Rename bitstream.h to get_bits.h.
[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 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
42 #undef NDEBUG
43 #include <assert.h>
44
45 typedef enum
46 {
47     I_F_Q = -1,    /*!< insufficient frame quality */
48     SILENCE,
49     RATE_OCTAVE,
50     RATE_QUARTER,
51     RATE_HALF,
52     RATE_FULL
53 } qcelp_packet_rate;
54
55 typedef struct
56 {
57     GetBitContext     gb;
58     qcelp_packet_rate bitrate;
59     QCELPFrame        frame;    /*!< unpacked data frame */
60
61     uint8_t  erasure_count;
62     uint8_t  octave_count;      /*!< count the consecutive RATE_OCTAVE frames */
63     float    prev_lspf[10];
64     float    predictor_lspf[10];/*!< LSP predictor for RATE_OCTAVE and I_F_Q */
65     float    pitch_synthesis_filter_mem[303];
66     float    pitch_pre_filter_mem[303];
67     float    rnd_fir_filter_mem[180];
68     float    formant_mem[170];
69     float    last_codebook_gain;
70     int      prev_g1[2];
71     int      prev_bitrate;
72     float    pitch_gain[4];
73     uint8_t  pitch_lag[4];
74     uint16_t first16bits;
75     uint8_t  warned_buf_mismatch_bitrate;
76 } QCELPContext;
77
78 /**
79  * Reconstructs LPC coefficients from the line spectral pair frequencies.
80  *
81  * TIA/EIA/IS-733 2.4.3.3.5
82  */
83 void ff_celp_lspf2lpc(const double *lspf, float *lpc);
84
85 /**
86  * Initialize the speech codec according to the specification.
87  *
88  * TIA/EIA/IS-733 2.4.9
89  */
90 static av_cold int qcelp_decode_init(AVCodecContext *avctx)
91 {
92     QCELPContext *q = avctx->priv_data;
93     int i;
94
95     avctx->sample_fmt = SAMPLE_FMT_FLT;
96
97     for(i=0; i<10; i++)
98         q->prev_lspf[i] = (i+1)/11.;
99
100     return 0;
101 }
102
103 /**
104  * Decodes the 10 quantized LSP frequencies from the LSPV/LSP
105  * transmission codes of any bitrate and checks for badly received packets.
106  *
107  * @param q the context
108  * @param lspf line spectral pair frequencies
109  *
110  * @return 0 on success, -1 if the packet is badly received
111  *
112  * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
113  */
114 static int decode_lspf(QCELPContext *q, float *lspf)
115 {
116     int i;
117     float tmp_lspf, smooth, erasure_coeff;
118     const float *predictors;
119
120     if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q)
121     {
122         predictors = (q->prev_bitrate != RATE_OCTAVE &&
123                        q->prev_bitrate != I_F_Q ?
124                        q->prev_lspf : q->predictor_lspf);
125
126         if(q->bitrate == RATE_OCTAVE)
127         {
128             q->octave_count++;
129
130             for(i=0; i<10; i++)
131             {
132                 q->predictor_lspf[i] =
133                              lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
134                                                          : -QCELP_LSP_SPREAD_FACTOR)
135                                      + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR
136                                      + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
137             }
138             smooth = (q->octave_count < 10 ? .875 : 0.1);
139         }else
140         {
141             erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
142
143             assert(q->bitrate == I_F_Q);
144
145             if(q->erasure_count > 1)
146                 erasure_coeff *= (q->erasure_count < 4 ? 0.9 : 0.7);
147
148             for(i=0; i<10; i++)
149             {
150                 q->predictor_lspf[i] =
151                              lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
152                                      + erasure_coeff * predictors[i];
153             }
154             smooth = 0.125;
155         }
156
157         // Check the stability of the LSP frequencies.
158         lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
159         for(i=1; i<10; i++)
160             lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR));
161
162         lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR));
163         for(i=9; i>0; i--)
164             lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
165
166         // Low-pass filter the LSP frequencies.
167         ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
168     }else
169     {
170         q->octave_count = 0;
171
172         tmp_lspf = 0.;
173         for(i=0; i<5 ; i++)
174         {
175             lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
176             lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
177         }
178
179         // Check for badly received packets.
180         if(q->bitrate == RATE_QUARTER)
181         {
182             if(lspf[9] <= .70 || lspf[9] >=  .97)
183                 return -1;
184             for(i=3; i<10; i++)
185                 if(fabs(lspf[i] - lspf[i-2]) < .08)
186                     return -1;
187         }else
188         {
189             if(lspf[9] <= .66 || lspf[9] >= .985)
190                 return -1;
191             for(i=4; i<10; i++)
192                 if (fabs(lspf[i] - lspf[i-4]) < .0931)
193                     return -1;
194         }
195     }
196     return 0;
197 }
198
199 /**
200  * Converts codebook transmission codes to GAIN and INDEX.
201  *
202  * @param q the context
203  * @param gain array holding the decoded gain
204  *
205  * TIA/EIA/IS-733 2.4.6.2
206  */
207 static void decode_gain_and_index(QCELPContext  *q,
208                                   float *gain) {
209     int   i, subframes_count, g1[16];
210     float slope;
211
212     if(q->bitrate >= RATE_QUARTER)
213     {
214         switch(q->bitrate)
215         {
216             case RATE_FULL: subframes_count = 16; break;
217             case RATE_HALF: subframes_count = 4;  break;
218             default:        subframes_count = 5;
219         }
220         for(i=0; i<subframes_count; i++)
221         {
222             g1[i] = 4 * q->frame.cbgain[i];
223             if(q->bitrate == RATE_FULL && !((i+1) & 3))
224             {
225                 g1[i] += av_clip((g1[i-1] + g1[i-2] + g1[i-3]) / 3 - 6, 0, 32);
226             }
227
228             gain[i] = qcelp_g12ga[g1[i]];
229
230             if(q->frame.cbsign[i])
231             {
232                 gain[i] = -gain[i];
233                 q->frame.cindex[i] = (q->frame.cindex[i]-89) & 127;
234             }
235         }
236
237         q->prev_g1[0] = g1[i-2];
238         q->prev_g1[1] = g1[i-1];
239         q->last_codebook_gain = qcelp_g12ga[g1[i-1]];
240
241         if(q->bitrate == RATE_QUARTER)
242         {
243             // Provide smoothing of the unvoiced excitation energy.
244             gain[7] =     gain[4];
245             gain[6] = 0.4*gain[3] + 0.6*gain[4];
246             gain[5] =     gain[3];
247             gain[4] = 0.8*gain[2] + 0.2*gain[3];
248             gain[3] = 0.2*gain[1] + 0.8*gain[2];
249             gain[2] =     gain[1];
250             gain[1] = 0.6*gain[0] + 0.4*gain[1];
251         }
252     }else
253     {
254         if(q->bitrate == RATE_OCTAVE)
255         {
256             g1[0] = 2 * q->frame.cbgain[0]
257                   + av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
258             subframes_count = 8;
259         }else
260         {
261             assert(q->bitrate == I_F_Q);
262
263             g1[0] = q->prev_g1[1];
264             switch(q->erasure_count)
265             {
266                 case 1 : break;
267                 case 2 : g1[0] -= 1; break;
268                 case 3 : g1[0] -= 2; break;
269                 default: g1[0] -= 6;
270             }
271             if(g1[0] < 0)
272                 g1[0] = 0;
273             subframes_count = 4;
274         }
275         // This interpolation is done to produce smoother background noise.
276         slope = 0.5*(qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
277         for(i=1; i<=subframes_count; i++)
278             gain[i-1] = q->last_codebook_gain + slope * i;
279
280         q->last_codebook_gain = gain[i-2];
281         q->prev_g1[0] = q->prev_g1[1];
282         q->prev_g1[1] = g1[0];
283     }
284 }
285
286 /**
287  * If the received packet is Rate 1/4 a further sanity check is made of the
288  * codebook gain.
289  *
290  * @param cbgain the unpacked cbgain array
291  * @return -1 if the sanity check fails, 0 otherwise
292  *
293  * TIA/EIA/IS-733 2.4.8.7.3
294  */
295 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
296 {
297     int i, diff, prev_diff=0;
298
299     for(i=1; i<5; i++)
300     {
301         diff = cbgain[i] - cbgain[i-1];
302         if(FFABS(diff) > 10)
303             return -1;
304         else if(FFABS(diff - prev_diff) > 12)
305             return -1;
306         prev_diff = diff;
307     }
308     return 0;
309 }
310
311 /**
312  * Computes the scaled codebook vector Cdn From INDEX and GAIN
313  * for all rates.
314  *
315  * The specification lacks some information here.
316  *
317  * TIA/EIA/IS-733 has an omission on the codebook index determination
318  * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
319  * you have to subtract the decoded index parameter from the given scaled
320  * codebook vector index 'n' to get the desired circular codebook index, but
321  * it does not mention that you have to clamp 'n' to [0-9] in order to get
322  * RI-compliant results.
323  *
324  * The reason for this mistake seems to be the fact they forgot to mention you
325  * have to do these calculations per codebook subframe and adjust given
326  * equation values accordingly.
327  *
328  * @param q the context
329  * @param gain array holding the 4 pitch subframe gain values
330  * @param cdn_vector array for the generated scaled codebook vector
331  */
332 static void compute_svector(QCELPContext *q, const float *gain,
333                             float *cdn_vector)
334 {
335     int      i, j, k;
336     uint16_t cbseed, cindex;
337     float    *rnd, tmp_gain, fir_filter_value;
338
339     switch(q->bitrate)
340     {
341         case RATE_FULL:
342             for(i=0; i<16; i++)
343             {
344                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
345                 cindex = -q->frame.cindex[i];
346                 for(j=0; j<10; j++)
347                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
348             }
349         break;
350         case RATE_HALF:
351             for(i=0; i<4; i++)
352             {
353                 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
354                 cindex = -q->frame.cindex[i];
355                 for (j = 0; j < 40; j++)
356                 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
357             }
358         break;
359         case RATE_QUARTER:
360             cbseed = (0x0003 & q->frame.lspv[4])<<14 |
361                      (0x003F & q->frame.lspv[3])<< 8 |
362                      (0x0060 & q->frame.lspv[2])<< 1 |
363                      (0x0007 & q->frame.lspv[1])<< 3 |
364                      (0x0038 & q->frame.lspv[0])>> 3 ;
365             rnd = q->rnd_fir_filter_mem + 20;
366             for(i=0; i<8; i++)
367             {
368                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
369                 for(k=0; k<20; k++)
370                 {
371                     cbseed = 521 * cbseed + 259;
372                     *rnd = (int16_t)cbseed;
373
374                     // FIR filter
375                     fir_filter_value = 0.0;
376                     for(j=0; j<10; j++)
377                         fir_filter_value += qcelp_rnd_fir_coefs[j ]
378                                           * (rnd[-j ] + rnd[-20+j]);
379
380                     fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
381                     *cdn_vector++ = tmp_gain * fir_filter_value;
382                     rnd++;
383                 }
384             }
385             memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
386         break;
387         case RATE_OCTAVE:
388             cbseed = q->first16bits;
389             for(i=0; i<8; i++)
390             {
391                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
392                 for(j=0; j<20; j++)
393                 {
394                     cbseed = 521 * cbseed + 259;
395                     *cdn_vector++ = tmp_gain * (int16_t)cbseed;
396                 }
397             }
398         break;
399         case I_F_Q:
400             cbseed = -44; // random codebook index
401             for(i=0; i<4; i++)
402             {
403                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
404                 for(j=0; j<40; j++)
405                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
406             }
407         break;
408         case SILENCE:
409             memset(cdn_vector, 0, 160 * sizeof(float));
410         break;
411     }
412 }
413
414 /**
415  * Compute the gain control
416  *
417  * @param v_in gain-controlled vector
418  * @param v_ref vector to control gain of
419  *
420  * @return gain control
421  *
422  * FIXME: If v_ref is a zero vector, it energy is zero
423  *        and the behavior of the gain control is
424  *        undefined in the specs.
425  *
426  * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
427  */
428 static float compute_gain_ctrl(const float *v_ref, const float *v_in, const int len)
429 {
430     float scalefactor = ff_dot_productf(v_in, v_in, len);
431
432     if(scalefactor)
433         scalefactor = sqrt(ff_dot_productf(v_ref, v_ref, len) / scalefactor);
434     else
435         ff_log_missing_feature(NULL, "Zero energy for gain control", 1);
436     return scalefactor;
437 }
438
439 /**
440  * Apply generic gain control.
441  *
442  * @param v_out output vector
443  * @param v_in gain-controlled vector
444  * @param v_ref vector to control gain of
445  *
446  * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
447  */
448 static void apply_gain_ctrl(float *v_out, const float *v_ref,
449                             const float *v_in)
450 {
451     int   i, j, len;
452     float scalefactor;
453
454     for(i=0, j=0; i<4; i++)
455     {
456         scalefactor = compute_gain_ctrl(v_ref + j, v_in + j, 40);
457         for(len=j+40; j<len; j++)
458             v_out[j] = scalefactor * v_in[j];
459     }
460 }
461
462 /**
463  * Apply filter in pitch-subframe steps.
464  *
465  * @param memory buffer for the previous state of the filter
466  *        - must be able to contain 303 elements
467  *        - the 143 first elements are from the previous state
468  *        - the next 160 are for output
469  * @param v_in input filter vector
470  * @param gain per-subframe gain array, each element is between 0.0 and 2.0
471  * @param lag per-subframe lag array, each element is
472  *        - between 16 and 143 if its corresponding pfrac is 0,
473  *        - between 16 and 139 otherwise
474  * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
475  *        otherwise
476  *
477  * @return filter output vector
478  */
479 static const float *do_pitchfilter(float memory[303], const float v_in[160],
480                                    const float gain[4], const uint8_t *lag,
481                                    const uint8_t pfrac[4])
482 {
483     int         i, j;
484     float       *v_lag, *v_out;
485     const float *v_len;
486
487     v_out = memory + 143; // Output vector starts at memory[143].
488
489     for(i=0; i<4; i++)
490     {
491         if(gain[i])
492         {
493             v_lag = memory + 143 + 40 * i - lag[i];
494             for(v_len=v_in+40; v_in<v_len; v_in++)
495             {
496                 if(pfrac[i]) // If it is a fractional lag...
497                 {
498                     for(j=0, *v_out=0.; j<4; j++)
499                         *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
500                 }else
501                     *v_out = *v_lag;
502
503                 *v_out = *v_in + gain[i] * *v_out;
504
505                 v_lag++;
506                 v_out++;
507             }
508         }else
509         {
510             memcpy(v_out, v_in, 40 * sizeof(float));
511             v_in  += 40;
512             v_out += 40;
513         }
514     }
515
516     memmove(memory, memory + 160, 143 * sizeof(float));
517     return memory + 143;
518 }
519
520 /**
521  * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
522  * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
523  *
524  * @param q the context
525  * @param cdn_vector the scaled codebook vector
526  */
527 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
528 {
529     int         i;
530     const float *v_synthesis_filtered, *v_pre_filtered;
531
532     if(q->bitrate >= RATE_HALF ||
533        q->bitrate == SILENCE ||
534        (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF)))
535     {
536
537         if(q->bitrate >= RATE_HALF)
538         {
539
540             // Compute gain & lag for the whole frame.
541             for(i=0; i<4; i++)
542             {
543                 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
544
545                 q->pitch_lag[i] = q->frame.plag[i] + 16;
546             }
547         }else
548         {
549             float max_pitch_gain;
550
551             if (q->bitrate == I_F_Q)
552             {
553                   if (q->erasure_count < 3)
554                       max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
555                   else
556                       max_pitch_gain = 0.0;
557             }else
558             {
559                 assert(q->bitrate == SILENCE);
560                 max_pitch_gain = 1.0;
561             }
562             for(i=0; i<4; i++)
563                 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
564
565             memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
566         }
567
568         // pitch synthesis filter
569         v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
570                                               cdn_vector, q->pitch_gain,
571                                               q->pitch_lag, q->frame.pfrac);
572
573         // pitch prefilter update
574         for(i=0; i<4; i++)
575             q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
576
577         v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
578                                         v_synthesis_filtered,
579                                         q->pitch_gain, q->pitch_lag,
580                                         q->frame.pfrac);
581
582         apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
583     }else
584     {
585         memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17,
586                143 * sizeof(float));
587         memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
588         memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
589         memset(q->pitch_lag,  0, sizeof(q->pitch_lag));
590     }
591 }
592
593 /**
594  * Reconstructs LPC coefficients from the line spectral pair frequencies
595  * and performs bandwidth expansion.
596  *
597  * @param lspf line spectral pair frequencies
598  * @param lpc linear predictive coding coefficients
599  *
600  * @note: bandwith_expansion_coeff could be precalculated into a table
601  *        but it seems to be slower on x86
602  *
603  * TIA/EIA/IS-733 2.4.3.3.5
604  */
605 static void lspf2lpc(const float *lspf, float *lpc)
606 {
607     double lsf[10];
608     double bandwith_expansion_coeff = QCELP_BANDWITH_EXPANSION_COEFF;
609     int   i;
610
611     for (i=0; i<10; i++)
612         lsf[i] = cos(M_PI * lspf[i]);
613
614     ff_celp_lspf2lpc(lsf, lpc);
615
616     for (i=0; i<10; i++)
617     {
618         lpc[i] *= bandwith_expansion_coeff;
619         bandwith_expansion_coeff *= QCELP_BANDWITH_EXPANSION_COEFF;
620     }
621 }
622
623 /**
624  * Interpolates LSP frequencies and computes LPC coefficients
625  * for a given bitrate & pitch subframe.
626  *
627  * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
628  *
629  * @param q the context
630  * @param curr_lspf LSP frequencies vector of the current frame
631  * @param lpc float vector for the resulting LPC
632  * @param subframe_num frame number in decoded stream
633  */
634 void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc,
635                      const int subframe_num)
636 {
637     float interpolated_lspf[10];
638     float weight;
639
640     if(q->bitrate >= RATE_QUARTER)
641         weight = 0.25 * (subframe_num + 1);
642     else if(q->bitrate == RATE_OCTAVE && !subframe_num)
643         weight = 0.625;
644     else
645         weight = 1.0;
646
647     if(weight != 1.0)
648     {
649         ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
650                                 weight, 1.0 - weight, 10);
651         lspf2lpc(interpolated_lspf, lpc);
652     }else if(q->bitrate >= RATE_QUARTER ||
653              (q->bitrate == I_F_Q && !subframe_num))
654         lspf2lpc(curr_lspf, lpc);
655     else if(q->bitrate == SILENCE && !subframe_num)
656         lspf2lpc(q->prev_lspf, lpc);
657 }
658
659 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
660 {
661     switch(buf_size)
662     {
663         case 35: return RATE_FULL;
664         case 17: return RATE_HALF;
665         case  8: return RATE_QUARTER;
666         case  4: return RATE_OCTAVE;
667         case  1: return SILENCE;
668     }
669
670     return I_F_Q;
671 }
672
673 /**
674  * Determine the bitrate from the frame size and/or the first byte of the frame.
675  *
676  * @param avctx the AV codec context
677  * @param buf_size length of the buffer
678  * @param buf the bufffer
679  *
680  * @return the bitrate on success,
681  *         I_F_Q  if the bitrate cannot be satisfactorily determined
682  *
683  * TIA/EIA/IS-733 2.4.8.7.1
684  */
685 static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx, const int buf_size,
686                              const uint8_t **buf)
687 {
688     qcelp_packet_rate bitrate;
689
690     if((bitrate = buf_size2bitrate(buf_size)) >= 0)
691     {
692         if(bitrate > **buf)
693         {
694             QCELPContext *q = avctx->priv_data;
695             if (!q->warned_buf_mismatch_bitrate)
696             {
697             av_log(avctx, AV_LOG_WARNING,
698                    "Claimed bitrate and buffer size mismatch.\n");
699                 q->warned_buf_mismatch_bitrate = 1;
700             }
701             bitrate = **buf;
702         }else if(bitrate < **buf)
703         {
704             av_log(avctx, AV_LOG_ERROR,
705                    "Buffer is too small for the claimed bitrate.\n");
706             return I_F_Q;
707         }
708         (*buf)++;
709     }else if((bitrate = buf_size2bitrate(buf_size + 1)) >= 0)
710     {
711         av_log(avctx, AV_LOG_WARNING,
712                "Bitrate byte is missing, guessing the bitrate from packet size.\n");
713     }else
714         return I_F_Q;
715
716     if(bitrate == SILENCE)
717     {
718         //FIXME: Remove experimental warning when tested with samples.
719         ff_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
720     }
721     return bitrate;
722 }
723
724 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
725                                             const char *message)
726 {
727     av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
728            message);
729 }
730
731 static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
732                               AVPacket *avpkt)
733 {
734     const uint8_t *buf = avpkt->data;
735     int buf_size = avpkt->size;
736     QCELPContext *q = avctx->priv_data;
737     float *outbuffer = data;
738     int   i;
739     float quantized_lspf[10], lpc[10];
740     float gain[16];
741     float *formant_mem;
742
743     if((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q)
744     {
745         warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
746         goto erasure;
747     }
748
749     if(q->bitrate == RATE_OCTAVE &&
750        (q->first16bits = AV_RB16(buf)) == 0xFFFF)
751     {
752         warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
753         goto erasure;
754     }
755
756     if(q->bitrate > SILENCE)
757     {
758         const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
759         const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate]
760                                        + qcelp_unpacking_bitmaps_lengths[q->bitrate];
761         uint8_t           *unpacked_data = (uint8_t *)&q->frame;
762
763         init_get_bits(&q->gb, buf, 8*buf_size);
764
765         memset(&q->frame, 0, sizeof(QCELPFrame));
766
767         for(; bitmaps < bitmaps_end; bitmaps++)
768             unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
769
770         // Check for erasures/blanks on rates 1, 1/4 and 1/8.
771         if(q->frame.reserved)
772         {
773             warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
774             goto erasure;
775         }
776         if(q->bitrate == RATE_QUARTER &&
777            codebook_sanity_check_for_rate_quarter(q->frame.cbgain))
778         {
779             warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
780             goto erasure;
781         }
782
783         if(q->bitrate >= RATE_HALF)
784         {
785             for(i=0; i<4; i++)
786             {
787                 if(q->frame.pfrac[i] && q->frame.plag[i] >= 124)
788                 {
789                     warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
790                     goto erasure;
791                 }
792             }
793         }
794     }
795
796     decode_gain_and_index(q, gain);
797     compute_svector(q, gain, outbuffer);
798
799     if(decode_lspf(q, quantized_lspf) < 0)
800     {
801         warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
802         goto erasure;
803     }
804
805
806     apply_pitch_filters(q, outbuffer);
807
808     if(q->bitrate == I_F_Q)
809     {
810 erasure:
811         q->bitrate = I_F_Q;
812         q->erasure_count++;
813         decode_gain_and_index(q, gain);
814         compute_svector(q, gain, outbuffer);
815         decode_lspf(q, quantized_lspf);
816         apply_pitch_filters(q, outbuffer);
817     }else
818         q->erasure_count = 0;
819
820     formant_mem = q->formant_mem + 10;
821     for(i=0; i<4; i++)
822     {
823         interpolate_lpc(q, quantized_lspf, lpc, i);
824         ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40,
825                                      10);
826         formant_mem += 40;
827     }
828     memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
829
830     // FIXME: postfilter and final gain control should be here.
831     // TIA/EIA/IS-733 2.4.8.6
832
833     formant_mem = q->formant_mem + 10;
834     for(i=0; i<160; i++)
835         *outbuffer++ = av_clipf(*formant_mem++, QCELP_CLIP_LOWER_BOUND,
836                                 QCELP_CLIP_UPPER_BOUND);
837
838     memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
839     q->prev_bitrate = q->bitrate;
840
841     *data_size = 160 * sizeof(*outbuffer);
842
843     return *data_size;
844 }
845
846 AVCodec qcelp_decoder =
847 {
848     .name   = "qcelp",
849     .type   = CODEC_TYPE_AUDIO,
850     .id     = CODEC_ID_QCELP,
851     .init   = qcelp_decode_init,
852     .decode = qcelp_decode_frame,
853     .priv_data_size = sizeof(QCELPContext),
854     .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
855 };