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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 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 "bitstream.h"
34
35 #include "qcelpdata.h"
36
37 #include "celp_math.h"
38 #include "celp_filters.h"
39
40 #undef NDEBUG
41 #include <assert.h>
42
43 typedef enum
44 {
45     I_F_Q = -1,    /*!< insufficient frame quality */
46     SILENCE,
47     RATE_OCTAVE,
48     RATE_QUARTER,
49     RATE_HALF,
50     RATE_FULL
51 } qcelp_packet_rate;
52
53 typedef struct
54 {
55     GetBitContext     gb;
56     qcelp_packet_rate bitrate;
57     QCELPFrame        frame;    /*!< unpacked data frame */
58
59     uint8_t  erasure_count;
60     uint8_t  octave_count;      /*!< count the consecutive RATE_OCTAVE frames */
61     float    prev_lspf[10];
62     float    predictor_lspf[10];/*!< LSP predictor for RATE_OCTAVE and I_F_Q */
63     float    pitch_synthesis_filter_mem[303];
64     float    pitch_pre_filter_mem[303];
65     float    rnd_fir_filter_mem[180];
66     float    formant_mem[170];
67     float    last_codebook_gain;
68     int      prev_g1[2];
69     int      prev_bitrate;
70     float    pitch_gain[4];
71     uint8_t  pitch_lag[4];
72     uint16_t first16bits;
73 } QCELPContext;
74
75 /**
76  * Reconstructs LPC coefficients from the line spectral pair frequencies.
77  *
78  * TIA/EIA/IS-733 2.4.3.3.5
79  */
80 void ff_qcelp_lspf2lpc(const float *lspf, float *lpc);
81
82 static void weighted_vector_sumf(float *out, const float *in_a,
83                                  const float *in_b, float weight_coeff_a,
84                                  float weight_coeff_b, int length)
85 {
86     int i;
87
88     for(i=0; i<length; i++)
89         out[i] = weight_coeff_a * in_a[i]
90                + weight_coeff_b * in_b[i];
91 }
92
93 /**
94  * Initialize the speech codec according to the specification.
95  *
96  * TIA/EIA/IS-733 2.4.9
97  */
98 static av_cold int qcelp_decode_init(AVCodecContext *avctx)
99 {
100     QCELPContext *q = avctx->priv_data;
101     int i;
102
103     avctx->sample_fmt = SAMPLE_FMT_FLT;
104
105     for(i=0; i<10; i++)
106         q->prev_lspf[i] = (i+1)/11.;
107
108     return 0;
109 }
110
111 /**
112  * Decodes the 10 quantized LSP frequencies from the LSPV/LSP
113  * transmission codes of any bitrate and checks for badly received packets.
114  *
115  * @param q the context
116  * @param lspf line spectral pair frequencies
117  *
118  * @return 0 on success, -1 if the packet is badly received
119  *
120  * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
121  */
122 static int decode_lspf(QCELPContext *q, float *lspf)
123 {
124     int i;
125     float tmp_lspf, smooth, erasure_coeff;
126     const float *predictors;
127
128     if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q)
129     {
130         predictors = (q->prev_bitrate != RATE_OCTAVE &&
131                        q->prev_bitrate != I_F_Q ?
132                        q->prev_lspf : q->predictor_lspf);
133
134         if(q->bitrate == RATE_OCTAVE)
135         {
136             q->octave_count++;
137
138             for(i=0; i<10; i++)
139             {
140                 q->predictor_lspf[i] =
141                              lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
142                                                          : -QCELP_LSP_SPREAD_FACTOR)
143                                      + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR
144                                      + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
145             }
146             smooth = (q->octave_count < 10 ? .875 : 0.1);
147         }else
148         {
149             erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
150
151             assert(q->bitrate == I_F_Q);
152
153             if(q->erasure_count > 1)
154                 erasure_coeff *= (q->erasure_count < 4 ? 0.9 : 0.7);
155
156             for(i=0; i<10; i++)
157             {
158                 q->predictor_lspf[i] =
159                              lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
160                                      + erasure_coeff * predictors[i];
161             }
162             smooth = 0.125;
163         }
164
165         // Check the stability of the LSP frequencies.
166         lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
167         for(i=1; i<10; i++)
168             lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR));
169
170         lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR));
171         for(i=9; i>0; i--)
172             lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
173
174         // Low-pass filter the LSP frequencies.
175         weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
176     }else
177     {
178         q->octave_count = 0;
179
180         tmp_lspf = 0.;
181         for(i=0; i<5 ; i++)
182         {
183             lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
184             lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
185         }
186
187         // Check for badly received packets.
188         if(q->bitrate == RATE_QUARTER)
189         {
190             if(lspf[9] <= .70 || lspf[9] >=  .97)
191                 return -1;
192             for(i=3; i<10; i++)
193                 if(fabs(lspf[i] - lspf[i-2]) < .08)
194                     return -1;
195         }else
196         {
197             if(lspf[9] <= .66 || lspf[9] >= .985)
198                 return -1;
199             for(i=4; i<10; i++)
200                 if (fabs(lspf[i] - lspf[i-4]) < .0931)
201                     return -1;
202         }
203     }
204     return 0;
205 }
206
207 /**
208  * Converts codebook transmission codes to GAIN and INDEX.
209  *
210  * @param q the context
211  * @param gain array holding the decoded gain
212  *
213  * TIA/EIA/IS-733 2.4.6.2
214  */
215 static void decode_gain_and_index(QCELPContext  *q,
216                                   float *gain) {
217     int   i, subframes_count, g1[16];
218     float slope;
219
220     if(q->bitrate >= RATE_QUARTER)
221     {
222         switch(q->bitrate)
223         {
224             case RATE_FULL: subframes_count = 16; break;
225             case RATE_HALF: subframes_count = 4;  break;
226             default:        subframes_count = 5;
227         }
228         for(i=0; i<subframes_count; i++)
229         {
230             g1[i] = 4 * q->frame.cbgain[i];
231             if(q->bitrate == RATE_FULL && !((i+1) & 3))
232             {
233                 g1[i] += av_clip((g1[i-1] + g1[i-2] + g1[i-3]) / 3 - 6, 0, 32);
234             }
235
236             gain[i] = qcelp_g12ga[g1[i]];
237
238             if(q->frame.cbsign[i])
239             {
240                 gain[i] = -gain[i];
241                 q->frame.cindex[i] = (q->frame.cindex[i]-89) & 127;
242             }
243         }
244
245         q->prev_g1[0] = g1[i-2];
246         q->prev_g1[1] = g1[i-1];
247         q->last_codebook_gain = qcelp_g12ga[g1[i-1]];
248
249         if(q->bitrate == RATE_QUARTER)
250         {
251             // Provide smoothing of the unvoiced excitation energy.
252             gain[7] =     gain[4];
253             gain[6] = 0.4*gain[3] + 0.6*gain[4];
254             gain[5] =     gain[3];
255             gain[4] = 0.8*gain[2] + 0.2*gain[3];
256             gain[3] = 0.2*gain[1] + 0.8*gain[2];
257             gain[2] =     gain[1];
258             gain[1] = 0.6*gain[0] + 0.4*gain[1];
259         }
260     }else
261     {
262         if(q->bitrate == RATE_OCTAVE)
263         {
264             g1[0] = 2 * q->frame.cbgain[0]
265                   + av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
266             subframes_count = 8;
267         }else
268         {
269             assert(q->bitrate == I_F_Q);
270
271             g1[0] = q->prev_g1[1];
272             switch(q->erasure_count)
273             {
274                 case 1 : break;
275                 case 2 : g1[0] -= 1; break;
276                 case 3 : g1[0] -= 2; break;
277                 default: g1[0] -= 6;
278             }
279             if(g1[0] < 0)
280                 g1[0] = 0;
281             subframes_count = 4;
282         }
283         // This interpolation is done to produce smoother background noise.
284         slope = 0.5*(qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
285         for(i=1; i<=subframes_count; i++)
286             gain[i-1] = q->last_codebook_gain + slope * i;
287
288         q->last_codebook_gain = gain[i-2];
289         q->prev_g1[0] = q->prev_g1[1];
290         q->prev_g1[1] = g1[0];
291     }
292 }
293
294 /**
295  * If the received packet is Rate 1/4 a further sanity check is made of the
296  * codebook gain.
297  *
298  * @param cbgain the unpacked cbgain array
299  * @return -1 if the sanity check fails, 0 otherwise
300  *
301  * TIA/EIA/IS-733 2.4.8.7.3
302  */
303 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
304 {
305     int i, diff, prev_diff=0;
306
307     for(i=1; i<5; i++)
308     {
309         diff = cbgain[i] - cbgain[i-1];
310         if(FFABS(diff) > 10)
311             return -1;
312         else if(FFABS(diff - prev_diff) > 12)
313             return -1;
314         prev_diff = diff;
315     }
316     return 0;
317 }
318
319 /**
320  * Computes the scaled codebook vector Cdn From INDEX and GAIN
321  * for all rates.
322  *
323  * The specification lacks some information here.
324  *
325  * TIA/EIA/IS-733 has an omission on the codebook index determination
326  * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
327  * you have to subtract the decoded index parameter from the given scaled
328  * codebook vector index 'n' to get the desired circular codebook index, but
329  * it does not mention that you have to clamp 'n' to [0-9] in order to get
330  * RI-compliant results.
331  *
332  * The reason for this mistake seems to be the fact they forgot to mention you
333  * have to do these calculations per codebook subframe and adjust given
334  * equation values accordingly.
335  *
336  * @param q the context
337  * @param gain array holding the 4 pitch subframe gain values
338  * @param cdn_vector array for the generated scaled codebook vector
339  */
340 static void compute_svector(QCELPContext *q, const float *gain,
341                             float *cdn_vector)
342 {
343     int      i, j, k;
344     uint16_t cbseed, cindex;
345     float    *rnd, tmp_gain, fir_filter_value;
346
347     switch(q->bitrate)
348     {
349         case RATE_FULL:
350             for(i=0; i<16; i++)
351             {
352                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
353                 cindex = -q->frame.cindex[i];
354                 for(j=0; j<10; j++)
355                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
356             }
357         break;
358         case RATE_HALF:
359             for(i=0; i<4; i++)
360             {
361                 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
362                 cindex = -q->frame.cindex[i];
363                 for (j = 0; j < 40; j++)
364                 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
365             }
366         break;
367         case RATE_QUARTER:
368             cbseed = (0x0003 & q->frame.lspv[4])<<14 |
369                      (0x003F & q->frame.lspv[3])<< 8 |
370                      (0x0060 & q->frame.lspv[2])<< 1 |
371                      (0x0007 & q->frame.lspv[1])<< 3 |
372                      (0x0038 & q->frame.lspv[0])>> 3 ;
373             rnd = q->rnd_fir_filter_mem + 20;
374             for(i=0; i<8; i++)
375             {
376                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
377                 for(k=0; k<20; k++)
378                 {
379                     cbseed = 521 * cbseed + 259;
380                     *rnd = (int16_t)cbseed;
381
382                     // FIR filter
383                     fir_filter_value = 0.0;
384                     for(j=0; j<10; j++)
385                         fir_filter_value += qcelp_rnd_fir_coefs[j ]
386                                           * (rnd[-j ] + rnd[-20+j]);
387
388                     fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
389                     *cdn_vector++ = tmp_gain * fir_filter_value;
390                     rnd++;
391                 }
392             }
393             memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
394         break;
395         case RATE_OCTAVE:
396             cbseed = q->first16bits;
397             for(i=0; i<8; i++)
398             {
399                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
400                 for(j=0; j<20; j++)
401                 {
402                     cbseed = 521 * cbseed + 259;
403                     *cdn_vector++ = tmp_gain * (int16_t)cbseed;
404                 }
405             }
406         break;
407         case I_F_Q:
408             cbseed = -44; // random codebook index
409             for(i=0; i<4; i++)
410             {
411                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
412                 for(j=0; j<40; j++)
413                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
414             }
415         break;
416     }
417 }
418
419 /**
420  * Apply generic gain control.
421  *
422  * @param v_out output vector
423  * @param v_in gain-controlled vector
424  * @param v_ref vector to control gain of
425  *
426  * FIXME: If v_ref is a zero vector, it energy is zero
427  *        and the behavior of the gain control is
428  *        undefined in the specs.
429  *
430  * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
431  */
432 static void apply_gain_ctrl(float *v_out, const float *v_ref,
433                             const float *v_in)
434 {
435     int   i, j, len;
436     float scalefactor;
437
438     for(i=0, j=0; i<4; i++)
439     {
440         scalefactor = ff_dot_productf(v_in + j, v_in + j, 40);
441         if(scalefactor)
442             scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40)
443                         / scalefactor);
444         else
445             av_log_missing_feature(NULL, "Zero energy for gain control", 1);
446         for(len=j+40; j<len; j++)
447             v_out[j] = scalefactor * v_in[j];
448     }
449 }
450
451 /**
452  * Apply filter in pitch-subframe steps.
453  *
454  * @param memory buffer for the previous state of the filter
455  *        - must be able to contain 303 elements
456  *        - the 143 first elements are from the previous state
457  *        - the next 160 are for output
458  * @param v_in input filter vector
459  * @param gain per-subframe gain array, each element is between 0.0 and 2.0
460  * @param lag per-subframe lag array, each element is
461  *        - between 16 and 143 if its corresponding pfrac is 0,
462  *        - between 16 and 139 otherwise
463  * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
464  *        otherwise
465  *
466  * @return filter output vector
467  */
468 static const float *do_pitchfilter(float memory[303], const float v_in[160],
469                                    const float gain[4], const uint8_t *lag,
470                                    const uint8_t pfrac[4])
471 {
472     int         i, j;
473     float       *v_lag, *v_out;
474     const float *v_len;
475
476     v_out = memory + 143; // Output vector starts at memory[143].
477
478     for(i=0; i<4; i++)
479     {
480         if(gain[i])
481         {
482             v_lag = memory + 143 + 40 * i - lag[i];
483             for(v_len=v_in+40; v_in<v_len; v_in++)
484             {
485                 if(pfrac[i]) // If it is a fractional lag...
486                 {
487                     for(j=0, *v_out=0.; j<4; j++)
488                         *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
489                 }else
490                     *v_out = *v_lag;
491
492                 *v_out = *v_in + gain[i] * *v_out;
493
494                 v_lag++;
495                 v_out++;
496             }
497         }else
498         {
499             memcpy(v_out, v_in, 40 * sizeof(float));
500             v_in  += 40;
501             v_out += 40;
502         }
503     }
504
505     memmove(memory, memory + 160, 143 * sizeof(float));
506     return memory + 143;
507 }
508
509 /**
510  * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
511  * TIA/EIA/IS-733 2.4.5.2
512  *
513  * @param q the context
514  * @param cdn_vector the scaled codebook vector
515  */
516 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
517 {
518     int         i;
519     const float *v_synthesis_filtered, *v_pre_filtered;
520
521     if(q->bitrate >= RATE_HALF ||
522        (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF)))
523     {
524
525         if(q->bitrate >= RATE_HALF)
526         {
527
528             // Compute gain & lag for the whole frame.
529             for(i=0; i<4; i++)
530             {
531                 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
532
533                 q->pitch_lag[i] = q->frame.plag[i] + 16;
534             }
535         }else
536         {
537             float max_pitch_gain = q->erasure_count < 3 ? 0.9 - 0.3 * (q->erasure_count - 1) : 0.0;
538             for(i=0; i<4; i++)
539                 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
540
541             memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
542         }
543
544         // pitch synthesis filter
545         v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
546                                               cdn_vector, q->pitch_gain,
547                                               q->pitch_lag, q->frame.pfrac);
548
549         // pitch prefilter update
550         for(i=0; i<4; i++)
551             q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
552
553         v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
554                                         v_synthesis_filtered,
555                                         q->pitch_gain, q->pitch_lag,
556                                         q->frame.pfrac);
557
558         apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
559     }else
560     {
561         memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17,
562                143 * sizeof(float));
563         memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
564         memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
565         memset(q->pitch_lag,  0, sizeof(q->pitch_lag));
566     }
567 }
568
569 /**
570  * Interpolates LSP frequencies and computes LPC coefficients
571  * for a given bitrate & pitch subframe.
572  *
573  * TIA/EIA/IS-733 2.4.3.3.4
574  *
575  * @param q the context
576  * @param curr_lspf LSP frequencies vector of the current frame
577  * @param lpc float vector for the resulting LPC
578  * @param subframe_num frame number in decoded stream
579  */
580 void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc,
581                      const int subframe_num)
582 {
583     float interpolated_lspf[10];
584     float weight;
585
586     if(q->bitrate >= RATE_QUARTER)
587         weight = 0.25 * (subframe_num + 1);
588     else if(q->bitrate == RATE_OCTAVE && !subframe_num)
589         weight = 0.625;
590     else
591         weight = 1.0;
592
593     if(weight != 1.0)
594     {
595         weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
596                              weight, 1.0 - weight, 10);
597         ff_qcelp_lspf2lpc(interpolated_lspf, lpc);
598     }else if(q->bitrate >= RATE_QUARTER ||
599              (q->bitrate == I_F_Q && !subframe_num))
600         ff_qcelp_lspf2lpc(curr_lspf, lpc);
601 }
602
603 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
604 {
605     switch(buf_size)
606     {
607         case 35: return RATE_FULL;
608         case 17: return RATE_HALF;
609         case  8: return RATE_QUARTER;
610         case  4: return RATE_OCTAVE;
611         case  1: return SILENCE;
612     }
613
614     return I_F_Q;
615 }
616
617 /**
618  * Determine the bitrate from the frame size and/or the first byte of the frame.
619  *
620  * @param avctx the AV codec context
621  * @param buf_size length of the buffer
622  * @param buf the bufffer
623  *
624  * @return the bitrate on success,
625  *         I_F_Q  if the bitrate cannot be satisfactorily determined
626  *
627  * TIA/EIA/IS-733 2.4.8.7.1
628  */
629 static int determine_bitrate(AVCodecContext *avctx, const int buf_size,
630                              const uint8_t **buf)
631 {
632     qcelp_packet_rate bitrate;
633
634     if((bitrate = buf_size2bitrate(buf_size)) >= 0)
635     {
636         if(bitrate > **buf)
637         {
638             av_log(avctx, AV_LOG_WARNING,
639                    "Claimed bitrate and buffer size mismatch.\n");
640             bitrate = **buf;
641         }else if(bitrate < **buf)
642         {
643             av_log(avctx, AV_LOG_ERROR,
644                    "Buffer is too small for the claimed bitrate.\n");
645             return I_F_Q;
646         }
647         (*buf)++;
648     }else if((bitrate = buf_size2bitrate(buf_size + 1)) >= 0)
649     {
650         av_log(avctx, AV_LOG_WARNING,
651                "Bitrate byte is missing, guessing the bitrate from packet size.\n");
652     }else
653         return I_F_Q;
654
655     if(bitrate == SILENCE)
656     {
657         // FIXME: the decoder should not handle SILENCE frames as I_F_Q frames
658         av_log_missing_feature(avctx, "Blank frame", 1);
659         bitrate = I_F_Q;
660     }
661     return bitrate;
662 }
663
664 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
665                                             const char *message)
666 {
667     av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
668            message);
669 }
670
671 static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
672                               const uint8_t *buf, int buf_size)
673 {
674     QCELPContext *q = avctx->priv_data;
675     float *outbuffer = data;
676     int   i;
677     float quantized_lspf[10], lpc[10];
678     float gain[16];
679     float *formant_mem;
680
681     if((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q)
682     {
683         warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
684         goto erasure;
685     }
686
687     if(q->bitrate == RATE_OCTAVE &&
688        (q->first16bits = AV_RB16(buf)) == 0xFFFF)
689     {
690         warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
691         goto erasure;
692     }
693
694     if(q->bitrate > SILENCE)
695     {
696         const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
697         const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate]
698                                        + qcelp_unpacking_bitmaps_lengths[q->bitrate];
699         uint8_t           *unpacked_data = (uint8_t *)&q->frame;
700
701         init_get_bits(&q->gb, buf, 8*buf_size);
702
703         memset(&q->frame, 0, sizeof(QCELPFrame));
704
705         for(; bitmaps < bitmaps_end; bitmaps++)
706             unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
707
708         // Check for erasures/blanks on rates 1, 1/4 and 1/8.
709         if(q->frame.reserved)
710         {
711             warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
712             goto erasure;
713         }
714         if(q->bitrate == RATE_QUARTER &&
715            codebook_sanity_check_for_rate_quarter(q->frame.cbgain))
716         {
717             warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
718             goto erasure;
719         }
720
721         if(q->bitrate >= RATE_HALF)
722         {
723             for(i=0; i<4; i++)
724             {
725                 if(q->frame.pfrac[i] && q->frame.plag[i] >= 124)
726                 {
727                     warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
728                     goto erasure;
729                 }
730             }
731         }
732     }
733
734     decode_gain_and_index(q, gain);
735     compute_svector(q, gain, outbuffer);
736
737     if(decode_lspf(q, quantized_lspf) < 0)
738     {
739         warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
740         goto erasure;
741     }
742
743
744     apply_pitch_filters(q, outbuffer);
745
746     if(q->bitrate == I_F_Q)
747     {
748 erasure:
749         q->bitrate = I_F_Q;
750         q->erasure_count++;
751         decode_gain_and_index(q, gain);
752         compute_svector(q, gain, outbuffer);
753         decode_lspf(q, quantized_lspf);
754         apply_pitch_filters(q, outbuffer);
755     }else
756         q->erasure_count = 0;
757
758     formant_mem = q->formant_mem + 10;
759     for(i=0; i<4; i++)
760     {
761         interpolate_lpc(q, quantized_lspf, lpc, i);
762         ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40,
763                                      10);
764         formant_mem += 40;
765     }
766     memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
767
768     // FIXME: postfilter and final gain control should be here.
769     // TIA/EIA/IS-733 2.4.8.6
770
771     formant_mem = q->formant_mem + 10;
772     for(i=0; i<160; i++)
773         *outbuffer++ = av_clipf(*formant_mem++, QCELP_CLIP_LOWER_BOUND,
774                                 QCELP_CLIP_UPPER_BOUND);
775
776     memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
777     q->prev_bitrate = q->bitrate;
778
779     *data_size = 160 * sizeof(*outbuffer);
780
781     return *data_size;
782 }
783
784 AVCodec qcelp_decoder =
785 {
786     .name   = "qcelp",
787     .type   = CODEC_TYPE_AUDIO,
788     .id     = CODEC_ID_QCELP,
789     .init   = qcelp_decode_init,
790     .decode = qcelp_decode_frame,
791     .priv_data_size = sizeof(QCELPContext),
792     .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
793 };