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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 "internal.h"
34 #include "bitstream.h"
35
36 #include "qcelpdata.h"
37
38 #include "celp_math.h"
39 #include "celp_filters.h"
40
41 #undef NDEBUG
42 #include <assert.h>
43
44 typedef enum
45 {
46     I_F_Q = -1,    /*!< insufficient frame quality */
47     SILENCE,
48     RATE_OCTAVE,
49     RATE_QUARTER,
50     RATE_HALF,
51     RATE_FULL
52 } qcelp_packet_rate;
53
54 typedef struct
55 {
56     GetBitContext     gb;
57     qcelp_packet_rate bitrate;
58     QCELPFrame        frame;    /*!< unpacked data frame */
59
60     uint8_t  erasure_count;
61     uint8_t  octave_count;      /*!< count the consecutive RATE_OCTAVE frames */
62     float    prev_lspf[10];
63     float    predictor_lspf[10];/*!< LSP predictor for RATE_OCTAVE and I_F_Q */
64     float    pitch_synthesis_filter_mem[303];
65     float    pitch_pre_filter_mem[303];
66     float    rnd_fir_filter_mem[180];
67     float    formant_mem[170];
68     float    last_codebook_gain;
69     int      prev_g1[2];
70     int      prev_bitrate;
71     float    pitch_gain[4];
72     uint8_t  pitch_lag[4];
73     uint16_t first16bits;
74 } QCELPContext;
75
76 /**
77  * Reconstructs LPC coefficients from the line spectral pair frequencies.
78  *
79  * TIA/EIA/IS-733 2.4.3.3.5
80  */
81 void ff_qcelp_lspf2lpc(const float *lspf, float *lpc);
82
83 static void weighted_vector_sumf(float *out, const float *in_a,
84                                  const float *in_b, float weight_coeff_a,
85                                  float weight_coeff_b, int length)
86 {
87     int i;
88
89     for(i=0; i<length; i++)
90         out[i] = weight_coeff_a * in_a[i]
91                + weight_coeff_b * in_b[i];
92 }
93
94 /**
95  * Initialize the speech codec according to the specification.
96  *
97  * TIA/EIA/IS-733 2.4.9
98  */
99 static av_cold int qcelp_decode_init(AVCodecContext *avctx)
100 {
101     QCELPContext *q = avctx->priv_data;
102     int i;
103
104     avctx->sample_fmt = SAMPLE_FMT_FLT;
105
106     for(i=0; i<10; i++)
107         q->prev_lspf[i] = (i+1)/11.;
108
109     return 0;
110 }
111
112 /**
113  * Decodes the 10 quantized LSP frequencies from the LSPV/LSP
114  * transmission codes of any bitrate and checks for badly received packets.
115  *
116  * @param q the context
117  * @param lspf line spectral pair frequencies
118  *
119  * @return 0 on success, -1 if the packet is badly received
120  *
121  * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
122  */
123 static int decode_lspf(QCELPContext *q, float *lspf)
124 {
125     int i;
126     float tmp_lspf, smooth, erasure_coeff;
127     const float *predictors;
128
129     if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q)
130     {
131         predictors = (q->prev_bitrate != RATE_OCTAVE &&
132                        q->prev_bitrate != I_F_Q ?
133                        q->prev_lspf : q->predictor_lspf);
134
135         if(q->bitrate == RATE_OCTAVE)
136         {
137             q->octave_count++;
138
139             for(i=0; i<10; i++)
140             {
141                 q->predictor_lspf[i] =
142                              lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
143                                                          : -QCELP_LSP_SPREAD_FACTOR)
144                                      + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR
145                                      + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
146             }
147             smooth = (q->octave_count < 10 ? .875 : 0.1);
148         }else
149         {
150             erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
151
152             assert(q->bitrate == I_F_Q);
153
154             if(q->erasure_count > 1)
155                 erasure_coeff *= (q->erasure_count < 4 ? 0.9 : 0.7);
156
157             for(i=0; i<10; i++)
158             {
159                 q->predictor_lspf[i] =
160                              lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
161                                      + erasure_coeff * predictors[i];
162             }
163             smooth = 0.125;
164         }
165
166         // Check the stability of the LSP frequencies.
167         lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
168         for(i=1; i<10; i++)
169             lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR));
170
171         lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR));
172         for(i=9; i>0; i--)
173             lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
174
175         // Low-pass filter the LSP frequencies.
176         weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
177     }else
178     {
179         q->octave_count = 0;
180
181         tmp_lspf = 0.;
182         for(i=0; i<5 ; i++)
183         {
184             lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
185             lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
186         }
187
188         // Check for badly received packets.
189         if(q->bitrate == RATE_QUARTER)
190         {
191             if(lspf[9] <= .70 || lspf[9] >=  .97)
192                 return -1;
193             for(i=3; i<10; i++)
194                 if(fabs(lspf[i] - lspf[i-2]) < .08)
195                     return -1;
196         }else
197         {
198             if(lspf[9] <= .66 || lspf[9] >= .985)
199                 return -1;
200             for(i=4; i<10; i++)
201                 if (fabs(lspf[i] - lspf[i-4]) < .0931)
202                     return -1;
203         }
204     }
205     return 0;
206 }
207
208 /**
209  * Converts codebook transmission codes to GAIN and INDEX.
210  *
211  * @param q the context
212  * @param gain array holding the decoded gain
213  *
214  * TIA/EIA/IS-733 2.4.6.2
215  */
216 static void decode_gain_and_index(QCELPContext  *q,
217                                   float *gain) {
218     int   i, subframes_count, g1[16];
219     float slope;
220
221     if(q->bitrate >= RATE_QUARTER)
222     {
223         switch(q->bitrate)
224         {
225             case RATE_FULL: subframes_count = 16; break;
226             case RATE_HALF: subframes_count = 4;  break;
227             default:        subframes_count = 5;
228         }
229         for(i=0; i<subframes_count; i++)
230         {
231             g1[i] = 4 * q->frame.cbgain[i];
232             if(q->bitrate == RATE_FULL && !((i+1) & 3))
233             {
234                 g1[i] += av_clip((g1[i-1] + g1[i-2] + g1[i-3]) / 3 - 6, 0, 32);
235             }
236
237             gain[i] = qcelp_g12ga[g1[i]];
238
239             if(q->frame.cbsign[i])
240             {
241                 gain[i] = -gain[i];
242                 q->frame.cindex[i] = (q->frame.cindex[i]-89) & 127;
243             }
244         }
245
246         q->prev_g1[0] = g1[i-2];
247         q->prev_g1[1] = g1[i-1];
248         q->last_codebook_gain = qcelp_g12ga[g1[i-1]];
249
250         if(q->bitrate == RATE_QUARTER)
251         {
252             // Provide smoothing of the unvoiced excitation energy.
253             gain[7] =     gain[4];
254             gain[6] = 0.4*gain[3] + 0.6*gain[4];
255             gain[5] =     gain[3];
256             gain[4] = 0.8*gain[2] + 0.2*gain[3];
257             gain[3] = 0.2*gain[1] + 0.8*gain[2];
258             gain[2] =     gain[1];
259             gain[1] = 0.6*gain[0] + 0.4*gain[1];
260         }
261     }else
262     {
263         if(q->bitrate == RATE_OCTAVE)
264         {
265             g1[0] = 2 * q->frame.cbgain[0]
266                   + av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
267             subframes_count = 8;
268         }else
269         {
270             assert(q->bitrate == I_F_Q);
271
272             g1[0] = q->prev_g1[1];
273             switch(q->erasure_count)
274             {
275                 case 1 : break;
276                 case 2 : g1[0] -= 1; break;
277                 case 3 : g1[0] -= 2; break;
278                 default: g1[0] -= 6;
279             }
280             if(g1[0] < 0)
281                 g1[0] = 0;
282             subframes_count = 4;
283         }
284         // This interpolation is done to produce smoother background noise.
285         slope = 0.5*(qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
286         for(i=1; i<=subframes_count; i++)
287             gain[i-1] = q->last_codebook_gain + slope * i;
288
289         q->last_codebook_gain = gain[i-2];
290         q->prev_g1[0] = q->prev_g1[1];
291         q->prev_g1[1] = g1[0];
292     }
293 }
294
295 /**
296  * If the received packet is Rate 1/4 a further sanity check is made of the
297  * codebook gain.
298  *
299  * @param cbgain the unpacked cbgain array
300  * @return -1 if the sanity check fails, 0 otherwise
301  *
302  * TIA/EIA/IS-733 2.4.8.7.3
303  */
304 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
305 {
306     int i, diff, prev_diff=0;
307
308     for(i=1; i<5; i++)
309     {
310         diff = cbgain[i] - cbgain[i-1];
311         if(FFABS(diff) > 10)
312             return -1;
313         else if(FFABS(diff - prev_diff) > 12)
314             return -1;
315         prev_diff = diff;
316     }
317     return 0;
318 }
319
320 /**
321  * Computes the scaled codebook vector Cdn From INDEX and GAIN
322  * for all rates.
323  *
324  * The specification lacks some information here.
325  *
326  * TIA/EIA/IS-733 has an omission on the codebook index determination
327  * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
328  * you have to subtract the decoded index parameter from the given scaled
329  * codebook vector index 'n' to get the desired circular codebook index, but
330  * it does not mention that you have to clamp 'n' to [0-9] in order to get
331  * RI-compliant results.
332  *
333  * The reason for this mistake seems to be the fact they forgot to mention you
334  * have to do these calculations per codebook subframe and adjust given
335  * equation values accordingly.
336  *
337  * @param q the context
338  * @param gain array holding the 4 pitch subframe gain values
339  * @param cdn_vector array for the generated scaled codebook vector
340  */
341 static void compute_svector(QCELPContext *q, const float *gain,
342                             float *cdn_vector)
343 {
344     int      i, j, k;
345     uint16_t cbseed, cindex;
346     float    *rnd, tmp_gain, fir_filter_value;
347
348     switch(q->bitrate)
349     {
350         case RATE_FULL:
351             for(i=0; i<16; i++)
352             {
353                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
354                 cindex = -q->frame.cindex[i];
355                 for(j=0; j<10; j++)
356                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
357             }
358         break;
359         case RATE_HALF:
360             for(i=0; i<4; i++)
361             {
362                 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
363                 cindex = -q->frame.cindex[i];
364                 for (j = 0; j < 40; j++)
365                 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
366             }
367         break;
368         case RATE_QUARTER:
369             cbseed = (0x0003 & q->frame.lspv[4])<<14 |
370                      (0x003F & q->frame.lspv[3])<< 8 |
371                      (0x0060 & q->frame.lspv[2])<< 1 |
372                      (0x0007 & q->frame.lspv[1])<< 3 |
373                      (0x0038 & q->frame.lspv[0])>> 3 ;
374             rnd = q->rnd_fir_filter_mem + 20;
375             for(i=0; i<8; i++)
376             {
377                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
378                 for(k=0; k<20; k++)
379                 {
380                     cbseed = 521 * cbseed + 259;
381                     *rnd = (int16_t)cbseed;
382
383                     // FIR filter
384                     fir_filter_value = 0.0;
385                     for(j=0; j<10; j++)
386                         fir_filter_value += qcelp_rnd_fir_coefs[j ]
387                                           * (rnd[-j ] + rnd[-20+j]);
388
389                     fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
390                     *cdn_vector++ = tmp_gain * fir_filter_value;
391                     rnd++;
392                 }
393             }
394             memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
395         break;
396         case RATE_OCTAVE:
397             cbseed = q->first16bits;
398             for(i=0; i<8; i++)
399             {
400                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
401                 for(j=0; j<20; j++)
402                 {
403                     cbseed = 521 * cbseed + 259;
404                     *cdn_vector++ = tmp_gain * (int16_t)cbseed;
405                 }
406             }
407         break;
408         case I_F_Q:
409             cbseed = -44; // random codebook index
410             for(i=0; i<4; i++)
411             {
412                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
413                 for(j=0; j<40; j++)
414                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
415             }
416         break;
417     }
418 }
419
420 /**
421  * Apply generic gain control.
422  *
423  * @param v_out output vector
424  * @param v_in gain-controlled vector
425  * @param v_ref vector to control gain of
426  *
427  * FIXME: If v_ref is a zero vector, it energy is zero
428  *        and the behavior of the gain control is
429  *        undefined in the specs.
430  *
431  * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
432  */
433 static void apply_gain_ctrl(float *v_out, const float *v_ref,
434                             const float *v_in)
435 {
436     int   i, j, len;
437     float scalefactor;
438
439     for(i=0, j=0; i<4; i++)
440     {
441         scalefactor = ff_dot_productf(v_in + j, v_in + j, 40);
442         if(scalefactor)
443             scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40)
444                         / scalefactor);
445         else
446             ff_log_missing_feature(NULL, "Zero energy for gain control", 1);
447         for(len=j+40; j<len; j++)
448             v_out[j] = scalefactor * v_in[j];
449     }
450 }
451
452 /**
453  * Apply filter in pitch-subframe steps.
454  *
455  * @param memory buffer for the previous state of the filter
456  *        - must be able to contain 303 elements
457  *        - the 143 first elements are from the previous state
458  *        - the next 160 are for output
459  * @param v_in input filter vector
460  * @param gain per-subframe gain array, each element is between 0.0 and 2.0
461  * @param lag per-subframe lag array, each element is
462  *        - between 16 and 143 if its corresponding pfrac is 0,
463  *        - between 16 and 139 otherwise
464  * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
465  *        otherwise
466  *
467  * @return filter output vector
468  */
469 static const float *do_pitchfilter(float memory[303], const float v_in[160],
470                                    const float gain[4], const uint8_t *lag,
471                                    const uint8_t pfrac[4])
472 {
473     int         i, j;
474     float       *v_lag, *v_out;
475     const float *v_len;
476
477     v_out = memory + 143; // Output vector starts at memory[143].
478
479     for(i=0; i<4; i++)
480     {
481         if(gain[i])
482         {
483             v_lag = memory + 143 + 40 * i - lag[i];
484             for(v_len=v_in+40; v_in<v_len; v_in++)
485             {
486                 if(pfrac[i]) // If it is a fractional lag...
487                 {
488                     for(j=0, *v_out=0.; j<4; j++)
489                         *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
490                 }else
491                     *v_out = *v_lag;
492
493                 *v_out = *v_in + gain[i] * *v_out;
494
495                 v_lag++;
496                 v_out++;
497             }
498         }else
499         {
500             memcpy(v_out, v_in, 40 * sizeof(float));
501             v_in  += 40;
502             v_out += 40;
503         }
504     }
505
506     memmove(memory, memory + 160, 143 * sizeof(float));
507     return memory + 143;
508 }
509
510 /**
511  * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
512  * TIA/EIA/IS-733 2.4.5.2
513  *
514  * @param q the context
515  * @param cdn_vector the scaled codebook vector
516  */
517 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
518 {
519     int         i;
520     const float *v_synthesis_filtered, *v_pre_filtered;
521
522     if(q->bitrate >= RATE_HALF ||
523        (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF)))
524     {
525
526         if(q->bitrate >= RATE_HALF)
527         {
528
529             // Compute gain & lag for the whole frame.
530             for(i=0; i<4; i++)
531             {
532                 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
533
534                 q->pitch_lag[i] = q->frame.plag[i] + 16;
535             }
536         }else
537         {
538             float max_pitch_gain = q->erasure_count < 3 ? 0.9 - 0.3 * (q->erasure_count - 1) : 0.0;
539             for(i=0; i<4; i++)
540                 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
541
542             memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
543         }
544
545         // pitch synthesis filter
546         v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
547                                               cdn_vector, q->pitch_gain,
548                                               q->pitch_lag, q->frame.pfrac);
549
550         // pitch prefilter update
551         for(i=0; i<4; i++)
552             q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
553
554         v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
555                                         v_synthesis_filtered,
556                                         q->pitch_gain, q->pitch_lag,
557                                         q->frame.pfrac);
558
559         apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
560     }else
561     {
562         memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17,
563                143 * sizeof(float));
564         memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
565         memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
566         memset(q->pitch_lag,  0, sizeof(q->pitch_lag));
567     }
568 }
569
570 /**
571  * Interpolates LSP frequencies and computes LPC coefficients
572  * for a given bitrate & pitch subframe.
573  *
574  * TIA/EIA/IS-733 2.4.3.3.4
575  *
576  * @param q the context
577  * @param curr_lspf LSP frequencies vector of the current frame
578  * @param lpc float vector for the resulting LPC
579  * @param subframe_num frame number in decoded stream
580  */
581 void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc,
582                      const int subframe_num)
583 {
584     float interpolated_lspf[10];
585     float weight;
586
587     if(q->bitrate >= RATE_QUARTER)
588         weight = 0.25 * (subframe_num + 1);
589     else if(q->bitrate == RATE_OCTAVE && !subframe_num)
590         weight = 0.625;
591     else
592         weight = 1.0;
593
594     if(weight != 1.0)
595     {
596         weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
597                              weight, 1.0 - weight, 10);
598         ff_qcelp_lspf2lpc(interpolated_lspf, lpc);
599     }else if(q->bitrate >= RATE_QUARTER ||
600              (q->bitrate == I_F_Q && !subframe_num))
601         ff_qcelp_lspf2lpc(curr_lspf, lpc);
602 }
603
604 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
605 {
606     switch(buf_size)
607     {
608         case 35: return RATE_FULL;
609         case 17: return RATE_HALF;
610         case  8: return RATE_QUARTER;
611         case  4: return RATE_OCTAVE;
612         case  1: return SILENCE;
613     }
614
615     return I_F_Q;
616 }
617
618 /**
619  * Determine the bitrate from the frame size and/or the first byte of the frame.
620  *
621  * @param avctx the AV codec context
622  * @param buf_size length of the buffer
623  * @param buf the bufffer
624  *
625  * @return the bitrate on success,
626  *         I_F_Q  if the bitrate cannot be satisfactorily determined
627  *
628  * TIA/EIA/IS-733 2.4.8.7.1
629  */
630 static int determine_bitrate(AVCodecContext *avctx, const int buf_size,
631                              const uint8_t **buf)
632 {
633     qcelp_packet_rate bitrate;
634
635     if((bitrate = buf_size2bitrate(buf_size)) >= 0)
636     {
637         if(bitrate > **buf)
638         {
639             av_log(avctx, AV_LOG_WARNING,
640                    "Claimed bitrate and buffer size mismatch.\n");
641             bitrate = **buf;
642         }else if(bitrate < **buf)
643         {
644             av_log(avctx, AV_LOG_ERROR,
645                    "Buffer is too small for the claimed bitrate.\n");
646             return I_F_Q;
647         }
648         (*buf)++;
649     }else if((bitrate = buf_size2bitrate(buf_size + 1)) >= 0)
650     {
651         av_log(avctx, AV_LOG_WARNING,
652                "Bitrate byte is missing, guessing the bitrate from packet size.\n");
653     }else
654         return I_F_Q;
655
656     if(bitrate == SILENCE)
657     {
658         // FIXME: the decoder should not handle SILENCE frames as I_F_Q frames
659         ff_log_missing_feature(avctx, "Blank frame", 1);
660         bitrate = I_F_Q;
661     }
662     return bitrate;
663 }
664
665 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
666                                             const char *message)
667 {
668     av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
669            message);
670 }
671
672 static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
673                               const uint8_t *buf, int buf_size)
674 {
675     QCELPContext *q = avctx->priv_data;
676     float *outbuffer = data;
677     int   i;
678     float quantized_lspf[10], lpc[10];
679     float gain[16];
680     float *formant_mem;
681
682     if((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q)
683     {
684         warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
685         goto erasure;
686     }
687
688     if(q->bitrate == RATE_OCTAVE &&
689        (q->first16bits = AV_RB16(buf)) == 0xFFFF)
690     {
691         warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
692         goto erasure;
693     }
694
695     if(q->bitrate > SILENCE)
696     {
697         const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
698         const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate]
699                                        + qcelp_unpacking_bitmaps_lengths[q->bitrate];
700         uint8_t           *unpacked_data = (uint8_t *)&q->frame;
701
702         init_get_bits(&q->gb, buf, 8*buf_size);
703
704         memset(&q->frame, 0, sizeof(QCELPFrame));
705
706         for(; bitmaps < bitmaps_end; bitmaps++)
707             unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
708
709         // Check for erasures/blanks on rates 1, 1/4 and 1/8.
710         if(q->frame.reserved)
711         {
712             warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
713             goto erasure;
714         }
715         if(q->bitrate == RATE_QUARTER &&
716            codebook_sanity_check_for_rate_quarter(q->frame.cbgain))
717         {
718             warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
719             goto erasure;
720         }
721
722         if(q->bitrate >= RATE_HALF)
723         {
724             for(i=0; i<4; i++)
725             {
726                 if(q->frame.pfrac[i] && q->frame.plag[i] >= 124)
727                 {
728                     warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
729                     goto erasure;
730                 }
731             }
732         }
733     }
734
735     decode_gain_and_index(q, gain);
736     compute_svector(q, gain, outbuffer);
737
738     if(decode_lspf(q, quantized_lspf) < 0)
739     {
740         warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
741         goto erasure;
742     }
743
744
745     apply_pitch_filters(q, outbuffer);
746
747     if(q->bitrate == I_F_Q)
748     {
749 erasure:
750         q->bitrate = I_F_Q;
751         q->erasure_count++;
752         decode_gain_and_index(q, gain);
753         compute_svector(q, gain, outbuffer);
754         decode_lspf(q, quantized_lspf);
755         apply_pitch_filters(q, outbuffer);
756     }else
757         q->erasure_count = 0;
758
759     formant_mem = q->formant_mem + 10;
760     for(i=0; i<4; i++)
761     {
762         interpolate_lpc(q, quantized_lspf, lpc, i);
763         ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40,
764                                      10);
765         formant_mem += 40;
766     }
767     memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
768
769     // FIXME: postfilter and final gain control should be here.
770     // TIA/EIA/IS-733 2.4.8.6
771
772     formant_mem = q->formant_mem + 10;
773     for(i=0; i<160; i++)
774         *outbuffer++ = av_clipf(*formant_mem++, QCELP_CLIP_LOWER_BOUND,
775                                 QCELP_CLIP_UPPER_BOUND);
776
777     memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
778     q->prev_bitrate = q->bitrate;
779
780     *data_size = 160 * sizeof(*outbuffer);
781
782     return *data_size;
783 }
784
785 AVCodec qcelp_decoder =
786 {
787     .name   = "qcelp",
788     .type   = CODEC_TYPE_AUDIO,
789     .id     = CODEC_ID_QCELP,
790     .init   = qcelp_decode_init,
791     .decode = qcelp_decode_frame,
792     .priv_data_size = sizeof(QCELPContext),
793     .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
794 };