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   1 /*
   2  * QCELP decoder
   3  * Copyright (c) 2007 Reynaldo H. Verdejo Pinochet
   4  *
   5  * This file is part of FFmpeg.
   6  *
   7  * FFmpeg 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  * FFmpeg 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 FFmpeg; 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 FFmpeg 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     GetBitContext     gb;
  60     qcelp_packet_rate bitrate;
  61     QCELPFrame        frame;    /*!< unpacked data frame */
  62
  63     uint8_t  erasure_count;
  64     uint8_t  octave_count;      /*!< count the consecutive RATE_OCTAVE frames */
  65     float    prev_lspf[10];
  66     float    predictor_lspf[10];/*!< LSP predictor for RATE_OCTAVE and I_F_Q */
  67     float    pitch_synthesis_filter_mem[303];
  68     float    pitch_pre_filter_mem[303];
  69     float    rnd_fir_filter_mem[180];
  70     float    formant_mem[170];
  71     float    last_codebook_gain;
  72     int      prev_g1[2];
  73     int      prev_bitrate;
  74     float    pitch_gain[4];
  75     uint8_t  pitch_lag[4];
  76     uint16_t first16bits;
  77     uint8_t  warned_buf_mismatch_bitrate;
  78
  79     /* postfilter */
  80     float    postfilter_synth_mem[10];
  81     float    postfilter_agc_mem;
  82     float    postfilter_tilt_mem;
  83 } QCELPContext;
  84
  85 /**
  86  * Initialize the speech codec according to the specification.
  87  *
  88  * TIA/EIA/IS-733 2.4.9
  89  */
  90 static av_cold int qcelp_decode_init(AVCodecContext *avctx)
  91 {
  92     QCELPContext *q = avctx->priv_data;
  93     int i;
  94
  95     avctx->sample_fmt = SAMPLE_FMT_FLT;
  96
  97     for(i=0; i<10; i++)
  98         q->prev_lspf[i] = (i+1)/11.;
  99
 100     return 0;
 101 }
 102
 103 /**
 104  * Decodes the 10 quantized LSP frequencies from the LSPV/LSP
 105  * transmission codes of any bitrate and checks for badly received packets.
 106  *
 107  * @param q the context
 108  * @param lspf line spectral pair frequencies
 109  *
 110  * @return 0 on success, -1 if the packet is badly received
 111  *
 112  * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
 113  */
 114 static int decode_lspf(QCELPContext *q, float *lspf)
 115 {
 116     int i;
 117     float tmp_lspf, smooth, erasure_coeff;
 118     const float *predictors;
 119
 120     if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q)
 121     {
 122         predictors = (q->prev_bitrate != RATE_OCTAVE &&
 123                        q->prev_bitrate != I_F_Q ?
 124                        q->prev_lspf : q->predictor_lspf);
 125
 126         if(q->bitrate == RATE_OCTAVE)
 127         {
 128             q->octave_count++;
 129
 130             for(i=0; i<10; i++)
 131             {
 132                 q->predictor_lspf[i] =
 133                              lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
 134                                                          : -QCELP_LSP_SPREAD_FACTOR)
 135                                      + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR
 136                                      + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
 137             }
 138             smooth = (q->octave_count < 10 ? .875 : 0.1);
 139         }else
 140         {
 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             {
 150                 q->predictor_lspf[i] =
 151                              lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
 152                                      + erasure_coeff * predictors[i];
 153             }
 154             smooth = 0.125;
 155         }
 156
 157         // Check the stability of the LSP frequencies.
 158         lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
 159         for(i=1; i<10; i++)
 160             lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR));
 161
 162         lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR));
 163         for(i=9; i>0; i--)
 164             lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
 165
 166         // Low-pass filter the LSP frequencies.
 167         ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
 168     }else
 169     {
 170         q->octave_count = 0;
 171
 172         tmp_lspf = 0.;
 173         for(i=0; i<5 ; i++)
 174         {
 175             lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
 176             lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
 177         }
 178
 179         // Check for badly received packets.
 180         if(q->bitrate == RATE_QUARTER)
 181         {
 182             if(lspf[9] <= .70 || lspf[9] >=  .97)
 183                 return -1;
 184             for(i=3; i<10; i++)
 185                 if(fabs(lspf[i] - lspf[i-2]) < .08)
 186                     return -1;
 187         }else
 188         {
 189             if(lspf[9] <= .66 || lspf[9] >= .985)
 190                 return -1;
 191             for(i=4; i<10; i++)
 192                 if (fabs(lspf[i] - lspf[i-4]) < .0931)
 193                     return -1;
 194         }
 195     }
 196     return 0;
 197 }
 198
 199 /**
 200  * Converts codebook transmission codes to GAIN and INDEX.
 201  *
 202  * @param q the context
 203  * @param gain array holding the decoded gain
 204  *
 205  * TIA/EIA/IS-733 2.4.6.2
 206  */
 207 static void decode_gain_and_index(QCELPContext  *q,
 208                                   float *gain) {
 209     int   i, subframes_count, g1[16];
 210     float slope;
 211
 212     if(q->bitrate >= RATE_QUARTER)
 213     {
 214         switch(q->bitrate)
 215         {
 216             case RATE_FULL: subframes_count = 16; break;
 217             case RATE_HALF: subframes_count = 4;  break;
 218             default:        subframes_count = 5;
 219         }
 220         for(i=0; i<subframes_count; i++)
 221         {
 222             g1[i] = 4 * q->frame.cbgain[i];
 223             if(q->bitrate == RATE_FULL && !((i+1) & 3))
 224             {
 225                 g1[i] += av_clip((g1[i-1] + g1[i-2] + g1[i-3]) / 3 - 6, 0, 32);
 226             }
 227
 228             gain[i] = qcelp_g12ga[g1[i]];
 229
 230             if(q->frame.cbsign[i])
 231             {
 232                 gain[i] = -gain[i];
 233                 q->frame.cindex[i] = (q->frame.cindex[i]-89) & 127;
 234             }
 235         }
 236
 237         q->prev_g1[0] = g1[i-2];
 238         q->prev_g1[1] = g1[i-1];
 239         q->last_codebook_gain = qcelp_g12ga[g1[i-1]];
 240
 241         if(q->bitrate == RATE_QUARTER)
 242         {
 243             // Provide smoothing of the unvoiced excitation energy.
 244             gain[7] =     gain[4];
 245             gain[6] = 0.4*gain[3] + 0.6*gain[4];
 246             gain[5] =     gain[3];
 247             gain[4] = 0.8*gain[2] + 0.2*gain[3];
 248             gain[3] = 0.2*gain[1] + 0.8*gain[2];
 249             gain[2] =     gain[1];
 250             gain[1] = 0.6*gain[0] + 0.4*gain[1];
 251         }
 252     }else if (q->bitrate != SILENCE)
 253     {
 254         if(q->bitrate == RATE_OCTAVE)
 255         {
 256             g1[0] = 2 * q->frame.cbgain[0]
 257                   + av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
 258             subframes_count = 8;
 259         }else
 260         {
 261             assert(q->bitrate == I_F_Q);
 262
 263             g1[0] = q->prev_g1[1];
 264             switch(q->erasure_count)
 265             {
 266                 case 1 : break;
 267                 case 2 : g1[0] -= 1; break;
 268                 case 3 : g1[0] -= 2; break;
 269                 default: g1[0] -= 6;
 270             }
 271             if(g1[0] < 0)
 272                 g1[0] = 0;
 273             subframes_count = 4;
 274         }
 275         // This interpolation is done to produce smoother background noise.
 276         slope = 0.5*(qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
 277         for(i=1; i<=subframes_count; i++)
 278             gain[i-1] = q->last_codebook_gain + slope * i;
 279
 280         q->last_codebook_gain = gain[i-2];
 281         q->prev_g1[0] = q->prev_g1[1];
 282         q->prev_g1[1] = g1[0];
 283     }
 284 }
 285
 286 /**
 287  * If the received packet is Rate 1/4 a further sanity check is made of the
 288  * codebook gain.
 289  *
 290  * @param cbgain the unpacked cbgain array
 291  * @return -1 if the sanity check fails, 0 otherwise
 292  *
 293  * TIA/EIA/IS-733 2.4.8.7.3
 294  */
 295 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
 296 {
 297     int i, diff, prev_diff=0;
 298
 299     for(i=1; i<5; i++)
 300     {
 301         diff = cbgain[i] - cbgain[i-1];
 302         if(FFABS(diff) > 10)
 303             return -1;
 304         else if(FFABS(diff - prev_diff) > 12)
 305             return -1;
 306         prev_diff = diff;
 307     }
 308     return 0;
 309 }
 310
 311 /**
 312  * Computes the scaled codebook vector Cdn From INDEX and GAIN
 313  * for all rates.
 314  *
 315  * The specification lacks some information here.
 316  *
 317  * TIA/EIA/IS-733 has an omission on the codebook index determination
 318  * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
 319  * you have to subtract the decoded index parameter from the given scaled
 320  * codebook vector index 'n' to get the desired circular codebook index, but
 321  * it does not mention that you have to clamp 'n' to [0-9] in order to get
 322  * RI-compliant results.
 323  *
 324  * The reason for this mistake seems to be the fact they forgot to mention you
 325  * have to do these calculations per codebook subframe and adjust given
 326  * equation values accordingly.
 327  *
 328  * @param q the context
 329  * @param gain array holding the 4 pitch subframe gain values
 330  * @param cdn_vector array for the generated scaled codebook vector
 331  */
 332 static void compute_svector(QCELPContext *q, const float *gain,
 333                             float *cdn_vector)
 334 {
 335     int      i, j, k;
 336     uint16_t cbseed, cindex;
 337     float    *rnd, tmp_gain, fir_filter_value;
 338
 339     switch(q->bitrate)
 340     {
 341         case RATE_FULL:
 342             for(i=0; i<16; i++)
 343             {
 344                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
 345                 cindex = -q->frame.cindex[i];
 346                 for(j=0; j<10; j++)
 347                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
 348             }
 349         break;
 350         case RATE_HALF:
 351             for(i=0; i<4; i++)
 352             {
 353                 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
 354                 cindex = -q->frame.cindex[i];
 355                 for (j = 0; j < 40; j++)
 356                 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
 357             }
 358         break;
 359         case RATE_QUARTER:
 360             cbseed = (0x0003 & q->frame.lspv[4])<<14 |
 361                      (0x003F & q->frame.lspv[3])<< 8 |
 362                      (0x0060 & q->frame.lspv[2])<< 1 |
 363                      (0x0007 & q->frame.lspv[1])<< 3 |
 364                      (0x0038 & q->frame.lspv[0])>> 3 ;
 365             rnd = q->rnd_fir_filter_mem + 20;
 366             for(i=0; i<8; i++)
 367             {
 368                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
 369                 for(k=0; k<20; k++)
 370                 {
 371                     cbseed = 521 * cbseed + 259;
 372                     *rnd = (int16_t)cbseed;
 373
 374                     // FIR filter
 375                     fir_filter_value = 0.0;
 376                     for(j=0; j<10; j++)
 377                         fir_filter_value += qcelp_rnd_fir_coefs[j ]
 378                                           * (rnd[-j ] + rnd[-20+j]);
 379
 380                     fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
 381                     *cdn_vector++ = tmp_gain * fir_filter_value;
 382                     rnd++;
 383                 }
 384             }
 385             memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
 386         break;
 387         case RATE_OCTAVE:
 388             cbseed = q->first16bits;
 389             for(i=0; i<8; i++)
 390             {
 391                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
 392                 for(j=0; j<20; j++)
 393                 {
 394                     cbseed = 521 * cbseed + 259;
 395                     *cdn_vector++ = tmp_gain * (int16_t)cbseed;
 396                 }
 397             }
 398         break;
 399         case I_F_Q:
 400             cbseed = -44; // random codebook index
 401             for(i=0; i<4; i++)
 402             {
 403                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
 404                 for(j=0; j<40; j++)
 405                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
 406             }
 407         break;
 408         case SILENCE:
 409             memset(cdn_vector, 0, 160 * sizeof(float));
 410         break;
 411     }
 412 }
 413
 414 /**
 415  * Apply generic gain control.
 416  *
 417  * @param v_out output vector
 418  * @param v_in gain-controlled vector
 419  * @param v_ref vector to control gain of
 420  *
 421  * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
 422  */
 423 static void apply_gain_ctrl(float *v_out, const float *v_ref,
 424                             const float *v_in)
 425 {
 426     int i;
 427
 428     for (i = 0; i < 160; i += 40)
 429         ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
 430                                                 ff_dot_productf(v_ref + i,
 431                                                                 v_ref + i, 40),
 432                                                 40);
 433 }
 434
 435 /**
 436  * Apply filter in pitch-subframe steps.
 437  *
 438  * @param memory buffer for the previous state of the filter
 439  *        - must be able to contain 303 elements
 440  *        - the 143 first elements are from the previous state
 441  *        - the next 160 are for output
 442  * @param v_in input filter vector
 443  * @param gain per-subframe gain array, each element is between 0.0 and 2.0
 444  * @param lag per-subframe lag array, each element is
 445  *        - between 16 and 143 if its corresponding pfrac is 0,
 446  *        - between 16 and 139 otherwise
 447  * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
 448  *        otherwise
 449  *
 450  * @return filter output vector
 451  */
 452 static const float *do_pitchfilter(float memory[303], const float v_in[160],
 453                                    const float gain[4], const uint8_t *lag,
 454                                    const uint8_t pfrac[4])
 455 {
 456     int         i, j;
 457     float       *v_lag, *v_out;
 458     const float *v_len;
 459
 460     v_out = memory + 143; // Output vector starts at memory[143].
 461
 462     for(i=0; i<4; i++)
 463     {
 464         if(gain[i])
 465         {
 466             v_lag = memory + 143 + 40 * i - lag[i];
 467             for(v_len=v_in+40; v_in<v_len; v_in++)
 468             {
 469                 if(pfrac[i]) // If it is a fractional lag...
 470                 {
 471                     for(j=0, *v_out=0.; j<4; j++)
 472                         *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
 473                 }else
 474                     *v_out = *v_lag;
 475
 476                 *v_out = *v_in + gain[i] * *v_out;
 477
 478                 v_lag++;
 479                 v_out++;
 480             }
 481         }else
 482         {
 483             memcpy(v_out, v_in, 40 * sizeof(float));
 484             v_in  += 40;
 485             v_out += 40;
 486         }
 487     }
 488
 489     memmove(memory, memory + 160, 143 * sizeof(float));
 490     return memory + 143;
 491 }
 492
 493 /**
 494  * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
 495  * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
 496  *
 497  * @param q the context
 498  * @param cdn_vector the scaled codebook vector
 499  */
 500 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
 501 {
 502     int         i;
 503     const float *v_synthesis_filtered, *v_pre_filtered;
 504
 505     if(q->bitrate >= RATE_HALF ||
 506        q->bitrate == SILENCE ||
 507        (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF)))
 508     {
 509
 510         if(q->bitrate >= RATE_HALF)
 511         {
 512
 513             // Compute gain & lag for the whole frame.
 514             for(i=0; i<4; i++)
 515             {
 516                 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
 517
 518                 q->pitch_lag[i] = q->frame.plag[i] + 16;
 519             }
 520         }else
 521         {
 522             float max_pitch_gain;
 523
 524             if (q->bitrate == I_F_Q)
 525             {
 526                   if (q->erasure_count < 3)
 527                       max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
 528                   else
 529                       max_pitch_gain = 0.0;
 530             }else
 531             {
 532                 assert(q->bitrate == SILENCE);
 533                 max_pitch_gain = 1.0;
 534             }
 535             for(i=0; i<4; i++)
 536                 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
 537
 538             memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
 539         }
 540
 541         // pitch synthesis filter
 542         v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
 543                                               cdn_vector, q->pitch_gain,
 544                                               q->pitch_lag, q->frame.pfrac);
 545
 546         // pitch prefilter update
 547         for(i=0; i<4; i++)
 548             q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
 549
 550         v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
 551                                         v_synthesis_filtered,
 552                                         q->pitch_gain, q->pitch_lag,
 553                                         q->frame.pfrac);
 554
 555         apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
 556     }else
 557     {
 558         memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17,
 559                143 * sizeof(float));
 560         memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
 561         memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
 562         memset(q->pitch_lag,  0, sizeof(q->pitch_lag));
 563     }
 564 }
 565
 566 /**
 567  * Reconstructs LPC coefficients from the line spectral pair frequencies
 568  * and performs bandwidth expansion.
 569  *
 570  * @param lspf line spectral pair frequencies
 571  * @param lpc linear predictive coding coefficients
 572  *
 573  * @note: bandwidth_expansion_coeff could be precalculated into a table
 574  *        but it seems to be slower on x86
 575  *
 576  * TIA/EIA/IS-733 2.4.3.3.5
 577  */
 578 static void lspf2lpc(const float *lspf, float *lpc)
 579 {
 580     double lsp[10];
 581     double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
 582     int   i;
 583
 584     for (i=0; i<10; i++)
 585         lsp[i] = cos(M_PI * lspf[i]);
 586
 587     ff_acelp_lspd2lpc(lsp, lpc, 5);
 588
 589     for (i=0; i<10; i++)
 590     {
 591         lpc[i] *= bandwidth_expansion_coeff;
 592         bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
 593     }
 594 }
 595
 596 /**
 597  * Interpolates LSP frequencies and computes LPC coefficients
 598  * for a given bitrate & pitch subframe.
 599  *
 600  * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
 601  *
 602  * @param q the context
 603  * @param curr_lspf LSP frequencies vector of the current frame
 604  * @param lpc float vector for the resulting LPC
 605  * @param subframe_num frame number in decoded stream
 606  */
 607 static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
 608                             float *lpc, const int subframe_num)
 609 {
 610     float interpolated_lspf[10];
 611     float weight;
 612
 613     if(q->bitrate >= RATE_QUARTER)
 614         weight = 0.25 * (subframe_num + 1);
 615     else if(q->bitrate == RATE_OCTAVE && !subframe_num)
 616         weight = 0.625;
 617     else
 618         weight = 1.0;
 619
 620     if(weight != 1.0)
 621     {
 622         ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
 623                                 weight, 1.0 - weight, 10);
 624         lspf2lpc(interpolated_lspf, lpc);
 625     }else if(q->bitrate >= RATE_QUARTER ||
 626              (q->bitrate == I_F_Q && !subframe_num))
 627         lspf2lpc(curr_lspf, lpc);
 628     else if(q->bitrate == SILENCE && !subframe_num)
 629         lspf2lpc(q->prev_lspf, lpc);
 630 }
 631
 632 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
 633 {
 634     switch(buf_size)
 635     {
 636         case 35: return RATE_FULL;
 637         case 17: return RATE_HALF;
 638         case  8: return RATE_QUARTER;
 639         case  4: return RATE_OCTAVE;
 640         case  1: return SILENCE;
 641     }
 642
 643     return I_F_Q;
 644 }
 645
 646 /**
 647  * Determine the bitrate from the frame size and/or the first byte of the frame.
 648  *
 649  * @param avctx the AV codec context
 650  * @param buf_size length of the buffer
 651  * @param buf the bufffer
 652  *
 653  * @return the bitrate on success,
 654  *         I_F_Q  if the bitrate cannot be satisfactorily determined
 655  *
 656  * TIA/EIA/IS-733 2.4.8.7.1
 657  */
 658 static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx, const int buf_size,
 659                              const uint8_t **buf)
 660 {
 661     qcelp_packet_rate bitrate;
 662
 663     if((bitrate = buf_size2bitrate(buf_size)) >= 0)
 664     {
 665         if(bitrate > **buf)
 666         {
 667             QCELPContext *q = avctx->priv_data;
 668             if (!q->warned_buf_mismatch_bitrate)
 669             {
 670             av_log(avctx, AV_LOG_WARNING,
 671                    "Claimed bitrate and buffer size mismatch.\n");
 672                 q->warned_buf_mismatch_bitrate = 1;
 673             }
 674             bitrate = **buf;
 675         }else if(bitrate < **buf)
 676         {
 677             av_log(avctx, AV_LOG_ERROR,
 678                    "Buffer is too small for the claimed bitrate.\n");
 679             return I_F_Q;
 680         }
 681         (*buf)++;
 682     }else if((bitrate = buf_size2bitrate(buf_size + 1)) >= 0)
 683     {
 684         av_log(avctx, AV_LOG_WARNING,
 685                "Bitrate byte is missing, guessing the bitrate from packet size.\n");
 686     }else
 687         return I_F_Q;
 688
 689     if(bitrate == SILENCE)
 690     {
 691         //FIXME: Remove experimental warning when tested with samples.
 692         av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
 693     }
 694     return bitrate;
 695 }
 696
 697 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
 698                                             const char *message)
 699 {
 700     av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
 701            message);
 702 }
 703
 704 static void postfilter(QCELPContext *q, float *samples, float *lpc)
 705 {
 706     static const float pow_0_775[10] = {
 707         0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
 708         0.216676, 0.167924, 0.130141, 0.100859, 0.078166
 709     }, pow_0_625[10] = {
 710         0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
 711         0.059605, 0.037253, 0.023283, 0.014552, 0.009095
 712     };
 713     float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
 714     int n;
 715
 716     for (n = 0; n < 10; n++) {
 717         lpc_s[n] = lpc[n] * pow_0_625[n];
 718         lpc_p[n] = lpc[n] * pow_0_775[n];
 719     }
 720
 721     ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
 722                                       q->formant_mem + 10, 160, 10);
 723     memcpy(pole_out, q->postfilter_synth_mem,       sizeof(float) * 10);
 724     ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
 725     memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
 726
 727     ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
 728
 729     ff_adaptive_gain_control(samples, pole_out + 10,
 730         ff_dot_productf(q->formant_mem + 10, q->formant_mem + 10, 160),
 731         160, 0.9375, &q->postfilter_agc_mem);
 732 }
 733
 734 static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
 735                               AVPacket *avpkt)
 736 {
 737     const uint8_t *buf = avpkt->data;
 738     int buf_size = avpkt->size;
 739     QCELPContext *q = avctx->priv_data;
 740     float *outbuffer = data;
 741     int   i;
 742     float quantized_lspf[10], lpc[10];
 743     float gain[16];
 744     float *formant_mem;
 745
 746     if((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q)
 747     {
 748         warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
 749         goto erasure;
 750     }
 751
 752     if(q->bitrate == RATE_OCTAVE &&
 753        (q->first16bits = AV_RB16(buf)) == 0xFFFF)
 754     {
 755         warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
 756         goto erasure;
 757     }
 758
 759     if(q->bitrate > SILENCE)
 760     {
 761         const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
 762         const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate]
 763                                        + qcelp_unpacking_bitmaps_lengths[q->bitrate];
 764         uint8_t           *unpacked_data = (uint8_t *)&q->frame;
 765
 766         init_get_bits(&q->gb, buf, 8*buf_size);
 767
 768         memset(&q->frame, 0, sizeof(QCELPFrame));
 769
 770         for(; bitmaps < bitmaps_end; bitmaps++)
 771             unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
 772
 773         // Check for erasures/blanks on rates 1, 1/4 and 1/8.
 774         if(q->frame.reserved)
 775         {
 776             warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
 777             goto erasure;
 778         }
 779         if(q->bitrate == RATE_QUARTER &&
 780            codebook_sanity_check_for_rate_quarter(q->frame.cbgain))
 781         {
 782             warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
 783             goto erasure;
 784         }
 785
 786         if(q->bitrate >= RATE_HALF)
 787         {
 788             for(i=0; i<4; i++)
 789             {
 790                 if(q->frame.pfrac[i] && q->frame.plag[i] >= 124)
 791                 {
 792                     warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
 793                     goto erasure;
 794                 }
 795             }
 796         }
 797     }
 798
 799     decode_gain_and_index(q, gain);
 800     compute_svector(q, gain, outbuffer);
 801
 802     if(decode_lspf(q, quantized_lspf) < 0)
 803     {
 804         warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
 805         goto erasure;
 806     }
 807
 808
 809     apply_pitch_filters(q, outbuffer);
 810
 811     if(q->bitrate == I_F_Q)
 812     {
 813 erasure:
 814         q->bitrate = I_F_Q;
 815         q->erasure_count++;
 816         decode_gain_and_index(q, gain);
 817         compute_svector(q, gain, outbuffer);
 818         decode_lspf(q, quantized_lspf);
 819         apply_pitch_filters(q, outbuffer);
 820     }else
 821         q->erasure_count = 0;
 822
 823     formant_mem = q->formant_mem + 10;
 824     for(i=0; i<4; i++)
 825     {
 826         interpolate_lpc(q, quantized_lspf, lpc, i);
 827         ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40,
 828                                      10);
 829         formant_mem += 40;
 830     }
 831
 832     // postfilter, as per TIA/EIA/IS-733 2.4.8.6
 833     postfilter(q, outbuffer, lpc);
 834
 835     memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
 836
 837     memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
 838     q->prev_bitrate = q->bitrate;
 839
 840     *data_size = 160 * sizeof(*outbuffer);
 841
 842     return *data_size;
 843 }
 844
 845 AVCodec qcelp_decoder =
 846 {
 847     .name   = "qcelp",
 848     .type   = AVMEDIA_TYPE_AUDIO,
 849     .id     = CODEC_ID_QCELP,
 850     .init   = qcelp_decode_init,
 851     .decode = qcelp_decode_frame,
 852     .priv_data_size = sizeof(QCELPContext),
 853     .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
 854 };