Rename y400a to gray8a.
[ffmpeg.git] / 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
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 = AV_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  * Decode the 10 quantized LSP frequencies from the LSPV/LSP
105  * transmission codes of any bitrate and check 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  * Convert 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  * Compute 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  * Reconstruct LPC coefficients from the line spectral pair frequencies
568  * and perform 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  * Interpolate LSP frequencies and compute 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 ff_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 };