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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
49typedef 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
82static inline int shift(int a,int b)
83{
84 return (a+(1<<(b-1))) >> b;
85}
86
87static inline int shift_down(int a,int b)
88{
89 return (a>>b)+(a<0);
90}
91
92static 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) \
96do{\
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
139static 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
160static 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
170static 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
180static 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
190static 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
204static 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
216static 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
232static 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
252static 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
349static 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
448static 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
465static 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
500static int 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 if (!state)
507 return AVERROR(ENOMEM);
508
509 memcpy(state, window, 4* window_entries);
510
511 for (i = 0; i < out_entries; i++)
512 {
513 int step = (i+1)*channels, k, j;
514 double xx = 0.0, xy = 0.0;
515#if 1
516 int *x_ptr = &(window[step]);
517 int *state_ptr = &(state[0]);
518 j = window_entries - step;
519 for (;j>0;j--,x_ptr++,state_ptr++)
520 {
521 double x_value = *x_ptr;
522 double state_value = *state_ptr;
523 xx += state_value*state_value;
524 xy += x_value*state_value;
525 }
526#else
527 for (j = 0; j <= (window_entries - step); j++);
528 {
529 double stepval = window[step+j];
530 double stateval = window[j];
531// xx += (double)window[j]*(double)window[j];
532// xy += (double)window[step+j]*(double)window[j];
533 xx += stateval*stateval;
534 xy += stepval*stateval;
535 }
536#endif
537 if (xx == 0.0)
538 k = 0;
539 else
540 k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5));
541
542 if (k > (LATTICE_FACTOR/tap_quant[i]))
543 k = LATTICE_FACTOR/tap_quant[i];
544 if (-k > (LATTICE_FACTOR/tap_quant[i]))
545 k = -(LATTICE_FACTOR/tap_quant[i]);
546
547 out[i] = k;
548 k *= tap_quant[i];
549
550#if 1
551 x_ptr = &(window[step]);
552 state_ptr = &(state[0]);
553 j = window_entries - step;
554 for (;j>0;j--,x_ptr++,state_ptr++)
555 {
556 int x_value = *x_ptr;
557 int state_value = *state_ptr;
558 *x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT);
559 *state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT);
560 }
561#else
562 for (j=0; j <= (window_entries - step); j++)
563 {
564 int stepval = window[step+j];
565 int stateval=state[j];
566 window[step+j] += shift_down(k * stateval, LATTICE_SHIFT);
567 state[j] += shift_down(k * stepval, LATTICE_SHIFT);
568 }
569#endif
570 }
571
572 av_free(state);
573 return 0;
574}
575
576static inline int code_samplerate(int samplerate)
577{
578 switch (samplerate)
579 {
580 case 44100: return 0;
581 case 22050: return 1;
582 case 11025: return 2;
583 case 96000: return 3;
584 case 48000: return 4;
585 case 32000: return 5;
586 case 24000: return 6;
587 case 16000: return 7;
588 case 8000: return 8;
589 }
590 return AVERROR(EINVAL);
591}
592
593static av_cold int sonic_encode_init(AVCodecContext *avctx)
594{
595 SonicContext *s = avctx->priv_data;
596 PutBitContext pb;
597 int i;
598
599 s->version = 2;
600
601 if (avctx->channels > MAX_CHANNELS)
602 {
603 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
604 return AVERROR(EINVAL); /* only stereo or mono for now */
605 }
606
607 if (avctx->channels == 2)
608 s->decorrelation = MID_SIDE;
609 else
610 s->decorrelation = 3;
611
612 if (avctx->codec->id == AV_CODEC_ID_SONIC_LS)
613 {
614 s->lossless = 1;
615 s->num_taps = 32;
616 s->downsampling = 1;
617 s->quantization = 0.0;
618 }
619 else
620 {
621 s->num_taps = 128;
622 s->downsampling = 2;
623 s->quantization = 1.0;
624 }
625
626 // max tap 2048
627 if (s->num_taps < 32 || s->num_taps > 1024 || s->num_taps % 32) {
628 av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n");
629 return AVERROR_INVALIDDATA;
630 }
631
632 // generate taps
633 s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant));
634 if (!s->tap_quant)
635 return AVERROR(ENOMEM);
636
637 for (i = 0; i < s->num_taps; i++)
638 s->tap_quant[i] = ff_sqrt(i+1);
639
640 s->channels = avctx->channels;
641 s->samplerate = avctx->sample_rate;
642
643 s->block_align = 2048LL*s->samplerate/(44100*s->downsampling);
644 s->frame_size = s->channels*s->block_align*s->downsampling;
645
646 s->tail_size = s->num_taps*s->channels;
647 s->tail = av_calloc(s->tail_size, sizeof(*s->tail));
648 if (!s->tail)
649 return AVERROR(ENOMEM);
650
651 s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k) );
652 if (!s->predictor_k)
653 return AVERROR(ENOMEM);
654
655 for (i = 0; i < s->channels; i++)
656 {
657 s->coded_samples[i] = av_calloc(s->block_align, sizeof(**s->coded_samples));
658 if (!s->coded_samples[i])
659 return AVERROR(ENOMEM);
660 }
661
662 s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples));
663
664 s->window_size = ((2*s->tail_size)+s->frame_size);
665 s->window = av_calloc(s->window_size, sizeof(*s->window));
666 if (!s->window || !s->int_samples)
667 return AVERROR(ENOMEM);
668
669 avctx->extradata = av_mallocz(16);
670 if (!avctx->extradata)
671 return AVERROR(ENOMEM);
672 init_put_bits(&pb, avctx->extradata, 16*8);
673
674 put_bits(&pb, 2, s->version); // version
675 if (s->version >= 1)
676 {
677 if (s->version >= 2) {
678 put_bits(&pb, 8, s->version);
679 put_bits(&pb, 8, s->minor_version);
680 }
681 put_bits(&pb, 2, s->channels);
682 put_bits(&pb, 4, code_samplerate(s->samplerate));
683 }
684 put_bits(&pb, 1, s->lossless);
685 if (!s->lossless)
686 put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision
687 put_bits(&pb, 2, s->decorrelation);
688 put_bits(&pb, 2, s->downsampling);
689 put_bits(&pb, 5, (s->num_taps >> 5)-1); // 32..1024
690 put_bits(&pb, 1, 0); // XXX FIXME: no custom tap quant table
691
692 flush_put_bits(&pb);
693 avctx->extradata_size = put_bits_count(&pb)/8;
694
695 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d.%d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
696 s->version, s->minor_version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
697
698 avctx->frame_size = s->block_align*s->downsampling;
699
700 return 0;
701}
702
703static av_cold int sonic_encode_close(AVCodecContext *avctx)
704{
705 SonicContext *s = avctx->priv_data;
706 int i;
707
708 for (i = 0; i < s->channels; i++)
709 av_freep(&s->coded_samples[i]);
710
711 av_freep(&s->predictor_k);
712 av_freep(&s->tail);
713 av_freep(&s->tap_quant);
714 av_freep(&s->window);
715 av_freep(&s->int_samples);
716
717 return 0;
718}
719
720static int sonic_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
721 const AVFrame *frame, int *got_packet_ptr)
722{
723 SonicContext *s = avctx->priv_data;
724 RangeCoder c;
725 int i, j, ch, quant = 0, x = 0;
726 int ret;
727 const short *samples = (const int16_t*)frame->data[0];
728 uint8_t state[32];
729
730 if ((ret = ff_alloc_packet2(avctx, avpkt, s->frame_size * 5 + 1000, 0)) < 0)
731 return ret;
732
733 ff_init_range_encoder(&c, avpkt->data, avpkt->size);
734 ff_build_rac_states(&c, 0.05*(1LL<<32), 256-8);
735 memset(state, 128, sizeof(state));
736
737 // short -> internal
738 for (i = 0; i < s->frame_size; i++)
739 s->int_samples[i] = samples[i];
740
741 if (!s->lossless)
742 for (i = 0; i < s->frame_size; i++)
743 s->int_samples[i] = s->int_samples[i] << SAMPLE_SHIFT;
744
745 switch(s->decorrelation)
746 {
747 case MID_SIDE:
748 for (i = 0; i < s->frame_size; i += s->channels)
749 {
750 s->int_samples[i] += s->int_samples[i+1];
751 s->int_samples[i+1] -= shift(s->int_samples[i], 1);
752 }
753 break;
754 case LEFT_SIDE:
755 for (i = 0; i < s->frame_size; i += s->channels)
756 s->int_samples[i+1] -= s->int_samples[i];
757 break;
758 case RIGHT_SIDE:
759 for (i = 0; i < s->frame_size; i += s->channels)
760 s->int_samples[i] -= s->int_samples[i+1];
761 break;
762 }
763
764 memset(s->window, 0, 4* s->window_size);
765
766 for (i = 0; i < s->tail_size; i++)
767 s->window[x++] = s->tail[i];
768
769 for (i = 0; i < s->frame_size; i++)
770 s->window[x++] = s->int_samples[i];
771
772 for (i = 0; i < s->tail_size; i++)
773 s->window[x++] = 0;
774
775 for (i = 0; i < s->tail_size; i++)
776 s->tail[i] = s->int_samples[s->frame_size - s->tail_size + i];
777
778 // generate taps
779 ret = modified_levinson_durbin(s->window, s->window_size,
780 s->predictor_k, s->num_taps, s->channels, s->tap_quant);
781 if (ret < 0)
782 return ret;
783
784 if ((ret = intlist_write(&c, state, s->predictor_k, s->num_taps, 0)) < 0)
785 return ret;
786
787 for (ch = 0; ch < s->channels; ch++)
788 {
789 x = s->tail_size+ch;
790 for (i = 0; i < s->block_align; i++)
791 {
792 int sum = 0;
793 for (j = 0; j < s->downsampling; j++, x += s->channels)
794 sum += s->window[x];
795 s->coded_samples[ch][i] = sum;
796 }
797 }
798
799 // simple rate control code
800 if (!s->lossless)
801 {
802 double energy1 = 0.0, energy2 = 0.0;
803 for (ch = 0; ch < s->channels; ch++)
804 {
805 for (i = 0; i < s->block_align; i++)
806 {
807 double sample = s->coded_samples[ch][i];
808 energy2 += sample*sample;
809 energy1 += fabs(sample);
810 }
811 }
812
813 energy2 = sqrt(energy2/(s->channels*s->block_align));
814 energy1 = M_SQRT2*energy1/(s->channels*s->block_align);
815
816 // increase bitrate when samples are like a gaussian distribution
817 // reduce bitrate when samples are like a two-tailed exponential distribution
818
819 if (energy2 > energy1)
820 energy2 += (energy2-energy1)*RATE_VARIATION;
821
822 quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR);
823// av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2);
824
825 quant = av_clip(quant, 1, 65534);
826
827 put_symbol(&c, state, quant, 0, NULL, NULL);
828
829 quant *= SAMPLE_FACTOR;
830 }
831
832 // write out coded samples
833 for (ch = 0; ch < s->channels; ch++)
834 {
835 if (!s->lossless)
836 for (i = 0; i < s->block_align; i++)
837 s->coded_samples[ch][i] = ROUNDED_DIV(s->coded_samples[ch][i], quant);
838
839 if ((ret = intlist_write(&c, state, s->coded_samples[ch], s->block_align, 1)) < 0)
840 return ret;
841 }
842
843// av_log(avctx, AV_LOG_DEBUG, "used bytes: %d\n", (put_bits_count(&pb)+7)/8);
844
845 avpkt->size = ff_rac_terminate(&c);
846 *got_packet_ptr = 1;
847 return 0;
848
849}
850#endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */
851
852#if CONFIG_SONIC_DECODER
853static const int samplerate_table[] =
854 { 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 };
855
856static av_cold int sonic_decode_init(AVCodecContext *avctx)
857{
858 SonicContext *s = avctx->priv_data;
859 GetBitContext gb;
860 int i;
861 int ret;
862
863 s->channels = avctx->channels;
864 s->samplerate = avctx->sample_rate;
865
866 if (!avctx->extradata)
867 {
868 av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n");
869 return AVERROR_INVALIDDATA;
870 }
871
872 ret = init_get_bits8(&gb, avctx->extradata, avctx->extradata_size);
873 if (ret < 0)
874 return ret;
875
876 s->version = get_bits(&gb, 2);
877 if (s->version >= 2) {
878 s->version = get_bits(&gb, 8);
879 s->minor_version = get_bits(&gb, 8);
880 }
881 if (s->version != 2)
882 {
883 av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n");
884 return AVERROR_INVALIDDATA;
885 }
886
887 if (s->version >= 1)
888 {
889 int sample_rate_index;
890 s->channels = get_bits(&gb, 2);
891 sample_rate_index = get_bits(&gb, 4);
892 if (sample_rate_index >= FF_ARRAY_ELEMS(samplerate_table)) {
893 av_log(avctx, AV_LOG_ERROR, "Invalid sample_rate_index %d\n", sample_rate_index);
894 return AVERROR_INVALIDDATA;
895 }
896 s->samplerate = samplerate_table[sample_rate_index];
897 av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n",
898 s->channels, s->samplerate);
899 }
900
901 if (s->channels > MAX_CHANNELS || s->channels < 1)
902 {
903 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n");
904 return AVERROR_INVALIDDATA;
905 }
906 avctx->channels = s->channels;
907
908 s->lossless = get_bits1(&gb);
909 if (!s->lossless)
910 skip_bits(&gb, 3); // XXX FIXME
911 s->decorrelation = get_bits(&gb, 2);
912 if (s->decorrelation != 3 && s->channels != 2) {
913 av_log(avctx, AV_LOG_ERROR, "invalid decorrelation %d\n", s->decorrelation);
914 return AVERROR_INVALIDDATA;
915 }
916
917 s->downsampling = get_bits(&gb, 2);
918 if (!s->downsampling) {
919 av_log(avctx, AV_LOG_ERROR, "invalid downsampling value\n");
920 return AVERROR_INVALIDDATA;
921 }
922
923 s->num_taps = (get_bits(&gb, 5)+1)<<5;
924 if (get_bits1(&gb)) // XXX FIXME
925 av_log(avctx, AV_LOG_INFO, "Custom quant table\n");
926
927 s->block_align = 2048LL*s->samplerate/(44100*s->downsampling);
928 s->frame_size = s->channels*s->block_align*s->downsampling;
929// avctx->frame_size = s->block_align;
930
931 if (s->num_taps * s->channels > s->frame_size) {
932 av_log(avctx, AV_LOG_ERROR,
933 "number of taps times channels (%d * %d) larger than frame size %d\n",
934 s->num_taps, s->channels, s->frame_size);
935 return AVERROR_INVALIDDATA;
936 }
937
938 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d.%d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n",
939 s->version, s->minor_version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling);
940
941 // generate taps
942 s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant));
943 if (!s->tap_quant)
944 return AVERROR(ENOMEM);
945
946 for (i = 0; i < s->num_taps; i++)
947 s->tap_quant[i] = ff_sqrt(i+1);
948
949 s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k));
950
951 for (i = 0; i < s->channels; i++)
952 {
953 s->predictor_state[i] = av_calloc(s->num_taps, sizeof(**s->predictor_state));
954 if (!s->predictor_state[i])
955 return AVERROR(ENOMEM);
956 }
957
958 for (i = 0; i < s->channels; i++)
959 {
960 s->coded_samples[i] = av_calloc(s->block_align, sizeof(**s->coded_samples));
961 if (!s->coded_samples[i])
962 return AVERROR(ENOMEM);
963 }
964 s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples));
965 if (!s->int_samples)
966 return AVERROR(ENOMEM);
967
968 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
969 return 0;
970}
971
972static av_cold int sonic_decode_close(AVCodecContext *avctx)
973{
974 SonicContext *s = avctx->priv_data;
975 int i;
976
977 av_freep(&s->int_samples);
978 av_freep(&s->tap_quant);
979 av_freep(&s->predictor_k);
980
981 for (i = 0; i < s->channels; i++)
982 {
983 av_freep(&s->predictor_state[i]);
984 av_freep(&s->coded_samples[i]);
985 }
986
987 return 0;
988}
989
990static int sonic_decode_frame(AVCodecContext *avctx,
991 void *data, int *got_frame_ptr,
992 AVPacket *avpkt)
993{
994 const uint8_t *buf = avpkt->data;
995 int buf_size = avpkt->size;
996 SonicContext *s = avctx->priv_data;
997 RangeCoder c;
998 uint8_t state[32];
999 int i, quant, ch, j, ret;
1000 int16_t *samples;
1001 AVFrame *frame = data;
1002
1003 if (buf_size == 0) return 0;
1004
1005 frame->nb_samples = s->frame_size / avctx->channels;
1006 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1007 return ret;
1008 samples = (int16_t *)frame->data[0];
1009
1010// av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size);
1011
1012 memset(state, 128, sizeof(state));
1013 ff_init_range_decoder(&c, buf, buf_size);
1014 ff_build_rac_states(&c, 0.05*(1LL<<32), 256-8);
1015
1016 intlist_read(&c, state, s->predictor_k, s->num_taps, 0);
1017
1018 // dequantize
1019 for (i = 0; i < s->num_taps; i++)
1020 s->predictor_k[i] *= s->tap_quant[i];
1021
1022 if (s->lossless)
1023 quant = 1;
1024 else
1025 quant = get_symbol(&c, state, 0) * SAMPLE_FACTOR;
1026
1027// av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant);
1028
1029 for (ch = 0; ch < s->channels; ch++)
1030 {
1031 int x = ch;
1032
1033 predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps);
1034
1035 intlist_read(&c, state, s->coded_samples[ch], s->block_align, 1);
1036
1037 for (i = 0; i < s->block_align; i++)
1038 {
1039 for (j = 0; j < s->downsampling - 1; j++)
1040 {
1041 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0);
1042 x += s->channels;
1043 }
1044
1045 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * quant);
1046 x += s->channels;
1047 }
1048
1049 for (i = 0; i < s->num_taps; i++)
1050 s->predictor_state[ch][i] = s->int_samples[s->frame_size - s->channels + ch - i*s->channels];
1051 }
1052
1053 switch(s->decorrelation)
1054 {
1055 case MID_SIDE:
1056 for (i = 0; i < s->frame_size; i += s->channels)
1057 {
1058 s->int_samples[i+1] += shift(s->int_samples[i], 1);
1059 s->int_samples[i] -= s->int_samples[i+1];
1060 }
1061 break;
1062 case LEFT_SIDE:
1063 for (i = 0; i < s->frame_size; i += s->channels)
1064 s->int_samples[i+1] += s->int_samples[i];
1065 break;
1066 case RIGHT_SIDE:
1067 for (i = 0; i < s->frame_size; i += s->channels)
1068 s->int_samples[i] += s->int_samples[i+1];
1069 break;
1070 }
1071
1072 if (!s->lossless)
1073 for (i = 0; i < s->frame_size; i++)
1074 s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT);
1075
1076 // internal -> short
1077 for (i = 0; i < s->frame_size; i++)
1078 samples[i] = av_clip_int16(s->int_samples[i]);
1079
1080 *got_frame_ptr = 1;
1081
1082 return buf_size;
1083}
1084
1085AVCodec ff_sonic_decoder = {
1086 .name = "sonic",
1087 .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
1088 .type = AVMEDIA_TYPE_AUDIO,
1089 .id = AV_CODEC_ID_SONIC,
1090 .priv_data_size = sizeof(SonicContext),
1091 .init = sonic_decode_init,
1092 .close = sonic_decode_close,
1093 .decode = sonic_decode_frame,
1094 .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_EXPERIMENTAL,
1095};
1096#endif /* CONFIG_SONIC_DECODER */
1097
1098#if CONFIG_SONIC_ENCODER
1099AVCodec ff_sonic_encoder = {
1100 .name = "sonic",
1101 .long_name = NULL_IF_CONFIG_SMALL("Sonic"),
1102 .type = AVMEDIA_TYPE_AUDIO,
1103 .id = AV_CODEC_ID_SONIC,
1104 .priv_data_size = sizeof(SonicContext),
1105 .init = sonic_encode_init,
1106 .encode2 = sonic_encode_frame,
1107 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
1108 .capabilities = AV_CODEC_CAP_EXPERIMENTAL,
1109 .close = sonic_encode_close,
1110};
1111#endif
1112
1113#if CONFIG_SONIC_LS_ENCODER
1114AVCodec ff_sonic_ls_encoder = {
1115 .name = "sonicls",
1116 .long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"),
1117 .type = AVMEDIA_TYPE_AUDIO,
1118 .id = AV_CODEC_ID_SONIC_LS,
1119 .priv_data_size = sizeof(SonicContext),
1120 .init = sonic_encode_init,
1121 .encode2 = sonic_encode_frame,
1122 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE },
1123 .capabilities = AV_CODEC_CAP_EXPERIMENTAL,
1124 .close = sonic_encode_close,
1125};
1126#endif
1127
generated by cgit at 2024-05-17 03:23:08 (GMT)