text: /*
   * SIPR / ACELP.NET decoder
   *
   * Copyright (c) 2008 Vladimir Voroshilov
   * Copyright (c) 2009 Vitor Sessak
   *
   * 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
   */
  
  #include <math.h>
  #include <stdint.h>
  
  #include "libavutil/mathematics.h"
  #include "avcodec.h"
  #define ALT_BITSTREAM_READER_LE
  #include "get_bits.h"
  #include "dsputil.h"
  
  #include "lsp.h"
  #include "celp_math.h"
  #include "acelp_vectors.h"
  #include "acelp_pitch_delay.h"
  #include "acelp_filters.h"
  #include "celp_filters.h"
  
  #define MAX_SUBFRAME_COUNT   5
  
  #include "sipr.h"
  #include "siprdata.h"
  
  typedef struct {
      const char *mode_name;
      uint16_t bits_per_frame;
      uint8_t subframe_count;
      uint8_t frames_per_packet;
      float pitch_sharp_factor;
  
      /* bitstream parameters */
      uint8_t number_of_fc_indexes;
      uint8_t ma_predictor_bits;  ///< size in bits of the switched MA predictor
  
      /** size in bits of the i-th stage vector of quantizer */
      uint8_t vq_indexes_bits[5];
  
      /** size in bits of the adaptive-codebook index for every subframe */
      uint8_t pitch_delay_bits[5];
  
      uint8_t gp_index_bits;
      uint8_t fc_index_bits[10]; ///< size in bits of the fixed codebook indexes
      uint8_t gc_index_bits;     ///< size in bits of the gain  codebook indexes
  } SiprModeParam;
  
  static const SiprModeParam modes[MODE_COUNT] = {
      [MODE_16k] = {
          .mode_name          = "16k",
          .bits_per_frame     = 160,
          .subframe_count     = SUBFRAME_COUNT_16k,
          .frames_per_packet  = 1,
          .pitch_sharp_factor = 0.00,
  
          .number_of_fc_indexes = 10,
          .ma_predictor_bits    = 1,
          .vq_indexes_bits      = {7, 8, 7, 7, 7},
          .pitch_delay_bits     = {9, 6},
          .gp_index_bits        = 4,
          .fc_index_bits        = {4, 5, 4, 5, 4, 5, 4, 5, 4, 5},
          .gc_index_bits        = 5
      },
  
      [MODE_8k5] = {
          .mode_name          = "8k5",
          .bits_per_frame     = 152,
          .subframe_count     = 3,
          .frames_per_packet  = 1,
          .pitch_sharp_factor = 0.8,
  
          .number_of_fc_indexes = 3,
          .ma_predictor_bits    = 0,
          .vq_indexes_bits      = {6, 7, 7, 7, 5},
          .pitch_delay_bits     = {8, 5, 5},
          .gp_index_bits        = 0,
          .fc_index_bits        = {9, 9, 9},
          .gc_index_bits        = 7
      },
  
      [MODE_6k5] = {
          .mode_name          = "6k5",
          .bits_per_frame     = 232,
          .subframe_count     = 3,
          .frames_per_packet  = 2,
          .pitch_sharp_factor = 0.8,
  
          .number_of_fc_indexes = 3,
          .ma_predictor_bits    = 0,
          .vq_indexes_bits      = {6, 7, 7, 7, 5},
          .pitch_delay_bits     = {8, 5, 5},
          .gp_index_bits        = 0,
          .fc_index_bits        = {5, 5, 5},
          .gc_index_bits        = 7
      },
  
      [MODE_5k0] = {
          .mode_name          = "5k0",
          .bits_per_frame     = 296,
          .subframe_count     = 5,
          .frames_per_packet  = 2,
          .pitch_sharp_factor = 0.85,
  
          .number_of_fc_indexes = 1,
          .ma_predictor_bits    = 0,
          .vq_indexes_bits      = {6, 7, 7, 7, 5},
          .pitch_delay_bits     = {8, 5, 8, 5, 5},
          .gp_index_bits        = 0,
          .fc_index_bits        = {10},
          .gc_index_bits        = 7
      }
  };
  
  const float ff_pow_0_5[] = {
      1.0/(1 <<  1), 1.0/(1 <<  2), 1.0/(1 <<  3), 1.0/(1 <<  4),
      1.0/(1 <<  5), 1.0/(1 <<  6), 1.0/(1 <<  7), 1.0/(1 <<  8),
      1.0/(1 <<  9), 1.0/(1 << 10), 1.0/(1 << 11), 1.0/(1 << 12),
      1.0/(1 << 13), 1.0/(1 << 14), 1.0/(1 << 15), 1.0/(1 << 16)
  };
  
  static void dequant(float *out, const int *idx, const float *cbs[])
  {
      int i;
      int stride  = 2;
      int num_vec = 5;
  
      for (i = 0; i < num_vec; i++)
          memcpy(out + stride*i, cbs[i] + stride*idx[i], stride*sizeof(float));
  
  }
  
  static void lsf_decode_fp(float *lsfnew, float *lsf_history,
                            const SiprParameters *parm)
  {
      int i;
      float lsf_tmp[LP_FILTER_ORDER];
  
      dequant(lsf_tmp, parm->vq_indexes, lsf_codebooks);
  
      for (i = 0; i < LP_FILTER_ORDER; i++)
          lsfnew[i] = lsf_history[i] * 0.33 + lsf_tmp[i] + mean_lsf[i];
  
      ff_sort_nearly_sorted_floats(lsfnew, LP_FILTER_ORDER - 1);
  
      /* Note that a minimum distance is not enforced between the last value and
         the previous one, contrary to what is done in ff_acelp_reorder_lsf() */
      ff_set_min_dist_lsf(lsfnew, LSFQ_DIFF_MIN, LP_FILTER_ORDER - 1);
      lsfnew[9] = FFMIN(lsfnew[LP_FILTER_ORDER - 1], 1.3 * M_PI);
  
      memcpy(lsf_history, lsf_tmp, LP_FILTER_ORDER * sizeof(*lsf_history));
  
      for (i = 0; i < LP_FILTER_ORDER - 1; i++)
          lsfnew[i] = cos(lsfnew[i]);
      lsfnew[LP_FILTER_ORDER - 1] *= 6.153848 / M_PI;
  }
  
  /** Apply pitch lag to the fixed vector (AMR section 6.1.2). */
  static void pitch_sharpening(int pitch_lag_int, float beta,
                               float *fixed_vector)
  {
      int i;
  
      for (i = pitch_lag_int; i < SUBFR_SIZE; i++)
          fixed_vector[i] += beta * fixed_vector[i - pitch_lag_int];
  }
  
  /**
   * Extracts decoding parameters from the input bitstream.
   * @param parms          parameters structure
   * @param pgb            pointer to initialized GetBitContext structure
   */
  static void decode_parameters(SiprParameters* parms, GetBitContext *pgb,
                                const SiprModeParam *p)
  {
      int i, j;
  
      parms->ma_pred_switch           = get_bits(pgb, p->ma_predictor_bits);
  
      for (i = 0; i < 5; i++)
          parms->vq_indexes[i]        = get_bits(pgb, p->vq_indexes_bits[i]);
  
      for (i = 0; i < p->subframe_count; i++) {
          parms->pitch_delay[i]       = get_bits(pgb, p->pitch_delay_bits[i]);
          parms->gp_index[i]          = get_bits(pgb, p->gp_index_bits);
  
          for (j = 0; j < p->number_of_fc_indexes; j++)
              parms->fc_indexes[i][j] = get_bits(pgb, p->fc_index_bits[j]);
  
          parms->gc_index[i]          = get_bits(pgb, p->gc_index_bits);
      }
  }
  
  static void lsp2lpc_sipr(const double *lsp, float *Az)
  {
      int lp_half_order = LP_FILTER_ORDER >> 1;
      double buf[(LP_FILTER_ORDER >> 1) + 1];
      double pa[(LP_FILTER_ORDER >> 1) + 1];
      double *qa = buf + 1;
      int i,j;
  
      qa[-1] = 0.0;
  
      ff_lsp2polyf(lsp    , pa, lp_half_order    );
      ff_lsp2polyf(lsp + 1, qa, lp_half_order - 1);
  
      for (i = 1, j = LP_FILTER_ORDER - 1; i < lp_half_order; i++, j--) {
          double paf =  pa[i]            * (1 + lsp[LP_FILTER_ORDER - 1]);
          double qaf = (qa[i] - qa[i-2]) * (1 - lsp[LP_FILTER_ORDER - 1]);
          Az[i-1]  = (paf + qaf) * 0.5;
          Az[j-1]  = (paf - qaf) * 0.5;
      }
  
      Az[lp_half_order - 1] = (1.0 + lsp[LP_FILTER_ORDER - 1]) *
          pa[lp_half_order] * 0.5;
  
      Az[LP_FILTER_ORDER - 1] = lsp[LP_FILTER_ORDER - 1];
  }
  
  static void sipr_decode_lp(float *lsfnew, const float *lsfold, float *Az,
                             int num_subfr)
  {
      double lsfint[LP_FILTER_ORDER];
      int i,j;
      float t, t0 = 1.0 / num_subfr;
  
      t = t0 * 0.5;
      for (i = 0; i < num_subfr; i++) {
          for (j = 0; j < LP_FILTER_ORDER; j++)
              lsfint[j] = lsfold[j] * (1 - t) + t * lsfnew[j];
  
          lsp2lpc_sipr(lsfint, Az);
          Az += LP_FILTER_ORDER;
          t += t0;
      }
  }
  
  /**
   * Evaluates the adaptive impulse response.
   */
  static void eval_ir(const float *Az, int pitch_lag, float *freq,
                      float pitch_sharp_factor)
  {
      float tmp1[SUBFR_SIZE+1], tmp2[LP_FILTER_ORDER+1];
      int i;
  
      tmp1[0] = 1.;
      for (i = 0; i < LP_FILTER_ORDER; i++) {
          tmp1[i+1] = Az[i] * ff_pow_0_55[i];
          tmp2[i  ] = Az[i] * ff_pow_0_7 [i];
      }
      memset(tmp1 + 11, 0, 37 * sizeof(float));
  
      ff_celp_lp_synthesis_filterf(freq, tmp2, tmp1, SUBFR_SIZE,
                                   LP_FILTER_ORDER);
  
      pitch_sharpening(pitch_lag, pitch_sharp_factor, freq);
  }
  
  /**
   * Evaluates the convolution of a vector with a sparse vector.
   */
  static void convolute_with_sparse(float *out, const AMRFixed *pulses,
                                    const float *shape, int length)
  {
      int i, j;
  
      memset(out, 0, length*sizeof(float));
      for (i = 0; i < pulses->n; i++)
          for (j = pulses->x[i]; j < length; j++)
              out[j] += pulses->y[i] * shape[j - pulses->x[i]];
  }
  
  /**
   * Apply postfilter, very similar to AMR one.
   */
  static void postfilter_5k0(SiprContext *ctx, const float *lpc, float *samples)
  {
      float buf[SUBFR_SIZE + LP_FILTER_ORDER];
      float *pole_out = buf + LP_FILTER_ORDER;
      float lpc_n[LP_FILTER_ORDER];
      float lpc_d[LP_FILTER_ORDER];
      int i;
  
      for (i = 0; i < LP_FILTER_ORDER; i++) {
          lpc_d[i] = lpc[i] * ff_pow_0_75[i];
          lpc_n[i] = lpc[i] * ff_pow_0_5 [i];
      };
  
      memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem,
             LP_FILTER_ORDER*sizeof(float));
  
      ff_celp_lp_synthesis_filterf(pole_out, lpc_d, samples, SUBFR_SIZE,
                                   LP_FILTER_ORDER);
  
      memcpy(ctx->postfilter_mem, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
             LP_FILTER_ORDER*sizeof(float));
  
      ff_tilt_compensation(&ctx->tilt_mem, 0.4, pole_out, SUBFR_SIZE);
  
      memcpy(pole_out - LP_FILTER_ORDER, ctx->postfilter_mem5k0,
             LP_FILTER_ORDER*sizeof(*pole_out));
  
      memcpy(ctx->postfilter_mem5k0, pole_out + SUBFR_SIZE - LP_FILTER_ORDER,
             LP_FILTER_ORDER*sizeof(*pole_out));
  
      ff_celp_lp_zero_synthesis_filterf(samples, lpc_n, pole_out, SUBFR_SIZE,
                                        LP_FILTER_ORDER);
  
  }
  
  static void decode_fixed_sparse(AMRFixed *fixed_sparse, const int16_t *pulses,
                                  SiprMode mode, int low_gain)
  {
      int i;
  
      switch (mode) {
      case MODE_6k5:
          for (i = 0; i < 3; i++) {
              fixed_sparse->x[i] = 3 * (pulses[i] & 0xf) + i;
              fixed_sparse->y[i] = pulses[i] & 0x10 ? -1 : 1;
          }
          fixed_sparse->n = 3;
          break;
      case MODE_8k5:
          for (i = 0; i < 3; i++) {
              fixed_sparse->x[2*i    ] = 3 * ((pulses[i] >> 4) & 0xf) + i;
              fixed_sparse->x[2*i + 1] = 3 * ( pulses[i]       & 0xf) + i;
  
              fixed_sparse->y[2*i    ] = (pulses[i] & 0x100) ? -1.0: 1.0;
  
              fixed_sparse->y[2*i + 1] =
                  (fixed_sparse->x[2*i + 1] < fixed_sparse->x[2*i]) ?
                  -fixed_sparse->y[2*i    ] : fixed_sparse->y[2*i];
          }
  
          fixed_sparse->n = 6;
          break;
      case MODE_5k0:
      default:
          if (low_gain) {
              int offset = (pulses[0] & 0x200) ? 2 : 0;
              int val = pulses[0];
  
              for (i = 0; i < 3; i++) {
                  int index = (val & 0x7) * 6 + 4 - i*2;
  
                  fixed_sparse->y[i] = (offset + index) & 0x3 ? -1 : 1;
                  fixed_sparse->x[i] = index;
  
                  val >>= 3;
              }
              fixed_sparse->n = 3;
          } else {
              int pulse_subset = (pulses[0] >> 8) & 1;
  
              fixed_sparse->x[0] = ((pulses[0] >> 4) & 15) * 3 + pulse_subset;
              fixed_sparse->x[1] = ( pulses[0]       & 15) * 3 + pulse_subset + 1;
  
              fixed_sparse->y[0] = pulses[0] & 0x200 ? -1 : 1;
              fixed_sparse->y[1] = -fixed_sparse->y[0];
              fixed_sparse->n = 2;
          }
          break;
      }
  }
  
  static void decode_frame(SiprContext *ctx, SiprParameters *params,
                           float *out_data)
  {
      int i, j;
      int subframe_count = modes[ctx->mode].subframe_count;
      int frame_size = subframe_count * SUBFR_SIZE;
      float Az[LP_FILTER_ORDER * MAX_SUBFRAME_COUNT];
      float *excitation;
      float ir_buf[SUBFR_SIZE + LP_FILTER_ORDER];
      float lsf_new[LP_FILTER_ORDER];
      float *impulse_response = ir_buf + LP_FILTER_ORDER;
      float *synth = ctx->synth_buf + 16; // 16 instead of LP_FILTER_ORDER for
                                          // memory alignment
      int t0_first = 0;
      AMRFixed fixed_cb;
  
      memset(ir_buf, 0, LP_FILTER_ORDER * sizeof(float));
      lsf_decode_fp(lsf_new, ctx->lsf_history, params);
  
      sipr_decode_lp(lsf_new, ctx->lsp_history, Az, subframe_count);
  
      memcpy(ctx->lsp_history, lsf_new, LP_FILTER_ORDER * sizeof(float));
  
      excitation = ctx->excitation + PITCH_DELAY_MAX + L_INTERPOL;
  
      for (i = 0; i < subframe_count; i++) {
          float *pAz = Az + i*LP_FILTER_ORDER;
          float fixed_vector[SUBFR_SIZE];
          int T0,T0_frac;
          float pitch_gain, gain_code, avg_energy;
  
          ff_decode_pitch_lag(&T0, &T0_frac, params->pitch_delay[i], t0_first, i,
                              ctx->mode == MODE_5k0, 6);
  
          if (i == 0 || (i == 2 && ctx->mode == MODE_5k0))
              t0_first = T0;
  
          ff_acelp_interpolatef(excitation, excitation - T0 + (T0_frac <= 0),
                                ff_b60_sinc, 6,
                                2 * ((2 + T0_frac)%3 + 1), LP_FILTER_ORDER,
                                SUBFR_SIZE);
  
          decode_fixed_sparse(&fixed_cb, params->fc_indexes[i], ctx->mode,
                              ctx->past_pitch_gain < 0.8);
  
          eval_ir(pAz, T0, impulse_response, modes[ctx->mode].pitch_sharp_factor);
  
          convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response,
                                SUBFR_SIZE);
  
          avg_energy =
              (0.01 + ff_dot_productf(fixed_vector, fixed_vector, SUBFR_SIZE))/
                  SUBFR_SIZE;
  
          ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0];
  
          gain_code = ff_amr_set_fixed_gain(gain_cb[params->gc_index[i]][1],
                                            avg_energy, ctx->energy_history,
                                            34 - 15.0/(0.05*M_LN10/M_LN2),
                                            pred);
  
          ff_weighted_vector_sumf(excitation, excitation, fixed_vector,
                                  pitch_gain, gain_code, SUBFR_SIZE);
  
          pitch_gain *= 0.5 * pitch_gain;
          pitch_gain = FFMIN(pitch_gain, 0.4);
  
          ctx->gain_mem = 0.7 * ctx->gain_mem + 0.3 * pitch_gain;
          ctx->gain_mem = FFMIN(ctx->gain_mem, pitch_gain);
          gain_code *= ctx->gain_mem;
  
          for (j = 0; j < SUBFR_SIZE; j++)
              fixed_vector[j] = excitation[j] - gain_code * fixed_vector[j];
  
          if (ctx->mode == MODE_5k0) {
              postfilter_5k0(ctx, pAz, fixed_vector);
  
              ff_celp_lp_synthesis_filterf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
                                           pAz, excitation, SUBFR_SIZE,
                                           LP_FILTER_ORDER);
          }
  
          ff_celp_lp_synthesis_filterf(synth + i*SUBFR_SIZE, pAz, fixed_vector,
                                       SUBFR_SIZE, LP_FILTER_ORDER);
  
          excitation += SUBFR_SIZE;
      }
  
      memcpy(synth - LP_FILTER_ORDER, synth + frame_size - LP_FILTER_ORDER,
             LP_FILTER_ORDER * sizeof(float));
  
      if (ctx->mode == MODE_5k0) {
          for (i = 0; i < subframe_count; i++) {
              float energy = ff_dot_productf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
                                             ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
                                             SUBFR_SIZE);
              ff_adaptive_gain_control(&synth[i * SUBFR_SIZE],
                                       &synth[i * SUBFR_SIZE], energy,
                                       SUBFR_SIZE, 0.9, &ctx->postfilter_agc);
          }
  
          memcpy(ctx->postfilter_syn5k0, ctx->postfilter_syn5k0 + frame_size,
                 LP_FILTER_ORDER*sizeof(float));
      }
      memcpy(ctx->excitation, excitation - PITCH_DELAY_MAX - L_INTERPOL,
             (PITCH_DELAY_MAX + L_INTERPOL) * sizeof(float));
  
      ff_acelp_apply_order_2_transfer_function(out_data, synth,
                                               (const float[2]) {-1.99997   , 1.000000000},
                                               (const float[2]) {-1.93307352, 0.935891986},
                                               0.939805806,
                                               ctx->highpass_filt_mem,
                                               frame_size);
  }
  
  static av_cold int sipr_decoder_init(AVCodecContext * avctx)
  {
      SiprContext *ctx = avctx->priv_data;
      int i;
  
      if      (avctx->bit_rate > 12200) ctx->mode = MODE_16k;
      else if (avctx->bit_rate > 7500 ) ctx->mode = MODE_8k5;
      else if (avctx->bit_rate > 5750 ) ctx->mode = MODE_6k5;
      else                              ctx->mode = MODE_5k0;
  
      av_log(avctx, AV_LOG_DEBUG, "Mode: %s\n", modes[ctx->mode].mode_name);
  
      if (ctx->mode == MODE_16k)
          ff_sipr_init_16k(ctx);
  
      for (i = 0; i < LP_FILTER_ORDER; i++)
          ctx->lsp_history[i] = cos((i+1) * M_PI / (LP_FILTER_ORDER + 1));
  
      for (i = 0; i < 4; i++)
          ctx->energy_history[i] = -14;
  
      avctx->sample_fmt = SAMPLE_FMT_FLT;
  
      dsputil_init(&ctx->dsp, avctx);
  
      return 0;
  }
  
  static int sipr_decode_frame(AVCodecContext *avctx, void *datap,
                               int *data_size, AVPacket *avpkt)
  {
      SiprContext *ctx = avctx->priv_data;
      const uint8_t *buf=avpkt->data;
      SiprParameters parm;
      const SiprModeParam *mode_par = &modes[ctx->mode];
      GetBitContext gb;
      float *data = datap;
      int subframe_size = ctx->mode == MODE_16k ? L_SUBFR_16k : SUBFR_SIZE;
      int i;
  
      ctx->avctx = avctx;
      if (avpkt->size < (mode_par->bits_per_frame >> 3)) {
          av_log(avctx, AV_LOG_ERROR,
                 "Error processing packet: packet size (%d) too small\n",
                 avpkt->size);
  
          *data_size = 0;
          return -1;
      }
      if (*data_size < subframe_size * mode_par->subframe_count * sizeof(float)) {
          av_log(avctx, AV_LOG_ERROR,
                 "Error processing packet: output buffer (%d) too small\n",
                 *data_size);
  
          *data_size = 0;
          return -1;
      }
  
      init_get_bits(&gb, buf, mode_par->bits_per_frame);
  
      for (i = 0; i < mode_par->frames_per_packet; i++) {
          decode_parameters(&parm, &gb, mode_par);
  
          if (ctx->mode == MODE_16k)
              ff_sipr_decode_frame_16k(ctx, &parm, data);
          else
              decode_frame(ctx, &parm, data);
  
          data += subframe_size * mode_par->subframe_count;
      }
  
      *data_size = mode_par->frames_per_packet * subframe_size *
          mode_par->subframe_count * sizeof(float);
  
      return mode_par->bits_per_frame >> 3;
  };
  
  AVCodec sipr_decoder = {
      "sipr",
      AVMEDIA_TYPE_AUDIO,
      CODEC_ID_SIPR,
      sizeof(SiprContext),
      sipr_decoder_init,
      NULL,
      NULL,
      sipr_decode_frame,
      .long_name = NULL_IF_CONFIG_SMALL("RealAudio SIPR / ACELP.NET"),
  };
  