libavcodec/qcelpdec.c

   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 libavcodec/qcelpdec.c
  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_vectors.h"
  41 #include "lsp.h"
  42
  43 #undef NDEBUG
  44 #include <assert.h>
  45
  46 typedef enum
  47 {
  48     I_F_Q = -1,    /*!< insufficient frame quality */
  49     SILENCE,
  50     RATE_OCTAVE,
  51     RATE_QUARTER,
  52     RATE_HALF,
  53     RATE_FULL
  54 } qcelp_packet_rate;
  55
  56 typedef struct
  57 {
  58     GetBitContext     gb;
  59     qcelp_packet_rate bitrate;
  60     QCELPFrame        frame;    /*!< unpacked data frame */
  61
  62     uint8_t  erasure_count;
  63     uint8_t  octave_count;      /*!< count the consecutive RATE_OCTAVE frames */
  64     float    prev_lspf[10];
  65     float    predictor_lspf[10];/*!< LSP predictor for RATE_OCTAVE and I_F_Q */
  66     float    pitch_synthesis_filter_mem[303];
  67     float    pitch_pre_filter_mem[303];
  68     float    rnd_fir_filter_mem[180];
  69     float    formant_mem[170];
  70     float    last_codebook_gain;
  71     int      prev_g1[2];
  72     int      prev_bitrate;
  73     float    pitch_gain[4];
  74     uint8_t  pitch_lag[4];
  75     uint16_t first16bits;
  76     uint8_t  warned_buf_mismatch_bitrate;
  77 } QCELPContext;
  78
  79 /**
  80  * Initialize the speech codec according to the specification.
  81  *
  82  * TIA/EIA/IS-733 2.4.9
  83  */
  84 static av_cold int qcelp_decode_init(AVCodecContext *avctx)
  85 {
  86     QCELPContext *q = avctx->priv_data;
  87     int i;
  88
  89     avctx->sample_fmt = SAMPLE_FMT_FLT;
  90
  91     for(i=0; i<10; i++)
  92         q->prev_lspf[i] = (i+1)/11.;
  93
  94     return 0;
  95 }
  96
  97 /**
  98  * Decodes the 10 quantized LSP frequencies from the LSPV/LSP
  99  * transmission codes of any bitrate and checks for badly received packets.
 100  *
 101  * @param q the context
 102  * @param lspf line spectral pair frequencies
 103  *
 104  * @return 0 on success, -1 if the packet is badly received
 105  *
 106  * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
 107  */
 108 static int decode_lspf(QCELPContext *q, float *lspf)
 109 {
 110     int i;
 111     float tmp_lspf, smooth, erasure_coeff;
 112     const float *predictors;
 113
 114     if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q)
 115     {
 116         predictors = (q->prev_bitrate != RATE_OCTAVE &&
 117                        q->prev_bitrate != I_F_Q ?
 118                        q->prev_lspf : q->predictor_lspf);
 119
 120         if(q->bitrate == RATE_OCTAVE)
 121         {
 122             q->octave_count++;
 123
 124             for(i=0; i<10; i++)
 125             {
 126                 q->predictor_lspf[i] =
 127                              lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
 128                                                          : -QCELP_LSP_SPREAD_FACTOR)
 129                                      + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR
 130                                      + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
 131             }
 132             smooth = (q->octave_count < 10 ? .875 : 0.1);
 133         }else
 134         {
 135             erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
 136
 137             assert(q->bitrate == I_F_Q);
 138
 139             if(q->erasure_count > 1)
 140                 erasure_coeff *= (q->erasure_count < 4 ? 0.9 : 0.7);
 141
 142             for(i=0; i<10; i++)
 143             {
 144                 q->predictor_lspf[i] =
 145                              lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
 146                                      + erasure_coeff * predictors[i];
 147             }
 148             smooth = 0.125;
 149         }
 150
 151         // Check the stability of the LSP frequencies.
 152         lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
 153         for(i=1; i<10; i++)
 154             lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR));
 155
 156         lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR));
 157         for(i=9; i>0; i--)
 158             lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
 159
 160         // Low-pass filter the LSP frequencies.
 161         ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
 162     }else
 163     {
 164         q->octave_count = 0;
 165
 166         tmp_lspf = 0.;
 167         for(i=0; i<5 ; i++)
 168         {
 169             lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
 170             lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
 171         }
 172
 173         // Check for badly received packets.
 174         if(q->bitrate == RATE_QUARTER)
 175         {
 176             if(lspf[9] <= .70 || lspf[9] >=  .97)
 177                 return -1;
 178             for(i=3; i<10; i++)
 179                 if(fabs(lspf[i] - lspf[i-2]) < .08)
 180                     return -1;
 181         }else
 182         {
 183             if(lspf[9] <= .66 || lspf[9] >= .985)
 184                 return -1;
 185             for(i=4; i<10; i++)
 186                 if (fabs(lspf[i] - lspf[i-4]) < .0931)
 187                     return -1;
 188         }
 189     }
 190     return 0;
 191 }
 192
 193 /**
 194  * Converts codebook transmission codes to GAIN and INDEX.
 195  *
 196  * @param q the context
 197  * @param gain array holding the decoded gain
 198  *
 199  * TIA/EIA/IS-733 2.4.6.2
 200  */
 201 static void decode_gain_and_index(QCELPContext  *q,
 202                                   float *gain) {
 203     int   i, subframes_count, g1[16];
 204     float slope;
 205
 206     if(q->bitrate >= RATE_QUARTER)
 207     {
 208         switch(q->bitrate)
 209         {
 210             case RATE_FULL: subframes_count = 16; break;
 211             case RATE_HALF: subframes_count = 4;  break;
 212             default:        subframes_count = 5;
 213         }
 214         for(i=0; i<subframes_count; i++)
 215         {
 216             g1[i] = 4 * q->frame.cbgain[i];
 217             if(q->bitrate == RATE_FULL && !((i+1) & 3))
 218             {
 219                 g1[i] += av_clip((g1[i-1] + g1[i-2] + g1[i-3]) / 3 - 6, 0, 32);
 220             }
 221
 222             gain[i] = qcelp_g12ga[g1[i]];
 223
 224             if(q->frame.cbsign[i])
 225             {
 226                 gain[i] = -gain[i];
 227                 q->frame.cindex[i] = (q->frame.cindex[i]-89) & 127;
 228             }
 229         }
 230
 231         q->prev_g1[0] = g1[i-2];
 232         q->prev_g1[1] = g1[i-1];
 233         q->last_codebook_gain = qcelp_g12ga[g1[i-1]];
 234
 235         if(q->bitrate == RATE_QUARTER)
 236         {
 237             // Provide smoothing of the unvoiced excitation energy.
 238             gain[7] =     gain[4];
 239             gain[6] = 0.4*gain[3] + 0.6*gain[4];
 240             gain[5] =     gain[3];
 241             gain[4] = 0.8*gain[2] + 0.2*gain[3];
 242             gain[3] = 0.2*gain[1] + 0.8*gain[2];
 243             gain[2] =     gain[1];
 244             gain[1] = 0.6*gain[0] + 0.4*gain[1];
 245         }
 246     }else if (q->bitrate != SILENCE)
 247     {
 248         if(q->bitrate == RATE_OCTAVE)
 249         {
 250             g1[0] = 2 * q->frame.cbgain[0]
 251                   + av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
 252             subframes_count = 8;
 253         }else
 254         {
 255             assert(q->bitrate == I_F_Q);
 256
 257             g1[0] = q->prev_g1[1];
 258             switch(q->erasure_count)
 259             {
 260                 case 1 : break;
 261                 case 2 : g1[0] -= 1; break;
 262                 case 3 : g1[0] -= 2; break;
 263                 default: g1[0] -= 6;
 264             }
 265             if(g1[0] < 0)
 266                 g1[0] = 0;
 267             subframes_count = 4;
 268         }
 269         // This interpolation is done to produce smoother background noise.
 270         slope = 0.5*(qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
 271         for(i=1; i<=subframes_count; i++)
 272             gain[i-1] = q->last_codebook_gain + slope * i;
 273
 274         q->last_codebook_gain = gain[i-2];
 275         q->prev_g1[0] = q->prev_g1[1];
 276         q->prev_g1[1] = g1[0];
 277     }
 278 }
 279
 280 /**
 281  * If the received packet is Rate 1/4 a further sanity check is made of the
 282  * codebook gain.
 283  *
 284  * @param cbgain the unpacked cbgain array
 285  * @return -1 if the sanity check fails, 0 otherwise
 286  *
 287  * TIA/EIA/IS-733 2.4.8.7.3
 288  */
 289 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
 290 {
 291     int i, diff, prev_diff=0;
 292
 293     for(i=1; i<5; i++)
 294     {
 295         diff = cbgain[i] - cbgain[i-1];
 296         if(FFABS(diff) > 10)
 297             return -1;
 298         else if(FFABS(diff - prev_diff) > 12)
 299             return -1;
 300         prev_diff = diff;
 301     }
 302     return 0;
 303 }
 304
 305 /**
 306  * Computes the scaled codebook vector Cdn From INDEX and GAIN
 307  * for all rates.
 308  *
 309  * The specification lacks some information here.
 310  *
 311  * TIA/EIA/IS-733 has an omission on the codebook index determination
 312  * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
 313  * you have to subtract the decoded index parameter from the given scaled
 314  * codebook vector index 'n' to get the desired circular codebook index, but
 315  * it does not mention that you have to clamp 'n' to [0-9] in order to get
 316  * RI-compliant results.
 317  *
 318  * The reason for this mistake seems to be the fact they forgot to mention you
 319  * have to do these calculations per codebook subframe and adjust given
 320  * equation values accordingly.
 321  *
 322  * @param q the context
 323  * @param gain array holding the 4 pitch subframe gain values
 324  * @param cdn_vector array for the generated scaled codebook vector
 325  */
 326 static void compute_svector(QCELPContext *q, const float *gain,
 327                             float *cdn_vector)
 328 {
 329     int      i, j, k;
 330     uint16_t cbseed, cindex;
 331     float    *rnd, tmp_gain, fir_filter_value;
 332
 333     switch(q->bitrate)
 334     {
 335         case RATE_FULL:
 336             for(i=0; i<16; i++)
 337             {
 338                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
 339                 cindex = -q->frame.cindex[i];
 340                 for(j=0; j<10; j++)
 341                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
 342             }
 343         break;
 344         case RATE_HALF:
 345             for(i=0; i<4; i++)
 346             {
 347                 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
 348                 cindex = -q->frame.cindex[i];
 349                 for (j = 0; j < 40; j++)
 350                 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
 351             }
 352         break;
 353         case RATE_QUARTER:
 354             cbseed = (0x0003 & q->frame.lspv[4])<<14 |
 355                      (0x003F & q->frame.lspv[3])<< 8 |
 356                      (0x0060 & q->frame.lspv[2])<< 1 |
 357                      (0x0007 & q->frame.lspv[1])<< 3 |
 358                      (0x0038 & q->frame.lspv[0])>> 3 ;
 359             rnd = q->rnd_fir_filter_mem + 20;
 360             for(i=0; i<8; i++)
 361             {
 362                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
 363                 for(k=0; k<20; k++)
 364                 {
 365                     cbseed = 521 * cbseed + 259;
 366                     *rnd = (int16_t)cbseed;
 367
 368                     // FIR filter
 369                     fir_filter_value = 0.0;
 370                     for(j=0; j<10; j++)
 371                         fir_filter_value += qcelp_rnd_fir_coefs[j ]
 372                                           * (rnd[-j ] + rnd[-20+j]);
 373
 374                     fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
 375                     *cdn_vector++ = tmp_gain * fir_filter_value;
 376                     rnd++;
 377                 }
 378             }
 379             memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
 380         break;
 381         case RATE_OCTAVE:
 382             cbseed = q->first16bits;
 383             for(i=0; i<8; i++)
 384             {
 385                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
 386                 for(j=0; j<20; j++)
 387                 {
 388                     cbseed = 521 * cbseed + 259;
 389                     *cdn_vector++ = tmp_gain * (int16_t)cbseed;
 390                 }
 391             }
 392         break;
 393         case I_F_Q:
 394             cbseed = -44; // random codebook index
 395             for(i=0; i<4; i++)
 396             {
 397                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
 398                 for(j=0; j<40; j++)
 399                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
 400             }
 401         break;
 402         case SILENCE:
 403             memset(cdn_vector, 0, 160 * sizeof(float));
 404         break;
 405     }
 406 }
 407
 408 /**
 409  * Apply generic gain control.
 410  *
 411  * @param v_out output vector
 412  * @param v_in gain-controlled vector
 413  * @param v_ref vector to control gain of
 414  *
 415  * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
 416  */
 417 static void apply_gain_ctrl(float *v_out, const float *v_ref,
 418                             const float *v_in)
 419 {
 420     int i;
 421
 422     for (i = 0; i < 160; i += 40)
 423         ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
 424                                                 ff_dot_productf(v_ref + i,
 425                                                                 v_ref + i, 40),
 426                                                 40);
 427 }
 428
 429 /**
 430  * Apply filter in pitch-subframe steps.
 431  *
 432  * @param memory buffer for the previous state of the filter
 433  *        - must be able to contain 303 elements
 434  *        - the 143 first elements are from the previous state
 435  *        - the next 160 are for output
 436  * @param v_in input filter vector
 437  * @param gain per-subframe gain array, each element is between 0.0 and 2.0
 438  * @param lag per-subframe lag array, each element is
 439  *        - between 16 and 143 if its corresponding pfrac is 0,
 440  *        - between 16 and 139 otherwise
 441  * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
 442  *        otherwise
 443  *
 444  * @return filter output vector
 445  */
 446 static const float *do_pitchfilter(float memory[303], const float v_in[160],
 447                                    const float gain[4], const uint8_t *lag,
 448                                    const uint8_t pfrac[4])
 449 {
 450     int         i, j;
 451     float       *v_lag, *v_out;
 452     const float *v_len;
 453
 454     v_out = memory + 143; // Output vector starts at memory[143].
 455
 456     for(i=0; i<4; i++)
 457     {
 458         if(gain[i])
 459         {
 460             v_lag = memory + 143 + 40 * i - lag[i];
 461             for(v_len=v_in+40; v_in<v_len; v_in++)
 462             {
 463                 if(pfrac[i]) // If it is a fractional lag...
 464                 {
 465                     for(j=0, *v_out=0.; j<4; j++)
 466                         *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
 467                 }else
 468                     *v_out = *v_lag;
 469
 470                 *v_out = *v_in + gain[i] * *v_out;
 471
 472                 v_lag++;
 473                 v_out++;
 474             }
 475         }else
 476         {
 477             memcpy(v_out, v_in, 40 * sizeof(float));
 478             v_in  += 40;
 479             v_out += 40;
 480         }
 481     }
 482
 483     memmove(memory, memory + 160, 143 * sizeof(float));
 484     return memory + 143;
 485 }
 486
 487 /**
 488  * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
 489  * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
 490  *
 491  * @param q the context
 492  * @param cdn_vector the scaled codebook vector
 493  */
 494 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
 495 {
 496     int         i;
 497     const float *v_synthesis_filtered, *v_pre_filtered;
 498
 499     if(q->bitrate >= RATE_HALF ||
 500        q->bitrate == SILENCE ||
 501        (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF)))
 502     {
 503
 504         if(q->bitrate >= RATE_HALF)
 505         {
 506
 507             // Compute gain & lag for the whole frame.
 508             for(i=0; i<4; i++)
 509             {
 510                 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
 511
 512                 q->pitch_lag[i] = q->frame.plag[i] + 16;
 513             }
 514         }else
 515         {
 516             float max_pitch_gain;
 517
 518             if (q->bitrate == I_F_Q)
 519             {
 520                   if (q->erasure_count < 3)
 521                       max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
 522                   else
 523                       max_pitch_gain = 0.0;
 524             }else
 525             {
 526                 assert(q->bitrate == SILENCE);
 527                 max_pitch_gain = 1.0;
 528             }
 529             for(i=0; i<4; i++)
 530                 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
 531
 532             memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
 533         }
 534
 535         // pitch synthesis filter
 536         v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
 537                                               cdn_vector, q->pitch_gain,
 538                                               q->pitch_lag, q->frame.pfrac);
 539
 540         // pitch prefilter update
 541         for(i=0; i<4; i++)
 542             q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
 543
 544         v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
 545                                         v_synthesis_filtered,
 546                                         q->pitch_gain, q->pitch_lag,
 547                                         q->frame.pfrac);
 548
 549         apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
 550     }else
 551     {
 552         memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17,
 553                143 * sizeof(float));
 554         memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
 555         memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
 556         memset(q->pitch_lag,  0, sizeof(q->pitch_lag));
 557     }
 558 }
 559
 560 /**
 561  * Reconstructs LPC coefficients from the line spectral pair frequencies
 562  * and performs bandwidth expansion.
 563  *
 564  * @param lspf line spectral pair frequencies
 565  * @param lpc linear predictive coding coefficients
 566  *
 567  * @note: bandwidth_expansion_coeff could be precalculated into a table
 568  *        but it seems to be slower on x86
 569  *
 570  * TIA/EIA/IS-733 2.4.3.3.5
 571  */
 572 static void lspf2lpc(const float *lspf, float *lpc)
 573 {
 574     double lsp[10];
 575     double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
 576     int   i;
 577
 578     for (i=0; i<10; i++)
 579         lsp[i] = cos(M_PI * lspf[i]);
 580
 581     ff_acelp_lspd2lpc(lsp, lpc, 5);
 582
 583     for (i=0; i<10; i++)
 584     {
 585         lpc[i] *= bandwidth_expansion_coeff;
 586         bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
 587     }
 588 }
 589
 590 /**
 591  * Interpolates LSP frequencies and computes LPC coefficients
 592  * for a given bitrate & pitch subframe.
 593  *
 594  * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
 595  *
 596  * @param q the context
 597  * @param curr_lspf LSP frequencies vector of the current frame
 598  * @param lpc float vector for the resulting LPC
 599  * @param subframe_num frame number in decoded stream
 600  */
 601 void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc,
 602                      const int subframe_num)
 603 {
 604     float interpolated_lspf[10];
 605     float weight;
 606
 607     if(q->bitrate >= RATE_QUARTER)
 608         weight = 0.25 * (subframe_num + 1);
 609     else if(q->bitrate == RATE_OCTAVE && !subframe_num)
 610         weight = 0.625;
 611     else
 612         weight = 1.0;
 613
 614     if(weight != 1.0)
 615     {
 616         ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
 617                                 weight, 1.0 - weight, 10);
 618         lspf2lpc(interpolated_lspf, lpc);
 619     }else if(q->bitrate >= RATE_QUARTER ||
 620              (q->bitrate == I_F_Q && !subframe_num))
 621         lspf2lpc(curr_lspf, lpc);
 622     else if(q->bitrate == SILENCE && !subframe_num)
 623         lspf2lpc(q->prev_lspf, lpc);
 624 }
 625
 626 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
 627 {
 628     switch(buf_size)
 629     {
 630         case 35: return RATE_FULL;
 631         case 17: return RATE_HALF;
 632         case  8: return RATE_QUARTER;
 633         case  4: return RATE_OCTAVE;
 634         case  1: return SILENCE;
 635     }
 636
 637     return I_F_Q;
 638 }
 639
 640 /**
 641  * Determine the bitrate from the frame size and/or the first byte of the frame.
 642  *
 643  * @param avctx the AV codec context
 644  * @param buf_size length of the buffer
 645  * @param buf the bufffer
 646  *
 647  * @return the bitrate on success,
 648  *         I_F_Q  if the bitrate cannot be satisfactorily determined
 649  *
 650  * TIA/EIA/IS-733 2.4.8.7.1
 651  */
 652 static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx, const int buf_size,
 653                              const uint8_t **buf)
 654 {
 655     qcelp_packet_rate bitrate;
 656
 657     if((bitrate = buf_size2bitrate(buf_size)) >= 0)
 658     {
 659         if(bitrate > **buf)
 660         {
 661             QCELPContext *q = avctx->priv_data;
 662             if (!q->warned_buf_mismatch_bitrate)
 663             {
 664             av_log(avctx, AV_LOG_WARNING,
 665                    "Claimed bitrate and buffer size mismatch.\n");
 666                 q->warned_buf_mismatch_bitrate = 1;
 667             }
 668             bitrate = **buf;
 669         }else if(bitrate < **buf)
 670         {
 671             av_log(avctx, AV_LOG_ERROR,
 672                    "Buffer is too small for the claimed bitrate.\n");
 673             return I_F_Q;
 674         }
 675         (*buf)++;
 676     }else if((bitrate = buf_size2bitrate(buf_size + 1)) >= 0)
 677     {
 678         av_log(avctx, AV_LOG_WARNING,
 679                "Bitrate byte is missing, guessing the bitrate from packet size.\n");
 680     }else
 681         return I_F_Q;
 682
 683     if(bitrate == SILENCE)
 684     {
 685         //FIXME: Remove experimental warning when tested with samples.
 686         av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
 687     }
 688     return bitrate;
 689 }
 690
 691 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
 692                                             const char *message)
 693 {
 694     av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
 695            message);
 696 }
 697
 698 static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
 699                               AVPacket *avpkt)
 700 {
 701     const uint8_t *buf = avpkt->data;
 702     int buf_size = avpkt->size;
 703     QCELPContext *q = avctx->priv_data;
 704     float *outbuffer = data;
 705     int   i;
 706     float quantized_lspf[10], lpc[10];
 707     float gain[16];
 708     float *formant_mem;
 709
 710     if((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q)
 711     {
 712         warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
 713         goto erasure;
 714     }
 715
 716     if(q->bitrate == RATE_OCTAVE &&
 717        (q->first16bits = AV_RB16(buf)) == 0xFFFF)
 718     {
 719         warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
 720         goto erasure;
 721     }
 722
 723     if(q->bitrate > SILENCE)
 724     {
 725         const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
 726         const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate]
 727                                        + qcelp_unpacking_bitmaps_lengths[q->bitrate];
 728         uint8_t           *unpacked_data = (uint8_t *)&q->frame;
 729
 730         init_get_bits(&q->gb, buf, 8*buf_size);
 731
 732         memset(&q->frame, 0, sizeof(QCELPFrame));
 733
 734         for(; bitmaps < bitmaps_end; bitmaps++)
 735             unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
 736
 737         // Check for erasures/blanks on rates 1, 1/4 and 1/8.
 738         if(q->frame.reserved)
 739         {
 740             warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
 741             goto erasure;
 742         }
 743         if(q->bitrate == RATE_QUARTER &&
 744            codebook_sanity_check_for_rate_quarter(q->frame.cbgain))
 745         {
 746             warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
 747             goto erasure;
 748         }
 749
 750         if(q->bitrate >= RATE_HALF)
 751         {
 752             for(i=0; i<4; i++)
 753             {
 754                 if(q->frame.pfrac[i] && q->frame.plag[i] >= 124)
 755                 {
 756                     warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
 757                     goto erasure;
 758                 }
 759             }
 760         }
 761     }
 762
 763     decode_gain_and_index(q, gain);
 764     compute_svector(q, gain, outbuffer);
 765
 766     if(decode_lspf(q, quantized_lspf) < 0)
 767     {
 768         warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
 769         goto erasure;
 770     }
 771
 772
 773     apply_pitch_filters(q, outbuffer);
 774
 775     if(q->bitrate == I_F_Q)
 776     {
 777 erasure:
 778         q->bitrate = I_F_Q;
 779         q->erasure_count++;
 780         decode_gain_and_index(q, gain);
 781         compute_svector(q, gain, outbuffer);
 782         decode_lspf(q, quantized_lspf);
 783         apply_pitch_filters(q, outbuffer);
 784     }else
 785         q->erasure_count = 0;
 786
 787     formant_mem = q->formant_mem + 10;
 788     for(i=0; i<4; i++)
 789     {
 790         interpolate_lpc(q, quantized_lspf, lpc, i);
 791         ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40,
 792                                      10);
 793         formant_mem += 40;
 794     }
 795     memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
 796
 797     // FIXME: postfilter and final gain control should be here.
 798     // TIA/EIA/IS-733 2.4.8.6
 799
 800     formant_mem = q->formant_mem + 10;
 801     for(i=0; i<160; i++)
 802         *outbuffer++ = av_clipf(*formant_mem++, QCELP_CLIP_LOWER_BOUND,
 803                                 QCELP_CLIP_UPPER_BOUND);
 804
 805     memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
 806     q->prev_bitrate = q->bitrate;
 807
 808     *data_size = 160 * sizeof(*outbuffer);
 809
 810     return *data_size;
 811 }
 812
 813 AVCodec qcelp_decoder =
 814 {
 815     .name   = "qcelp",
 816     .type   = CODEC_TYPE_AUDIO,
 817     .id     = CODEC_ID_QCELP,
 818     .init   = qcelp_decode_init,
 819     .decode = qcelp_decode_frame,
 820     .priv_data_size = sizeof(QCELPContext),
 821     .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
 822 };