Merge remote-tracking branch 'qatar/master'
[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         predictors = (q->prev_bitrate != RATE_OCTAVE &&
122                        q->prev_bitrate != I_F_Q ?
123                        q->prev_lspf : q->predictor_lspf);
124
125         if (q->bitrate == RATE_OCTAVE) {
126             q->octave_count++;
127
128             for (i=0; i<10; i++) {
129                 q->predictor_lspf[i] =
130                              lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
131                                                          : -QCELP_LSP_SPREAD_FACTOR)
132                                      + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR
133                                      + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
134             }
135             smooth = (q->octave_count < 10 ? .875 : 0.1);
136         } else {
137             erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
138
139             assert(q->bitrate == I_F_Q);
140
141             if(q->erasure_count > 1)
142                 erasure_coeff *= (q->erasure_count < 4 ? 0.9 : 0.7);
143
144             for(i = 0; i < 10; i++) {
145                 q->predictor_lspf[i] =
146                              lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
147                                      + erasure_coeff * predictors[i];
148             }
149             smooth = 0.125;
150         }
151
152         // Check the stability of the LSP frequencies.
153         lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
154         for(i=1; i<10; i++)
155             lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR));
156
157         lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR));
158         for(i=9; i>0; i--)
159             lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
160
161         // Low-pass filter the LSP frequencies.
162         ff_weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
163     } else {
164         q->octave_count = 0;
165
166         tmp_lspf = 0.;
167         for (i = 0; i < 5; i++) {
168             lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
169             lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
170         }
171
172         // Check for badly received packets.
173         if (q->bitrate == RATE_QUARTER) {
174             if(lspf[9] <= .70 || lspf[9] >=  .97)
175                 return -1;
176             for(i=3; i<10; i++)
177                 if(fabs(lspf[i] - lspf[i-2]) < .08)
178                     return -1;
179         } else {
180             if(lspf[9] <= .66 || lspf[9] >= .985)
181                 return -1;
182             for(i=4; i<10; i++)
183                 if (fabs(lspf[i] - lspf[i-4]) < .0931)
184                     return -1;
185         }
186     }
187     return 0;
188 }
189
190 /**
191  * Convert codebook transmission codes to GAIN and INDEX.
192  *
193  * @param q the context
194  * @param gain array holding the decoded gain
195  *
196  * TIA/EIA/IS-733 2.4.6.2
197  */
198 static void decode_gain_and_index(QCELPContext  *q,
199                                   float *gain) {
200     int   i, subframes_count, g1[16];
201     float slope;
202
203     if (q->bitrate >= RATE_QUARTER) {
204         switch (q->bitrate) {
205             case RATE_FULL: subframes_count = 16; break;
206             case RATE_HALF: subframes_count = 4;  break;
207             default:        subframes_count = 5;
208         }
209         for(i = 0; i < subframes_count; i++) {
210             g1[i] = 4 * q->frame.cbgain[i];
211             if (q->bitrate == RATE_FULL && !((i+1) & 3)) {
212                 g1[i] += av_clip((g1[i-1] + g1[i-2] + g1[i-3]) / 3 - 6, 0, 32);
213             }
214
215             gain[i] = qcelp_g12ga[g1[i]];
216
217             if (q->frame.cbsign[i]) {
218                 gain[i] = -gain[i];
219                 q->frame.cindex[i] = (q->frame.cindex[i]-89) & 127;
220             }
221         }
222
223         q->prev_g1[0] = g1[i-2];
224         q->prev_g1[1] = g1[i-1];
225         q->last_codebook_gain = qcelp_g12ga[g1[i-1]];
226
227         if (q->bitrate == RATE_QUARTER) {
228             // Provide smoothing of the unvoiced excitation energy.
229             gain[7] =     gain[4];
230             gain[6] = 0.4*gain[3] + 0.6*gain[4];
231             gain[5] =     gain[3];
232             gain[4] = 0.8*gain[2] + 0.2*gain[3];
233             gain[3] = 0.2*gain[1] + 0.8*gain[2];
234             gain[2] =     gain[1];
235             gain[1] = 0.6*gain[0] + 0.4*gain[1];
236         }
237     } else if (q->bitrate != SILENCE) {
238         if (q->bitrate == RATE_OCTAVE) {
239             g1[0] = 2 * q->frame.cbgain[0]
240                   + av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
241             subframes_count = 8;
242         } else {
243             assert(q->bitrate == I_F_Q);
244
245             g1[0] = q->prev_g1[1];
246             switch (q->erasure_count) {
247                 case 1 : break;
248                 case 2 : g1[0] -= 1; break;
249                 case 3 : g1[0] -= 2; break;
250                 default: g1[0] -= 6;
251             }
252             if(g1[0] < 0)
253                 g1[0] = 0;
254             subframes_count = 4;
255         }
256         // This interpolation is done to produce smoother background noise.
257         slope = 0.5*(qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
258         for(i=1; i<=subframes_count; i++)
259             gain[i-1] = q->last_codebook_gain + slope * i;
260
261         q->last_codebook_gain = gain[i-2];
262         q->prev_g1[0] = q->prev_g1[1];
263         q->prev_g1[1] = g1[0];
264     }
265 }
266
267 /**
268  * If the received packet is Rate 1/4 a further sanity check is made of the
269  * codebook gain.
270  *
271  * @param cbgain the unpacked cbgain array
272  * @return -1 if the sanity check fails, 0 otherwise
273  *
274  * TIA/EIA/IS-733 2.4.8.7.3
275  */
276 static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
277 {
278     int i, diff, prev_diff=0;
279
280     for(i=1; i<5; i++) {
281         diff = cbgain[i] - cbgain[i-1];
282         if(FFABS(diff) > 10)
283             return -1;
284         else if(FFABS(diff - prev_diff) > 12)
285             return -1;
286         prev_diff = diff;
287     }
288     return 0;
289 }
290
291 /**
292  * Compute the scaled codebook vector Cdn From INDEX and GAIN
293  * for all rates.
294  *
295  * The specification lacks some information here.
296  *
297  * TIA/EIA/IS-733 has an omission on the codebook index determination
298  * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
299  * you have to subtract the decoded index parameter from the given scaled
300  * codebook vector index 'n' to get the desired circular codebook index, but
301  * it does not mention that you have to clamp 'n' to [0-9] in order to get
302  * RI-compliant results.
303  *
304  * The reason for this mistake seems to be the fact they forgot to mention you
305  * have to do these calculations per codebook subframe and adjust given
306  * equation values accordingly.
307  *
308  * @param q the context
309  * @param gain array holding the 4 pitch subframe gain values
310  * @param cdn_vector array for the generated scaled codebook vector
311  */
312 static void compute_svector(QCELPContext *q, const float *gain,
313                             float *cdn_vector)
314 {
315     int      i, j, k;
316     uint16_t cbseed, cindex;
317     float    *rnd, tmp_gain, fir_filter_value;
318
319     switch (q->bitrate) {
320         case RATE_FULL:
321             for (i = 0; i < 16; i++) {
322                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
323                 cindex = -q->frame.cindex[i];
324                 for(j=0; j<10; j++)
325                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
326             }
327         break;
328         case RATE_HALF:
329             for (i = 0; i < 4; i++) {
330                 tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
331                 cindex = -q->frame.cindex[i];
332                 for (j = 0; j < 40; j++)
333                 *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
334             }
335         break;
336         case RATE_QUARTER:
337             cbseed = (0x0003 & q->frame.lspv[4])<<14 |
338                      (0x003F & q->frame.lspv[3])<< 8 |
339                      (0x0060 & q->frame.lspv[2])<< 1 |
340                      (0x0007 & q->frame.lspv[1])<< 3 |
341                      (0x0038 & q->frame.lspv[0])>> 3 ;
342             rnd = q->rnd_fir_filter_mem + 20;
343             for (i = 0; i < 8; i++) {
344                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
345                 for (k = 0; k < 20; k++) {
346                     cbseed = 521 * cbseed + 259;
347                     *rnd = (int16_t)cbseed;
348
349                     // FIR filter
350                     fir_filter_value = 0.0;
351                     for(j=0; j<10; j++)
352                         fir_filter_value += qcelp_rnd_fir_coefs[j ]
353                                           * (rnd[-j ] + rnd[-20+j]);
354
355                     fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
356                     *cdn_vector++ = tmp_gain * fir_filter_value;
357                     rnd++;
358                 }
359             }
360             memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
361         break;
362         case RATE_OCTAVE:
363             cbseed = q->first16bits;
364             for (i = 0; i < 8; i++) {
365                 tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
366                 for (j = 0; j < 20; j++) {
367                     cbseed = 521 * cbseed + 259;
368                     *cdn_vector++ = tmp_gain * (int16_t)cbseed;
369                 }
370             }
371         break;
372         case I_F_Q:
373             cbseed = -44; // random codebook index
374             for (i = 0; i < 4; i++) {
375                 tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
376                 for(j=0; j<40; j++)
377                     *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
378             }
379         break;
380         case SILENCE:
381             memset(cdn_vector, 0, 160 * sizeof(float));
382         break;
383     }
384 }
385
386 /**
387  * Apply generic gain control.
388  *
389  * @param v_out output vector
390  * @param v_in gain-controlled vector
391  * @param v_ref vector to control gain of
392  *
393  * TIA/EIA/IS-733 2.4.8.3, 2.4.8.6
394  */
395 static void apply_gain_ctrl(float *v_out, const float *v_ref,
396                             const float *v_in)
397 {
398     int i;
399
400     for (i = 0; i < 160; i += 40)
401         ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
402                                                 ff_dot_productf(v_ref + i,
403                                                                 v_ref + i, 40),
404                                                 40);
405 }
406
407 /**
408  * Apply filter in pitch-subframe steps.
409  *
410  * @param memory buffer for the previous state of the filter
411  *        - must be able to contain 303 elements
412  *        - the 143 first elements are from the previous state
413  *        - the next 160 are for output
414  * @param v_in input filter vector
415  * @param gain per-subframe gain array, each element is between 0.0 and 2.0
416  * @param lag per-subframe lag array, each element is
417  *        - between 16 and 143 if its corresponding pfrac is 0,
418  *        - between 16 and 139 otherwise
419  * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
420  *        otherwise
421  *
422  * @return filter output vector
423  */
424 static const float *do_pitchfilter(float memory[303], const float v_in[160],
425                                    const float gain[4], const uint8_t *lag,
426                                    const uint8_t pfrac[4])
427 {
428     int         i, j;
429     float       *v_lag, *v_out;
430     const float *v_len;
431
432     v_out = memory + 143; // Output vector starts at memory[143].
433
434     for (i = 0; i < 4; i++) {
435         if (gain[i]) {
436             v_lag = memory + 143 + 40 * i - lag[i];
437             for (v_len = v_in + 40; v_in < v_len; v_in++) {
438                 if (pfrac[i]) { // If it is a fractional lag...
439                     for(j=0, *v_out=0.; j<4; j++)
440                         *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
441                 }else
442                     *v_out = *v_lag;
443
444                 *v_out = *v_in + gain[i] * *v_out;
445
446                 v_lag++;
447                 v_out++;
448             }
449         } else {
450             memcpy(v_out, v_in, 40 * sizeof(float));
451             v_in  += 40;
452             v_out += 40;
453         }
454     }
455
456     memmove(memory, memory + 160, 143 * sizeof(float));
457     return memory + 143;
458 }
459
460 /**
461  * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
462  * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
463  *
464  * @param q the context
465  * @param cdn_vector the scaled codebook vector
466  */
467 static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
468 {
469     int         i;
470     const float *v_synthesis_filtered, *v_pre_filtered;
471
472     if(q->bitrate >= RATE_HALF ||
473        q->bitrate == SILENCE ||
474       (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF))) {
475
476         if(q->bitrate >= RATE_HALF) {
477
478             // Compute gain & lag for the whole frame.
479             for (i = 0; i < 4; i++) {
480                 q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
481
482                 q->pitch_lag[i] = q->frame.plag[i] + 16;
483             }
484         } else {
485             float max_pitch_gain;
486
487             if (q->bitrate == I_F_Q) {
488                   if (q->erasure_count < 3)
489                       max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
490                   else
491                       max_pitch_gain = 0.0;
492             } else {
493                 assert(q->bitrate == SILENCE);
494                 max_pitch_gain = 1.0;
495             }
496             for(i=0; i<4; i++)
497                 q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
498
499             memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
500         }
501
502         // pitch synthesis filter
503         v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
504                                               cdn_vector, q->pitch_gain,
505                                               q->pitch_lag, q->frame.pfrac);
506
507         // pitch prefilter update
508         for(i=0; i<4; i++)
509             q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
510
511         v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
512                                         v_synthesis_filtered,
513                                         q->pitch_gain, q->pitch_lag,
514                                         q->frame.pfrac);
515
516         apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
517     } else {
518         memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17,
519                143 * sizeof(float));
520         memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
521         memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
522         memset(q->pitch_lag,  0, sizeof(q->pitch_lag));
523     }
524 }
525
526 /**
527  * Reconstruct LPC coefficients from the line spectral pair frequencies
528  * and perform bandwidth expansion.
529  *
530  * @param lspf line spectral pair frequencies
531  * @param lpc linear predictive coding coefficients
532  *
533  * @note: bandwidth_expansion_coeff could be precalculated into a table
534  *        but it seems to be slower on x86
535  *
536  * TIA/EIA/IS-733 2.4.3.3.5
537  */
538 static void lspf2lpc(const float *lspf, float *lpc)
539 {
540     double lsp[10];
541     double bandwidth_expansion_coeff = QCELP_BANDWIDTH_EXPANSION_COEFF;
542     int   i;
543
544     for (i=0; i<10; i++)
545         lsp[i] = cos(M_PI * lspf[i]);
546
547     ff_acelp_lspd2lpc(lsp, lpc, 5);
548
549     for (i = 0; i < 10; i++) {
550         lpc[i] *= bandwidth_expansion_coeff;
551         bandwidth_expansion_coeff *= QCELP_BANDWIDTH_EXPANSION_COEFF;
552     }
553 }
554
555 /**
556  * Interpolate LSP frequencies and compute LPC coefficients
557  * for a given bitrate & pitch subframe.
558  *
559  * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
560  *
561  * @param q the context
562  * @param curr_lspf LSP frequencies vector of the current frame
563  * @param lpc float vector for the resulting LPC
564  * @param subframe_num frame number in decoded stream
565  */
566 static void interpolate_lpc(QCELPContext *q, const float *curr_lspf,
567                             float *lpc, const int subframe_num)
568 {
569     float interpolated_lspf[10];
570     float weight;
571
572     if(q->bitrate >= RATE_QUARTER)
573         weight = 0.25 * (subframe_num + 1);
574     else if(q->bitrate == RATE_OCTAVE && !subframe_num)
575         weight = 0.625;
576     else
577         weight = 1.0;
578
579     if (weight != 1.0) {
580         ff_weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
581                                 weight, 1.0 - weight, 10);
582         lspf2lpc(interpolated_lspf, lpc);
583     }else if(q->bitrate >= RATE_QUARTER ||
584              (q->bitrate == I_F_Q && !subframe_num))
585         lspf2lpc(curr_lspf, lpc);
586     else if(q->bitrate == SILENCE && !subframe_num)
587         lspf2lpc(q->prev_lspf, lpc);
588 }
589
590 static qcelp_packet_rate buf_size2bitrate(const int buf_size)
591 {
592     switch (buf_size) {
593         case 35: return RATE_FULL;
594         case 17: return RATE_HALF;
595         case  8: return RATE_QUARTER;
596         case  4: return RATE_OCTAVE;
597         case  1: return SILENCE;
598     }
599
600     return I_F_Q;
601 }
602
603 /**
604  * Determine the bitrate from the frame size and/or the first byte of the frame.
605  *
606  * @param avctx the AV codec context
607  * @param buf_size length of the buffer
608  * @param buf the bufffer
609  *
610  * @return the bitrate on success,
611  *         I_F_Q  if the bitrate cannot be satisfactorily determined
612  *
613  * TIA/EIA/IS-733 2.4.8.7.1
614  */
615 static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx, const int buf_size,
616                              const uint8_t **buf)
617 {
618     qcelp_packet_rate bitrate;
619
620     if ((bitrate = buf_size2bitrate(buf_size)) >= 0) {
621         if (bitrate > **buf) {
622             QCELPContext *q = avctx->priv_data;
623             if (!q->warned_buf_mismatch_bitrate) {
624             av_log(avctx, AV_LOG_WARNING,
625                    "Claimed bitrate and buffer size mismatch.\n");
626                 q->warned_buf_mismatch_bitrate = 1;
627             }
628             bitrate = **buf;
629         } else if (bitrate < **buf) {
630             av_log(avctx, AV_LOG_ERROR,
631                    "Buffer is too small for the claimed bitrate.\n");
632             return I_F_Q;
633         }
634         (*buf)++;
635     } else if ((bitrate = buf_size2bitrate(buf_size + 1)) >= 0) {
636         av_log(avctx, AV_LOG_WARNING,
637                "Bitrate byte is missing, guessing the bitrate from packet size.\n");
638     }else
639         return I_F_Q;
640
641     if (bitrate == SILENCE) {
642         //FIXME: Remove experimental warning when tested with samples.
643         av_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
644     }
645     return bitrate;
646 }
647
648 static void warn_insufficient_frame_quality(AVCodecContext *avctx,
649                                             const char *message)
650 {
651     av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
652            message);
653 }
654
655 static void postfilter(QCELPContext *q, float *samples, float *lpc)
656 {
657     static const float pow_0_775[10] = {
658         0.775000, 0.600625, 0.465484, 0.360750, 0.279582,
659         0.216676, 0.167924, 0.130141, 0.100859, 0.078166
660     }, pow_0_625[10] = {
661         0.625000, 0.390625, 0.244141, 0.152588, 0.095367,
662         0.059605, 0.037253, 0.023283, 0.014552, 0.009095
663     };
664     float lpc_s[10], lpc_p[10], pole_out[170], zero_out[160];
665     int n;
666
667     for (n = 0; n < 10; n++) {
668         lpc_s[n] = lpc[n] * pow_0_625[n];
669         lpc_p[n] = lpc[n] * pow_0_775[n];
670     }
671
672     ff_celp_lp_zero_synthesis_filterf(zero_out, lpc_s,
673                                       q->formant_mem + 10, 160, 10);
674     memcpy(pole_out, q->postfilter_synth_mem,       sizeof(float) * 10);
675     ff_celp_lp_synthesis_filterf(pole_out + 10, lpc_p, zero_out, 160, 10);
676     memcpy(q->postfilter_synth_mem, pole_out + 160, sizeof(float) * 10);
677
678     ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
679
680     ff_adaptive_gain_control(samples, pole_out + 10,
681         ff_dot_productf(q->formant_mem + 10, q->formant_mem + 10, 160),
682         160, 0.9375, &q->postfilter_agc_mem);
683 }
684
685 static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
686                               AVPacket *avpkt)
687 {
688     const uint8_t *buf = avpkt->data;
689     int buf_size = avpkt->size;
690     QCELPContext *q = avctx->priv_data;
691     float *outbuffer = data;
692     int   i, out_size;
693     float quantized_lspf[10], lpc[10];
694     float gain[16];
695     float *formant_mem;
696
697     out_size = 160 * av_get_bytes_per_sample(avctx->sample_fmt);
698     if (*data_size < out_size) {
699         av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
700         return AVERROR(EINVAL);
701     }
702
703     if ((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q) {
704         warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
705         goto erasure;
706     }
707
708     if(q->bitrate == RATE_OCTAVE &&
709        (q->first16bits = AV_RB16(buf)) == 0xFFFF) {
710         warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
711         goto erasure;
712     }
713
714     if (q->bitrate > SILENCE) {
715         const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
716         const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate]
717                                        + qcelp_unpacking_bitmaps_lengths[q->bitrate];
718         uint8_t           *unpacked_data = (uint8_t *)&q->frame;
719
720         init_get_bits(&q->gb, buf, 8*buf_size);
721
722         memset(&q->frame, 0, sizeof(QCELPFrame));
723
724         for(; bitmaps < bitmaps_end; bitmaps++)
725             unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
726
727         // Check for erasures/blanks on rates 1, 1/4 and 1/8.
728         if (q->frame.reserved) {
729             warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
730             goto erasure;
731         }
732         if(q->bitrate == RATE_QUARTER &&
733            codebook_sanity_check_for_rate_quarter(q->frame.cbgain)) {
734             warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
735             goto erasure;
736         }
737
738         if (q->bitrate >= RATE_HALF) {
739             for (i = 0; i < 4; i++) {
740                 if (q->frame.pfrac[i] && q->frame.plag[i] >= 124) {
741                     warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
742                     goto erasure;
743                 }
744             }
745         }
746     }
747
748     decode_gain_and_index(q, gain);
749     compute_svector(q, gain, outbuffer);
750
751     if (decode_lspf(q, quantized_lspf) < 0) {
752         warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
753         goto erasure;
754     }
755
756
757     apply_pitch_filters(q, outbuffer);
758
759     if (q->bitrate == I_F_Q) {
760 erasure:
761         q->bitrate = I_F_Q;
762         q->erasure_count++;
763         decode_gain_and_index(q, gain);
764         compute_svector(q, gain, outbuffer);
765         decode_lspf(q, quantized_lspf);
766         apply_pitch_filters(q, outbuffer);
767     }else
768         q->erasure_count = 0;
769
770     formant_mem = q->formant_mem + 10;
771     for (i = 0; i < 4; i++) {
772         interpolate_lpc(q, quantized_lspf, lpc, i);
773         ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40,
774                                      10);
775         formant_mem += 40;
776     }
777
778     // postfilter, as per TIA/EIA/IS-733 2.4.8.6
779     postfilter(q, outbuffer, lpc);
780
781     memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
782
783     memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
784     q->prev_bitrate = q->bitrate;
785
786     *data_size = out_size;
787
788     return buf_size;
789 }
790
791 AVCodec ff_qcelp_decoder =
792 {
793     .name   = "qcelp",
794     .type   = AVMEDIA_TYPE_AUDIO,
795     .id     = CODEC_ID_QCELP,
796     .init   = qcelp_decode_init,
797     .decode = qcelp_decode_frame,
798     .priv_data_size = sizeof(QCELPContext),
799     .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
800 };