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 22#include "config_components.h" 23 24#include "avcodec.h" 25#include "codec_internal.h" 26#include "encode.h" 27#include "get_bits.h" 28#include "golomb.h" 29#include "internal.h" 30#include "put_golomb.h" 31#include "rangecoder.h" 32 33 34/** 35 * @file 36 * Simple free lossless/lossy audio codec 37 * Based on Paul Francis Harrison's Bonk (http://www.logarithmic.net/pfh/bonk) 38 * Written and designed by Alex Beregszaszi 39 * 40 * TODO: 41 * - CABAC put/get_symbol 42 * - independent quantizer for channels 43 * - >2 channels support 44 * - more decorrelation types 45 * - more tap_quant tests 46 * - selectable intlist writers/readers (bonk-style, golomb, cabac) 47 */ 48 49#define MAX_CHANNELS 2 50 51#define MID_SIDE 0 52#define LEFT_SIDE 1 53#define RIGHT_SIDE 2 54 55typedef struct SonicContext { 56 int version; 57 int minor_version; 58 int lossless, decorrelation; 59 60 int num_taps, downsampling; 61 double quantization; 62 63 int channels, samplerate, block_align, frame_size; 64 65 int *tap_quant; 66 int *int_samples; 67 int *coded_samples[MAX_CHANNELS]; 68 69 // for encoding 70 int *tail; 71 int tail_size; 72 int *window; 73 int window_size; 74 75 // for decoding 76 int *predictor_k; 77 int *predictor_state[MAX_CHANNELS]; 78} SonicContext; 79 80#define LATTICE_SHIFT 10 81#define SAMPLE_SHIFT 4 82#define LATTICE_FACTOR (1 << LATTICE_SHIFT) 83#define SAMPLE_FACTOR (1 << SAMPLE_SHIFT) 84 85#define BASE_QUANT 0.6 86#define RATE_VARIATION 3.0 87 88static inline int shift(int a,int b) 89{ 90 return (a+(1<<(b-1))) >> b; 91} 92 93static inline int shift_down(int a,int b) 94{ 95 return (a>>b)+(a<0); 96} 97 98static 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]){ 99 int i; 100 101#define put_rac(C,S,B) \ 102do{\ 103 if(rc_stat){\ 104 rc_stat[*(S)][B]++;\ 105 rc_stat2[(S)-state][B]++;\ 106 }\ 107 put_rac(C,S,B);\ 108}while(0) 109 110 if(v){ 111 const int a= FFABS(v); 112 const int e= av_log2(a); 113 put_rac(c, state+0, 0); 114 if(e<=9){ 115 for(i=0; i<e; i++){ 116 put_rac(c, state+1+i, 1); //1..10 117 } 118 put_rac(c, state+1+i, 0); 119 120 for(i=e-1; i>=0; i--){ 121 put_rac(c, state+22+i, (a>>i)&1); //22..31 122 } 123 124 if(is_signed) 125 put_rac(c, state+11 + e, v < 0); //11..21 126 }else{ 127 for(i=0; i<e; i++){ 128 put_rac(c, state+1+FFMIN(i,9), 1); //1..10 129 } 130 put_rac(c, state+1+9, 0); 131 132 for(i=e-1; i>=0; i--){ 133 put_rac(c, state+22+FFMIN(i,9), (a>>i)&1); //22..31 134 } 135 136 if(is_signed) 137 put_rac(c, state+11 + 10, v < 0); //11..21 138 } 139 }else{ 140 put_rac(c, state+0, 1); 141 } 142#undef put_rac 143} 144 145static inline av_flatten int get_symbol(RangeCoder *c, uint8_t *state, int is_signed){ 146 if(get_rac(c, state+0)) 147 return 0; 148 else{ 149 int i, e; 150 unsigned a; 151 e= 0; 152 while(get_rac(c, state+1 + FFMIN(e,9))){ //1..10 153 e++; 154 if (e > 31) 155 return AVERROR_INVALIDDATA; 156 } 157 158 a= 1; 159 for(i=e-1; i>=0; i--){ 160 a += a + get_rac(c, state+22 + FFMIN(i,9)); //22..31 161 } 162 163 e= -(is_signed && get_rac(c, state+11 + FFMIN(e, 10))); //11..21 164 return (a^e)-e; 165 } 166} 167 168#if 1 169static inline int intlist_write(RangeCoder *c, uint8_t *state, int *buf, int entries, int base_2_part) 170{ 171 int i; 172 173 for (i = 0; i < entries; i++) 174 put_symbol(c, state, buf[i], 1, NULL, NULL); 175 176 return 1; 177} 178 179static inline int intlist_read(RangeCoder *c, uint8_t *state, int *buf, int entries, int base_2_part) 180{ 181 int i; 182 183 for (i = 0; i < entries; i++) 184 buf[i] = get_symbol(c, state, 1); 185 186 return 1; 187} 188#elif 1 189static inline int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part) 190{ 191 int i; 192 193 for (i = 0; i < entries; i++) 194 set_se_golomb(pb, buf[i]); 195 196 return 1; 197} 198 199static inline int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part) 200{ 201 int i; 202 203 for (i = 0; i < entries; i++) 204 buf[i] = get_se_golomb(gb); 205 206 return 1; 207} 208 209#else 210 211#define ADAPT_LEVEL 8 212 213static int bits_to_store(uint64_t x) 214{ 215 int res = 0; 216 217 while(x) 218 { 219 res++; 220 x >>= 1; 221 } 222 return res; 223} 224 225static void write_uint_max(PutBitContext *pb, unsigned int value, unsigned int max) 226{ 227 int i, bits; 228 229 if (!max) 230 return; 231 232 bits = bits_to_store(max); 233 234 for (i = 0; i < bits-1; i++) 235 put_bits(pb, 1, value & (1 << i)); 236 237 if ( (value | (1 << (bits-1))) <= max) 238 put_bits(pb, 1, value & (1 << (bits-1))); 239} 240 241static unsigned int read_uint_max(GetBitContext *gb, int max) 242{ 243 int i, bits, value = 0; 244 245 if (!max) 246 return 0; 247 248 bits = bits_to_store(max); 249 250 for (i = 0; i < bits-1; i++) 251 if (get_bits1(gb)) 252 value += 1 << i; 253 254 if ( (value | (1<<(bits-1))) <= max) 255 if (get_bits1(gb)) 256 value += 1 << (bits-1); 257 258 return value; 259} 260 261static int intlist_write(PutBitContext *pb, int *buf, int entries, int base_2_part) 262{ 263 int i, j, x = 0, low_bits = 0, max = 0; 264 int step = 256, pos = 0, dominant = 0, any = 0; 265 int *copy, *bits; 266 267 copy = av_calloc(entries, sizeof(*copy)); 268 if (!copy) 269 return AVERROR(ENOMEM); 270 271 if (base_2_part) 272 { 273 int energy = 0; 274 275 for (i = 0; i < entries; i++) 276 energy += abs(buf[i]); 277 278 low_bits = bits_to_store(energy / (entries * 2)); 279 if (low_bits > 15) 280 low_bits = 15; 281 282 put_bits(pb, 4, low_bits); 283 } 284 285 for (i = 0; i < entries; i++) 286 { 287 put_bits(pb, low_bits, abs(buf[i])); 288 copy[i] = abs(buf[i]) >> low_bits; 289 if (copy[i] > max) 290 max = abs(copy[i]); 291 } 292 293 bits = av_calloc(entries*max, sizeof(*bits)); 294 if (!bits) 295 { 296 av_free(copy); 297 return AVERROR(ENOMEM); 298 } 299 300 for (i = 0; i <= max; i++) 301 { 302 for (j = 0; j < entries; j++) 303 if (copy[j] >= i) 304 bits[x++] = copy[j] > i; 305 } 306 307 // store bitstream 308 while (pos < x) 309 { 310 int steplet = step >> 8; 311 312 if (pos + steplet > x) 313 steplet = x - pos; 314 315 for (i = 0; i < steplet; i++) 316 if (bits[i+pos] != dominant) 317 any = 1; 318 319 put_bits(pb, 1, any); 320 321 if (!any) 322 { 323 pos += steplet; 324 step += step / ADAPT_LEVEL; 325 } 326 else 327 { 328 int interloper = 0; 329 330 while (((pos + interloper) < x) && (bits[pos + interloper] == dominant)) 331 interloper++; 332 333 // note change 334 write_uint_max(pb, interloper, (step >> 8) - 1); 335 336 pos += interloper + 1; 337 step -= step / ADAPT_LEVEL; 338 } 339 340 if (step < 256) 341 { 342 step = 65536 / step; 343 dominant = !dominant; 344 } 345 } 346 347 // store signs 348 for (i = 0; i < entries; i++) 349 if (buf[i]) 350 put_bits(pb, 1, buf[i] < 0); 351 352 av_free(bits); 353 av_free(copy); 354 355 return 0; 356} 357 358static int intlist_read(GetBitContext *gb, int *buf, int entries, int base_2_part) 359{ 360 int i, low_bits = 0, x = 0; 361 int n_zeros = 0, step = 256, dominant = 0; 362 int pos = 0, level = 0; 363 int *bits = av_calloc(entries, sizeof(*bits)); 364 365 if (!bits) 366 return AVERROR(ENOMEM); 367 368 if (base_2_part) 369 { 370 low_bits = get_bits(gb, 4); 371 372 if (low_bits) 373 for (i = 0; i < entries; i++) 374 buf[i] = get_bits(gb, low_bits); 375 } 376 377// av_log(NULL, AV_LOG_INFO, "entries: %d, low bits: %d\n", entries, low_bits); 378 379 while (n_zeros < entries) 380 { 381 int steplet = step >> 8; 382 383 if (!get_bits1(gb)) 384 { 385 for (i = 0; i < steplet; i++) 386 bits[x++] = dominant; 387 388 if (!dominant) 389 n_zeros += steplet; 390 391 step += step / ADAPT_LEVEL; 392 } 393 else 394 { 395 int actual_run = read_uint_max(gb, steplet-1); 396 397// av_log(NULL, AV_LOG_INFO, "actual run: %d\n", actual_run); 398 399 for (i = 0; i < actual_run; i++) 400 bits[x++] = dominant; 401 402 bits[x++] = !dominant; 403 404 if (!dominant) 405 n_zeros += actual_run; 406 else 407 n_zeros++; 408 409 step -= step / ADAPT_LEVEL; 410 } 411 412 if (step < 256) 413 { 414 step = 65536 / step; 415 dominant = !dominant; 416 } 417 } 418 419 // reconstruct unsigned values 420 n_zeros = 0; 421 for (i = 0; n_zeros < entries; i++) 422 { 423 while(1) 424 { 425 if (pos >= entries) 426 { 427 pos = 0; 428 level += 1 << low_bits; 429 } 430 431 if (buf[pos] >= level) 432 break; 433 434 pos++; 435 } 436 437 if (bits[i]) 438 buf[pos] += 1 << low_bits; 439 else 440 n_zeros++; 441 442 pos++; 443 } 444 av_free(bits); 445 446 // read signs 447 for (i = 0; i < entries; i++) 448 if (buf[i] && get_bits1(gb)) 449 buf[i] = -buf[i]; 450 451// av_log(NULL, AV_LOG_INFO, "zeros: %d pos: %d\n", n_zeros, pos); 452 453 return 0; 454} 455#endif 456 457static void predictor_init_state(int *k, int *state, int order) 458{ 459 int i; 460 461 for (i = order-2; i >= 0; i--) 462 { 463 int j, p, x = state[i]; 464 465 for (j = 0, p = i+1; p < order; j++,p++) 466 { 467 int tmp = x + shift_down(k[j] * (unsigned)state[p], LATTICE_SHIFT); 468 state[p] += shift_down(k[j]* (unsigned)x, LATTICE_SHIFT); 469 x = tmp; 470 } 471 } 472} 473 474static int predictor_calc_error(int *k, int *state, int order, int error) 475{ 476 int i, x = error - (unsigned)shift_down(k[order-1] * (unsigned)state[order-1], LATTICE_SHIFT); 477 478#if 1 479 int *k_ptr = &(k[order-2]), 480 *state_ptr = &(state[order-2]); 481 for (i = order-2; i >= 0; i--, k_ptr--, state_ptr--) 482 { 483 int k_value = *k_ptr, state_value = *state_ptr; 484 x -= (unsigned)shift_down(k_value * (unsigned)state_value, LATTICE_SHIFT); 485 state_ptr[1] = state_value + shift_down(k_value * (unsigned)x, LATTICE_SHIFT); 486 } 487#else 488 for (i = order-2; i >= 0; i--) 489 { 490 x -= (unsigned)shift_down(k[i] * state[i], LATTICE_SHIFT); 491 state[i+1] = state[i] + shift_down(k[i] * x, LATTICE_SHIFT); 492 } 493#endif 494 495 // don't drift too far, to avoid overflows 496 if (x > (SAMPLE_FACTOR<<16)) x = (SAMPLE_FACTOR<<16); 497 if (x < -(SAMPLE_FACTOR<<16)) x = -(SAMPLE_FACTOR<<16); 498 499 state[0] = x; 500 501 return x; 502} 503 504#if CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER 505// Heavily modified Levinson-Durbin algorithm which 506// copes better with quantization, and calculates the 507// actual whitened result as it goes. 508 509static void modified_levinson_durbin(int *window, int window_entries, 510 int *out, int out_entries, int channels, int *tap_quant) 511{ 512 int i; 513 int *state = window + window_entries; 514 515 memcpy(state, window, window_entries * sizeof(*state)); 516 517 for (i = 0; i < out_entries; i++) 518 { 519 int step = (i+1)*channels, k, j; 520 double xx = 0.0, xy = 0.0; 521#if 1 522 int *x_ptr = &(window[step]); 523 int *state_ptr = &(state[0]); 524 j = window_entries - step; 525 for (;j>0;j--,x_ptr++,state_ptr++) 526 { 527 double x_value = *x_ptr; 528 double state_value = *state_ptr; 529 xx += state_value*state_value; 530 xy += x_value*state_value; 531 } 532#else 533 for (j = 0; j <= (window_entries - step); j++); 534 { 535 double stepval = window[step+j]; 536 double stateval = window[j]; 537// xx += (double)window[j]*(double)window[j]; 538// xy += (double)window[step+j]*(double)window[j]; 539 xx += stateval*stateval; 540 xy += stepval*stateval; 541 } 542#endif 543 if (xx == 0.0) 544 k = 0; 545 else 546 k = (int)(floor(-xy/xx * (double)LATTICE_FACTOR / (double)(tap_quant[i]) + 0.5)); 547 548 if (k > (LATTICE_FACTOR/tap_quant[i])) 549 k = LATTICE_FACTOR/tap_quant[i]; 550 if (-k > (LATTICE_FACTOR/tap_quant[i])) 551 k = -(LATTICE_FACTOR/tap_quant[i]); 552 553 out[i] = k; 554 k *= tap_quant[i]; 555 556#if 1 557 x_ptr = &(window[step]); 558 state_ptr = &(state[0]); 559 j = window_entries - step; 560 for (;j>0;j--,x_ptr++,state_ptr++) 561 { 562 int x_value = *x_ptr; 563 int state_value = *state_ptr; 564 *x_ptr = x_value + shift_down(k*state_value,LATTICE_SHIFT); 565 *state_ptr = state_value + shift_down(k*x_value, LATTICE_SHIFT); 566 } 567#else 568 for (j=0; j <= (window_entries - step); j++) 569 { 570 int stepval = window[step+j]; 571 int stateval=state[j]; 572 window[step+j] += shift_down(k * stateval, LATTICE_SHIFT); 573 state[j] += shift_down(k * stepval, LATTICE_SHIFT); 574 } 575#endif 576 } 577} 578 579static inline int code_samplerate(int samplerate) 580{ 581 switch (samplerate) 582 { 583 case 44100: return 0; 584 case 22050: return 1; 585 case 11025: return 2; 586 case 96000: return 3; 587 case 48000: return 4; 588 case 32000: return 5; 589 case 24000: return 6; 590 case 16000: return 7; 591 case 8000: return 8; 592 } 593 return AVERROR(EINVAL); 594} 595 596static av_cold int sonic_encode_init(AVCodecContext *avctx) 597{ 598 SonicContext *s = avctx->priv_data; 599 int *coded_samples; 600 PutBitContext pb; 601 int i; 602 603 s->version = 2; 604 605 if (avctx->ch_layout.nb_channels > MAX_CHANNELS) 606 { 607 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); 608 return AVERROR(EINVAL); /* only stereo or mono for now */ 609 } 610 611 if (avctx->ch_layout.nb_channels == 2) 612 s->decorrelation = MID_SIDE; 613 else 614 s->decorrelation = 3; 615 616 if (avctx->codec->id == AV_CODEC_ID_SONIC_LS) 617 { 618 s->lossless = 1; 619 s->num_taps = 32; 620 s->downsampling = 1; 621 s->quantization = 0.0; 622 } 623 else 624 { 625 s->num_taps = 128; 626 s->downsampling = 2; 627 s->quantization = 1.0; 628 } 629 630 // max tap 2048 631 if (s->num_taps < 32 || s->num_taps > 1024 || s->num_taps % 32) { 632 av_log(avctx, AV_LOG_ERROR, "Invalid number of taps\n"); 633 return AVERROR_INVALIDDATA; 634 } 635 636 // generate taps 637 s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant)); 638 if (!s->tap_quant) 639 return AVERROR(ENOMEM); 640 641 for (i = 0; i < s->num_taps; i++) 642 s->tap_quant[i] = ff_sqrt(i+1); 643 644 s->channels = avctx->ch_layout.nb_channels; 645 s->samplerate = avctx->sample_rate; 646 647 s->block_align = 2048LL*s->samplerate/(44100*s->downsampling); 648 s->frame_size = s->channels*s->block_align*s->downsampling; 649 650 s->tail_size = s->num_taps*s->channels; 651 s->tail = av_calloc(s->tail_size, sizeof(*s->tail)); 652 if (!s->tail) 653 return AVERROR(ENOMEM); 654 655 s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k) ); 656 if (!s->predictor_k) 657 return AVERROR(ENOMEM); 658 659 coded_samples = av_calloc(s->block_align, s->channels * sizeof(**s->coded_samples)); 660 if (!coded_samples) 661 return AVERROR(ENOMEM); 662 for (i = 0; i < s->channels; i++, coded_samples += s->block_align) 663 s->coded_samples[i] = coded_samples; 664 665 s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples)); 666 667 s->window_size = ((2*s->tail_size)+s->frame_size); 668 s->window = av_calloc(s->window_size, 2 * sizeof(*s->window)); 669 if (!s->window || !s->int_samples) 670 return AVERROR(ENOMEM); 671 672 avctx->extradata = av_mallocz(16); 673 if (!avctx->extradata) 674 return AVERROR(ENOMEM); 675 init_put_bits(&pb, avctx->extradata, 16*8); 676 677 put_bits(&pb, 2, s->version); // version 678 if (s->version >= 1) 679 { 680 if (s->version >= 2) { 681 put_bits(&pb, 8, s->version); 682 put_bits(&pb, 8, s->minor_version); 683 } 684 put_bits(&pb, 2, s->channels); 685 put_bits(&pb, 4, code_samplerate(s->samplerate)); 686 } 687 put_bits(&pb, 1, s->lossless); 688 if (!s->lossless) 689 put_bits(&pb, 3, SAMPLE_SHIFT); // XXX FIXME: sample precision 690 put_bits(&pb, 2, s->decorrelation); 691 put_bits(&pb, 2, s->downsampling); 692 put_bits(&pb, 5, (s->num_taps >> 5)-1); // 32..1024 693 put_bits(&pb, 1, 0); // XXX FIXME: no custom tap quant table 694 695 flush_put_bits(&pb); 696 avctx->extradata_size = put_bytes_output(&pb); 697 698 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d.%d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", 699 s->version, s->minor_version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); 700 701 avctx->frame_size = s->block_align*s->downsampling; 702 703 return 0; 704} 705 706static av_cold int sonic_encode_close(AVCodecContext *avctx) 707{ 708 SonicContext *s = avctx->priv_data; 709 710 av_freep(&s->coded_samples[0]); 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_packet(avctx, avpkt, s->frame_size * 5 + 1000)) < 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, s->window_size * sizeof(*s->window)); 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 modified_levinson_durbin(s->window, s->window_size, 780 s->predictor_k, s->num_taps, s->channels, s->tap_quant); 781 782 if ((ret = intlist_write(&c, state, s->predictor_k, s->num_taps, 0)) < 0) 783 return ret; 784 785 for (ch = 0; ch < s->channels; ch++) 786 { 787 x = s->tail_size+ch; 788 for (i = 0; i < s->block_align; i++) 789 { 790 int sum = 0; 791 for (j = 0; j < s->downsampling; j++, x += s->channels) 792 sum += s->window[x]; 793 s->coded_samples[ch][i] = sum; 794 } 795 } 796 797 // simple rate control code 798 if (!s->lossless) 799 { 800 double energy1 = 0.0, energy2 = 0.0; 801 for (ch = 0; ch < s->channels; ch++) 802 { 803 for (i = 0; i < s->block_align; i++) 804 { 805 double sample = s->coded_samples[ch][i]; 806 energy2 += sample*sample; 807 energy1 += fabs(sample); 808 } 809 } 810 811 energy2 = sqrt(energy2/(s->channels*s->block_align)); 812 energy1 = M_SQRT2*energy1/(s->channels*s->block_align); 813 814 // increase bitrate when samples are like a gaussian distribution 815 // reduce bitrate when samples are like a two-tailed exponential distribution 816 817 if (energy2 > energy1) 818 energy2 += (energy2-energy1)*RATE_VARIATION; 819 820 quant = (int)(BASE_QUANT*s->quantization*energy2/SAMPLE_FACTOR); 821// av_log(avctx, AV_LOG_DEBUG, "quant: %d energy: %f / %f\n", quant, energy1, energy2); 822 823 quant = av_clip(quant, 1, 65534); 824 825 put_symbol(&c, state, quant, 0, NULL, NULL); 826 827 quant *= SAMPLE_FACTOR; 828 } 829 830 // write out coded samples 831 for (ch = 0; ch < s->channels; ch++) 832 { 833 if (!s->lossless) 834 for (i = 0; i < s->block_align; i++) 835 s->coded_samples[ch][i] = ROUNDED_DIV(s->coded_samples[ch][i], quant); 836 837 if ((ret = intlist_write(&c, state, s->coded_samples[ch], s->block_align, 1)) < 0) 838 return ret; 839 } 840 841 avpkt->size = ff_rac_terminate(&c, 0); 842 *got_packet_ptr = 1; 843 return 0; 844 845} 846#endif /* CONFIG_SONIC_ENCODER || CONFIG_SONIC_LS_ENCODER */ 847 848#if CONFIG_SONIC_DECODER 849static const int samplerate_table[] = 850 { 44100, 22050, 11025, 96000, 48000, 32000, 24000, 16000, 8000 }; 851 852static av_cold int sonic_decode_init(AVCodecContext *avctx) 853{ 854 SonicContext *s = avctx->priv_data; 855 int *tmp; 856 GetBitContext gb; 857 int i; 858 int ret; 859 860 s->channels = avctx->ch_layout.nb_channels; 861 s->samplerate = avctx->sample_rate; 862 863 if (!avctx->extradata) 864 { 865 av_log(avctx, AV_LOG_ERROR, "No mandatory headers present\n"); 866 return AVERROR_INVALIDDATA; 867 } 868 869 ret = init_get_bits8(&gb, avctx->extradata, avctx->extradata_size); 870 if (ret < 0) 871 return ret; 872 873 s->version = get_bits(&gb, 2); 874 if (s->version >= 2) { 875 s->version = get_bits(&gb, 8); 876 s->minor_version = get_bits(&gb, 8); 877 } 878 if (s->version != 2) 879 { 880 av_log(avctx, AV_LOG_ERROR, "Unsupported Sonic version, please report\n"); 881 return AVERROR_INVALIDDATA; 882 } 883 884 if (s->version >= 1) 885 { 886 int sample_rate_index; 887 s->channels = get_bits(&gb, 2); 888 sample_rate_index = get_bits(&gb, 4); 889 if (sample_rate_index >= FF_ARRAY_ELEMS(samplerate_table)) { 890 av_log(avctx, AV_LOG_ERROR, "Invalid sample_rate_index %d\n", sample_rate_index); 891 return AVERROR_INVALIDDATA; 892 } 893 s->samplerate = samplerate_table[sample_rate_index]; 894 av_log(avctx, AV_LOG_INFO, "Sonicv2 chans: %d samprate: %d\n", 895 s->channels, s->samplerate); 896 } 897 898 if (s->channels > MAX_CHANNELS || s->channels < 1) 899 { 900 av_log(avctx, AV_LOG_ERROR, "Only mono and stereo streams are supported by now\n"); 901 return AVERROR_INVALIDDATA; 902 } 903 av_channel_layout_uninit(&avctx->ch_layout); 904 avctx->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC; 905 avctx->ch_layout.nb_channels = s->channels; 906 907 s->lossless = get_bits1(&gb); 908 if (!s->lossless) 909 skip_bits(&gb, 3); // XXX FIXME 910 s->decorrelation = get_bits(&gb, 2); 911 if (s->decorrelation != 3 && s->channels != 2) { 912 av_log(avctx, AV_LOG_ERROR, "invalid decorrelation %d\n", s->decorrelation); 913 return AVERROR_INVALIDDATA; 914 } 915 916 s->downsampling = get_bits(&gb, 2); 917 if (!s->downsampling) { 918 av_log(avctx, AV_LOG_ERROR, "invalid downsampling value\n"); 919 return AVERROR_INVALIDDATA; 920 } 921 922 s->num_taps = (get_bits(&gb, 5)+1)<<5; 923 if (get_bits1(&gb)) // XXX FIXME 924 av_log(avctx, AV_LOG_INFO, "Custom quant table\n"); 925 926 s->block_align = 2048LL*s->samplerate/(44100*s->downsampling); 927 s->frame_size = s->channels*s->block_align*s->downsampling; 928// avctx->frame_size = s->block_align; 929 930 if (s->num_taps * s->channels > s->frame_size) { 931 av_log(avctx, AV_LOG_ERROR, 932 "number of taps times channels (%d * %d) larger than frame size %d\n", 933 s->num_taps, s->channels, s->frame_size); 934 return AVERROR_INVALIDDATA; 935 } 936 937 av_log(avctx, AV_LOG_INFO, "Sonic: ver: %d.%d ls: %d dr: %d taps: %d block: %d frame: %d downsamp: %d\n", 938 s->version, s->minor_version, s->lossless, s->decorrelation, s->num_taps, s->block_align, s->frame_size, s->downsampling); 939 940 // generate taps 941 s->tap_quant = av_calloc(s->num_taps, sizeof(*s->tap_quant)); 942 if (!s->tap_quant) 943 return AVERROR(ENOMEM); 944 945 for (i = 0; i < s->num_taps; i++) 946 s->tap_quant[i] = ff_sqrt(i+1); 947 948 s->predictor_k = av_calloc(s->num_taps, sizeof(*s->predictor_k)); 949 950 tmp = av_calloc(s->num_taps, s->channels * sizeof(**s->predictor_state)); 951 if (!tmp) 952 return AVERROR(ENOMEM); 953 for (i = 0; i < s->channels; i++, tmp += s->num_taps) 954 s->predictor_state[i] = tmp; 955 956 tmp = av_calloc(s->block_align, s->channels * sizeof(**s->coded_samples)); 957 if (!tmp) 958 return AVERROR(ENOMEM); 959 for (i = 0; i < s->channels; i++, tmp += s->block_align) 960 s->coded_samples[i] = tmp; 961 962 s->int_samples = av_calloc(s->frame_size, sizeof(*s->int_samples)); 963 if (!s->int_samples) 964 return AVERROR(ENOMEM); 965 966 avctx->sample_fmt = AV_SAMPLE_FMT_S16; 967 return 0; 968} 969 970static av_cold int sonic_decode_close(AVCodecContext *avctx) 971{ 972 SonicContext *s = avctx->priv_data; 973 974 av_freep(&s->int_samples); 975 av_freep(&s->tap_quant); 976 av_freep(&s->predictor_k); 977 av_freep(&s->predictor_state[0]); 978 av_freep(&s->coded_samples[0]); 979 980 return 0; 981} 982 983static int sonic_decode_frame(AVCodecContext *avctx, AVFrame *frame, 984 int *got_frame_ptr, AVPacket *avpkt) 985{ 986 const uint8_t *buf = avpkt->data; 987 int buf_size = avpkt->size; 988 SonicContext *s = avctx->priv_data; 989 RangeCoder c; 990 uint8_t state[32]; 991 int i, quant, ch, j, ret; 992 int16_t *samples; 993 994 if (buf_size == 0) return 0; 995 996 frame->nb_samples = s->frame_size / avctx->ch_layout.nb_channels; 997 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) 998 return ret; 999 samples = (int16_t *)frame->data[0]; 1000 1001// av_log(NULL, AV_LOG_INFO, "buf_size: %d\n", buf_size); 1002 1003 memset(state, 128, sizeof(state)); 1004 ff_init_range_decoder(&c, buf, buf_size); 1005 ff_build_rac_states(&c, 0.05*(1LL<<32), 256-8); 1006 1007 intlist_read(&c, state, s->predictor_k, s->num_taps, 0); 1008 1009 // dequantize 1010 for (i = 0; i < s->num_taps; i++) 1011 s->predictor_k[i] *= (unsigned) s->tap_quant[i]; 1012 1013 if (s->lossless) 1014 quant = 1; 1015 else 1016 quant = get_symbol(&c, state, 0) * (unsigned)SAMPLE_FACTOR; 1017 1018// av_log(NULL, AV_LOG_INFO, "quant: %d\n", quant); 1019 1020 for (ch = 0; ch < s->channels; ch++) 1021 { 1022 int x = ch; 1023 1024 if (c.overread > MAX_OVERREAD) 1025 return AVERROR_INVALIDDATA; 1026 1027 predictor_init_state(s->predictor_k, s->predictor_state[ch], s->num_taps); 1028 1029 intlist_read(&c, state, s->coded_samples[ch], s->block_align, 1); 1030 1031 for (i = 0; i < s->block_align; i++) 1032 { 1033 for (j = 0; j < s->downsampling - 1; j++) 1034 { 1035 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, 0); 1036 x += s->channels; 1037 } 1038 1039 s->int_samples[x] = predictor_calc_error(s->predictor_k, s->predictor_state[ch], s->num_taps, s->coded_samples[ch][i] * (unsigned)quant); 1040 x += s->channels; 1041 } 1042 1043 for (i = 0; i < s->num_taps; i++) 1044 s->predictor_state[ch][i] = s->int_samples[s->frame_size - s->channels + ch - i*s->channels]; 1045 } 1046 1047 switch(s->decorrelation) 1048 { 1049 case MID_SIDE: 1050 for (i = 0; i < s->frame_size; i += s->channels) 1051 { 1052 s->int_samples[i+1] += shift(s->int_samples[i], 1); 1053 s->int_samples[i] -= s->int_samples[i+1]; 1054 } 1055 break; 1056 case LEFT_SIDE: 1057 for (i = 0; i < s->frame_size; i += s->channels) 1058 s->int_samples[i+1] += s->int_samples[i]; 1059 break; 1060 case RIGHT_SIDE: 1061 for (i = 0; i < s->frame_size; i += s->channels) 1062 s->int_samples[i] += s->int_samples[i+1]; 1063 break; 1064 } 1065 1066 if (!s->lossless) 1067 for (i = 0; i < s->frame_size; i++) 1068 s->int_samples[i] = shift(s->int_samples[i], SAMPLE_SHIFT); 1069 1070 // internal -> short 1071 for (i = 0; i < s->frame_size; i++) 1072 samples[i] = av_clip_int16(s->int_samples[i]); 1073 1074 *got_frame_ptr = 1; 1075 1076 return buf_size; 1077} 1078 1079const FFCodec ff_sonic_decoder = { 1080 .p.name = "sonic", 1081 .p.long_name = NULL_IF_CONFIG_SMALL("Sonic"), 1082 .p.type = AVMEDIA_TYPE_AUDIO, 1083 .p.id = AV_CODEC_ID_SONIC, 1084 .priv_data_size = sizeof(SonicContext), 1085 .init = sonic_decode_init, 1086 .close = sonic_decode_close, 1087 FF_CODEC_DECODE_CB(sonic_decode_frame), 1088 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_EXPERIMENTAL | AV_CODEC_CAP_CHANNEL_CONF, 1089 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, 1090}; 1091#endif /* CONFIG_SONIC_DECODER */ 1092 1093#if CONFIG_SONIC_ENCODER 1094const FFCodec ff_sonic_encoder = { 1095 .p.name = "sonic", 1096 .p.long_name = NULL_IF_CONFIG_SMALL("Sonic"), 1097 .p.type = AVMEDIA_TYPE_AUDIO, 1098 .p.id = AV_CODEC_ID_SONIC, 1099 .priv_data_size = sizeof(SonicContext), 1100 .init = sonic_encode_init, 1101 FF_CODEC_ENCODE_CB(sonic_encode_frame), 1102 .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, 1103 .p.capabilities = AV_CODEC_CAP_EXPERIMENTAL, 1104 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, 1105 .close = sonic_encode_close, 1106}; 1107#endif 1108 1109#if CONFIG_SONIC_LS_ENCODER 1110const FFCodec ff_sonic_ls_encoder = { 1111 .p.name = "sonicls", 1112 .p.long_name = NULL_IF_CONFIG_SMALL("Sonic lossless"), 1113 .p.type = AVMEDIA_TYPE_AUDIO, 1114 .p.id = AV_CODEC_ID_SONIC_LS, 1115 .priv_data_size = sizeof(SonicContext), 1116 .init = sonic_encode_init, 1117 FF_CODEC_ENCODE_CB(sonic_encode_frame), 1118 .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_NONE }, 1119 .p.capabilities = AV_CODEC_CAP_EXPERIMENTAL, 1120 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, 1121 .close = sonic_encode_close, 1122}; 1123#endif 1124