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   1 /*
   2  * QCELP decoder
   3  * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
   4  *
   5  * This file is part of Libav.
   6  *
   7  * Libav is free software; you can redistribute it and/or
   8  * modify it under the terms of the GNU Lesser General Public
   9  * License as published by the Free Software Foundation; either
  10  * version 2.1 of the License, or (at your option) any later version.
  11  *
  12  * Libav is distributed in the hope that it will be useful,
  13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15  * Lesser General Public License for more details.
  16  *
  17  * You should have received a copy of the GNU Lesser General Public
  18  * License along with Libav; if not, write to the Free Software
  19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20  */
  21
  22 /**
  23  * @file
  24  * QCELP decoder
  25  * @author Reynaldo H. Verdejo Pinochet
  26  * @remark Libav merging spearheaded by Kenan Gillet
  27  * @remark Development mentored by Benjamin Larson
  28  */
  29
  30 #include <stddef.h>
  31
  32 #include "avcodec.h"
  33 #include "internal.h"
  34 #include "get_bits.h"
  35
  36 #include "qcelpdata.h"
  37
  38 #include "celp_math.h"
  39 #include "celp_filters.h"
  40 #include "acelp_filters.h"
  41 #include "acelp_vectors.h"
  42 #include "lsp.h"
  43
  44 #undef NDEBUG
  45 #include <assert.h>
  46
  47 typedef enum
  48 {
  49     I_F_Q = -1,    /**< insufficient frame quality */
  50     SILENCE,
  51     RATE_OCTAVE,
  52     RATE_QUARTER,
  53     RATE_HALF,
  54     RATE_FULL
  55 } qcelp_packet_rate;
  56
  57 typedef struct
  58 {
  59     AVFrame           avframe;
  60     GetBitContext     gb;
  61     qcelp_packet_rate bitrate;
  62     QCELPFrame        frame;    /**< unpacked data frame */
  63
  64     uint8_t  erasure_count;
  65     uint8_t  octave_count;      /**< count the consecutive RATE_OCTAVE frames */
  66     float    prev_lspf[10];
  67     float    predictor_lspf[10];/**< LSP predictor for RATE_OCTAVE and I_F_Q */
  68     float    pitch_synthesis_filter_mem[303];
  69     float    pitch_pre_filter_mem[303];
  70     float    rnd_fir_filter_mem[180];
  71     float    formant_mem[170];
  72     float    last_codebook_gain;
  73     int      prev_g1[2];
  74     int      prev_bitrate;
  75     float    pitch_gain[4];
  76     uint8_t  pitch_lag[4];
  77     uint16_t first16bits;
  78     uint8_t  warned_buf_mismatch_bitrate;
  79
  80     /* postfilter */
  81     float    postfilter_synth_mem[10];
  82     float    postfilter_agc_mem;
  83     float    postfilter_tilt_mem;
  84 } QCELPContext;
  85
  86 /**
  87  * Initialize the speech codec according to the specification.
  88  *
  89  * TIA/EIA/IS-733 2.4.9
  90  */
  91 static av_cold int qcelp_decode_init(AVCodecContext *avctx)
  92 {
  93     QCELPContext *q = avctx->priv_data;
  94     int i;
  95
  96     avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
  97
  98     for(i=0; i<10; i++)
  99         q->prev_lspf[i] = (i+1)/11.;
 100
 101     avcodec_get_frame_defaults(&q->avframe);
 102     avctx->coded_frame = &q->avframe;
 103
 104     return 0;
 105 }
 106
 107 /**
 108  * Decode the 10 quantized LSP frequencies from the LSPV/LSP
 109  * transmission codes of any bitrate and check for badly received packets.
 110  *
 111  * @param q the context
 112  * @param lspf line spectral pair frequencies
 113  *
 114  * @return 0 on success, -1 if the packet is badly received
 115  *
 116  * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
 117  */
 118 static int decode_lspf(QCELPContext *q, float *lspf)
 119 {
 120     int i;
 121     float tmp_lspf, smooth, erasure_coeff;
 122     const float *predictors;
 123
 124     if (q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q) {
 125         predictors = (q->prev_bitrate != RATE_OCTAVE &&
 126                        q->prev_bitrate != I_F_Q ?
 127                        q->prev_lspf : q->predictor_lspf);
 128
 129         if (q->bitrate == RATE_OCTAVE) {
 130             q->octave_count++;
 131
 132             for (i=0; i<10; i++) {
 133                 q->predictor_lspf[i] =
 134                              lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
 135                                                          : -QCELP_LSP_SPREAD_FACTOR)
 136                                      + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR
 137                                      + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
 138             }
 139             smooth = (q->octave_count < 10 ? .875 : 0.1);
 140         } else {
 141             erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
 142
 143             assert(q->bitrate == I_F_Q);
 144
 145             if(q->erasure_count > 1)
 146                 erasure_coeff *= (q->erasure_count < 4 ? 0.9 : 0.7);
 147
 148             for(i = 0; i < 10; i++) {
 149                 q->predictor_lspf[i] =
 150                              lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
 151                                      + erasure_coeff * predictors[i];
 152             }
 153             smooth = 0.125;
 154         }
 155
 156         // Check the stability of the LSP frequencies.
 157         lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
 158         for(i=1; i<10; i++)
 159             lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR));
 160
 161         lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR));
 162         for(i=9; i>0; i--)
 163             lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
 164
 165         // Low-pass filter the LSP frequencies.
 166         ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
 167     } else {
 168         q->octave_count = 0;
 169
 170         tmp_lspf = 0.;
 171         for (i = 0; i < 5; i++) {
 172             lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
 173             lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
 174         }
 175
 176         // Check for badly received packets.
 177         if (q->bitrate == RATE_QUARTER) {
 178             if(lspf[9] <= .70 || lspf[9] >=  .97)
 179                 return -1;
 180             for(i=3; i<10; i++)
 181                 if(fabs(lspf[i] - lspf[i-2]) < .08)
 182                     return -1;
 183         } else {
 184             if(lspf[9] <= .66 || lspf[9] >= .985)
 185                 return -1;
 186             for(i=4; i<10; i++)
 187                 if (fabs(lspf[i] - lspf[i-4]) < .0931)
 188                     return -1;
 189         }
 190     }
 191     return 0;
 192 }
 193
 194 /**
 195  * Convert codebook transmission codes to GAIN and INDEX.
 196  *
 197  * @param q the context
 198  * @param gain array holding the decoded gain
 199  *
 200  * TIA/EIA/IS-733 2.4.6.2
 201  */
 202 static void decode_gain_and_index(QCELPContext  *q,
 203                                   float *gain) {
 204     int   i, subframes_count, g1[16];
 205     float slope;
 206
 207     if (q->bitrate >= RATE_QUARTER) {
 208         switch (q->bitrate) {
 209             case RATE_FULL: subframes_count = 16; break;
 210             case RATE_HALF: subframes_count = 4;  break;
 211             default:        subframes_count = 5;
 212         }
 213         for(i = 0; i < subframes_count; i++) {
 214             g1[i] = 4 * q->frame.cbgain[i];
 215             if (q->bitrate == RATE_FULL && !((i+1) & 3)) {
 216                 g1[i] += av_clip((g1[i-1] + g1[i-2] + g1[i-3]) / 3 - 6, 0, 32);
 217             }
 218
 219             gain[i] = qcelp_g12ga[g1[i]];
 220
 221             if (q->frame.cbsign[i]) {
 222                 gain[i] = -gain[i];
 223                 q->frame.cindex[i] = (q->frame.cindex[i]-89) & 127;
 224             }
 225         }
 226
 227         q->prev_g1[0] = g1[i-2];
 228         q->prev_g1[1] = g1[i-1];
 229         q->last_codebook_gain = qcelp_g12ga[g1[i-1]];
 230
 231         if (q->bitrate == RATE_QUARTER) {
 232             // Provide smoothing of the unvoiced excitation energy.
 233             gain[7] =     gain[4];
 234             gain[6] = 0.4*gain[3] + 0.6*gain[4];
 235             gain[5] =     gain[3];
 236             gain[4] = 0.8*gain[2] + 0.2*gain[3];
 237             gain[3] = 0.2*gain[1] + 0.8*gain[2];
 238             gain[2] =     gain[1];
 239             gain[1] = 0.6*gain[0] + 0.4*gain[1];
 240         }
 241     } else if (q->bitrate != SILENCE) {
 242         if (q->bitrate == RATE_OCTAVE) {
 243             g1[0] = 2 * q->frame.cbgain[0]
 244                   + av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
 245             subframes_count = 8;
 246         } else {
 247             assert(q->bitrate == I_F_Q);
 248
 249             g1[0] = q->prev_g1[1];
 250             switch (q->erasure_count) {
 251                 case 1 : break;
 252                 case 2 : g1[0] -= 1; break;
 253                 case 3 : g1[0] -= 2; break;
 254                 default: g1[0] -= 6;
 255             }
 256             if(g1[0] < 0)
 257                 g1[0] = 0;
 258             subframes_count = 4;
 259         }
 260         // This interpolation is done to produce smoother background noise.
 261         slope = 0.5*(qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
 262         for(i=1; i<=subframes_count; i++)
 263             gain[i-1] = q->last_codebook_gain + slope * i;
 264
 265         q->last_codebook_gain = gain[i-2];
 266         q->prev_g1[0] = q->prev_g1[1];
 267         q->prev_g1[1] = g1[0];
 268     }
 269 }
 270
 271 /**
 272  * If the received packet is Rate 1/4 a further sanity check is made of the
 273  * codebook gain.
 274  *
 275  * @param cbgain the unpacked cbgain array
 276  * @return -1 if the sanity check fails, 0 otherwise
 277  *
 278  * TIA/EIA/IS-733 2.4.8.7.3
 279  */
 280 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
 281 {
 282     int i, diff, prev_diff=0;
 283
 284     for(i=1; i<5; i++) {
 285         diff = cbgain[i] - cbgain[i-1];
 286         if(FFABS(diff) > 10)
 287             return -1;
 288         else if(FFABS(diff - prev_diff) > 12)
 289             return -1;
 290         prev_diff = diff;
 291     }
 292     return 0;
 293 }
 294
 295 /**
 296  * Compute the scaled codebook vector Cdn From INDEX and GAIN
 297  * for all rates.
 298  *
 299  * The specification lacks some information here.
 300  *
 301  * TIA/EIA/IS-733 has an omission on the codebook index determination
 302  * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
 303  * you have to subtract the decoded index parameter from the given scaled
 304  * codebook vector index 'n' to get the desired circular codebook index, but
 305  * it does not mention that you have to clamp 'n' to [0-9] in order to get
 306  * RI-compliant results.
 307  *
 308  * The reason for this mistake seems to be the fact they forgot to mention you
 309  * have to do these calculations per codebook subframe and adjust given
 310  * equation values accordingly.
 311  *
 312  * @param q the context
 313  * @param gain array holding the 4 pitch subframe gain values
 314  * @param cdn_vector array for the generated scaled codebook vector
 315  */
 316 static void compute_svector(QCELPContext *q, const float *gain,
 317                             float *cdn_vector)
 318 {
 319     int      i, j, k;
 320     uint16_t cbseed, cindex;
 321     float    *rnd, tmp_gain, fir_filter_value;
 322
 323     switch (q->bitrate) {
 324         case RATE_FULL:
 325             for (i = 0; i < 16; i++) {
 326                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
 327                 cindex = -q->frame.cindex[i];
 328                 for(j=0; j<10; j++)
 329                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
 330             }
 331         break;
 332         case RATE_HALF:
 333             for (i = 0; i < 4; i++) {
 334                 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
 335                 cindex = -q->frame.cindex[i];
 336                 for (j = 0; j < 40; j++)
 337                 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
 338             }
 339         break;
 340         case RATE_QUARTER:
 341             cbseed = (0x0003 & q->frame.lspv[4])<<14 |
 342                      (0x003F & q->frame.lspv[3])<< 8 |
 343                      (0x0060 & q->frame.lspv[2])<< 1 |
 344                      (0x0007 & q->frame.lspv[1])<< 3 |
 345                      (0x0038 & q->frame.lspv[0])>> 3 ;
 346             rnd = q->rnd_fir_filter_mem + 20;
 347             for (i = 0; i < 8; i++) {
 348                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
 349                 for (k = 0; k < 20; k++) {
 350                     cbseed = 521 * cbseed + 259;
 351                     *rnd = (int16_t)cbseed;
 352
 353                     // FIR filter
 354                     fir_filter_value = 0.0;
 355                     for(j=0; j<10; j++)
 356                         fir_filter_value += qcelp_rnd_fir_coefs[j ]
 357                                           * (rnd[-j ] + rnd[-20+j]);
 358
 359                     fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
 360                     *cdn_vector++ = tmp_gain * fir_filter_value;
 361                     rnd++;
 362                 }
 363             }
 364             memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
 365         break;
 366         case RATE_OCTAVE:
 367             cbseed = q->first16bits;
 368             for (i = 0; i < 8; i++) {
 369                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
 370                 for (j = 0; j < 20; j++) {
 371                     cbseed = 521 * cbseed + 259;
 372                     *cdn_vector++ = tmp_gain * (int16_t)cbseed;
 373                 }
 374             }
 375         break;
 376         case I_F_Q:
 377             cbseed = -44; // random codebook index
 378             for (i = 0; i < 4; i++) {
 379                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
 380                 for(j=0; j<40; j++)
 381                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
 382             }
 383         break;
 384         case SILENCE:
 385             memset(cdn_vector, 0, 160 * sizeof(float));
 386         break;
 387     }
 388 }
 389
 390 /**
 391  * Apply generic gain control.
 392  *
 393  * @param v_out output vector
 394  * @param v_in gain-controlled vector
 395  * @param v_ref vector to control gain of
 396  *
 397  * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
 398  */
 399 static void apply_gain_ctrl(float *v_out, const float *v_ref,
 400                             const float *v_in)
 401 {
 402     int i;
 403
 404     for (i = 0; i < 160; i += 40)
 405         ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
 406                                                 ff_dot_productf(v_ref + i,
 407                                                                 v_ref + i, 40),
 408                                                 40);
 409 }
 410
 411 /**
 412  * Apply filter in pitch-subframe steps.
 413  *
 414  * @param memory buffer for the previous state of the filter
 415  *        - must be able to contain 303 elements
 416  *        - the 143 first elements are from the previous state
 417  *        - the next 160 are for output
 418  * @param v_in input filter vector
 419  * @param gain per-subframe gain array, each element is between 0.0 and 2.0
 420  * @param lag per-subframe lag array, each element is
 421  *        - between 16 and 143 if its corresponding pfrac is 0,
 422  *        - between 16 and 139 otherwise
 423  * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
 424  *        otherwise
 425  *
 426  * @return filter output vector
 427  */
 428 static const float *do_pitchfilter(float memory[303], const float v_in[160],
 429                                    const float gain[4], const uint8_t *lag,
 430                                    const uint8_t pfrac[4])
 431 {
 432     int         i, j;
 433     float       *v_lag, *v_out;
 434     const float *v_len;
 435
 436     v_out = memory + 143; // Output vector starts at memory[143].
 437
 438     for (i = 0; i < 4; i++) {
 439         if (gain[i]) {
 440             v_lag = memory + 143 + 40 * i - lag[i];
 441             for (v_len = v_in + 40; v_in < v_len; v_in++) {
 442                 if (pfrac[i]) { // If it is a fractional lag...
 443                     for(j=0, *v_out=0.; j<4; j++)
 444                         *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
 445                 }else
 446                     *v_out = *v_lag;
 447
 448                 *v_out = *v_in + gain[i] * *v_out;
 449
 450                 v_lag++;
 451                 v_out++;
 452             }
 453         } else {
 454             memcpy(v_out, v_in, 40 * sizeof(float));
 455             v_in  += 40;
 456             v_out += 40;
 457         }
 458     }
 459
 460     memmove(memory, memory + 160, 143 * sizeof(float));
 461     return memory + 143;
 462 }
 463
 464 /**
 465  * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
 466  * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
 467  *
 468  * @param q the context
 469  * @param cdn_vector the scaled codebook vector
 470  */
 471 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
 472 {
 473     int         i;
 474     const float *v_synthesis_filtered, *v_pre_filtered;
 475
 476     if(q->bitrate >= RATE_HALF ||
 477        q->bitrate == SILENCE ||
 478       (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {
 479
 480         if(q->bitrate >= RATE_HALF) {
 481
 482             // Compute gain & lag for the whole frame.
 483             for (i = 0; i < 4; i++) {
 484                 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
 485
 486                 q->pitch_lag[i] = q->frame.plag[i] + 16;
 487             }
 488         } else {
 489             float max_pitch_gain;
 490
 491             if (q->bitrate == I_F_Q) {
 492                   if (q->erasure_count < 3)
 493                       max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
 494                   else
 495                       max_pitch_gain = 0.0;
 496             } else {
 497                 assert(q->bitrate == SILENCE);
 498                 max_pitch_gain = 1.0;
 499             }
 500             for(i=0; i<4; i++)
 501                 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
 502
 503             memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
 504         }
 505
 506         // pitch synthesis filter
 507         v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
 508                                               cdn_vector, q->pitch_gain,
 509                                               q->pitch_lag, q->frame.pfrac);
 510
 511         // pitch prefilter update
 512         for(i=0; i<4; i++)
 513             q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
 514
 515         v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
 516                                         v_synthesis_filtered,
 517                                         q->pitch_gain, q->pitch_lag,
 518                                         q->frame.pfrac);
 519
 520         apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
 521     } else {
 522         memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17,
 523                143 * sizeof(float));
 524         memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
 525         memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
 526         memset(q->pitch_lag,  0, sizeof(q->pitch_lag));
 527     }
 528 }
 529
 530 /**
 531  * Reconstruct LPC coefficients from the line spectral pair frequencies
 532  * and perform bandwidth expansion.
 533  *
 534  * @param lspf line spectral pair frequencies
 535  * @param lpc linear predictive coding coefficients
 536  *
 537  * @note: bandwidth_expansion_coeff could be precalculated into a table
 538  *        but it seems to be slower on x86
 539  *
 540  * TIA/EIA/IS-733 2.4.3.3.5
 541  */
 542 static void lspf2lpc(const float *lspf, float *lpc)
 543 {
 544     double lsp[10];
 545     double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
 546     int   i;
 547
 548     for (i=0; i<10; i++)
 549         lsp[i] = cos(M_PI * lspf[i]);
 550
 551     ff_acelp_lspd2lpc(lsp, lpc, 5);
 552
 553     for (i = 0; i < 10; i++) {
 554         lpc[i] *= bandwidth_expansion_coeff;
 555         bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
 556     }
 557 }
 558
 559 /**
 560  * Interpolate LSP frequencies and compute LPC coefficients
 561  * for a given bitrate & pitch subframe.
 562  *
 563  * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
 564  *
 565  * @param q the context
 566  * @param curr_lspf LSP frequencies vector of the current frame
 567  * @param lpc float vector for the resulting LPC
 568  * @param subframe_num frame number in decoded stream
 569  */
 570 static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
 571                             float *lpc, const int subframe_num)
 572 {
 573     float interpolated_lspf[10];
 574     float weight;
 575
 576     if(q->bitrate >= RATE_QUARTER)
 577         weight = 0.25 * (subframe_num + 1);
 578     else if(q->bitrate == RATE_OCTAVE && !subframe_num)
 579         weight = 0.625;
 580     else
 581         weight = 1.0;
 582
 583     if (weight != 1.0) {
 584         ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
 585                                 weight, 1.0 - weight, 10);
 586         lspf2lpc(interpolated_lspf, lpc);
 587     }else if(q->bitrate >= RATE_QUARTER ||
 588              (q->bitrate == I_F_Q && !subframe_num))
 589         lspf2lpc(curr_lspf, lpc);
 590     else if(q->bitrate == SILENCE && !subframe_num)
 591         lspf2lpc(q->prev_lspf, lpc);
 592 }
 593
 594 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
 595 {
 596     switch (buf_size) {
 597         case 35: return RATE_FULL;
 598         case 17: return RATE_HALF;
 599         case  8: return RATE_QUARTER;
 600         case  4: return RATE_OCTAVE;
 601         case  1: return SILENCE;
 602     }
 603
 604     return I_F_Q;
 605 }
 606
 607 /**
 608  * Determine the bitrate from the frame size and/or the first byte of the frame.
 609  *
 610  * @param avctx the AV codec context
 611  * @param buf_size length of the buffer
 612  * @param buf the bufffer
 613  *
 614  * @return the bitrate on success,
 615  *         I_F_Q  if the bitrate cannot be satisfactorily determined
 616  *
 617  * TIA/EIA/IS-733 2.4.8.7.1
 618  */
 619 static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx, const int buf_size,
 620                              const uint8_t **buf)
 621 {
 622     qcelp_packet_rate bitrate;
 623
 624     if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {
 625         if (bitrate > **buf) {
 626             QCELPContext *q = avctx->priv_data;
 627             if (!q->warned_buf_mismatch_bitrate) {
 628             av_log(avctx, AV_LOG_WARNING,
 629                    "Claimed bitrate and buffer size mismatch.\n");
 630                 q->warned_buf_mismatch_bitrate = 1;
 631             }
 632             bitrate = **buf;
 633         } else if (bitrate < **buf) {
 634             av_log(avctx, AV_LOG_ERROR,
 635                    "Buffer is too small for the claimed bitrate.\n");
 636             return I_F_Q;
 637         }
 638         (*buf)++;
 639     } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) {
 640         av_log(avctx, AV_LOG_WARNING,
 641                "Bitrate byte is missing, guessing the bitrate from packet size.\n");
 642     }else
 643         return I_F_Q;
 644
 645     if (bitrate == SILENCE) {
 646         //FIXME: Remove experimental warning when tested with samples.
 647         av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
 648     }
 649     return bitrate;
 650 }
 651
 652 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
 653                                             const char *message)
 654 {
 655     av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
 656            message);
 657 }
 658
 659 static void postfilter(QCELPContext *q, float *samples, float *lpc)
 660 {
 661     static const float pow_0_775[10] = {
 662         0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
 663         0.216676, 0.167924, 0.130141, 0.100859, 0.078166
 664     }, pow_0_625[10] = {
 665         0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
 666         0.059605, 0.037253, 0.023283, 0.014552, 0.009095
 667     };
 668     float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
 669     int n;
 670
 671     for (n = 0; n < 10; n++) {
 672         lpc_s[n] = lpc[n] * pow_0_625[n];
 673         lpc_p[n] = lpc[n] * pow_0_775[n];
 674     }
 675
 676     ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
 677                                       q->formant_mem + 10, 160, 10);
 678     memcpy(pole_out, q->postfilter_synth_mem,       sizeof(float) * 10);
 679     ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
 680     memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
 681
 682     ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
 683
 684     ff_adaptive_gain_control(samples, pole_out + 10,
 685         ff_dot_productf(q->formant_mem + 10, q->formant_mem + 10, 160),
 686         160, 0.9375, &q->postfilter_agc_mem);
 687 }
 688
 689 static int qcelp_decode_frame(AVCodecContext *avctx, void *data,
 690                               int *got_frame_ptr, AVPacket *avpkt)
 691 {
 692     const uint8_t *buf = avpkt->data;
 693     int buf_size = avpkt->size;
 694     QCELPContext *q = avctx->priv_data;
 695     float *outbuffer;
 696     int   i, ret;
 697     float quantized_lspf[10], lpc[10];
 698     float gain[16];
 699     float *formant_mem;
 700
 701     /* get output buffer */
 702     q->avframe.nb_samples = 160;
 703     if ((ret = avctx->get_buffer(avctx, &q->avframe)) < 0) {
 704         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
 705         return ret;
 706     }
 707     outbuffer = (float *)q->avframe.data[0];
 708
 709     if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
 710         warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
 711         goto erasure;
 712     }
 713
 714     if(q->bitrate == RATE_OCTAVE &&
 715        (q->first16bits = AV_RB16(buf)) == 0xFFFF) {
 716         warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
 717         goto erasure;
 718     }
 719
 720     if (q->bitrate > SILENCE) {
 721         const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
 722         const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate]
 723                                        + qcelp_unpacking_bitmaps_lengths[q->bitrate];
 724         uint8_t           *unpacked_data = (uint8_t *)&q->frame;
 725
 726         init_get_bits(&q->gb, buf, 8*buf_size);
 727
 728         memset(&q->frame, 0, sizeof(QCELPFrame));
 729
 730         for(; bitmaps < bitmaps_end; bitmaps++)
 731             unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
 732
 733         // Check for erasures/blanks on rates 1, 1/4 and 1/8.
 734         if (q->frame.reserved) {
 735             warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
 736             goto erasure;
 737         }
 738         if(q->bitrate == RATE_QUARTER &&
 739            codebook_sanity_check_for_rate_quarter(q->frame.cbgain)) {
 740             warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
 741             goto erasure;
 742         }
 743
 744         if (q->bitrate >= RATE_HALF) {
 745             for (i = 0; i < 4; i++) {
 746                 if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) {
 747                     warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
 748                     goto erasure;
 749                 }
 750             }
 751         }
 752     }
 753
 754     decode_gain_and_index(q, gain);
 755     compute_svector(q, gain, outbuffer);
 756
 757     if (decode_lspf(q, quantized_lspf) < 0) {
 758         warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
 759         goto erasure;
 760     }
 761
 762
 763     apply_pitch_filters(q, outbuffer);
 764
 765     if (q->bitrate == I_F_Q) {
 766 erasure:
 767         q->bitrate = I_F_Q;
 768         q->erasure_count++;
 769         decode_gain_and_index(q, gain);
 770         compute_svector(q, gain, outbuffer);
 771         decode_lspf(q, quantized_lspf);
 772         apply_pitch_filters(q, outbuffer);
 773     }else
 774         q->erasure_count = 0;
 775
 776     formant_mem = q->formant_mem + 10;
 777     for (i = 0; i < 4; i++) {
 778         interpolate_lpc(q, quantized_lspf, lpc, i);
 779         ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40,
 780                                      10);
 781         formant_mem += 40;
 782     }
 783
 784     // postfilter, as per TIA/EIA/IS-733 2.4.8.6
 785     postfilter(q, outbuffer, lpc);
 786
 787     memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
 788
 789     memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
 790     q->prev_bitrate = q->bitrate;
 791
 792     *got_frame_ptr   = 1;
 793     *(AVFrame *)data = q->avframe;
 794
 795     return buf_size;
 796 }
 797
 798 AVCodec ff_qcelp_decoder =
 799 {
 800     .name   = "qcelp",
 801     .type   = AVMEDIA_TYPE_AUDIO,
 802     .id     = CODEC_ID_QCELP,
 803     .init   = qcelp_decode_init,
 804     .decode = qcelp_decode_frame,
 805     .capabilities = CODEC_CAP_DR1,
 806     .priv_data_size = sizeof(QCELPContext),
 807     .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
 808 };