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 "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
44typedef 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
54typedef 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    uint8_t  warned_buf_mismatch_bitrate;
75} QCELPContext;
76
77/**
78 * Reconstructs LPC coefficients from the line spectral pair frequencies.
79 *
80 * TIA/EIA/IS-733 2.4.3.3.5
81 */
82void ff_qcelp_lspf2lpc(const float *lspf, float *lpc);
83
84static void weighted_vector_sumf(float *out, const float *in_a,
85                                 const float *in_b, float weight_coeff_a,
86                                 float weight_coeff_b, int length)
87{
88    int i;
89
90    for(i=0; i<length; i++)
91        out[i] = weight_coeff_a * in_a[i]
92               + weight_coeff_b * in_b[i];
93}
94
95/**
96 * Initialize the speech codec according to the specification.
97 *
98 * TIA/EIA/IS-733 2.4.9
99 */
100static av_cold int qcelp_decode_init(AVCodecContext *avctx)
101{
102    QCELPContext *q = avctx->priv_data;
103    int i;
104
105    avctx->sample_fmt = SAMPLE_FMT_FLT;
106
107    for(i=0; i<10; i++)
108        q->prev_lspf[i] = (i+1)/11.;
109
110    return 0;
111}
112
113/**
114 * Decodes the 10 quantized LSP frequencies from the LSPV/LSP
115 * transmission codes of any bitrate and checks for badly received packets.
116 *
117 * @param q the context
118 * @param lspf line spectral pair frequencies
119 *
120 * @return 0 on success, -1 if the packet is badly received
121 *
122 * TIA/EIA/IS-733 2.4.3.2.6.2-2, 2.4.8.7.3
123 */
124static int decode_lspf(QCELPContext *q, float *lspf)
125{
126    int i;
127    float tmp_lspf, smooth, erasure_coeff;
128    const float *predictors;
129
130    if(q->bitrate == RATE_OCTAVE || q->bitrate == I_F_Q)
131    {
132        predictors = (q->prev_bitrate != RATE_OCTAVE &&
133                       q->prev_bitrate != I_F_Q ?
134                       q->prev_lspf : q->predictor_lspf);
135
136        if(q->bitrate == RATE_OCTAVE)
137        {
138            q->octave_count++;
139
140            for(i=0; i<10; i++)
141            {
142                q->predictor_lspf[i] =
143                             lspf[i] = (q->frame.lspv[i] ?  QCELP_LSP_SPREAD_FACTOR
144                                                         : -QCELP_LSP_SPREAD_FACTOR)
145                                     + predictors[i] * QCELP_LSP_OCTAVE_PREDICTOR
146                                     + (i + 1) * ((1 - QCELP_LSP_OCTAVE_PREDICTOR)/11);
147            }
148            smooth = (q->octave_count < 10 ? .875 : 0.1);
149        }else
150        {
151            erasure_coeff = QCELP_LSP_OCTAVE_PREDICTOR;
152
153            assert(q->bitrate == I_F_Q);
154
155            if(q->erasure_count > 1)
156                erasure_coeff *= (q->erasure_count < 4 ? 0.9 : 0.7);
157
158            for(i=0; i<10; i++)
159            {
160                q->predictor_lspf[i] =
161                             lspf[i] = (i + 1) * ( 1 - erasure_coeff)/11
162                                     + erasure_coeff * predictors[i];
163            }
164            smooth = 0.125;
165        }
166
167        // Check the stability of the LSP frequencies.
168        lspf[0] = FFMAX(lspf[0], QCELP_LSP_SPREAD_FACTOR);
169        for(i=1; i<10; i++)
170            lspf[i] = FFMAX(lspf[i], (lspf[i-1] + QCELP_LSP_SPREAD_FACTOR));
171
172        lspf[9] = FFMIN(lspf[9], (1.0 - QCELP_LSP_SPREAD_FACTOR));
173        for(i=9; i>0; i--)
174            lspf[i-1] = FFMIN(lspf[i-1], (lspf[i] - QCELP_LSP_SPREAD_FACTOR));
175
176        // Low-pass filter the LSP frequencies.
177        weighted_vector_sumf(lspf, lspf, q->prev_lspf, smooth, 1.0-smooth, 10);
178    }else
179    {
180        q->octave_count = 0;
181
182        tmp_lspf = 0.;
183        for(i=0; i<5 ; i++)
184        {
185            lspf[2*i+0] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][0] * 0.0001;
186            lspf[2*i+1] = tmp_lspf += qcelp_lspvq[i][q->frame.lspv[i]][1] * 0.0001;
187        }
188
189        // Check for badly received packets.
190        if(q->bitrate == RATE_QUARTER)
191        {
192            if(lspf[9] <= .70 || lspf[9] >=  .97)
193                return -1;
194            for(i=3; i<10; i++)
195                if(fabs(lspf[i] - lspf[i-2]) < .08)
196                    return -1;
197        }else
198        {
199            if(lspf[9] <= .66 || lspf[9] >= .985)
200                return -1;
201            for(i=4; i<10; i++)
202                if (fabs(lspf[i] - lspf[i-4]) < .0931)
203                    return -1;
204        }
205    }
206    return 0;
207}
208
209/**
210 * Converts codebook transmission codes to GAIN and INDEX.
211 *
212 * @param q the context
213 * @param gain array holding the decoded gain
214 *
215 * TIA/EIA/IS-733 2.4.6.2
216 */
217static void decode_gain_and_index(QCELPContext  *q,
218                                  float *gain) {
219    int   i, subframes_count, g1[16];
220    float slope;
221
222    if(q->bitrate >= RATE_QUARTER)
223    {
224        switch(q->bitrate)
225        {
226            case RATE_FULL: subframes_count = 16; break;
227            case RATE_HALF: subframes_count = 4;  break;
228            default:        subframes_count = 5;
229        }
230        for(i=0; i<subframes_count; i++)
231        {
232            g1[i] = 4 * q->frame.cbgain[i];
233            if(q->bitrate == RATE_FULL && !((i+1) & 3))
234            {
235                g1[i] += av_clip((g1[i-1] + g1[i-2] + g1[i-3]) / 3 - 6, 0, 32);
236            }
237
238            gain[i] = qcelp_g12ga[g1[i]];
239
240            if(q->frame.cbsign[i])
241            {
242                gain[i] = -gain[i];
243                q->frame.cindex[i] = (q->frame.cindex[i]-89) & 127;
244            }
245        }
246
247        q->prev_g1[0] = g1[i-2];
248        q->prev_g1[1] = g1[i-1];
249        q->last_codebook_gain = qcelp_g12ga[g1[i-1]];
250
251        if(q->bitrate == RATE_QUARTER)
252        {
253            // Provide smoothing of the unvoiced excitation energy.
254            gain[7] =     gain[4];
255            gain[6] = 0.4*gain[3] + 0.6*gain[4];
256            gain[5] =     gain[3];
257            gain[4] = 0.8*gain[2] + 0.2*gain[3];
258            gain[3] = 0.2*gain[1] + 0.8*gain[2];
259            gain[2] =     gain[1];
260            gain[1] = 0.6*gain[0] + 0.4*gain[1];
261        }
262    }else
263    {
264        if(q->bitrate == RATE_OCTAVE)
265        {
266            g1[0] = 2 * q->frame.cbgain[0]
267                  + av_clip((q->prev_g1[0] + q->prev_g1[1]) / 2 - 5, 0, 54);
268            subframes_count = 8;
269        }else
270        {
271            assert(q->bitrate == I_F_Q);
272
273            g1[0] = q->prev_g1[1];
274            switch(q->erasure_count)
275            {
276                case 1 : break;
277                case 2 : g1[0] -= 1; break;
278                case 3 : g1[0] -= 2; break;
279                default: g1[0] -= 6;
280            }
281            if(g1[0] < 0)
282                g1[0] = 0;
283            subframes_count = 4;
284        }
285        // This interpolation is done to produce smoother background noise.
286        slope = 0.5*(qcelp_g12ga[g1[0]] - q->last_codebook_gain) / subframes_count;
287        for(i=1; i<=subframes_count; i++)
288            gain[i-1] = q->last_codebook_gain + slope * i;
289
290        q->last_codebook_gain = gain[i-2];
291        q->prev_g1[0] = q->prev_g1[1];
292        q->prev_g1[1] = g1[0];
293    }
294}
295
296/**
297 * If the received packet is Rate 1/4 a further sanity check is made of the
298 * codebook gain.
299 *
300 * @param cbgain the unpacked cbgain array
301 * @return -1 if the sanity check fails, 0 otherwise
302 *
303 * TIA/EIA/IS-733 2.4.8.7.3
304 */
305static int codebook_sanity_check_for_rate_quarter(const uint8_t *cbgain)
306{
307    int i, diff, prev_diff=0;
308
309    for(i=1; i<5; i++)
310    {
311        diff = cbgain[i] - cbgain[i-1];
312        if(FFABS(diff) > 10)
313            return -1;
314        else if(FFABS(diff - prev_diff) > 12)
315            return -1;
316        prev_diff = diff;
317    }
318    return 0;
319}
320
321/**
322 * Computes the scaled codebook vector Cdn From INDEX and GAIN
323 * for all rates.
324 *
325 * The specification lacks some information here.
326 *
327 * TIA/EIA/IS-733 has an omission on the codebook index determination
328 * formula for RATE_FULL and RATE_HALF frames at section 2.4.8.1.1. It says
329 * you have to subtract the decoded index parameter from the given scaled
330 * codebook vector index 'n' to get the desired circular codebook index, but
331 * it does not mention that you have to clamp 'n' to [0-9] in order to get
332 * RI-compliant results.
333 *
334 * The reason for this mistake seems to be the fact they forgot to mention you
335 * have to do these calculations per codebook subframe and adjust given
336 * equation values accordingly.
337 *
338 * @param q the context
339 * @param gain array holding the 4 pitch subframe gain values
340 * @param cdn_vector array for the generated scaled codebook vector
341 */
342static void compute_svector(QCELPContext *q, const float *gain,
343                            float *cdn_vector)
344{
345    int      i, j, k;
346    uint16_t cbseed, cindex;
347    float    *rnd, tmp_gain, fir_filter_value;
348
349    switch(q->bitrate)
350    {
351        case RATE_FULL:
352            for(i=0; i<16; i++)
353            {
354                tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
355                cindex = -q->frame.cindex[i];
356                for(j=0; j<10; j++)
357                    *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cindex++ & 127];
358            }
359        break;
360        case RATE_HALF:
361            for(i=0; i<4; i++)
362            {
363                tmp_gain = gain[i] * QCELP_RATE_HALF_CODEBOOK_RATIO;
364                cindex = -q->frame.cindex[i];
365                for (j = 0; j < 40; j++)
366                *cdn_vector++ = tmp_gain * qcelp_rate_half_codebook[cindex++ & 127];
367            }
368        break;
369        case RATE_QUARTER:
370            cbseed = (0x0003 & q->frame.lspv[4])<<14 |
371                     (0x003F & q->frame.lspv[3])<< 8 |
372                     (0x0060 & q->frame.lspv[2])<< 1 |
373                     (0x0007 & q->frame.lspv[1])<< 3 |
374                     (0x0038 & q->frame.lspv[0])>> 3 ;
375            rnd = q->rnd_fir_filter_mem + 20;
376            for(i=0; i<8; i++)
377            {
378                tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
379                for(k=0; k<20; k++)
380                {
381                    cbseed = 521 * cbseed + 259;
382                    *rnd = (int16_t)cbseed;
383
384                    // FIR filter
385                    fir_filter_value = 0.0;
386                    for(j=0; j<10; j++)
387                        fir_filter_value += qcelp_rnd_fir_coefs[j ]
388                                          * (rnd[-j ] + rnd[-20+j]);
389
390                    fir_filter_value += qcelp_rnd_fir_coefs[10] * rnd[-10];
391                    *cdn_vector++ = tmp_gain * fir_filter_value;
392                    rnd++;
393                }
394            }
395            memcpy(q->rnd_fir_filter_mem, q->rnd_fir_filter_mem + 160, 20 * sizeof(float));
396        break;
397        case RATE_OCTAVE:
398            cbseed = q->first16bits;
399            for(i=0; i<8; i++)
400            {
401                tmp_gain = gain[i] * (QCELP_SQRT1887 / 32768.0);
402                for(j=0; j<20; j++)
403                {
404                    cbseed = 521 * cbseed + 259;
405                    *cdn_vector++ = tmp_gain * (int16_t)cbseed;
406                }
407            }
408        break;
409        case I_F_Q:
410            cbseed = -44; // random codebook index
411            for(i=0; i<4; i++)
412            {
413                tmp_gain = gain[i] * QCELP_RATE_FULL_CODEBOOK_RATIO;
414                for(j=0; j<40; j++)
415                    *cdn_vector++ = tmp_gain * qcelp_rate_full_codebook[cbseed++ & 127];
416            }
417        break;
418        case SILENCE:
419            memset(cdn_vector, 0, 160 * sizeof(float));
420        break;
421    }
422}
423
424/**
425 * Apply generic gain control.
426 *
427 * @param v_out output vector
428 * @param v_in gain-controlled vector
429 * @param v_ref vector to control gain of
430 *
431 * FIXME: If v_ref is a zero vector, it energy is zero
432 *        and the behavior of the gain control is
433 *        undefined in the specs.
434 *
435 * TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
436 */
437static void apply_gain_ctrl(float *v_out, const float *v_ref,
438                            const float *v_in)
439{
440    int   i, j, len;
441    float scalefactor;
442
443    for(i=0, j=0; i<4; i++)
444    {
445        scalefactor = ff_dot_productf(v_in + j, v_in + j, 40);
446        if(scalefactor)
447            scalefactor = sqrt(ff_dot_productf(v_ref + j, v_ref + j, 40)
448                        / scalefactor);
449        else
450            ff_log_missing_feature(NULL, "Zero energy for gain control", 1);
451        for(len=j+40; j<len; j++)
452            v_out[j] = scalefactor * v_in[j];
453    }
454}
455
456/**
457 * Apply filter in pitch-subframe steps.
458 *
459 * @param memory buffer for the previous state of the filter
460 *        - must be able to contain 303 elements
461 *        - the 143 first elements are from the previous state
462 *        - the next 160 are for output
463 * @param v_in input filter vector
464 * @param gain per-subframe gain array, each element is between 0.0 and 2.0
465 * @param lag per-subframe lag array, each element is
466 *        - between 16 and 143 if its corresponding pfrac is 0,
467 *        - between 16 and 139 otherwise
468 * @param pfrac per-subframe boolean array, 1 if the lag is fractional, 0
469 *        otherwise
470 *
471 * @return filter output vector
472 */
473static const float *do_pitchfilter(float memory[303], const float v_in[160],
474                                   const float gain[4], const uint8_t *lag,
475                                   const uint8_t pfrac[4])
476{
477    int         i, j;
478    float       *v_lag, *v_out;
479    const float *v_len;
480
481    v_out = memory + 143; // Output vector starts at memory[143].
482
483    for(i=0; i<4; i++)
484    {
485        if(gain[i])
486        {
487            v_lag = memory + 143 + 40 * i - lag[i];
488            for(v_len=v_in+40; v_in<v_len; v_in++)
489            {
490                if(pfrac[i]) // If it is a fractional lag...
491                {
492                    for(j=0, *v_out=0.; j<4; j++)
493                        *v_out += qcelp_hammsinc_table[j] * (v_lag[j-4] + v_lag[3-j]);
494                }else
495                    *v_out = *v_lag;
496
497                *v_out = *v_in + gain[i] * *v_out;
498
499                v_lag++;
500                v_out++;
501            }
502        }else
503        {
504            memcpy(v_out, v_in, 40 * sizeof(float));
505            v_in  += 40;
506            v_out += 40;
507        }
508    }
509
510    memmove(memory, memory + 160, 143 * sizeof(float));
511    return memory + 143;
512}
513
514/**
515 * Apply pitch synthesis filter and pitch prefilter to the scaled codebook vector.
516 * TIA/EIA/IS-733 2.4.5.2, 2.4.8.7.2
517 *
518 * @param q the context
519 * @param cdn_vector the scaled codebook vector
520 */
521static void apply_pitch_filters(QCELPContext *q, float *cdn_vector)
522{
523    int         i;
524    const float *v_synthesis_filtered, *v_pre_filtered;
525
526    if(q->bitrate >= RATE_HALF ||
527       q->bitrate == SILENCE ||
528       (q->bitrate == I_F_Q && (q->prev_bitrate >= RATE_HALF)))
529    {
530
531        if(q->bitrate >= RATE_HALF)
532        {
533
534            // Compute gain & lag for the whole frame.
535            for(i=0; i<4; i++)
536            {
537                q->pitch_gain[i] = q->frame.plag[i] ? (q->frame.pgain[i] + 1) * 0.25 : 0.0;
538
539                q->pitch_lag[i] = q->frame.plag[i] + 16;
540            }
541        }else
542        {
543            float max_pitch_gain;
544
545            if (q->bitrate == I_F_Q)
546            {
547                  if (q->erasure_count < 3)
548                      max_pitch_gain = 0.9 - 0.3 * (q->erasure_count - 1);
549                  else
550                      max_pitch_gain = 0.0;
551            }else
552            {
553                assert(q->bitrate == SILENCE);
554                max_pitch_gain = 1.0;
555            }
556            for(i=0; i<4; i++)
557                q->pitch_gain[i] = FFMIN(q->pitch_gain[i], max_pitch_gain);
558
559            memset(q->frame.pfrac, 0, sizeof(q->frame.pfrac));
560        }
561
562        // pitch synthesis filter
563        v_synthesis_filtered = do_pitchfilter(q->pitch_synthesis_filter_mem,
564                                              cdn_vector, q->pitch_gain,
565                                              q->pitch_lag, q->frame.pfrac);
566
567        // pitch prefilter update
568        for(i=0; i<4; i++)
569            q->pitch_gain[i] = 0.5 * FFMIN(q->pitch_gain[i], 1.0);
570
571        v_pre_filtered = do_pitchfilter(q->pitch_pre_filter_mem,
572                                        v_synthesis_filtered,
573                                        q->pitch_gain, q->pitch_lag,
574                                        q->frame.pfrac);
575
576        apply_gain_ctrl(cdn_vector, v_synthesis_filtered, v_pre_filtered);
577    }else
578    {
579        memcpy(q->pitch_synthesis_filter_mem, cdn_vector + 17,
580               143 * sizeof(float));
581        memcpy(q->pitch_pre_filter_mem, cdn_vector + 17, 143 * sizeof(float));
582        memset(q->pitch_gain, 0, sizeof(q->pitch_gain));
583        memset(q->pitch_lag,  0, sizeof(q->pitch_lag));
584    }
585}
586
587/**
588 * Interpolates LSP frequencies and computes LPC coefficients
589 * for a given bitrate & pitch subframe.
590 *
591 * TIA/EIA/IS-733 2.4.3.3.4, 2.4.8.7.2
592 *
593 * @param q the context
594 * @param curr_lspf LSP frequencies vector of the current frame
595 * @param lpc float vector for the resulting LPC
596 * @param subframe_num frame number in decoded stream
597 */
598void interpolate_lpc(QCELPContext *q, const float *curr_lspf, float *lpc,
599                     const int subframe_num)
600{
601    float interpolated_lspf[10];
602    float weight;
603
604    if(q->bitrate >= RATE_QUARTER)
605        weight = 0.25 * (subframe_num + 1);
606    else if(q->bitrate == RATE_OCTAVE && !subframe_num)
607        weight = 0.625;
608    else
609        weight = 1.0;
610
611    if(weight != 1.0)
612    {
613        weighted_vector_sumf(interpolated_lspf, curr_lspf, q->prev_lspf,
614                             weight, 1.0 - weight, 10);
615        ff_qcelp_lspf2lpc(interpolated_lspf, lpc);
616    }else if(q->bitrate >= RATE_QUARTER ||
617             (q->bitrate == I_F_Q && !subframe_num))
618        ff_qcelp_lspf2lpc(curr_lspf, lpc);
619    else if(q->bitrate == SILENCE && !subframe_num)
620        ff_qcelp_lspf2lpc(q->prev_lspf, lpc);
621}
622
623static qcelp_packet_rate buf_size2bitrate(const int buf_size)
624{
625    switch(buf_size)
626    {
627        case 35: return RATE_FULL;
628        case 17: return RATE_HALF;
629        case  8: return RATE_QUARTER;
630        case  4: return RATE_OCTAVE;
631        case  1: return SILENCE;
632    }
633
634    return I_F_Q;
635}
636
637/**
638 * Determine the bitrate from the frame size and/or the first byte of the frame.
639 *
640 * @param avctx the AV codec context
641 * @param buf_size length of the buffer
642 * @param buf the bufffer
643 *
644 * @return the bitrate on success,
645 *         I_F_Q  if the bitrate cannot be satisfactorily determined
646 *
647 * TIA/EIA/IS-733 2.4.8.7.1
648 */
649static qcelp_packet_rate determine_bitrate(AVCodecContext *avctx, const int buf_size,
650                             const uint8_t **buf)
651{
652    qcelp_packet_rate bitrate;
653
654    if((bitrate = buf_size2bitrate(buf_size)) >= 0)
655    {
656        if(bitrate > **buf)
657        {
658            QCELPContext *q = avctx->priv_data;
659            if (!q->warned_buf_mismatch_bitrate)
660            {
661            av_log(avctx, AV_LOG_WARNING,
662                   "Claimed bitrate and buffer size mismatch.\n");
663                q->warned_buf_mismatch_bitrate = 1;
664            }
665            bitrate = **buf;
666        }else if(bitrate < **buf)
667        {
668            av_log(avctx, AV_LOG_ERROR,
669                   "Buffer is too small for the claimed bitrate.\n");
670            return I_F_Q;
671        }
672        (*buf)++;
673    }else if((bitrate = buf_size2bitrate(buf_size + 1)) >= 0)
674    {
675        av_log(avctx, AV_LOG_WARNING,
676               "Bitrate byte is missing, guessing the bitrate from packet size.\n");
677    }else
678        return I_F_Q;
679
680    if(bitrate == SILENCE)
681    {
682        //FIXME: Remove experimental warning when tested with samples.
683        ff_log_ask_for_sample(avctx, "'Blank frame handling is experimental.");
684    }
685    return bitrate;
686}
687
688static void warn_insufficient_frame_quality(AVCodecContext *avctx,
689                                            const char *message)
690{
691    av_log(avctx, AV_LOG_WARNING, "Frame #%d, IFQ: %s\n", avctx->frame_number,
692           message);
693}
694
695static int qcelp_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
696                              const uint8_t *buf, int buf_size)
697{
698    QCELPContext *q = avctx->priv_data;
699    float *outbuffer = data;
700    int   i;
701    float quantized_lspf[10], lpc[10];
702    float gain[16];
703    float *formant_mem;
704
705    if((q->bitrate = determine_bitrate(avctx, buf_size, &buf)) == I_F_Q)
706    {
707        warn_insufficient_frame_quality(avctx, "bitrate cannot be determined.");
708        goto erasure;
709    }
710
711    if(q->bitrate == RATE_OCTAVE &&
712       (q->first16bits = AV_RB16(buf)) == 0xFFFF)
713    {
714        warn_insufficient_frame_quality(avctx, "Bitrate is 1/8 and first 16 bits are on.");
715        goto erasure;
716    }
717
718    if(q->bitrate > SILENCE)
719    {
720        const QCELPBitmap *bitmaps     = qcelp_unpacking_bitmaps_per_rate[q->bitrate];
721        const QCELPBitmap *bitmaps_end = qcelp_unpacking_bitmaps_per_rate[q->bitrate]
722                                       + qcelp_unpacking_bitmaps_lengths[q->bitrate];
723        uint8_t           *unpacked_data = (uint8_t *)&q->frame;
724
725        init_get_bits(&q->gb, buf, 8*buf_size);
726
727        memset(&q->frame, 0, sizeof(QCELPFrame));
728
729        for(; bitmaps < bitmaps_end; bitmaps++)
730            unpacked_data[bitmaps->index] |= get_bits(&q->gb, bitmaps->bitlen) << bitmaps->bitpos;
731
732        // Check for erasures/blanks on rates 1, 1/4 and 1/8.
733        if(q->frame.reserved)
734        {
735            warn_insufficient_frame_quality(avctx, "Wrong data in reserved frame area.");
736            goto erasure;
737        }
738        if(q->bitrate == RATE_QUARTER &&
739           codebook_sanity_check_for_rate_quarter(q->frame.cbgain))
740        {
741            warn_insufficient_frame_quality(avctx, "Codebook gain sanity check failed.");
742            goto erasure;
743        }
744
745        if(q->bitrate >= RATE_HALF)
746        {
747            for(i=0; i<4; i++)
748            {
749                if(q->frame.pfrac[i] && q->frame.plag[i] >= 124)
750                {
751                    warn_insufficient_frame_quality(avctx, "Cannot initialize pitch filter.");
752                    goto erasure;
753                }
754            }
755        }
756    }
757
758    decode_gain_and_index(q, gain);
759    compute_svector(q, gain, outbuffer);
760
761    if(decode_lspf(q, quantized_lspf) < 0)
762    {
763        warn_insufficient_frame_quality(avctx, "Badly received packets in frame.");
764        goto erasure;
765    }
766
767
768    apply_pitch_filters(q, outbuffer);
769
770    if(q->bitrate == I_F_Q)
771    {
772erasure:
773        q->bitrate = I_F_Q;
774        q->erasure_count++;
775        decode_gain_and_index(q, gain);
776        compute_svector(q, gain, outbuffer);
777        decode_lspf(q, quantized_lspf);
778        apply_pitch_filters(q, outbuffer);
779    }else
780        q->erasure_count = 0;
781
782    formant_mem = q->formant_mem + 10;
783    for(i=0; i<4; i++)
784    {
785        interpolate_lpc(q, quantized_lspf, lpc, i);
786        ff_celp_lp_synthesis_filterf(formant_mem, lpc, outbuffer + i * 40, 40,
787                                     10);
788        formant_mem += 40;
789    }
790    memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
791
792    // FIXME: postfilter and final gain control should be here.
793    // TIA/EIA/IS-733 2.4.8.6
794
795    formant_mem = q->formant_mem + 10;
796    for(i=0; i<160; i++)
797        *outbuffer++ = av_clipf(*formant_mem++, QCELP_CLIP_LOWER_BOUND,
798                                QCELP_CLIP_UPPER_BOUND);
799
800    memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
801    q->prev_bitrate = q->bitrate;
802
803    *data_size = 160 * sizeof(*outbuffer);
804
805    return *data_size;
806}
807
808AVCodec qcelp_decoder =
809{
810    .name   = "qcelp",
811    .type   = CODEC_TYPE_AUDIO,
812    .id     = CODEC_ID_QCELP,
813    .init   = qcelp_decode_init,
814    .decode = qcelp_decode_frame,
815    .priv_data_size = sizeof(QCELPContext),
816    .long_name = NULL_IF_CONFIG_SMALL("QCELP / PureVoice"),
817};
818