text: /*
   * 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_data.h"
  #include "internal.h"
  
  #define CNG_RANDOM_SEED 12345
  
  typedef struct g723_1_context {
      AVClass *class;
  
      G723_1_Subframe subframe[4];
      enum FrameType cur_frame_type;
      enum FrameType past_frame_type;
      enum Rate cur_rate;
      uint8_t lsp_index[LSP_BANDS];
      int pitch_lag[2];
      int erased_frames;
  
      int16_t prev_lsp[LPC_ORDER];
      int16_t sid_lsp[LPC_ORDER];
      int16_t prev_excitation[PITCH_MAX];
      int16_t excitation[PITCH_MAX + FRAME_LEN + 4];
      int16_t synth_mem[LPC_ORDER];
      int16_t fir_mem[LPC_ORDER];
      int     iir_mem[LPC_ORDER];
  
      int random_seed;
      int cng_random_seed;
      int interp_index;
      int interp_gain;
      int sid_gain;
      int cur_gain;
      int reflection_coef;
      int pf_gain;                 ///< formant postfilter
                                   ///< gain scaling unit memory
      int postfilter;
  
      int16_t audio[FRAME_LEN + LPC_ORDER + PITCH_MAX + 4];
      int16_t prev_data[HALF_FRAME_LEN];
      int16_t prev_weight_sig[PITCH_MAX];
  
  
      int16_t hpf_fir_mem;                   ///< highpass filter fir
      int     hpf_iir_mem;                   ///< and iir memories
      int16_t perf_fir_mem[LPC_ORDER];       ///< perceptual filter fir
      int16_t perf_iir_mem[LPC_ORDER];       ///< and iir memories
  
      int16_t harmonic_mem[PITCH_MAX];
  } G723_1_Context;
  
  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);
  }
  
  /**
   * Calculate the number of left-shifts required for normalizing the input.
   *
   * @param num   input number
   * @param width width of the input, 15 or 31 bits
   */
  static int normalize_bits(int num, int width)
  {
      return width - av_log2(num) - 1;
  }
  
  #define normalize_bits_int16(num) normalize_bits(num, 15)
  #define normalize_bits_int32(num) normalize_bits(num, 31)
  
  /**
   * Scale vector contents based on the largest of their absolutes.
   */
  static int scale_vector(int16_t *dst, const int16_t *vector, int length)
  {
      int bits, max = 0;
      int i;
  
      for (i = 0; i < length; i++)
          max |= FFABS(vector[i]);
  
      bits= 14 - av_log2_16bit(max);
      bits= FFMAX(bits, 0);
  
      for (i = 0; i < length; i++)
          dst[i] = vector[i] << bits >> 3;
  
      return bits - 3;
  }
  
  /**
   * Perform inverse quantization of LSP frequencies.
   *
   * @param cur_lsp    the current LSP vector
   * @param prev_lsp   the previous LSP vector
   * @param lsp_index  VQ indices
   * @param bad_frame  bad frame flag
   */
  static void inverse_quant(int16_t *cur_lsp, int16_t *prev_lsp,
                            uint8_t *lsp_index, int bad_frame)
  {
      int min_dist, pred;
      int i, j, temp, stable;
  
      /* Check for frame erasure */
      if (!bad_frame) {
          min_dist     = 0x100;
          pred         = 12288;
      } else {
          min_dist     = 0x200;
          pred         = 23552;
          lsp_index[0] = lsp_index[1] = lsp_index[2] = 0;
      }
  
      /* Get the VQ table entry corresponding to the transmitted index */
      cur_lsp[0] = lsp_band0[lsp_index[0]][0];
      cur_lsp[1] = lsp_band0[lsp_index[0]][1];
      cur_lsp[2] = lsp_band0[lsp_index[0]][2];
      cur_lsp[3] = lsp_band1[lsp_index[1]][0];
      cur_lsp[4] = lsp_band1[lsp_index[1]][1];
      cur_lsp[5] = lsp_band1[lsp_index[1]][2];
      cur_lsp[6] = lsp_band2[lsp_index[2]][0];
      cur_lsp[7] = lsp_band2[lsp_index[2]][1];
      cur_lsp[8] = lsp_band2[lsp_index[2]][2];
      cur_lsp[9] = lsp_band2[lsp_index[2]][3];
  
      /* Add predicted vector & DC component to the previously quantized vector */
      for (i = 0; i < LPC_ORDER; i++) {
          temp        = ((prev_lsp[i] - dc_lsp[i]) * pred + (1 << 14)) >> 15;
          cur_lsp[i] += dc_lsp[i] + temp;
      }
  
      for (i = 0; i < LPC_ORDER; i++) {
          cur_lsp[0]             = FFMAX(cur_lsp[0],  0x180);
          cur_lsp[LPC_ORDER - 1] = FFMIN(cur_lsp[LPC_ORDER - 1], 0x7e00);
  
          /* Stability check */
          for (j = 1; j < LPC_ORDER; j++) {
              temp = min_dist + cur_lsp[j - 1] - cur_lsp[j];
              if (temp > 0) {
                  temp >>= 1;
                  cur_lsp[j - 1] -= temp;
                  cur_lsp[j]     += temp;
              }
          }
          stable = 1;
          for (j = 1; j < LPC_ORDER; j++) {
              temp = cur_lsp[j - 1] + min_dist - cur_lsp[j] - 4;
              if (temp > 0) {
                  stable = 0;
                  break;
              }
          }
          if (stable)
              break;
      }
      if (!stable)
          memcpy(cur_lsp, prev_lsp, LPC_ORDER * sizeof(*cur_lsp));
  }
  
  /**
   * Bitexact implementation of 2ab scaled by 1/2^16.
   *
   * @param a 32 bit multiplicand
   * @param b 16 bit multiplier
   */
  #define MULL2(a, b) \
          MULL(a,b,15)
  
  /**
   * Convert LSP frequencies to LPC coefficients.
   *
   * @param lpc buffer for LPC coefficients
   */
  static void lsp2lpc(int16_t *lpc)
  {
      int f1[LPC_ORDER / 2 + 1];
      int f2[LPC_ORDER / 2 + 1];
      int i, j;
  
      /* Calculate negative cosine */
      for (j = 0; j < LPC_ORDER; j++) {
          int index     = (lpc[j] >> 7) & 0x1FF;
          int offset    = lpc[j] & 0x7f;
          int temp1     = cos_tab[index] << 16;
          int temp2     = (cos_tab[index + 1] - cos_tab[index]) *
                            ((offset << 8) + 0x80) << 1;
  
          lpc[j] = -(av_sat_dadd32(1 << 15, temp1 + temp2) >> 16);
      }
  
      /*
       * Compute sum and difference polynomial coefficients
       * (bitexact alternative to lsp2poly() in lsp.c)
       */
      /* Initialize with values in Q28 */
      f1[0] = 1 << 28;
      f1[1] = (lpc[0] << 14) + (lpc[2] << 14);
      f1[2] = lpc[0] * lpc[2] + (2 << 28);
  
      f2[0] = 1 << 28;
      f2[1] = (lpc[1] << 14) + (lpc[3] << 14);
      f2[2] = lpc[1] * lpc[3] + (2 << 28);
  
      /*
       * Calculate and scale the coefficients by 1/2 in
       * each iteration for a final scaling factor of Q25
       */
      for (i = 2; i < LPC_ORDER / 2; i++) {
          f1[i + 1] = f1[i - 1] + MULL2(f1[i], lpc[2 * i]);
          f2[i + 1] = f2[i - 1] + MULL2(f2[i], lpc[2 * i + 1]);
  
          for (j = i; j >= 2; j--) {
              f1[j] = MULL2(f1[j - 1], lpc[2 * i]) +
                      (f1[j] >> 1) + (f1[j - 2] >> 1);
              f2[j] = MULL2(f2[j - 1], lpc[2 * i + 1]) +
                      (f2[j] >> 1) + (f2[j - 2] >> 1);
          }
  
          f1[0] >>= 1;
          f2[0] >>= 1;
          f1[1] = ((lpc[2 * i]     << 16 >> i) + f1[1]) >> 1;
          f2[1] = ((lpc[2 * i + 1] << 16 >> i) + f2[1]) >> 1;
      }
  
      /* Convert polynomial coefficients to LPC coefficients */
      for (i = 0; i < LPC_ORDER / 2; i++) {
          int64_t ff1 = f1[i + 1] + f1[i];
          int64_t ff2 = f2[i + 1] - f2[i];
  
          lpc[i] = av_clipl_int32(((ff1 + ff2) << 3) + (1 << 15)) >> 16;
          lpc[LPC_ORDER - i - 1] = av_clipl_int32(((ff1 - ff2) << 3) +
                                                  (1 << 15)) >> 16;
      }
  }
  
  /**
   * Quantize LSP frequencies by interpolation and convert them to
   * the corresponding LPC coefficients.
   *
   * @param lpc      buffer for LPC coefficients
   * @param cur_lsp  the current LSP vector
   * @param prev_lsp the previous LSP vector
   */
  static void lsp_interpolate(int16_t *lpc, int16_t *cur_lsp, int16_t *prev_lsp)
  {
      int i;
      int16_t *lpc_ptr = lpc;
  
      /* cur_lsp * 0.25 + prev_lsp * 0.75 */
      ff_acelp_weighted_vector_sum(lpc, cur_lsp, prev_lsp,
                                   4096, 12288, 1 << 13, 14, LPC_ORDER);
      ff_acelp_weighted_vector_sum(lpc + LPC_ORDER, cur_lsp, prev_lsp,
                                   8192, 8192, 1 << 13, 14, LPC_ORDER);
      ff_acelp_weighted_vector_sum(lpc + 2 * LPC_ORDER, cur_lsp, prev_lsp,
                                   12288, 4096, 1 << 13, 14, LPC_ORDER);
      memcpy(lpc + 3 * LPC_ORDER, cur_lsp, LPC_ORDER * sizeof(*lpc));
  
      for (i = 0; i < SUBFRAMES; i++) {
          lsp2lpc(lpc_ptr);
          lpc_ptr += LPC_ORDER;
      }
  }
  
  /**
   * Generate a train of dirac functions with period as pitch lag.
   */
  static void gen_dirac_train(int16_t *buf, int pitch_lag)
  {
      int16_t vector[SUBFRAME_LEN];
      int i, j;
  
      memcpy(vector, buf, SUBFRAME_LEN * sizeof(*vector));
      for (i = pitch_lag; i < SUBFRAME_LEN; i += pitch_lag) {
          for (j = 0; j < SUBFRAME_LEN - i; j++)
              buf[i + j] += vector[j];
      }
  }
  
  /**
   * 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)
              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;
          }
      }
  }
  
  /**
   * Get delayed contribution from the previous excitation vector.
   */
  static void get_residual(int16_t *residual, int16_t *prev_excitation, int lag)
  {
      int offset = PITCH_MAX - PITCH_ORDER / 2 - lag;
      int i;
  
      residual[0] = prev_excitation[offset];
      residual[1] = prev_excitation[offset + 1];
  
      offset += 2;
      for (i = 2; i < SUBFRAME_LEN + PITCH_ORDER - 1; i++)
          residual[i] = prev_excitation[offset + (i - 2) % lag];
  }
  
  static int dot_product(const int16_t *a, const int16_t *b, int length)
  {
      int sum = ff_dot_product(a,b,length);
      return av_sat_add32(sum, sum);
  }
  
  /**
   * Generate adaptive codebook excitation.
   */
  static void gen_acb_excitation(int16_t *vector, int16_t *prev_excitation,
                                 int pitch_lag, G723_1_Subframe *subfrm,
                                 enum Rate cur_rate)
  {
      int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
      const int16_t *cb_ptr;
      int lag = pitch_lag + subfrm->ad_cb_lag - 1;
  
      int i;
      int sum;
  
      get_residual(residual, prev_excitation, lag);
  
      /* Select quantization table */
      if (cur_rate == RATE_6300 && pitch_lag < SUBFRAME_LEN - 2) {
          cb_ptr = adaptive_cb_gain85;
      } else
          cb_ptr = adaptive_cb_gain170;
  
      /* Calculate adaptive vector */
      cb_ptr += subfrm->ad_cb_gain * 20;
      for (i = 0; i < SUBFRAME_LEN; i++) {
          sum = ff_dot_product(residual + i, cb_ptr, PITCH_ORDER);
          vector[i] = av_sat_dadd32(1 << 15, av_sat_add32(sum, sum)) >> 16;
      }
  }
  
  /**
   * 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 = 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] = dot_product(buf, buf, SUBFRAME_LEN);
  
      /* Compute forward residual energy */
      if (fwd_lag)
          energy[2] = dot_product(buf + fwd_lag, buf + fwd_lag, SUBFRAME_LEN);
  
      /* Compute backward residual energy */
      if (back_lag)
          energy[4] = 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 = 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 = 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  = 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 = 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;
  }
  
  /**
   * Peform 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   = normalize_bits(num,   31);
          bits2   = 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 = scale_vector(dst, buf, SUBFRAME_LEN);
  
          /* Compute auto correlation coefficients */
          auto_corr[0] = dot_product(dst, dst + 1, SUBFRAME_LEN - 1);
          auto_corr[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) {
          gen_acb_excitation(vector_ptr, vector_ptr,
                             p->pitch_lag[i >> 1], &p->subframe[i],
                             p->cur_rate);
          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++;
  
          inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, bad_frame);
          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);
                  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);
  
              /* Peform 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);
              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);
          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", "postfilter on/off", OFFSET(postfilter), AV_OPT_TYPE_INT,
        { .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   = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
      .priv_class     = &g723_1dec_class,
  };
  
  #if CONFIG_G723_1_ENCODER
  #define BITSTREAM_WRITER_LE
  #include "put_bits.h"
  
  static av_cold int g723_1_encode_init(AVCodecContext *avctx)
  {
      G723_1_Context *p = avctx->priv_data;
  
      if (avctx->sample_rate != 8000) {
          av_log(avctx, AV_LOG_ERROR, "Only 8000Hz sample rate supported\n");
          return -1;
      }
  
      if (avctx->channels != 1) {
          av_log(avctx, AV_LOG_ERROR, "Only mono supported\n");
          return AVERROR(EINVAL);
      }
  
      if (avctx->bit_rate == 6300) {
          p->cur_rate = RATE_6300;
      } else if (avctx->bit_rate == 5300) {
          av_log(avctx, AV_LOG_ERROR, "Bitrate not supported yet, use 6.3k\n");
          return AVERROR_PATCHWELCOME;
      } else {
          av_log(avctx, AV_LOG_ERROR,
                 "Bitrate not supported, use 6.3k\n");
          return AVERROR(EINVAL);
      }
      avctx->frame_size = 240;
      memcpy(p->prev_lsp, dc_lsp, LPC_ORDER * sizeof(int16_t));
  
      return 0;
  }
  
  /**
   * Remove DC component from the input signal.
   *
   * @param buf input signal
   * @param fir zero memory
   * @param iir pole memory
   */
  static void highpass_filter(int16_t *buf, int16_t *fir, int *iir)
  {
      int i;
      for (i = 0; i < FRAME_LEN; i++) {
          *iir   = (buf[i] << 15) + ((-*fir) << 15) + MULL2(*iir, 0x7f00);
          *fir   = buf[i];
          buf[i] = av_clipl_int32((int64_t)*iir + (1 << 15)) >> 16;
      }
  }
  
  /**
   * Estimate autocorrelation of the input vector.
   *
   * @param buf      input buffer
   * @param autocorr autocorrelation coefficients vector
   */
  static void comp_autocorr(int16_t *buf, int16_t *autocorr)
  {
      int i, scale, temp;
      int16_t vector[LPC_FRAME];
  
      scale_vector(vector, buf, LPC_FRAME);
  
      /* Apply the Hamming window */
      for (i = 0; i < LPC_FRAME; i++)
          vector[i] = (vector[i] * hamming_window[i] + (1 << 14)) >> 15;
  
      /* Compute the first autocorrelation coefficient */
      temp = ff_dot_product(vector, vector, LPC_FRAME);
  
      /* Apply a white noise correlation factor of (1025/1024) */
      temp += temp >> 10;
  
      /* Normalize */
      scale = normalize_bits_int32(temp);
      autocorr[0] = av_clipl_int32((int64_t)(temp << scale) +
                                   (1 << 15)) >> 16;
  
      /* Compute the remaining coefficients */
      if (!autocorr[0]) {
          memset(autocorr + 1, 0, LPC_ORDER * sizeof(int16_t));
      } else {
          for (i = 1; i <= LPC_ORDER; i++) {
             temp = ff_dot_product(vector, vector + i, LPC_FRAME - i);
             temp = MULL2((temp << scale), binomial_window[i - 1]);
             autocorr[i] = av_clipl_int32((int64_t)temp + (1 << 15)) >> 16;
          }
      }
  }
  
  /**
   * Use Levinson-Durbin recursion to compute LPC coefficients from
   * autocorrelation values.
   *
   * @param lpc      LPC coefficients vector
   * @param autocorr autocorrelation coefficients vector
   * @param error    prediction error
   */
  static void levinson_durbin(int16_t *lpc, int16_t *autocorr, int16_t error)
  {
      int16_t vector[LPC_ORDER];
      int16_t partial_corr;
      int i, j, temp;
  
      memset(lpc, 0, LPC_ORDER * sizeof(int16_t));
  
      for (i = 0; i < LPC_ORDER; i++) {
          /* Compute the partial correlation coefficient */
          temp = 0;
          for (j = 0; j < i; j++)
              temp -= lpc[j] * autocorr[i - j - 1];
          temp = ((autocorr[i] << 13) + temp) << 3;
  
          if (FFABS(temp) >= (error << 16))
              break;
  
          partial_corr = temp / (error << 1);
  
          lpc[i] = av_clipl_int32((int64_t)(partial_corr << 14) +
                                  (1 << 15)) >> 16;
  
          /* Update the prediction error */
          temp  = MULL2(temp, partial_corr);
          error = av_clipl_int32((int64_t)(error << 16) - temp +
                                  (1 << 15)) >> 16;
  
          memcpy(vector, lpc, i * sizeof(int16_t));
          for (j = 0; j < i; j++) {
              temp = partial_corr * vector[i - j - 1] << 1;
              lpc[j] = av_clipl_int32((int64_t)(lpc[j] << 16) - temp +
                                      (1 << 15)) >> 16;
          }
      }
  }
  
  /**
   * Calculate LPC coefficients for the current frame.
   *
   * @param buf       current frame
   * @param prev_data 2 trailing subframes of the previous frame
   * @param lpc       LPC coefficients vector
   */
  static void comp_lpc_coeff(int16_t *buf, int16_t *lpc)
  {
      int16_t autocorr[(LPC_ORDER + 1) * SUBFRAMES];
      int16_t *autocorr_ptr = autocorr;
      int16_t *lpc_ptr      = lpc;
      int i, j;
  
      for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
          comp_autocorr(buf + i, autocorr_ptr);
          levinson_durbin(lpc_ptr, autocorr_ptr + 1, autocorr_ptr[0]);
  
          lpc_ptr += LPC_ORDER;
          autocorr_ptr += LPC_ORDER + 1;
      }
  }
  
  static void lpc2lsp(int16_t *lpc, int16_t *prev_lsp, int16_t *lsp)
  {
      int f[LPC_ORDER + 2]; ///< coefficients of the sum and difference
                            ///< polynomials (F1, F2) ordered as
                            ///< f1[0], f2[0], ...., f1[5], f2[5]
  
      int max, shift, cur_val, prev_val, count, p;
      int i, j;
      int64_t temp;
  
      /* Initialize f1[0] and f2[0] to 1 in Q25 */
      for (i = 0; i < LPC_ORDER; i++)
          lsp[i] = (lpc[i] * bandwidth_expand[i] + (1 << 14)) >> 15;
  
      /* Apply bandwidth expansion on the LPC coefficients */
      f[0] = f[1] = 1 << 25;
  
      /* Compute the remaining coefficients */
      for (i = 0; i < LPC_ORDER / 2; i++) {
          /* f1 */
          f[2 * i + 2] = -f[2 * i] - ((lsp[i] + lsp[LPC_ORDER - 1 - i]) << 12);
          /* f2 */
          f[2 * i + 3] = f[2 * i + 1] - ((lsp[i] - lsp[LPC_ORDER - 1 - i]) << 12);
      }
  
      /* Divide f1[5] and f2[5] by 2 for use in polynomial evaluation */
      f[LPC_ORDER] >>= 1;
      f[LPC_ORDER + 1] >>= 1;
  
      /* Normalize and shorten */
      max = FFABS(f[0]);
      for (i = 1; i < LPC_ORDER + 2; i++)
          max = FFMAX(max, FFABS(f[i]));
  
      shift = normalize_bits_int32(max);
  
      for (i = 0; i < LPC_ORDER + 2; i++)
          f[i] = av_clipl_int32((int64_t)(f[i] << shift) + (1 << 15)) >> 16;
  
      /**
       * Evaluate F1 and F2 at uniform intervals of pi/256 along the
       * unit circle and check for zero crossings.
       */
      p    = 0;
      temp = 0;
      for (i = 0; i <= LPC_ORDER / 2; i++)
          temp += f[2 * i] * cos_tab[0];
      prev_val = av_clipl_int32(temp << 1);
      count    = 0;
      for ( i = 1; i < COS_TBL_SIZE / 2; i++) {
          /* Evaluate */
          temp = 0;
          for (j = 0; j <= LPC_ORDER / 2; j++)
              temp += f[LPC_ORDER - 2 * j + p] * cos_tab[i * j % COS_TBL_SIZE];
          cur_val = av_clipl_int32(temp << 1);
  
          /* Check for sign change, indicating a zero crossing */
          if ((cur_val ^ prev_val) < 0) {
              int abs_cur  = FFABS(cur_val);
              int abs_prev = FFABS(prev_val);
              int sum      = abs_cur + abs_prev;
  
              shift        = normalize_bits_int32(sum);
              sum          <<= shift;
              abs_prev     = abs_prev << shift >> 8;
              lsp[count++] = ((i - 1) << 7) + (abs_prev >> 1) / (sum >> 16);
  
              if (count == LPC_ORDER)
                  break;
  
              /* Switch between sum and difference polynomials */
              p ^= 1;
  
              /* Evaluate */
              temp = 0;
              for (j = 0; j <= LPC_ORDER / 2; j++){
                  temp += f[LPC_ORDER - 2 * j + p] *
                          cos_tab[i * j % COS_TBL_SIZE];
              }
              cur_val = av_clipl_int32(temp<<1);
          }
          prev_val = cur_val;
      }
  
      if (count != LPC_ORDER)
          memcpy(lsp, prev_lsp, LPC_ORDER * sizeof(int16_t));
  }
  
  /**
   * Quantize the current LSP subvector.
   *
   * @param num    band number
   * @param offset offset of the current subvector in an LPC_ORDER vector
   * @param size   size of the current subvector
   */
  #define get_index(num, offset, size) \
  {\
      int error, max = -1;\
      int16_t temp[4];\
      int i, j;\
      for (i = 0; i < LSP_CB_SIZE; i++) {\
          for (j = 0; j < size; j++){\
              temp[j] = (weight[j + (offset)] * lsp_band##num[i][j] +\
                        (1 << 14)) >> 15;\
          }\
          error =  dot_product(lsp + (offset), temp, size) << 1;\
          error -= dot_product(lsp_band##num[i], temp, size);\
          if (error > max) {\
              max = error;\
              lsp_index[num] = i;\
          }\
      }\
  }
  
  /**
   * Vector quantize the LSP frequencies.
   *
   * @param lsp      the current lsp vector
   * @param prev_lsp the previous lsp vector
   */
  static void lsp_quantize(uint8_t *lsp_index, int16_t *lsp, int16_t *prev_lsp)
  {
      int16_t weight[LPC_ORDER];
      int16_t min, max;
      int shift, i;
  
      /* Calculate the VQ weighting vector */
      weight[0] = (1 << 20) / (lsp[1] - lsp[0]);
      weight[LPC_ORDER - 1] = (1 << 20) /
                              (lsp[LPC_ORDER - 1] - lsp[LPC_ORDER - 2]);
  
      for (i = 1; i < LPC_ORDER - 1; i++) {
          min  = FFMIN(lsp[i] - lsp[i - 1], lsp[i + 1] - lsp[i]);
          if (min > 0x20)
              weight[i] = (1 << 20) / min;
          else
              weight[i] = INT16_MAX;
      }
  
      /* Normalize */
      max = 0;
      for (i = 0; i < LPC_ORDER; i++)
          max = FFMAX(weight[i], max);
  
      shift = normalize_bits_int16(max);
      for (i = 0; i < LPC_ORDER; i++) {
          weight[i] <<= shift;
      }
  
      /* Compute the VQ target vector */
      for (i = 0; i < LPC_ORDER; i++) {
          lsp[i] -= dc_lsp[i] +
                    (((prev_lsp[i] - dc_lsp[i]) * 12288 + (1 << 14)) >> 15);
      }
  
      get_index(0, 0, 3);
      get_index(1, 3, 3);
      get_index(2, 6, 4);
  }
  
  /**
   * Apply the formant perceptual weighting filter.
   *
   * @param flt_coef filter coefficients
   * @param unq_lpc  unquantized lpc vector
   */
  static void perceptual_filter(G723_1_Context *p, int16_t *flt_coef,
                                int16_t *unq_lpc, int16_t *buf)
  {
      int16_t vector[FRAME_LEN + LPC_ORDER];
      int i, j, k, l = 0;
  
      memcpy(buf, p->iir_mem, sizeof(int16_t) * LPC_ORDER);
      memcpy(vector, p->fir_mem, sizeof(int16_t) * LPC_ORDER);
      memcpy(vector + LPC_ORDER, buf + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
  
      for (i = LPC_ORDER, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++) {
          for (k = 0; k < LPC_ORDER; k++) {
              flt_coef[k + 2 * l] = (unq_lpc[k + l] * percept_flt_tbl[0][k] +
                                    (1 << 14)) >> 15;
              flt_coef[k + 2 * l + LPC_ORDER] = (unq_lpc[k + l] *
                                               percept_flt_tbl[1][k] +
                                               (1 << 14)) >> 15;
          }
          iir_filter(flt_coef + 2 * l, flt_coef + 2 * l + LPC_ORDER, vector + i,
                     buf + i, 0);
          l += LPC_ORDER;
      }
      memcpy(p->iir_mem, buf + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
      memcpy(p->fir_mem, vector + FRAME_LEN, sizeof(int16_t) * LPC_ORDER);
  }
  
  /**
   * Estimate the open loop pitch period.
   *
   * @param buf   perceptually weighted speech
   * @param start estimation is carried out from this position
   */
  static int estimate_pitch(int16_t *buf, int start)
  {
      int max_exp = 32;
      int max_ccr = 0x4000;
      int max_eng = 0x7fff;
      int index   = PITCH_MIN;
      int offset  = start - PITCH_MIN + 1;
  
      int ccr, eng, orig_eng, ccr_eng, exp;
      int diff, temp;
  
      int i;
  
      orig_eng = ff_dot_product(buf + offset, buf + offset, HALF_FRAME_LEN);
  
      for (i = PITCH_MIN; i <= PITCH_MAX - 3; i++) {
          offset--;
  
          /* Update energy and compute correlation */
          orig_eng += buf[offset] * buf[offset] -
                      buf[offset + HALF_FRAME_LEN] * buf[offset + HALF_FRAME_LEN];
          ccr      =  ff_dot_product(buf + start, buf + offset, HALF_FRAME_LEN);
          if (ccr <= 0)
              continue;
  
          /* Split into mantissa and exponent to maintain precision */
          exp  =   normalize_bits_int32(ccr);
          ccr  =   av_clipl_int32((int64_t)(ccr << exp) + (1 << 15)) >> 16;
          exp  <<= 1;
          ccr  *=  ccr;
          temp =   normalize_bits_int32(ccr);
          ccr  =   ccr << temp >> 16;
          exp  +=  temp;
  
          temp =   normalize_bits_int32(orig_eng);
          eng  =   av_clipl_int32((int64_t)(orig_eng << temp) + (1 << 15)) >> 16;
          exp  -=  temp;
  
          if (ccr >= eng) {
              exp--;
              ccr >>= 1;
          }
          if (exp > max_exp)
              continue;
  
          if (exp + 1 < max_exp)
              goto update;
  
          /* Equalize exponents before comparison */
          if (exp + 1 == max_exp)
              temp = max_ccr >> 1;
          else
              temp = max_ccr;
          ccr_eng = ccr * max_eng;
          diff    = ccr_eng - eng * temp;
          if (diff > 0 && (i - index < PITCH_MIN || diff > ccr_eng >> 2)) {
  update:
              index   = i;
              max_exp = exp;
              max_ccr = ccr;
              max_eng = eng;
          }
      }
      return index;
  }
  
  /**
   * Compute harmonic noise filter parameters.
   *
   * @param buf       perceptually weighted speech
   * @param pitch_lag open loop pitch period
   * @param hf        harmonic filter parameters
   */
  static void comp_harmonic_coeff(int16_t *buf, int16_t pitch_lag, HFParam *hf)
  {
      int ccr, eng, max_ccr, max_eng;
      int exp, max, diff;
      int energy[15];
      int i, j;
  
      for (i = 0, j = pitch_lag - 3; j <= pitch_lag + 3; i++, j++) {
          /* Compute residual energy */
          energy[i << 1] = ff_dot_product(buf - j, buf - j, SUBFRAME_LEN);
          /* Compute correlation */
          energy[(i << 1) + 1] = ff_dot_product(buf, buf - j, SUBFRAME_LEN);
      }
  
      /* Compute target energy */
      energy[14] = ff_dot_product(buf, buf, SUBFRAME_LEN);
  
      /* Normalize */
      max = 0;
      for (i = 0; i < 15; i++)
          max = FFMAX(max, FFABS(energy[i]));
  
      exp = normalize_bits_int32(max);
      for (i = 0; i < 15; i++) {
          energy[i] = av_clipl_int32((int64_t)(energy[i] << exp) +
                                     (1 << 15)) >> 16;
      }
  
      hf->index = -1;
      hf->gain  =  0;
      max_ccr   =  1;
      max_eng   =  0x7fff;
  
      for (i = 0; i <= 6; i++) {
          eng = energy[i << 1];
          ccr = energy[(i << 1) + 1];
  
          if (ccr <= 0)
              continue;
  
          ccr  = (ccr * ccr + (1 << 14)) >> 15;
          diff = ccr * max_eng - eng * max_ccr;
          if (diff > 0) {
              max_ccr   = ccr;
              max_eng   = eng;
              hf->index = i;
          }
      }
  
      if (hf->index == -1) {
          hf->index = pitch_lag;
          return;
      }
  
      eng = energy[14] * max_eng;
      eng = (eng >> 2) + (eng >> 3);
      ccr = energy[(hf->index << 1) + 1] * energy[(hf->index << 1) + 1];
      if (eng < ccr) {
          eng = energy[(hf->index << 1) + 1];
  
          if (eng >= max_eng)
              hf->gain = 0x2800;
          else
              hf->gain = ((eng << 15) / max_eng * 0x2800 + (1 << 14)) >> 15;
      }
      hf->index += pitch_lag - 3;
  }
  
  /**
   * Apply the harmonic noise shaping filter.
   *
   * @param hf filter parameters
   */
  static void harmonic_filter(HFParam *hf, const int16_t *src, int16_t *dest)
  {
      int i;
  
      for (i = 0; i < SUBFRAME_LEN; i++) {
          int64_t temp = hf->gain * src[i - hf->index] << 1;
          dest[i] = av_clipl_int32((src[i] << 16) - temp + (1 << 15)) >> 16;
      }
  }
  
  static void harmonic_noise_sub(HFParam *hf, const int16_t *src, int16_t *dest)
  {
      int i;
      for (i = 0; i < SUBFRAME_LEN; i++) {
          int64_t temp = hf->gain * src[i - hf->index] << 1;
          dest[i] = av_clipl_int32(((dest[i] - src[i]) << 16) + temp +
                                   (1 << 15)) >> 16;
  
      }
  }
  
  /**
   * Combined synthesis and formant perceptual weighting filer.
   *
   * @param qnt_lpc  quantized lpc coefficients
   * @param perf_lpc perceptual filter coefficients
   * @param perf_fir perceptual filter fir memory
   * @param perf_iir perceptual filter iir memory
   * @param scale    the filter output will be scaled by 2^scale
   */
  static void synth_percept_filter(int16_t *qnt_lpc, int16_t *perf_lpc,
                                   int16_t *perf_fir, int16_t *perf_iir,
                                   const int16_t *src, int16_t *dest, int scale)
  {
      int i, j;
      int16_t buf_16[SUBFRAME_LEN + LPC_ORDER];
      int64_t buf[SUBFRAME_LEN];
  
      int16_t *bptr_16 = buf_16 + LPC_ORDER;
  
      memcpy(buf_16, perf_fir, sizeof(int16_t) * LPC_ORDER);
      memcpy(dest - LPC_ORDER, perf_iir, sizeof(int16_t) * LPC_ORDER);
  
      for (i = 0; i < SUBFRAME_LEN; i++) {
          int64_t temp = 0;
          for (j = 1; j <= LPC_ORDER; j++)
              temp -= qnt_lpc[j - 1] * bptr_16[i - j];
  
          buf[i]     = (src[i] << 15) + (temp << 3);
          bptr_16[i] = av_clipl_int32(buf[i] + (1 << 15)) >> 16;
      }
  
      for (i = 0; i < SUBFRAME_LEN; i++) {
          int64_t fir = 0, iir = 0;
          for (j = 1; j <= LPC_ORDER; j++) {
              fir -= perf_lpc[j - 1] * bptr_16[i - j];
              iir += perf_lpc[j + LPC_ORDER - 1] * dest[i - j];
          }
          dest[i] = av_clipl_int32(((buf[i] + (fir << 3)) << scale) + (iir << 3) +
                                   (1 << 15)) >> 16;
      }
      memcpy(perf_fir, buf_16 + SUBFRAME_LEN, sizeof(int16_t) * LPC_ORDER);
      memcpy(perf_iir, dest + SUBFRAME_LEN - LPC_ORDER,
             sizeof(int16_t) * LPC_ORDER);
  }
  
  /**
   * Compute the adaptive codebook contribution.
   *
   * @param buf   input signal
   * @param index the current subframe index
   */
  static void acb_search(G723_1_Context *p, int16_t *residual,
                         int16_t *impulse_resp, const int16_t *buf,
                         int index)
  {
  
      int16_t flt_buf[PITCH_ORDER][SUBFRAME_LEN];
  
      const int16_t *cb_tbl = adaptive_cb_gain85;
  
      int ccr_buf[PITCH_ORDER * SUBFRAMES << 2];
  
      int pitch_lag = p->pitch_lag[index >> 1];
      int acb_lag   = 1;
      int acb_gain  = 0;
      int odd_frame = index & 1;
      int iter      = 3 + odd_frame;
      int count     = 0;
      int tbl_size  = 85;
  
      int i, j, k, l, max;
      int64_t temp;
  
      if (!odd_frame) {
          if (pitch_lag == PITCH_MIN)
              pitch_lag++;
          else
              pitch_lag = FFMIN(pitch_lag, PITCH_MAX - 5);
      }
  
      for (i = 0; i < iter; i++) {
          get_residual(residual, p->prev_excitation, pitch_lag + i - 1);
  
          for (j = 0; j < SUBFRAME_LEN; j++) {
              temp = 0;
              for (k = 0; k <= j; k++)
                  temp += residual[PITCH_ORDER - 1 + k] * impulse_resp[j - k];
              flt_buf[PITCH_ORDER - 1][j] = av_clipl_int32((temp << 1) +
                                                           (1 << 15)) >> 16;
          }
  
          for (j = PITCH_ORDER - 2; j >= 0; j--) {
              flt_buf[j][0] = ((residual[j] << 13) + (1 << 14)) >> 15;
              for (k = 1; k < SUBFRAME_LEN; k++) {
                  temp = (flt_buf[j + 1][k - 1] << 15) +
                         residual[j] * impulse_resp[k];
                  flt_buf[j][k] = av_clipl_int32((temp << 1) + (1 << 15)) >> 16;
              }
          }
  
          /* Compute crosscorrelation with the signal */
          for (j = 0; j < PITCH_ORDER; j++) {
              temp = ff_dot_product(buf, flt_buf[j], SUBFRAME_LEN);
              ccr_buf[count++] = av_clipl_int32(temp << 1);
          }
  
          /* Compute energies */
          for (j = 0; j < PITCH_ORDER; j++) {
              ccr_buf[count++] = dot_product(flt_buf[j], flt_buf[j],
                                             SUBFRAME_LEN);
          }
  
          for (j = 1; j < PITCH_ORDER; j++) {
              for (k = 0; k < j; k++) {
                  temp = ff_dot_product(flt_buf[j], flt_buf[k], SUBFRAME_LEN);
                  ccr_buf[count++] = av_clipl_int32(temp<<2);
              }
          }
      }
  
      /* Normalize and shorten */
      max = 0;
      for (i = 0; i < 20 * iter; i++)
          max = FFMAX(max, FFABS(ccr_buf[i]));
  
      temp = normalize_bits_int32(max);
  
      for (i = 0; i < 20 * iter; i++){
          ccr_buf[i] = av_clipl_int32((int64_t)(ccr_buf[i] << temp) +
                                      (1 << 15)) >> 16;
      }
  
      max = 0;
      for (i = 0; i < iter; i++) {
          /* Select quantization table */
          if (!odd_frame && pitch_lag + i - 1 >= SUBFRAME_LEN - 2 ||
              odd_frame && pitch_lag >= SUBFRAME_LEN - 2) {
              cb_tbl = adaptive_cb_gain170;
              tbl_size = 170;
          }
  
          for (j = 0, k = 0; j < tbl_size; j++, k += 20) {
              temp = 0;
              for (l = 0; l < 20; l++)
                  temp += ccr_buf[20 * i + l] * cb_tbl[k + l];
              temp =  av_clipl_int32(temp);
  
              if (temp > max) {
                  max      = temp;
                  acb_gain = j;
                  acb_lag  = i;
              }
          }
      }
  
      if (!odd_frame) {
          pitch_lag += acb_lag - 1;
          acb_lag   =  1;
      }
  
      p->pitch_lag[index >> 1]      = pitch_lag;
      p->subframe[index].ad_cb_lag  = acb_lag;
      p->subframe[index].ad_cb_gain = acb_gain;
  }
  
  /**
   * Subtract the adaptive codebook contribution from the input
   * to obtain the residual.
   *
   * @param buf target vector
   */
  static void sub_acb_contrib(const int16_t *residual, const int16_t *impulse_resp,
                              int16_t *buf)
  {
      int i, j;
      /* Subtract adaptive CB contribution to obtain the residual */
      for (i = 0; i < SUBFRAME_LEN; i++) {
          int64_t temp = buf[i] << 14;
          for (j = 0; j <= i; j++)
              temp -= residual[j] * impulse_resp[i - j];
  
          buf[i] = av_clipl_int32((temp << 2) + (1 << 15)) >> 16;
      }
  }
  
  /**
   * Quantize the residual signal using the fixed codebook (MP-MLQ).
   *
   * @param optim optimized fixed codebook parameters
   * @param buf   excitation vector
   */
  static void get_fcb_param(FCBParam *optim, int16_t *impulse_resp,
                            int16_t *buf, int pulse_cnt, int pitch_lag)
  {
      FCBParam param;
      int16_t impulse_r[SUBFRAME_LEN];
      int16_t temp_corr[SUBFRAME_LEN];
      int16_t impulse_corr[SUBFRAME_LEN];
  
      int ccr1[SUBFRAME_LEN];
      int ccr2[SUBFRAME_LEN];
      int amp, err, max, max_amp_index, min, scale, i, j, k, l;
  
      int64_t temp;
  
      /* Update impulse response */
      memcpy(impulse_r, impulse_resp, sizeof(int16_t) * SUBFRAME_LEN);
      param.dirac_train = 0;
      if (pitch_lag < SUBFRAME_LEN - 2) {
          param.dirac_train = 1;
          gen_dirac_train(impulse_r, pitch_lag);
      }
  
      for (i = 0; i < SUBFRAME_LEN; i++)
          temp_corr[i] = impulse_r[i] >> 1;
  
      /* Compute impulse response autocorrelation */
      temp = dot_product(temp_corr, temp_corr, SUBFRAME_LEN);
  
      scale = normalize_bits_int32(temp);
      impulse_corr[0] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
  
      for (i = 1; i < SUBFRAME_LEN; i++) {
          temp = dot_product(temp_corr + i, temp_corr, SUBFRAME_LEN - i);
          impulse_corr[i] = av_clipl_int32((temp << scale) + (1 << 15)) >> 16;
      }
  
      /* Compute crosscorrelation of impulse response with residual signal */
      scale -= 4;
      for (i = 0; i < SUBFRAME_LEN; i++){
          temp = dot_product(buf + i, impulse_r, SUBFRAME_LEN - i);
          if (scale < 0)
              ccr1[i] = temp >> -scale;
          else
              ccr1[i] = av_clipl_int32(temp << scale);
      }
  
      /* Search loop */
      for (i = 0; i < GRID_SIZE; i++) {
          /* Maximize the crosscorrelation */
          max = 0;
          for (j = i; j < SUBFRAME_LEN; j += GRID_SIZE) {
              temp = FFABS(ccr1[j]);
              if (temp >= max) {
                  max = temp;
                  param.pulse_pos[0] = j;
              }
          }
  
          /* Quantize the gain (max crosscorrelation/impulse_corr[0]) */
          amp = max;
          min = 1 << 30;
          max_amp_index = GAIN_LEVELS - 2;
          for (j = max_amp_index; j >= 2; j--) {
              temp = av_clipl_int32((int64_t)fixed_cb_gain[j] *
                                    impulse_corr[0] << 1);
              temp = FFABS(temp - amp);
              if (temp < min) {
                  min = temp;
                  max_amp_index = j;
              }
          }
  
          max_amp_index--;
          /* Select additional gain values */
          for (j = 1; j < 5; j++) {
              for (k = i; k < SUBFRAME_LEN; k += GRID_SIZE) {
                  temp_corr[k] = 0;
                  ccr2[k]      = ccr1[k];
              }
              param.amp_index = max_amp_index + j - 2;
              amp = fixed_cb_gain[param.amp_index];
  
              param.pulse_sign[0] = (ccr2[param.pulse_pos[0]] < 0) ? -amp : amp;
              temp_corr[param.pulse_pos[0]] = 1;
  
              for (k = 1; k < pulse_cnt; k++) {
                  max = -1 << 30;
                  for (l = i; l < SUBFRAME_LEN; l += GRID_SIZE) {
                      if (temp_corr[l])
                          continue;
                      temp = impulse_corr[FFABS(l - param.pulse_pos[k - 1])];
                      temp = av_clipl_int32((int64_t)temp *
                                            param.pulse_sign[k - 1] << 1);
                      ccr2[l] -= temp;
                      temp = FFABS(ccr2[l]);
                      if (temp > max) {
                          max = temp;
                          param.pulse_pos[k] = l;
                      }
                  }
  
                  param.pulse_sign[k] = (ccr2[param.pulse_pos[k]] < 0) ?
                                        -amp : amp;
                  temp_corr[param.pulse_pos[k]] = 1;
              }
  
              /* Create the error vector */
              memset(temp_corr, 0, sizeof(int16_t) * SUBFRAME_LEN);
  
              for (k = 0; k < pulse_cnt; k++)
                  temp_corr[param.pulse_pos[k]] = param.pulse_sign[k];
  
              for (k = SUBFRAME_LEN - 1; k >= 0; k--) {
                  temp = 0;
                  for (l = 0; l <= k; l++) {
                      int prod = av_clipl_int32((int64_t)temp_corr[l] *
                                                impulse_r[k - l] << 1);
                      temp     = av_clipl_int32(temp + prod);
                  }
                  temp_corr[k] = temp << 2 >> 16;
              }
  
              /* Compute square of error */
              err = 0;
              for (k = 0; k < SUBFRAME_LEN; k++) {
                  int64_t prod;
                  prod = av_clipl_int32((int64_t)buf[k] * temp_corr[k] << 1);
                  err  = av_clipl_int32(err - prod);
                  prod = av_clipl_int32((int64_t)temp_corr[k] * temp_corr[k]);
                  err  = av_clipl_int32(err + prod);
              }
  
              /* Minimize */
              if (err < optim->min_err) {
                  optim->min_err     = err;
                  optim->grid_index  = i;
                  optim->amp_index   = param.amp_index;
                  optim->dirac_train = param.dirac_train;
  
                  for (k = 0; k < pulse_cnt; k++) {
                      optim->pulse_sign[k] = param.pulse_sign[k];
                      optim->pulse_pos[k]  = param.pulse_pos[k];
                  }
              }
          }
      }
  }
  
  /**
   * Encode the pulse position and gain of the current subframe.
   *
   * @param optim optimized fixed CB parameters
   * @param buf   excitation vector
   */
  static void pack_fcb_param(G723_1_Subframe *subfrm, FCBParam *optim,
                             int16_t *buf, int pulse_cnt)
  {
      int i, j;
  
      j = PULSE_MAX - pulse_cnt;
  
      subfrm->pulse_sign = 0;
      subfrm->pulse_pos  = 0;
  
      for (i = 0; i < SUBFRAME_LEN >> 1; i++) {
          int val = buf[optim->grid_index + (i << 1)];
          if (!val) {
              subfrm->pulse_pos += combinatorial_table[j][i];
          } else {
              subfrm->pulse_sign <<= 1;
              if (val < 0) subfrm->pulse_sign++;
              j++;
  
              if (j == PULSE_MAX) break;
          }
      }
      subfrm->amp_index   = optim->amp_index;
      subfrm->grid_index  = optim->grid_index;
      subfrm->dirac_train = optim->dirac_train;
  }
  
  /**
   * Compute the fixed codebook excitation.
   *
   * @param buf          target vector
   * @param impulse_resp impulse response of the combined filter
   */
  static void fcb_search(G723_1_Context *p, int16_t *impulse_resp,
                         int16_t *buf, int index)
  {
      FCBParam optim;
      int pulse_cnt = pulses[index];
      int i;
  
      optim.min_err = 1 << 30;
      get_fcb_param(&optim, impulse_resp, buf, pulse_cnt, SUBFRAME_LEN);
  
      if (p->pitch_lag[index >> 1] < SUBFRAME_LEN - 2) {
          get_fcb_param(&optim, impulse_resp, buf, pulse_cnt,
                        p->pitch_lag[index >> 1]);
      }
  
      /* Reconstruct the excitation */
      memset(buf, 0, sizeof(int16_t) * SUBFRAME_LEN);
      for (i = 0; i < pulse_cnt; i++)
          buf[optim.pulse_pos[i]] = optim.pulse_sign[i];
  
      pack_fcb_param(&p->subframe[index], &optim, buf, pulse_cnt);
  
      if (optim.dirac_train)
          gen_dirac_train(buf, p->pitch_lag[index >> 1]);
  }
  
  /**
   * Pack the frame parameters into output bitstream.
   *
   * @param frame output buffer
   * @param size  size of the buffer
   */
  static int pack_bitstream(G723_1_Context *p, unsigned char *frame, int size)
  {
      PutBitContext pb;
      int info_bits, i, temp;
  
      init_put_bits(&pb, frame, size);
  
      if (p->cur_rate == RATE_6300) {
          info_bits = 0;
          put_bits(&pb, 2, info_bits);
      }else
          av_assert0(0);
  
      put_bits(&pb, 8, p->lsp_index[2]);
      put_bits(&pb, 8, p->lsp_index[1]);
      put_bits(&pb, 8, p->lsp_index[0]);
  
      put_bits(&pb, 7, p->pitch_lag[0] - PITCH_MIN);
      put_bits(&pb, 2, p->subframe[1].ad_cb_lag);
      put_bits(&pb, 7, p->pitch_lag[1] - PITCH_MIN);
      put_bits(&pb, 2, p->subframe[3].ad_cb_lag);
  
      /* Write 12 bit combined gain */
      for (i = 0; i < SUBFRAMES; i++) {
          temp = p->subframe[i].ad_cb_gain * GAIN_LEVELS +
                 p->subframe[i].amp_index;
          if (p->cur_rate ==  RATE_6300)
              temp += p->subframe[i].dirac_train << 11;
          put_bits(&pb, 12, temp);
      }
  
      put_bits(&pb, 1, p->subframe[0].grid_index);
      put_bits(&pb, 1, p->subframe[1].grid_index);
      put_bits(&pb, 1, p->subframe[2].grid_index);
      put_bits(&pb, 1, p->subframe[3].grid_index);
  
      if (p->cur_rate == RATE_6300) {
          skip_put_bits(&pb, 1); /* reserved bit */
  
          /* Write 13 bit combined position index */
          temp = (p->subframe[0].pulse_pos >> 16) * 810 +
                 (p->subframe[1].pulse_pos >> 14) *  90 +
                 (p->subframe[2].pulse_pos >> 16) *   9 +
                 (p->subframe[3].pulse_pos >> 14);
          put_bits(&pb, 13, temp);
  
          put_bits(&pb, 16, p->subframe[0].pulse_pos & 0xffff);
          put_bits(&pb, 14, p->subframe[1].pulse_pos & 0x3fff);
          put_bits(&pb, 16, p->subframe[2].pulse_pos & 0xffff);
          put_bits(&pb, 14, p->subframe[3].pulse_pos & 0x3fff);
  
          put_bits(&pb, 6, p->subframe[0].pulse_sign);
          put_bits(&pb, 5, p->subframe[1].pulse_sign);
          put_bits(&pb, 6, p->subframe[2].pulse_sign);
          put_bits(&pb, 5, p->subframe[3].pulse_sign);
      }
  
      flush_put_bits(&pb);
      return frame_size[info_bits];
  }
  
  static int g723_1_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                              const AVFrame *frame, int *got_packet_ptr)
  {
      G723_1_Context *p = avctx->priv_data;
      int16_t unq_lpc[LPC_ORDER * SUBFRAMES];
      int16_t qnt_lpc[LPC_ORDER * SUBFRAMES];
      int16_t cur_lsp[LPC_ORDER];
      int16_t weighted_lpc[LPC_ORDER * SUBFRAMES << 1];
      int16_t vector[FRAME_LEN + PITCH_MAX];
      int offset, ret;
      int16_t *in_orig = av_memdup(frame->data[0], frame->nb_samples * sizeof(int16_t));
      int16_t *in = in_orig;
  
      HFParam hf[4];
      int i, j;
  
      if (!in)
          return AVERROR(ENOMEM);
  
      highpass_filter(in, &p->hpf_fir_mem, &p->hpf_iir_mem);
  
      memcpy(vector, p->prev_data, HALF_FRAME_LEN * sizeof(int16_t));
      memcpy(vector + HALF_FRAME_LEN, in, FRAME_LEN * sizeof(int16_t));
  
      comp_lpc_coeff(vector, unq_lpc);
      lpc2lsp(&unq_lpc[LPC_ORDER * 3], p->prev_lsp, cur_lsp);
      lsp_quantize(p->lsp_index, cur_lsp, p->prev_lsp);
  
      /* Update memory */
      memcpy(vector + LPC_ORDER, p->prev_data + SUBFRAME_LEN,
             sizeof(int16_t) * SUBFRAME_LEN);
      memcpy(vector + LPC_ORDER + SUBFRAME_LEN, in,
             sizeof(int16_t) * (HALF_FRAME_LEN + SUBFRAME_LEN));
      memcpy(p->prev_data, in + HALF_FRAME_LEN,
             sizeof(int16_t) * HALF_FRAME_LEN);
      memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
  
      perceptual_filter(p, weighted_lpc, unq_lpc, vector);
  
      memcpy(in, vector + LPC_ORDER, sizeof(int16_t) * FRAME_LEN);
      memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
      memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
  
      scale_vector(vector, vector, FRAME_LEN + PITCH_MAX);
  
      p->pitch_lag[0] = estimate_pitch(vector, PITCH_MAX);
      p->pitch_lag[1] = estimate_pitch(vector, PITCH_MAX + HALF_FRAME_LEN);
  
      for (i = PITCH_MAX, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
          comp_harmonic_coeff(vector + i, p->pitch_lag[j >> 1], hf + j);
  
      memcpy(vector, p->prev_weight_sig, sizeof(int16_t) * PITCH_MAX);
      memcpy(vector + PITCH_MAX, in, sizeof(int16_t) * FRAME_LEN);
      memcpy(p->prev_weight_sig, vector + FRAME_LEN, sizeof(int16_t) * PITCH_MAX);
  
      for (i = 0, j = 0; j < SUBFRAMES; i += SUBFRAME_LEN, j++)
          harmonic_filter(hf + j, vector + PITCH_MAX + i, in + i);
  
      inverse_quant(cur_lsp, p->prev_lsp, p->lsp_index, 0);
      lsp_interpolate(qnt_lpc, cur_lsp, p->prev_lsp);
  
      memcpy(p->prev_lsp, cur_lsp, sizeof(int16_t) * LPC_ORDER);
  
      offset = 0;
      for (i = 0; i < SUBFRAMES; i++) {
          int16_t impulse_resp[SUBFRAME_LEN];
          int16_t residual[SUBFRAME_LEN + PITCH_ORDER - 1];
          int16_t flt_in[SUBFRAME_LEN];
          int16_t zero[LPC_ORDER], fir[LPC_ORDER], iir[LPC_ORDER];
  
          /**
           * Compute the combined impulse response of the synthesis filter,
           * formant perceptual weighting filter and harmonic noise shaping filter
           */
          memset(zero, 0, sizeof(int16_t) * LPC_ORDER);
          memset(vector, 0, sizeof(int16_t) * PITCH_MAX);
          memset(flt_in, 0, sizeof(int16_t) * SUBFRAME_LEN);
  
          flt_in[0] = 1 << 13; /* Unit impulse */
          synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
                               zero, zero, flt_in, vector + PITCH_MAX, 1);
          harmonic_filter(hf + i, vector + PITCH_MAX, impulse_resp);
  
           /* Compute the combined zero input response */
          flt_in[0] = 0;
          memcpy(fir, p->perf_fir_mem, sizeof(int16_t) * LPC_ORDER);
          memcpy(iir, p->perf_iir_mem, sizeof(int16_t) * LPC_ORDER);
  
          synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
                               fir, iir, flt_in, vector + PITCH_MAX, 0);
          memcpy(vector, p->harmonic_mem, sizeof(int16_t) * PITCH_MAX);
          harmonic_noise_sub(hf + i, vector + PITCH_MAX, in);
  
          acb_search(p, residual, impulse_resp, in, i);
          gen_acb_excitation(residual, p->prev_excitation,p->pitch_lag[i >> 1],
                             &p->subframe[i], p->cur_rate);
          sub_acb_contrib(residual, impulse_resp, in);
  
          fcb_search(p, impulse_resp, in, i);
  
          /* Reconstruct the excitation */
          gen_acb_excitation(impulse_resp, p->prev_excitation, p->pitch_lag[i >> 1],
                             &p->subframe[i], RATE_6300);
  
          memmove(p->prev_excitation, p->prev_excitation + SUBFRAME_LEN,
                 sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
          for (j = 0; j < SUBFRAME_LEN; j++)
              in[j] = av_clip_int16((in[j] << 1) + impulse_resp[j]);
          memcpy(p->prev_excitation + PITCH_MAX - SUBFRAME_LEN, in,
                 sizeof(int16_t) * SUBFRAME_LEN);
  
          /* Update filter memories */
          synth_percept_filter(qnt_lpc + offset, weighted_lpc + (offset << 1),
                               p->perf_fir_mem, p->perf_iir_mem,
                               in, vector + PITCH_MAX, 0);
          memmove(p->harmonic_mem, p->harmonic_mem + SUBFRAME_LEN,
                  sizeof(int16_t) * (PITCH_MAX - SUBFRAME_LEN));
          memcpy(p->harmonic_mem + PITCH_MAX - SUBFRAME_LEN, vector + PITCH_MAX,
                 sizeof(int16_t) * SUBFRAME_LEN);
  
          in += SUBFRAME_LEN;
          offset += LPC_ORDER;
      }
  
      av_freep(&in_orig); in = NULL;
  
      if ((ret = ff_alloc_packet2(avctx, avpkt, 24)) < 0)
          return ret;
  
      *got_packet_ptr = 1;
      avpkt->size = pack_bitstream(p, avpkt->data, avpkt->size);
      return 0;
  }
  
  AVCodec ff_g723_1_encoder = {
      .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_encode_init,
      .encode2        = g723_1_encode_frame,
      .sample_fmts    = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
                                                      AV_SAMPLE_FMT_NONE},
  };
  #endif
  