Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / sonic.c
1 /*
2 * Simple free lossless/lossy audio codec
3 * Copyright (c) 2004 Alex Beregszaszi
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 #include "avcodec.h"
22 #include "get_bits.h"
23 #include "golomb.h"
24 #include "internal.h"
25 #include "rangecoder.h"
26
27
28 /**
29 * @file
30 * Simple free lossless/lossy audio codec
31 * Based on Paul Francis Harrison's Bonk (http://www.logarithmic.net/pfh/bonk)
32 * Written and designed by Alex Beregszaszi
33 *
34 * TODO:
35 * - CABAC put/get_symbol
36 * - independent quantizer for channels
37 * - >2 channels support
38 * - more decorrelation types
39 * - more tap_quant tests
40 * - selectable intlist writers/readers (bonk-style, golomb, cabac)
41 */
42
43 #define MAX_CHANNELS 2
44
45 #define MID_SIDE 0
46 #define LEFT_SIDE 1
47 #define RIGHT_SIDE 2
48
49 typedef struct SonicContext {
50 int version;
51 int minor_version;
52 int lossless, decorrelation;
53
54 int num_taps, downsampling;
55 double quantization;
56
57 int channels, samplerate, block_align, frame_size;
58
59 int *tap_quant;
60 int *int_samples;
61 int *coded_samples[MAX_CHANNELS];
62
63 // for encoding
64 int *tail;
65 int tail_size;
66 int *window;
67 int window_size;
68
69 // for decoding
70 int *predictor_k;
71 int *predictor_state[MAX_CHANNELS];
72 } SonicContext;
73
74 #define LATTICE_SHIFT 10
75 #define SAMPLE_SHIFT 4
76 #define LATTICE_FACTOR (1 << LATTICE_SHIFT)
77 #define SAMPLE_FACTOR (1 << SAMPLE_SHIFT)
78
79 #define BASE_QUANT 0.6
80 #define RATE_VARIATION 3.0
81
82 static inline int shift(int a,int b)
83 {
84 return (a+(1<<(b-1))) >> b;
85 }
86
87 static inline int shift_down(int a,int b)
88 {
89 return (a>>b)+(a<0);
90 }
91
92 static av_always_inline av_flatten void put_symbol(RangeCoder *c, uint8_t *state, int v, int is_signed, uint64_t rc_stat[256][2], uint64_t rc_stat2[32][2]){
93 int i;
94
95 #define put_rac(C,S,B) \
96 do{\
97 if(rc_stat){\
98 rc_stat[*(S)][B]++;\
99 rc_stat2[(S)-state][B]++;\
100 }\
101 put_rac(C,S,B);\
102 }while(0)
103
104 if(v){
105 const int a= FFABS(v);
106 const int e= av_log2(a);
107 put_rac(c, state+0, 0);
108 if(e<=9){
109 for(i=0; i<e; i++){
110 put_rac(c, state+1+i, 1); //1..10
111 }
112 put_rac(c, state+1+i, 0);
113
114 for(i=e-1; i>=0; i--){
115 put_rac(c, state+22+i, (a>>i)&1); //22..31
116 }
117
118 if(is_signed)
119 put_rac(c, state+11 + e, v < 0); //11..21
120 }else{
121 for(i=0; i<e; i++){
122 put_rac(c, state+1+FFMIN(i,9), 1); //1..10
123 }
124 put_rac(c, state+1+9, 0);
125
126 for(i=e-1; i>=0; i--){
127 put_rac(c, state+22+FFMIN(i,9), (a>>i)&1); //22..31
128 }
129
130 if(is_signed)
131 put_rac(c, state+11 + 10, v < 0); //11..21
132 }
133 }else{
134 put_rac(c, state+0, 1);
135 }
136 #undef put_rac
137 }
138
139 static inline av_flatten int get_symbol(RangeCoder *c, uint8_t *state, int is_signed){
140 if(get_rac(c, state+0))
141 return 0;
142 else{
143 int i, e, a;
144 e= 0;
145 while(get_rac(c, state+1 + FFMIN(e,9))){ //1..10
146 e++;
147 }
148
149 a= 1;
150 for(i=e-1; i>=0; i--){
151 a += a + get_rac(c, state+22 + FFMIN(i,9)); //22..31
152 }
153
154 e= -(is_signed && get_rac(c, state+11 + FFMIN(e, 10))); //11..21
155 return (a^e)-e;
156 }
157 }
158
159 #if 1
160 static inline int intlist_write(RangeCoder *c, uint8_t *state, int *buf, int entries, int base_2_part)
161 {
162 int i;
163
164 for (i = 0; i < entries; i++)
165 put_symbol(c, state, buf[i], 1, NULL, NULL);
166
167 return 1;
168 }
169
170 static inline int intlist_read(RangeCoder *c, uint8_t *state, int *buf, int entries, int base_2_part)
171 {
172 int i;
173
174 for (i = 0; i < entries; i++)
175 buf[i] = get_symbol(c, state, 1);
176
177 return 1;
178 }
179 #elif 1
180 static inline int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part)
181 {
182 int i;
183
184 for (i = 0; i < entries; i++)
185 set_se_golomb(pb, buf[i]);
186
187 return 1;
188 }
189
190 static inline int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
191 {
192 int i;
193
194 for (i = 0; i < entries; i++)
195 buf[i] = get_se_golomb(gb);
196
197 return 1;
198 }
199
200 #else
201
202 #define ADAPT_LEVEL 8
203
204 static int bits_to_store(uint64_t x)
205 {
206 int res = 0;
207
208 while(x)
209 {
210 res++;
211 x >>= 1;
212 }
213 return res;
214 }
215
216 static void write_uint_max(PutBitContext *pb, unsigned int value, unsigned int max)
217 {
218 int i, bits;
219
220 if (!max)
221 return;
222
223 bits = bits_to_store(max);
224
225 for (i = 0; i < bits-1; i++)
226 put_bits(pb, 1, value & (1 << i));
227
228 if ( (value | (1 << (bits-1))) <= max)
229 put_bits(pb, 1, value & (1 << (bits-1)));
230 }
231
232 static unsigned int read_uint_max(GetBitContext *gb, int max)
233 {
234 int i, bits, value = 0;
235
236 if (!max)
237 return 0;
238
239 bits = bits_to_store(max);
240
241 for (i = 0; i < bits-1; i++)
242 if (get_bits1(gb))
243 value += 1 << i;
244
245 if ( (value | (1<<(bits-1))) <= max)
246 if (get_bits1(gb))
247 value += 1 << (bits-1);
248
249 return value;
250 }
251
252 static int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part)
253 {
254 int i, j, x = 0, low_bits = 0, max = 0;
255 int step = 256, pos = 0, dominant = 0, any = 0;
256 int *copy, *bits;
257
258 copy = av_calloc(entries, sizeof(*copy));
259 if (!copy)
260 return AVERROR(ENOMEM);
261
262 if (base_2_part)
263 {
264 int energy = 0;
265
266 for (i = 0; i < entries; i++)
267 energy += abs(buf[i]);
268
269 low_bits = bits_to_store(energy / (entries * 2));
270 if (low_bits > 15)
271 low_bits = 15;
272
273 put_bits(pb, 4, low_bits);
274 }
275
276 for (i = 0; i < entries; i++)
277 {
278 put_bits(pb, low_bits, abs(buf[i]));
279 copy[i] = abs(buf[i]) >> low_bits;
280 if (copy[i] > max)
281 max = abs(copy[i]);
282 }
283
284 bits = av_calloc(entries*max, sizeof(*bits));
285 if (!bits)
286 {
287 av_free(copy);
288 return AVERROR(ENOMEM);
289 }
290
291 for (i = 0; i <= max; i++)
292 {
293 for (j = 0; j < entries; j++)
294 if (copy[j] >= i)
295 bits[x++] = copy[j] > i;
296 }
297
298 // store bitstream
299 while (pos < x)
300 {
301 int steplet = step >> 8;
302
303 if (pos + steplet > x)
304 steplet = x - pos;
305
306 for (i = 0; i < steplet; i++)
307 if (bits[i+pos] != dominant)
308 any = 1;
309
310 put_bits(pb, 1, any);
311
312 if (!any)
313 {
314 pos += steplet;
315 step += step / ADAPT_LEVEL;
316 }
317 else
318 {
319 int interloper = 0;
320
321 while (((pos + interloper) < x) && (bits[pos + interloper] == dominant))
322 interloper++;
323
324 // note change
325 write_uint_max(pb, interloper, (step >> 8) - 1);
326
327 pos += interloper + 1;
328 step -= step / ADAPT_LEVEL;
329 }
330
331 if (step < 256)
332 {
333 step = 65536 / step;
334 dominant = !dominant;
335 }
336 }
337
338 // store signs
339 for (i = 0; i < entries; i++)
340 if (buf[i])
341 put_bits(pb, 1, buf[i] < 0);
342
343 av_free(bits);
344 av_free(copy);
345
346 return 0;
347 }
348
349 static int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part)
350 {
351 int i, low_bits = 0, x = 0;
352 int n_zeros = 0, step = 256, dominant = 0;
353 int pos = 0, level = 0;
354 int *bits = av_calloc(entries, sizeof(*bits));
355
356 if (!bits)
357 return AVERROR(ENOMEM);
358
359 if (base_2_part)
360 {
361 low_bits = get_bits(gb, 4);
362
363 if (low_bits)
364 for (i = 0; i < entries; i++)
365 buf[i] = get_bits(gb, low_bits);
366 }
367
368 // av_log(NULL, AV_LOG_INFO, "entries: %d, low bits: %d\n", entries, low_bits);
369
370 while (n_zeros < entries)
371 {
372 int steplet = step >> 8;
373
374 if (!get_bits1(gb))
375 {
376 for (i = 0; i < steplet; i++)
377 bits[x++] = dominant;
378
379 if (!dominant)
380 n_zeros += steplet;
381
382 step += step / ADAPT_LEVEL;
383 }
384 else
385 {
386 int actual_run = read_uint_max(gb, steplet-1);
387
388 // av_log(NULL, AV_LOG_INFO, "actual run: %d\n", actual_run);
389
390 for (i = 0; i < actual_run; i++)
391 bits[x++] = dominant;
392
393 bits[x++] = !dominant;
394
395 if (!dominant)
396 n_zeros += actual_run;
397 else
398 n_zeros++;
399
400 step -= step / ADAPT_LEVEL;
401 }
402
403 if (step < 256)
404 {
405 step = 65536 / step;
406 dominant = !dominant;
407 }
408 }
409
410 // reconstruct unsigned values
411 n_zeros = 0;
412 for (i = 0; n_zeros < entries; i++)
413 {
414 while(1)
415 {
416 if (pos >= entries)
417 {
418 pos = 0;
419 level += 1 << low_bits;
420 }
421
422 if (buf[pos] >= level)
423 break;
424
425 pos++;
426 }
427
428 if (bits[i])
429 buf[pos] += 1 << low_bits;
430 else
431 n_zeros++;
432
433 pos++;
434 }
435 av_free(bits);
436
437 // read signs
438 for (i = 0; i < entries; i++)
439 if (buf[i] && get_bits1(gb))
440 buf[i] = -buf[i];
441
442 // av_log(NULL, AV_LOG_INFO, "zeros: %d pos: %d\n", n_zeros, pos);
443
444 return 0;
445 }
446 #endif
447
448 static void predictor_init_state(int *k, int *state, int order)
449 {
450 int i;
451
452 for (i = order-2; i >= 0; i--)
453 {
454 int j, p, x = state[i];
455
456 for (j = 0, p = i+1; p < order; j++,p++)
457 {
458 int tmp = x + shift_down(k[j] * state[p], LATTICE_SHIFT);
459 state[p] += shift_down(k[j]*x, LATTICE_SHIFT);
460 x = tmp;
461 }
462 }
463 }
464
465 static int predictor_calc_error(int *k, int *state, int order, int error)
466 {
467 int i, x = error - shift_down(k[order-1] * state[order-1], LATTICE_SHIFT);
468
469 #if 1
470 int *k_ptr = &(k[order-2]),
471 *state_ptr = &(state[order-2]);
472 for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--)
473 {
474 int k_value = *k_ptr, state_value = *state_ptr;
475 x -= shift_down(k_value * state_value, LATTICE_SHIFT);
476 state_ptr[1] = state_value + shift_down(k_value * x, LATTICE_SHIFT);
477 }
478 #else
479 for (i = order-2; i >= 0; i--)
480 {
481 x -= shift_down(k[i] * state[i], LATTICE_SHIFT);
482 state[i+1] = state[i] + shift_down(k[i] * x, LATTICE_SHIFT);
483 }
484 #endif
485
486 // don't drift too far, to avoid overflows
487 if (x > (SAMPLE_FACTOR<<16)) x = (SAMPLE_FACTOR<<16);
488 if (x < -(SAMPLE_FACTOR<<16)) x = -(SAMPLE_FACTOR<<16);
489
490 state[0] = x;
491
492 return x;
493 }
494
495 #if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER
496 // Heavily modified Levinson-Durbin algorithm which
497 // copes better with quantization, and calculates the
498 // actual whitened result as it goes.
499
500 static void modified_levinson_durbin(int *window, int window_entries,
501 int *out, int out_entries, int channels, int *tap_quant)
502 {
503 int i;
504 int *state = av_calloc(window_entries, sizeof(*state));
505
506 memcpy(state, window, 4* window_entries);
507
508 for (i = 0; i < out_entries; i++)
509 {
510 int step = (i+1)*channels, k, j;
511 double xx = 0.0, xy = 0.0;
512 #if 1
513 int *x_ptr = &(window[step]);
514 int *state_ptr = &(state[0]);
515 j = window_entries - step;
516 for (;j>0;j--,x_ptr++,state_ptr++)
517 {
518 double x_value = *x_ptr;
519 double state_value = *state_ptr;
520 xx += state_value*state_value;
521 xy += x_value*state_value;
522 }
523 #else
524 for (j = 0; j <= (window_entries - step); j++);
525 {
526 double stepval = window[step+j];
527 double stateval = window[j];
528 // xx += (double)window[j]*(double)window[j];
529 // xy += (double)window[step+j]*(double)window[j];
530 xx += stateval*stateval;
531 xy += stepval*stateval;
532 }
533 #endif
534 if (xx == 0.0)
535 k = 0;
536 else
537 k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5));
538
539 if (k > (LATTICE_FACTOR/tap_quant[i]))
540 k = LATTICE_FACTOR/tap_quant[i];
541 if (-k > (LATTICE_FACTOR/tap_quant[i]))
542 k = -(LATTICE_FACTOR/tap_quant[i]);
543
544 out[i] = k;
545 k *= tap_quant[i];
546
547 #if 1
548 x_ptr = &(window[step]);
549 state_ptr = &(state[0]);
550 j = window_entries - step;
551 for (;j>0;j--,x_ptr++,state_ptr++)
552 {
553 int x_value = *x_ptr;
554 int state_value = *state_ptr;
555 *x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT);
556 *state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT);
557 }
558 #else
559 for (j=0; j <= (window_entries - step); j++)
560 {
561 int stepval = window[step+j];
562 int stateval=state[j];
563 window[step+j] += shift_down(k * stateval, LATTICE_SHIFT);
564 state[j] += shift_down(k * stepval, LATTICE_SHIFT);
565 }
566 #endif
567 }
568
569 av_free(state);
570 }
571
572 static inline int code_samplerate(int samplerate)
573 {
574 switch (samplerate)
575 {
576 case 44100: return 0;
577 case 22050: return 1;
578 case 11025: return 2;
579 case 96000: return 3;
580 case 48000: return 4;
581 case 32000: return 5;
582 case 24000: return 6;
583 case 16000: return 7;
584 case 8000: return 8;
585 }
586 return AVERROR(EINVAL);
587 }
588
589 static av_cold int sonic_encode_init(AVCodecContext *avctx)
590 {
591 SonicContext *s = avctx->priv_data;
592 PutBitContext pb;
593 int i;
594
595 s->version = 2;
596
597 if (avctx->channels > MAX_CHANNELS)
598 {
599 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
600 return AVERROR(EINVAL); /* only stereo or mono for now */
601 }
602
603 if (avctx->channels == 2)
604 s->decorrelation = MID_SIDE;
605 else
606 s->decorrelation = 3;
607
608 if (avctx->codec->id == AV_CODEC_ID_SONIC_LS)
609 {
610 s->lossless = 1;
611 s->num_taps = 32;
612 s->downsampling = 1;
613 s->quantization = 0.0;
614 }
615 else
616 {
617 s->num_taps = 128;
618 s->downsampling = 2;
619 s->quantization = 1.0;
620 }
621
622 // max tap 2048
623 if (s->num_taps < 32 || s->num_taps > 1024 || s->num_taps % 32) {
624 av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n");
625 return AVERROR_INVALIDDATA;
626 }
627
628 // generate taps
629 s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant));
630 for (i = 0; i < s->num_taps; i++)
631 s->tap_quant[i] = ff_sqrt(i+1);
632
633 s->channels = avctx->channels;
634 s->samplerate = avctx->sample_rate;
635
636 s->block_align = 2048LL*s->samplerate/(44100*s->downsampling);
637 s->frame_size = s->channels*s->block_align*s->downsampling;
638
639 s->tail_size = s->num_taps*s->channels;
640 s->tail = av_calloc(s->tail_size, sizeof(*s->tail));
641 if (!s->tail)
642 return AVERROR(ENOMEM);
643
644 s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k) );
645 if (!s->predictor_k)
646 return AVERROR(ENOMEM);
647
648 for (i = 0; i < s->channels; i++)
649 {
650 s->coded_samples[i] = av_calloc(s->block_align, sizeof(**s->coded_samples));
651 if (!s->coded_samples[i])
652 return AVERROR(ENOMEM);
653 }
654
655 s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples));
656
657 s->window_size = ((2*s->tail_size)+s->frame_size);
658 s->window = av_calloc(s->window_size, sizeof(*s->window));
659 if (!s->window)
660 return AVERROR(ENOMEM);
661
662 avctx->extradata = av_mallocz(16);
663 if (!avctx->extradata)
664 return AVERROR(ENOMEM);
665 init_put_bits(&pb, avctx->extradata, 16*8);
666
667 put_bits(&pb, 2, s->version); // version
668 if (s->version >= 1)
669 {
670 if (s->version >= 2) {
671 put_bits(&pb, 8, s->version);
672 put_bits(&pb, 8, s->minor_version);
673 }
674 put_bits(&pb, 2, s->channels);
675 put_bits(&pb, 4, code_samplerate(s->samplerate));
676 }
677 put_bits(&pb, 1, s->lossless);
678 if (!s->lossless)
679 put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision
680 put_bits(&pb, 2, s->decorrelation);
681 put_bits(&pb, 2, s->downsampling);
682 put_bits(&pb, 5, (s->num_taps >> 5)-1); // 32..1024
683 put_bits(&pb, 1, 0); // XXX FIXME: no custom tap quant table
684
685 flush_put_bits(&pb);
686 avctx->extradata_size = put_bits_count(&pb)/8;
687
688 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d.%d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
689 s->version, s->minor_version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
690
691 avctx->frame_size = s->block_align*s->downsampling;
692
693 return 0;
694 }
695
696 static av_cold int sonic_encode_close(AVCodecContext *avctx)
697 {
698 SonicContext *s = avctx->priv_data;
699 int i;
700
701 for (i = 0; i < s->channels; i++)
702 av_freep(&s->coded_samples[i]);
703
704 av_freep(&s->predictor_k);
705 av_freep(&s->tail);
706 av_freep(&s->tap_quant);
707 av_freep(&s->window);
708 av_freep(&s->int_samples);
709
710 return 0;
711 }
712
713 static int sonic_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
714 const AVFrame *frame, int *got_packet_ptr)
715 {
716 SonicContext *s = avctx->priv_data;
717 RangeCoder c;
718 int i, j, ch, quant = 0, x = 0;
719 int ret;
720 const short *samples = (const int16_t*)frame->data[0];
721 uint8_t state[32];
722
723 if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size * 5 + 1000)) < 0)
724 return ret;
725
726 ff_init_range_encoder(&c, avpkt->data, avpkt->size);
727 ff_build_rac_states(&c, 0.05*(1LL<<32), 256-8);
728 memset(state, 128, sizeof(state));
729
730 // short -> internal
731 for (i = 0; i < s->frame_size; i++)
732 s->int_samples[i] = samples[i];
733
734 if (!s->lossless)
735 for (i = 0; i < s->frame_size; i++)
736 s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT;
737
738 switch(s->decorrelation)
739 {
740 case MID_SIDE:
741 for (i = 0; i < s->frame_size; i += s->channels)
742 {
743 s->int_samples[i] += s->int_samples[i+1];
744 s->int_samples[i+1] -= shift(s->int_samples[i], 1);
745 }
746 break;
747 case LEFT_SIDE:
748 for (i = 0; i < s->frame_size; i += s->channels)
749 s->int_samples[i+1] -= s->int_samples[i];
750 break;
751 case RIGHT_SIDE:
752 for (i = 0; i < s->frame_size; i += s->channels)
753 s->int_samples[i] -= s->int_samples[i+1];
754 break;
755 }
756
757 memset(s->window, 0, 4* s->window_size);
758
759 for (i = 0; i < s->tail_size; i++)
760 s->window[x++] = s->tail[i];
761
762 for (i = 0; i < s->frame_size; i++)
763 s->window[x++] = s->int_samples[i];
764
765 for (i = 0; i < s->tail_size; i++)
766 s->window[x++] = 0;
767
768 for (i = 0; i < s->tail_size; i++)
769 s->tail[i] = s->int_samples[s->frame_size - s->tail_size + i];
770
771 // generate taps
772 modified_levinson_durbin(s->window, s->window_size,
773 s->predictor_k, s->num_taps, s->channels, s->tap_quant);
774 if ((ret = intlist_write(&c, state, s->predictor_k, s->num_taps, 0)) < 0)
775 return ret;
776
777 for (ch = 0; ch < s->channels; ch++)
778 {
779 x = s->tail_size+ch;
780 for (i = 0; i < s->block_align; i++)
781 {
782 int sum = 0;
783 for (j = 0; j < s->downsampling; j++, x += s->channels)
784 sum += s->window[x];
785 s->coded_samples[ch][i] = sum;
786 }
787 }
788
789 // simple rate control code
790 if (!s->lossless)
791 {
792 double energy1 = 0.0, energy2 = 0.0;
793 for (ch = 0; ch < s->channels; ch++)
794 {
795 for (i = 0; i < s->block_align; i++)
796 {
797 double sample = s->coded_samples[ch][i];
798 energy2 += sample*sample;
799 energy1 += fabs(sample);
800 }
801 }
802
803 energy2 = sqrt(energy2/(s->channels*s->block_align));
804 energy1 = M_SQRT2*energy1/(s->channels*s->block_align);
805
806 // increase bitrate when samples are like a gaussian distribution
807 // reduce bitrate when samples are like a two-tailed exponential distribution
808
809 if (energy2 > energy1)
810 energy2 += (energy2-energy1)*RATE_VARIATION;
811
812 quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR);
813 // av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2);
814
815 quant = av_clip(quant, 1, 65534);
816
817 put_symbol(&c, state, quant, 0, NULL, NULL);
818
819 quant *= SAMPLE_FACTOR;
820 }
821
822 // write out coded samples
823 for (ch = 0; ch < s->channels; ch++)
824 {
825 if (!s->lossless)
826 for (i = 0; i < s->block_align; i++)
827 s->coded_samples[ch][i] = ROUNDED_DIV(s->coded_samples[ch][i], quant);
828
829 if ((ret = intlist_write(&c, state, s->coded_samples[ch], s->block_align, 1)) < 0)
830 return ret;
831 }
832
833 // av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8);
834
835 avpkt->size = ff_rac_terminate(&c);
836 *got_packet_ptr = 1;
837 return 0;
838
839 }
840 #endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */
841
842 #if CONFIG_SONIC_DECODER
843 static const int samplerate_table[] =
844 { 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 };
845
846 static av_cold int sonic_decode_init(AVCodecContext *avctx)
847 {
848 SonicContext *s = avctx->priv_data;
849 GetBitContext gb;
850 int i;
851
852 s->channels = avctx->channels;
853 s->samplerate = avctx->sample_rate;
854
855 if (!avctx->extradata)
856 {
857 av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n");
858 return AVERROR_INVALIDDATA;
859 }
860
861 init_get_bits8(&gb, avctx->extradata, avctx->extradata_size);
862
863 s->version = get_bits(&gb, 2);
864 if (s->version >= 2) {
865 s->version = get_bits(&gb, 8);
866 s->minor_version = get_bits(&gb, 8);
867 }
868 if (s->version != 2)
869 {
870 av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n");
871 return AVERROR_INVALIDDATA;
872 }
873
874 if (s->version >= 1)
875 {
876 s->channels = get_bits(&gb, 2);
877 s->samplerate = samplerate_table[get_bits(&gb, 4)];
878 av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n",
879 s->channels, s->samplerate);
880 }
881
882 if (s->channels > MAX_CHANNELS)
883 {
884 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
885 return AVERROR_INVALIDDATA;
886 }
887
888 s->lossless = get_bits1(&gb);
889 if (!s->lossless)
890 skip_bits(&gb, 3); // XXX FIXME
891 s->decorrelation = get_bits(&gb, 2);
892 if (s->decorrelation != 3 && s->channels != 2) {
893 av_log(avctx, AV_LOG_ERROR, "invalid decorrelation %d\n", s->decorrelation);
894 return AVERROR_INVALIDDATA;
895 }
896
897 s->downsampling = get_bits(&gb, 2);
898 if (!s->downsampling) {
899 av_log(avctx, AV_LOG_ERROR, "invalid downsampling value\n");
900 return AVERROR_INVALIDDATA;
901 }
902
903 s->num_taps = (get_bits(&gb, 5)+1)<<5;
904 if (get_bits1(&gb)) // XXX FIXME
905 av_log(avctx, AV_LOG_INFO, "Custom quant table\n");
906
907 s->block_align = 2048LL*s->samplerate/(44100*s->downsampling);
908 s->frame_size = s->channels*s->block_align*s->downsampling;
909 // avctx->frame_size = s->block_align;
910
911 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d.%d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
912 s->version, s->minor_version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
913
914 // generate taps
915 s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant));
916 for (i = 0; i < s->num_taps; i++)
917 s->tap_quant[i] = ff_sqrt(i+1);
918
919 s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k));
920
921 for (i = 0; i < s->channels; i++)
922 {
923 s->predictor_state[i] = av_calloc(s->num_taps, sizeof(**s->predictor_state));
924 if (!s->predictor_state[i])
925 return AVERROR(ENOMEM);
926 }
927
928 for (i = 0; i < s->channels; i++)
929 {
930 s->coded_samples[i] = av_calloc(s->block_align, sizeof(**s->coded_samples));
931 if (!s->coded_samples[i])
932 return AVERROR(ENOMEM);
933 }
934 s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples));
935
936 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
937 return 0;
938 }
939
940 static av_cold int sonic_decode_close(AVCodecContext *avctx)
941 {
942 SonicContext *s = avctx->priv_data;
943 int i;
944
945 av_freep(&s->int_samples);
946 av_freep(&s->tap_quant);
947 av_freep(&s->predictor_k);
948
949 for (i = 0; i < s->channels; i++)
950 {
951 av_freep(&s->predictor_state[i]);
952 av_freep(&s->coded_samples[i]);
953 }
954
955 return 0;
956 }
957
958 static int sonic_decode_frame(AVCodecContext *avctx,
959 void *data, int *got_frame_ptr,
960 AVPacket *avpkt)
961 {
962 const uint8_t *buf = avpkt->data;
963 int buf_size = avpkt->size;
964 SonicContext *s = avctx->priv_data;
965 RangeCoder c;
966 uint8_t state[32];
967 int i, quant, ch, j, ret;
968 int16_t *samples;
969 AVFrame *frame = data;
970
971 if (buf_size == 0) return 0;
972
973 frame->nb_samples = s->frame_size / avctx->channels;
974 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
975 return ret;
976 samples = (int16_t *)frame->data[0];
977
978 // av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size);
979
980 memset(state, 128, sizeof(state));
981 ff_init_range_decoder(&c, buf, buf_size);
982 ff_build_rac_states(&c, 0.05*(1LL<<32), 256-8);
983
984 intlist_read(&c, state, s->predictor_k, s->num_taps, 0);
985
986 // dequantize
987 for (i = 0; i < s->num_taps; i++)
988 s->predictor_k[i] *= s->tap_quant[i];
989
990 if (s->lossless)
991 quant = 1;
992 else
993 quant = get_symbol(&c, state, 0) * SAMPLE_FACTOR;
994
995 // av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant);
996
997 for (ch = 0; ch < s->channels; ch++)
998 {
999 int x = ch;
1000
1001 predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps);
1002
1003 intlist_read(&c, state, s->coded_samples[ch], s->block_align, 1);
1004
1005 for (i = 0; i < s->block_align; i++)
1006 {
1007 for (j = 0; j < s->downsampling - 1; j++)
1008 {
1009 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0);
1010 x += s->channels;
1011 }
1012
1013 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * quant);
1014 x += s->channels;
1015 }
1016
1017 for (i = 0; i < s->num_taps; i++)
1018 s->predictor_state[ch][i] = s->int_samples[s->frame_size - s->channels + ch - i*s->channels];
1019 }
1020
1021 switch(s->decorrelation)
1022 {
1023 case MID_SIDE:
1024 for (i = 0; i < s->frame_size; i += s->channels)
1025 {
1026 s->int_samples[i+1] += shift(s->int_samples[i], 1);
1027 s->int_samples[i] -= s->int_samples[i+1];
1028 }
1029 break;
1030 case LEFT_SIDE:
1031 for (i = 0; i < s->frame_size; i += s->channels)
1032 s->int_samples[i+1] += s->int_samples[i];
1033 break;
1034 case RIGHT_SIDE:
1035 for (i = 0; i < s->frame_size; i += s->channels)
1036 s->int_samples[i] += s->int_samples[i+1];
1037 break;
1038 }
1039
1040 if (!s->lossless)
1041 for (i = 0; i < s->frame_size; i++)
1042 s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT);
1043
1044 // internal -> short
1045 for (i = 0; i < s->frame_size; i++)
1046 samples[i] = av_clip_int16(s->int_samples[i]);
1047
1048 *got_frame_ptr = 1;
1049
1050 return buf_size;
1051 }
1052
1053 AVCodec ff_sonic_decoder = {
1054 .name = "sonic",
1055 .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
1056 .type = AVMEDIA_TYPE_AUDIO,
1057 .id = AV_CODEC_ID_SONIC,
1058 .priv_data_size = sizeof(SonicContext),
1059 .init = sonic_decode_init,
1060 .close = sonic_decode_close,
1061 .decode = sonic_decode_frame,
1062 .capabilities = CODEC_CAP_DR1 | CODEC_CAP_EXPERIMENTAL,
1063 };
1064 #endif /* CONFIG_SONIC_DECODER */
1065
1066 #if CONFIG_SONIC_ENCODER
1067 AVCodec ff_sonic_encoder = {
1068 .name = "sonic",
1069 .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
1070 .type = AVMEDIA_TYPE_AUDIO,
1071 .id = AV_CODEC_ID_SONIC,
1072 .priv_data_size = sizeof(SonicContext),
1073 .init = sonic_encode_init,
1074 .encode2 = sonic_encode_frame,
1075 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
1076 .capabilities = CODEC_CAP_EXPERIMENTAL,
1077 .close = sonic_encode_close,
1078 };
1079 #endif
1080
1081 #if CONFIG_SONIC_LS_ENCODER
1082 AVCodec ff_sonic_ls_encoder = {
1083 .name = "sonicls",
1084 .long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"),
1085 .type = AVMEDIA_TYPE_AUDIO,
1086 .id = AV_CODEC_ID_SONIC_LS,
1087 .priv_data_size = sizeof(SonicContext),
1088 .init = sonic_encode_init,
1089 .encode2 = sonic_encode_frame,
1090 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
1091 .capabilities = CODEC_CAP_EXPERIMENTAL,
1092 .close = sonic_encode_close,
1093 };
1094 #endif