_1_gen_acb_excitation(vector_ptr + SUBFRAME_LEN,
                                      vector_ptr + SUBFRAME_LEN,
                                      p->pitch_lag[i >> 1], &p->subframe[i + 1],
                                      p->cur_rate);

 
         t = 0;
         for (j = 0; j < SUBFRAME_LEN * 2; j++)
             t |= FFABS(vector_ptr[j]);
         t = FFMIN(t, 0x7FFF);
         if (!t) {
             shift = 0;
         } else {
             shift = -10 + av_log2(t);
             if (shift < -2)
                 shift = -2;
         }
         sum = 0;
         if (shift < 0) {
            for (j = 0; j < SUBFRAME_LEN * 2; j++) {
                t      = vector_ptr[j] << -shift;
                sum   += t * t;
                tmp[j] = t;
            }
         } else {
            for (j = 0; j < SUBFRAME_LEN * 2; j++) {
                t      = vector_ptr[j] >> shift;
                sum   += t * t;
                tmp[j] = t;
            }
         }
 
         b0 = 0;
         for (j = 0; j < 11; j++)
             b0 += tmp[pos[(i / 2) * 11 + j]] * signs[(i / 2) * 11 + j];
         b0 = b0 * 2 * 2979LL + (1 << 29) >> 30; // approximated division by 11
 
         c = p->cur_gain * (p->cur_gain * SUBFRAME_LEN >> 5);
         if (shift * 2 + 3 >= 0)
             c >>= shift * 2 + 3;
         else
             c <<= -(shift * 2 + 3);
         c = (av_clipl_int32(sum << 1) - c) * 2979LL >> 15;
 
         delta = b0 * b0 * 2 - c;
         if (delta <= 0) {
             x = -b0;
         } else {
             delta = square_root(delta);
             x     = delta - b0;
             t     = delta + b0;
             if (FFABS(t) < FFABS(x))
                 x = -t;
         }
         shift++;
         if (shift < 0)
            x >>= -shift;
         else
            x <<= shift;
         x = av_clip(x, -10000, 10000);
 
         for (j = 0; j < 11; j++) {
             idx = (i / 2) * 11 + j;
             vector_ptr[pos[idx]] = av_clip_int16(vector_ptr[pos[idx]] +
                                                  (x * signs[idx] >> 15));
         }
 
         /* copy decoded data to serve as a history for the next decoded subframes */
         memcpy(vector_ptr + PITCH_MAX, vector_ptr,
                sizeof(*vector_ptr) * SUBFRAME_LEN * 2);
         vector_ptr += SUBFRAME_LEN * 2;
     }
     /* Save the excitation for the next frame */
     memcpy(p->prev_excitation, p->audio + LPC_ORDER + FRAME_LEN,
            PITCH_MAX * sizeof(*p->excitation));
 }
 

 static int g723_1_decode_frame(AVCodecContext *avctx, void *data,

                                int *got_frame_ptr, AVPacket *avpkt)

 {
     G723_1_Context *p  = avctx->priv_data;

     AVFrame *frame     = data;

     const uint8_t *buf = avpkt->data;
     int buf_size       = avpkt->size;
     int dec_mode       = buf[0] & 3;
 
     PPFParam ppf[SUBFRAMES];
     int16_t cur_lsp[LPC_ORDER];
     int16_t lpc[SUBFRAMES * LPC_ORDER];
     int16_t acb_vector[SUBFRAME_LEN];

     int16_t *out;

     int bad_frame = 0, i, j, ret;

     int16_t *audio = p->audio;

     if (buf_size < frame_size[dec_mode]) {
         if (buf_size)
             av_log(avctx, AV_LOG_WARNING,
                    "Expected %d bytes, got %d - skipping packet\n",
                    frame_size[dec_mode], buf_size);

         *got_frame_ptr = 0;

         return buf_size;
     }
 
     if (unpack_bitstream(p, buf, buf_size) < 0) {

         bad_frame = 1;
         if (p->past_frame_type == ACTIVE_FRAME)
             p->cur_frame_type = ACTIVE_FRAME;
         else
             p->cur_frame_type = UNTRANSMITTED_FRAME;

     }
 

     frame->nb_samples = FRAME_LEN;

     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
         return ret;

     out = (int16_t *)frame->data[0];

     if (p->cur_frame_type == ACTIVE_FRAME) {

         if (!bad_frame)

             p->erased_frames = 0;

         else if (p->erased_frames != 3)

             p->erased_frames++;
 

         ff_g723_1_inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
         ff_g723_1_lsp_interpolate(lpc, cur_lsp, p->prev_lsp);

 
         /* Save the lsp_vector for the next frame */

         memcpy(p->prev_lsp, cur_lsp, LPC_ORDER * sizeof(*p->prev_lsp));

 
         /* Generate the excitation for the frame */

         memcpy(p->excitation, p->prev_excitation,
                PITCH_MAX * sizeof(*p->excitation));

         if (!p->erased_frames) {

             int16_t *vector_ptr = p->excitation + PITCH_MAX;
 

             /* Update interpolation gain memory */
             p->interp_gain = fixed_cb_gain[(p->subframe[2].amp_index +
                                             p->subframe[3].amp_index) >> 1];
             for (i = 0; i < SUBFRAMES; i++) {

                 gen_fcb_excitation(vector_ptr, &p->subframe[i], p->cur_rate,

                                    p->pitch_lag[i >> 1], i);

                 ff_g723_1_gen_acb_excitation(acb_vector,
                                              &p->excitation[SUBFRAME_LEN * i],
                                              p->pitch_lag[i >> 1],
                                              &p->subframe[i], p->cur_rate);

                 /* Get the total excitation */
                 for (j = 0; j < SUBFRAME_LEN; j++) {

                     int v = av_clip_int16(vector_ptr[j] << 1);
                     vector_ptr[j] = av_clip_int16(v + acb_vector[j]);

                 }
                 vector_ptr += SUBFRAME_LEN;
             }
 
             vector_ptr = p->excitation + PITCH_MAX;
 
             p->interp_index = comp_interp_index(p, p->pitch_lag[1],
                                                 &p->sid_gain, &p->cur_gain);
 

             /* Perform pitch postfiltering */

             if (p->postfilter) {

                 i = PITCH_MAX;
                 for (j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
                     comp_ppf_coeff(p, i, p->pitch_lag[j >> 1],
                                    ppf + j, p->cur_rate);

                 for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)

                     ff_acelp_weighted_vector_sum(p->audio + LPC_ORDER + i,

                                                  vector_ptr + i,
                                                  vector_ptr + i + ppf[j].index,
                                                  ppf[j].sc_gain,
                                                  ppf[j].opt_gain,
                                                  1 << 14, 15, SUBFRAME_LEN);

             } else {
                 audio = vector_ptr - LPC_ORDER;
             }

             /* Save the excitation for the next frame */
             memcpy(p->prev_excitation, p->excitation + FRAME_LEN,
                    PITCH_MAX * sizeof(*p->excitation));

         } else {
             p->interp_gain = (p->interp_gain * 3 + 2) >> 2;
             if (p->erased_frames == 3) {
                 /* Mute output */
                 memset(p->excitation, 0,

                        (FRAME_LEN + PITCH_MAX) * sizeof(*p->excitation));

                 memset(p->prev_excitation, 0,
                        PITCH_MAX * sizeof(*p->excitation));

                 memset(frame->data[0], 0,

                        (FRAME_LEN + LPC_ORDER) * sizeof(int16_t));

             } else {

                 int16_t *buf = p->audio + LPC_ORDER;
 

                 /* Regenerate frame */

                 residual_interp(p->excitation, buf, p->interp_index,

                                 p->interp_gain, &p->random_seed);

 
                 /* Save the excitation for the next frame */
                 memcpy(p->prev_excitation, buf + (FRAME_LEN - PITCH_MAX),
                        PITCH_MAX * sizeof(*p->excitation));

             }
         }

         p->cng_random_seed = CNG_RANDOM_SEED;

     } else {

         if (p->cur_frame_type == SID_FRAME) {
             p->sid_gain = sid_gain_to_lsp_index(p->subframe[0].amp_index);

             ff_g723_1_inverse_quant(p->sid_lsp, p->prev_lsp, p->lsp_index, 0);

         } else if (p->past_frame_type == ACTIVE_FRAME) {
             p->sid_gain = estimate_sid_gain(p);
         }

         if (p->past_frame_type == ACTIVE_FRAME)
             p->cur_gain = p->sid_gain;
         else
             p->cur_gain = (p->cur_gain * 7 + p->sid_gain) >> 3;
         generate_noise(p);

         ff_g723_1_lsp_interpolate(lpc, p->sid_lsp, p->prev_lsp);

         /* Save the lsp_vector for the next frame */
         memcpy(p->prev_lsp, p->sid_lsp, LPC_ORDER * sizeof(*p->prev_lsp));

     }
 
     p->past_frame_type = p->cur_frame_type;
 

     memcpy(p->audio, p->synth_mem, LPC_ORDER * sizeof(*p->audio));

     for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)

         ff_celp_lp_synthesis_filter(p->audio + i, &lpc[j * LPC_ORDER],

                                     audio + i, SUBFRAME_LEN, LPC_ORDER,

                                     0, 1, 1 << 12);

     memcpy(p->synth_mem, p->audio + FRAME_LEN, LPC_ORDER * sizeof(*p->audio));

     if (p->postfilter) {

         formant_postfilter(p, lpc, p->audio, out);

     } else { // if output is not postfiltered it should be scaled by 2
         for (i = 0; i < FRAME_LEN; i++)

             out[i] = av_clip_int16(p->audio[LPC_ORDER + i] << 1);

     }

     *got_frame_ptr = 1;

 
     return frame_size[dec_mode];
 }
 

 #define OFFSET(x) offsetof(G723_1_Context, x)
 #define AD     AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
 
 static const AVOption options[] = {

     { "postfilter", "enable postfilter", OFFSET(postfilter), AV_OPT_TYPE_BOOL,

       { .i64 = 1 }, 0, 1, AD },

     { NULL }
 };
 

 static const AVClass g723_1dec_class = {
     .class_name = "G.723.1 decoder",
     .item_name  = av_default_item_name,
     .option     = options,
     .version    = LIBAVUTIL_VERSION_INT,
 };
 

 AVCodec ff_g723_1_decoder = {
     .name           = "g723_1",

     .long_name      = NULL_IF_CONFIG_SMALL("G.723.1"),

     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_G723_1,

     .priv_data_size = sizeof(G723_1_Context),
     .init           = g723_1_decode_init,
     .decode         = g723_1_decode_frame,

     .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,

     .priv_class     = &g723_1dec_class,

 };