3 * Copyright (c) 2001, 2002 Fabrice Bellard
5 * This file is part of Libav.
7 * Libav 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.
12 * Libav 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.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
27 #include "libavutil/audioconvert.h"
35 * - test lsf / mpeg25 extensively.
38 #include "mpegaudio.h"
39 #include "mpegaudiodecheader.h"
42 # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
43 # define compute_antialias compute_antialias_float
44 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
45 # define FIXR(x) ((float)(x))
46 # define FIXHR(x) ((float)(x))
47 # define MULH3(x, y, s) ((s)*(y)*(x))
48 # define MULLx(x, y, s) ((y)*(x))
49 # define RENAME(a) a ## _float
51 # define SHR(a,b) ((a)>>(b))
52 # define compute_antialias compute_antialias_integer
53 /* WARNING: only correct for posititive numbers */
54 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
55 # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
56 # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
57 # define MULH3(x, y, s) MULH((s)*(x), y)
58 # define MULLx(x, y, s) MULL(x,y,s)
66 #include "mpegaudiodata.h"
67 #include "mpegaudiodectab.h"
75 static void compute_antialias(MPADecodeContext *s, GranuleDef *g);
76 static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
77 int *dither_state, OUT_INT *samples, int incr);
79 /* vlc structure for decoding layer 3 huffman tables */
80 static VLC huff_vlc[16];
81 static VLC_TYPE huff_vlc_tables[
82 0+128+128+128+130+128+154+166+
83 142+204+190+170+542+460+662+414
85 static const int huff_vlc_tables_sizes[16] = {
86 0, 128, 128, 128, 130, 128, 154, 166,
87 142, 204, 190, 170, 542, 460, 662, 414
89 static VLC huff_quad_vlc[2];
90 static VLC_TYPE huff_quad_vlc_tables[128+16][2];
91 static const int huff_quad_vlc_tables_sizes[2] = {
94 /* computed from band_size_long */
95 static uint16_t band_index_long[9][23];
96 #include "mpegaudio_tablegen.h"
97 /* intensity stereo coef table */
98 static INTFLOAT is_table[2][16];
99 static INTFLOAT is_table_lsf[2][2][16];
100 static int32_t csa_table[8][4];
101 static float csa_table_float[8][4];
102 static INTFLOAT mdct_win[8][36];
104 static int16_t division_tab3[1<<6 ];
105 static int16_t division_tab5[1<<8 ];
106 static int16_t division_tab9[1<<11];
108 static int16_t * const division_tabs[4] = {
109 division_tab3, division_tab5, NULL, division_tab9
112 /* lower 2 bits: modulo 3, higher bits: shift */
113 static uint16_t scale_factor_modshift[64];
114 /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
115 static int32_t scale_factor_mult[15][3];
116 /* mult table for layer 2 group quantization */
118 #define SCALE_GEN(v) \
119 { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
121 static const int32_t scale_factor_mult2[3][3] = {
122 SCALE_GEN(4.0 / 3.0), /* 3 steps */
123 SCALE_GEN(4.0 / 5.0), /* 5 steps */
124 SCALE_GEN(4.0 / 9.0), /* 9 steps */
127 DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256];
130 * Convert region offsets to region sizes and truncate
131 * size to big_values.
133 static void ff_region_offset2size(GranuleDef *g){
135 g->region_size[2] = (576 / 2);
137 k = FFMIN(g->region_size[i], g->big_values);
138 g->region_size[i] = k - j;
143 static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g){
144 if (g->block_type == 2)
145 g->region_size[0] = (36 / 2);
147 if (s->sample_rate_index <= 2)
148 g->region_size[0] = (36 / 2);
149 else if (s->sample_rate_index != 8)
150 g->region_size[0] = (54 / 2);
152 g->region_size[0] = (108 / 2);
154 g->region_size[1] = (576 / 2);
157 static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2){
160 band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
161 /* should not overflow */
162 l = FFMIN(ra1 + ra2 + 2, 22);
164 band_index_long[s->sample_rate_index][l] >> 1;
167 static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g){
168 if (g->block_type == 2) {
169 if (g->switch_point) {
170 /* if switched mode, we handle the 36 first samples as
171 long blocks. For 8000Hz, we handle the 48 first
172 exponents as long blocks (XXX: check this!) */
173 if (s->sample_rate_index <= 2)
175 else if (s->sample_rate_index != 8)
178 g->long_end = 4; /* 8000 Hz */
180 g->short_start = 2 + (s->sample_rate_index != 8);
191 /* layer 1 unscaling */
192 /* n = number of bits of the mantissa minus 1 */
193 static inline int l1_unscale(int n, int mant, int scale_factor)
198 shift = scale_factor_modshift[scale_factor];
201 val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
203 /* NOTE: at this point, 1 <= shift >= 21 + 15 */
204 return (int)((val + (1LL << (shift - 1))) >> shift);
207 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
211 shift = scale_factor_modshift[scale_factor];
215 val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
216 /* NOTE: at this point, 0 <= shift <= 21 */
218 val = (val + (1 << (shift - 1))) >> shift;
222 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
223 static inline int l3_unscale(int value, int exponent)
228 e = table_4_3_exp [4*value + (exponent&3)];
229 m = table_4_3_value[4*value + (exponent&3)];
230 e -= (exponent >> 2);
234 m = (m + (1 << (e-1))) >> e;
239 /* all integer n^(4/3) computation code */
242 #define POW_FRAC_BITS 24
243 #define POW_FRAC_ONE (1 << POW_FRAC_BITS)
244 #define POW_FIX(a) ((int)((a) * POW_FRAC_ONE))
245 #define POW_MULL(a,b) (((int64_t)(a) * (int64_t)(b)) >> POW_FRAC_BITS)
247 static int dev_4_3_coefs[DEV_ORDER];
249 static av_cold void int_pow_init(void)
254 for(i=0;i<DEV_ORDER;i++) {
255 a = POW_MULL(a, POW_FIX(4.0 / 3.0) - i * POW_FIX(1.0)) / (i + 1);
256 dev_4_3_coefs[i] = a;
260 static av_cold int decode_init(AVCodecContext * avctx)
262 MPADecodeContext *s = avctx->priv_data;
267 s->apply_window_mp3 = apply_window_mp3_c;
268 #if HAVE_MMX && CONFIG_FLOAT
269 ff_mpegaudiodec_init_mmx(s);
272 ff_dct_init(&s->dct, 5, DCT_II);
274 if (HAVE_ALTIVEC && CONFIG_FLOAT) ff_mpegaudiodec_init_altivec(s);
276 avctx->sample_fmt= OUT_FMT;
277 s->error_recognition= avctx->error_recognition;
279 if (!init && !avctx->parse_only) {
282 /* scale factors table for layer 1/2 */
285 /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
288 scale_factor_modshift[i] = mod | (shift << 2);
291 /* scale factor multiply for layer 1 */
295 norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
296 scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
297 scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
298 scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
299 av_dlog(avctx, "%d: norm=%x s=%x %x %x\n",
301 scale_factor_mult[i][0],
302 scale_factor_mult[i][1],
303 scale_factor_mult[i][2]);
306 RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
308 /* huffman decode tables */
311 const HuffTable *h = &mpa_huff_tables[i];
313 uint8_t tmp_bits [512];
314 uint16_t tmp_codes[512];
316 memset(tmp_bits , 0, sizeof(tmp_bits ));
317 memset(tmp_codes, 0, sizeof(tmp_codes));
322 for(x=0;x<xsize;x++) {
323 for(y=0;y<xsize;y++){
324 tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
325 tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
330 huff_vlc[i].table = huff_vlc_tables+offset;
331 huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
332 init_vlc(&huff_vlc[i], 7, 512,
333 tmp_bits, 1, 1, tmp_codes, 2, 2,
334 INIT_VLC_USE_NEW_STATIC);
335 offset += huff_vlc_tables_sizes[i];
337 assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
341 huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
342 huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
343 init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
344 mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
345 INIT_VLC_USE_NEW_STATIC);
346 offset += huff_quad_vlc_tables_sizes[i];
348 assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
353 band_index_long[i][j] = k;
354 k += band_size_long[i][j];
356 band_index_long[i][22] = k;
359 /* compute n ^ (4/3) and store it in mantissa/exp format */
362 mpegaudio_tableinit();
364 for (i = 0; i < 4; i++)
365 if (ff_mpa_quant_bits[i] < 0)
366 for (j = 0; j < (1<<(-ff_mpa_quant_bits[i]+1)); j++) {
367 int val1, val2, val3, steps;
369 steps = ff_mpa_quant_steps[i];
374 division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
382 f = tan((double)i * M_PI / 12.0);
383 v = FIXR(f / (1.0 + f));
388 is_table[1][6 - i] = v;
392 is_table[0][i] = is_table[1][i] = 0.0;
399 e = -(j + 1) * ((i + 1) >> 1);
400 f = pow(2.0, e / 4.0);
402 is_table_lsf[j][k ^ 1][i] = FIXR(f);
403 is_table_lsf[j][k][i] = FIXR(1.0);
404 av_dlog(avctx, "is_table_lsf %d %d: %x %x\n",
405 i, j, is_table_lsf[j][0][i], is_table_lsf[j][1][i]);
412 cs = 1.0 / sqrt(1.0 + ci * ci);
414 csa_table[i][0] = FIXHR(cs/4);
415 csa_table[i][1] = FIXHR(ca/4);
416 csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
417 csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
418 csa_table_float[i][0] = cs;
419 csa_table_float[i][1] = ca;
420 csa_table_float[i][2] = ca + cs;
421 csa_table_float[i][3] = ca - cs;
424 /* compute mdct windows */
432 d= sin(M_PI * (i + 0.5) / 36.0);
435 else if(i>=24) d= sin(M_PI * (i - 18 + 0.5) / 12.0);
439 else if(i< 12) d= sin(M_PI * (i - 6 + 0.5) / 12.0);
442 //merge last stage of imdct into the window coefficients
443 d*= 0.5 / cos(M_PI*(2*i + 19)/72);
446 mdct_win[j][i/3] = FIXHR((d / (1<<5)));
448 mdct_win[j][i ] = FIXHR((d / (1<<5)));
452 /* NOTE: we do frequency inversion adter the MDCT by changing
453 the sign of the right window coefs */
456 mdct_win[j + 4][i] = mdct_win[j][i];
457 mdct_win[j + 4][i + 1] = -mdct_win[j][i + 1];
464 if (avctx->codec_id == CODEC_ID_MP3ADU)
471 static inline float round_sample(float *sum)
478 /* signed 16x16 -> 32 multiply add accumulate */
479 #define MACS(rt, ra, rb) rt+=(ra)*(rb)
481 /* signed 16x16 -> 32 multiply */
482 #define MULS(ra, rb) ((ra)*(rb))
484 #define MLSS(rt, ra, rb) rt-=(ra)*(rb)
488 static inline int round_sample(int64_t *sum)
491 sum1 = (int)((*sum) >> OUT_SHIFT);
492 *sum &= (1<<OUT_SHIFT)-1;
493 return av_clip(sum1, OUT_MIN, OUT_MAX);
496 # define MULS(ra, rb) MUL64(ra, rb)
497 # define MACS(rt, ra, rb) MAC64(rt, ra, rb)
498 # define MLSS(rt, ra, rb) MLS64(rt, ra, rb)
501 #define SUM8(op, sum, w, p) \
503 op(sum, (w)[0 * 64], (p)[0 * 64]); \
504 op(sum, (w)[1 * 64], (p)[1 * 64]); \
505 op(sum, (w)[2 * 64], (p)[2 * 64]); \
506 op(sum, (w)[3 * 64], (p)[3 * 64]); \
507 op(sum, (w)[4 * 64], (p)[4 * 64]); \
508 op(sum, (w)[5 * 64], (p)[5 * 64]); \
509 op(sum, (w)[6 * 64], (p)[6 * 64]); \
510 op(sum, (w)[7 * 64], (p)[7 * 64]); \
513 #define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
517 op1(sum1, (w1)[0 * 64], tmp);\
518 op2(sum2, (w2)[0 * 64], tmp);\
520 op1(sum1, (w1)[1 * 64], tmp);\
521 op2(sum2, (w2)[1 * 64], tmp);\
523 op1(sum1, (w1)[2 * 64], tmp);\
524 op2(sum2, (w2)[2 * 64], tmp);\
526 op1(sum1, (w1)[3 * 64], tmp);\
527 op2(sum2, (w2)[3 * 64], tmp);\
529 op1(sum1, (w1)[4 * 64], tmp);\
530 op2(sum2, (w2)[4 * 64], tmp);\
532 op1(sum1, (w1)[5 * 64], tmp);\
533 op2(sum2, (w2)[5 * 64], tmp);\
535 op1(sum1, (w1)[6 * 64], tmp);\
536 op2(sum2, (w2)[6 * 64], tmp);\
538 op1(sum1, (w1)[7 * 64], tmp);\
539 op2(sum2, (w2)[7 * 64], tmp);\
542 void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window)
546 /* max = 18760, max sum over all 16 coefs : 44736 */
549 v = ff_mpa_enwindow[i];
551 v *= 1.0 / (1LL<<(16 + FRAC_BITS));
560 // Needed for avoiding shuffles in ASM implementations
562 for(j=0; j < 16; j++)
563 window[512+16*i+j] = window[64*i+32-j];
566 for(j=0; j < 16; j++)
567 window[512+128+16*i+j] = window[64*i+48-j];
570 static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
571 int *dither_state, OUT_INT *samples, int incr)
573 register const MPA_INT *w, *w2, *p;
582 /* copy to avoid wrap */
583 memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf));
585 samples2 = samples + 31 * incr;
591 SUM8(MACS, sum, w, p);
593 SUM8(MLSS, sum, w + 32, p);
594 *samples = round_sample(&sum);
598 /* we calculate two samples at the same time to avoid one memory
599 access per two sample */
602 p = synth_buf + 16 + j;
603 SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
604 p = synth_buf + 48 - j;
605 SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
607 *samples = round_sample(&sum);
610 *samples2 = round_sample(&sum);
617 SUM8(MLSS, sum, w + 32, p);
618 *samples = round_sample(&sum);
623 /* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
625 /* XXX: optimize by avoiding ring buffer usage */
627 void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
628 MPA_INT *window, int *dither_state,
629 OUT_INT *samples, int incr,
630 INTFLOAT sb_samples[SBLIMIT])
632 register MPA_INT *synth_buf;
635 offset = *synth_buf_offset;
636 synth_buf = synth_buf_ptr + offset;
638 dct32(synth_buf, sb_samples);
639 apply_window_mp3_c(synth_buf, window, dither_state, samples, incr);
641 offset = (offset - 32) & 511;
642 *synth_buf_offset = offset;
646 #define C3 FIXHR(0.86602540378443864676/2)
648 /* 0.5 / cos(pi*(2*i+1)/36) */
649 static const INTFLOAT icos36[9] = {
650 FIXR(0.50190991877167369479),
651 FIXR(0.51763809020504152469), //0
652 FIXR(0.55168895948124587824),
653 FIXR(0.61038729438072803416),
654 FIXR(0.70710678118654752439), //1
655 FIXR(0.87172339781054900991),
656 FIXR(1.18310079157624925896),
657 FIXR(1.93185165257813657349), //2
658 FIXR(5.73685662283492756461),
661 /* 0.5 / cos(pi*(2*i+1)/36) */
662 static const INTFLOAT icos36h[9] = {
663 FIXHR(0.50190991877167369479/2),
664 FIXHR(0.51763809020504152469/2), //0
665 FIXHR(0.55168895948124587824/2),
666 FIXHR(0.61038729438072803416/2),
667 FIXHR(0.70710678118654752439/2), //1
668 FIXHR(0.87172339781054900991/2),
669 FIXHR(1.18310079157624925896/4),
670 FIXHR(1.93185165257813657349/4), //2
671 // FIXHR(5.73685662283492756461),
674 /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
676 static void imdct12(INTFLOAT *out, INTFLOAT *in)
678 INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
681 in1= in[1*3] + in[0*3];
682 in2= in[2*3] + in[1*3];
683 in3= in[3*3] + in[2*3];
684 in4= in[4*3] + in[3*3];
685 in5= in[5*3] + in[4*3];
689 in2= MULH3(in2, C3, 2);
690 in3= MULH3(in3, C3, 4);
693 t2 = MULH3(in1 - in5, icos36h[4], 2);
703 in1 = MULH3(in5 + in3, icos36h[1], 1);
710 in5 = MULH3(in5 - in3, icos36h[7], 2);
718 #define C1 FIXHR(0.98480775301220805936/2)
719 #define C2 FIXHR(0.93969262078590838405/2)
720 #define C3 FIXHR(0.86602540378443864676/2)
721 #define C4 FIXHR(0.76604444311897803520/2)
722 #define C5 FIXHR(0.64278760968653932632/2)
723 #define C6 FIXHR(0.5/2)
724 #define C7 FIXHR(0.34202014332566873304/2)
725 #define C8 FIXHR(0.17364817766693034885/2)
728 /* using Lee like decomposition followed by hand coded 9 points DCT */
729 static void imdct36(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win)
732 INTFLOAT t0, t1, t2, t3, s0, s1, s2, s3;
733 INTFLOAT tmp[18], *tmp1, *in1;
744 t2 = in1[2*4] + in1[2*8] - in1[2*2];
746 t3 = in1[2*0] + SHR(in1[2*6],1);
747 t1 = in1[2*0] - in1[2*6];
748 tmp1[ 6] = t1 - SHR(t2,1);
751 t0 = MULH3(in1[2*2] + in1[2*4] , C2, 2);
752 t1 = MULH3(in1[2*4] - in1[2*8] , -2*C8, 1);
753 t2 = MULH3(in1[2*2] + in1[2*8] , -C4, 2);
755 tmp1[10] = t3 - t0 - t2;
756 tmp1[ 2] = t3 + t0 + t1;
757 tmp1[14] = t3 + t2 - t1;
759 tmp1[ 4] = MULH3(in1[2*5] + in1[2*7] - in1[2*1], -C3, 2);
760 t2 = MULH3(in1[2*1] + in1[2*5], C1, 2);
761 t3 = MULH3(in1[2*5] - in1[2*7], -2*C7, 1);
762 t0 = MULH3(in1[2*3], C3, 2);
764 t1 = MULH3(in1[2*1] + in1[2*7], -C5, 2);
766 tmp1[ 0] = t2 + t3 + t0;
767 tmp1[12] = t2 + t1 - t0;
768 tmp1[ 8] = t3 - t1 - t0;
780 s1 = MULH3(t3 + t2, icos36h[j], 2);
781 s3 = MULLx(t3 - t2, icos36[8 - j], FRAC_BITS);
785 out[(9 + j)*SBLIMIT] = MULH3(t1, win[9 + j], 1) + buf[9 + j];
786 out[(8 - j)*SBLIMIT] = MULH3(t1, win[8 - j], 1) + buf[8 - j];
787 buf[9 + j] = MULH3(t0, win[18 + 9 + j], 1);
788 buf[8 - j] = MULH3(t0, win[18 + 8 - j], 1);
792 out[(9 + 8 - j)*SBLIMIT] = MULH3(t1, win[9 + 8 - j], 1) + buf[9 + 8 - j];
793 out[( j)*SBLIMIT] = MULH3(t1, win[ j], 1) + buf[ j];
794 buf[9 + 8 - j] = MULH3(t0, win[18 + 9 + 8 - j], 1);
795 buf[ + j] = MULH3(t0, win[18 + j], 1);
800 s1 = MULH3(tmp[17], icos36h[4], 2);
803 out[(9 + 4)*SBLIMIT] = MULH3(t1, win[9 + 4], 1) + buf[9 + 4];
804 out[(8 - 4)*SBLIMIT] = MULH3(t1, win[8 - 4], 1) + buf[8 - 4];
805 buf[9 + 4] = MULH3(t0, win[18 + 9 + 4], 1);
806 buf[8 - 4] = MULH3(t0, win[18 + 8 - 4], 1);
809 /* return the number of decoded frames */
810 static int mp_decode_layer1(MPADecodeContext *s)
812 int bound, i, v, n, ch, j, mant;
813 uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
814 uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
816 if (s->mode == MPA_JSTEREO)
817 bound = (s->mode_ext + 1) * 4;
821 /* allocation bits */
822 for(i=0;i<bound;i++) {
823 for(ch=0;ch<s->nb_channels;ch++) {
824 allocation[ch][i] = get_bits(&s->gb, 4);
827 for(i=bound;i<SBLIMIT;i++) {
828 allocation[0][i] = get_bits(&s->gb, 4);
832 for(i=0;i<bound;i++) {
833 for(ch=0;ch<s->nb_channels;ch++) {
834 if (allocation[ch][i])
835 scale_factors[ch][i] = get_bits(&s->gb, 6);
838 for(i=bound;i<SBLIMIT;i++) {
839 if (allocation[0][i]) {
840 scale_factors[0][i] = get_bits(&s->gb, 6);
841 scale_factors[1][i] = get_bits(&s->gb, 6);
845 /* compute samples */
847 for(i=0;i<bound;i++) {
848 for(ch=0;ch<s->nb_channels;ch++) {
849 n = allocation[ch][i];
851 mant = get_bits(&s->gb, n + 1);
852 v = l1_unscale(n, mant, scale_factors[ch][i]);
856 s->sb_samples[ch][j][i] = v;
859 for(i=bound;i<SBLIMIT;i++) {
860 n = allocation[0][i];
862 mant = get_bits(&s->gb, n + 1);
863 v = l1_unscale(n, mant, scale_factors[0][i]);
864 s->sb_samples[0][j][i] = v;
865 v = l1_unscale(n, mant, scale_factors[1][i]);
866 s->sb_samples[1][j][i] = v;
868 s->sb_samples[0][j][i] = 0;
869 s->sb_samples[1][j][i] = 0;
876 static int mp_decode_layer2(MPADecodeContext *s)
878 int sblimit; /* number of used subbands */
879 const unsigned char *alloc_table;
880 int table, bit_alloc_bits, i, j, ch, bound, v;
881 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
882 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
883 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
884 int scale, qindex, bits, steps, k, l, m, b;
886 /* select decoding table */
887 table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
888 s->sample_rate, s->lsf);
889 sblimit = ff_mpa_sblimit_table[table];
890 alloc_table = ff_mpa_alloc_tables[table];
892 if (s->mode == MPA_JSTEREO)
893 bound = (s->mode_ext + 1) * 4;
897 av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
900 if( bound > sblimit ) bound = sblimit;
902 /* parse bit allocation */
904 for(i=0;i<bound;i++) {
905 bit_alloc_bits = alloc_table[j];
906 for(ch=0;ch<s->nb_channels;ch++) {
907 bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
909 j += 1 << bit_alloc_bits;
911 for(i=bound;i<sblimit;i++) {
912 bit_alloc_bits = alloc_table[j];
913 v = get_bits(&s->gb, bit_alloc_bits);
916 j += 1 << bit_alloc_bits;
920 for(i=0;i<sblimit;i++) {
921 for(ch=0;ch<s->nb_channels;ch++) {
922 if (bit_alloc[ch][i])
923 scale_code[ch][i] = get_bits(&s->gb, 2);
928 for(i=0;i<sblimit;i++) {
929 for(ch=0;ch<s->nb_channels;ch++) {
930 if (bit_alloc[ch][i]) {
931 sf = scale_factors[ch][i];
932 switch(scale_code[ch][i]) {
935 sf[0] = get_bits(&s->gb, 6);
936 sf[1] = get_bits(&s->gb, 6);
937 sf[2] = get_bits(&s->gb, 6);
940 sf[0] = get_bits(&s->gb, 6);
945 sf[0] = get_bits(&s->gb, 6);
946 sf[2] = get_bits(&s->gb, 6);
950 sf[0] = get_bits(&s->gb, 6);
951 sf[2] = get_bits(&s->gb, 6);
963 for(i=0;i<bound;i++) {
964 bit_alloc_bits = alloc_table[j];
965 for(ch=0;ch<s->nb_channels;ch++) {
966 b = bit_alloc[ch][i];
968 scale = scale_factors[ch][i][k];
969 qindex = alloc_table[j+b];
970 bits = ff_mpa_quant_bits[qindex];
973 /* 3 values at the same time */
974 v = get_bits(&s->gb, -bits);
975 v2 = division_tabs[qindex][v];
976 steps = ff_mpa_quant_steps[qindex];
978 s->sb_samples[ch][k * 12 + l + 0][i] =
979 l2_unscale_group(steps, v2 & 15, scale);
980 s->sb_samples[ch][k * 12 + l + 1][i] =
981 l2_unscale_group(steps, (v2 >> 4) & 15, scale);
982 s->sb_samples[ch][k * 12 + l + 2][i] =
983 l2_unscale_group(steps, v2 >> 8 , scale);
986 v = get_bits(&s->gb, bits);
987 v = l1_unscale(bits - 1, v, scale);
988 s->sb_samples[ch][k * 12 + l + m][i] = v;
992 s->sb_samples[ch][k * 12 + l + 0][i] = 0;
993 s->sb_samples[ch][k * 12 + l + 1][i] = 0;
994 s->sb_samples[ch][k * 12 + l + 2][i] = 0;
997 /* next subband in alloc table */
998 j += 1 << bit_alloc_bits;
1000 /* XXX: find a way to avoid this duplication of code */
1001 for(i=bound;i<sblimit;i++) {
1002 bit_alloc_bits = alloc_table[j];
1003 b = bit_alloc[0][i];
1005 int mant, scale0, scale1;
1006 scale0 = scale_factors[0][i][k];
1007 scale1 = scale_factors[1][i][k];
1008 qindex = alloc_table[j+b];
1009 bits = ff_mpa_quant_bits[qindex];
1011 /* 3 values at the same time */
1012 v = get_bits(&s->gb, -bits);
1013 steps = ff_mpa_quant_steps[qindex];
1016 s->sb_samples[0][k * 12 + l + 0][i] =
1017 l2_unscale_group(steps, mant, scale0);
1018 s->sb_samples[1][k * 12 + l + 0][i] =
1019 l2_unscale_group(steps, mant, scale1);
1022 s->sb_samples[0][k * 12 + l + 1][i] =
1023 l2_unscale_group(steps, mant, scale0);
1024 s->sb_samples[1][k * 12 + l + 1][i] =
1025 l2_unscale_group(steps, mant, scale1);
1026 s->sb_samples[0][k * 12 + l + 2][i] =
1027 l2_unscale_group(steps, v, scale0);
1028 s->sb_samples[1][k * 12 + l + 2][i] =
1029 l2_unscale_group(steps, v, scale1);
1032 mant = get_bits(&s->gb, bits);
1033 s->sb_samples[0][k * 12 + l + m][i] =
1034 l1_unscale(bits - 1, mant, scale0);
1035 s->sb_samples[1][k * 12 + l + m][i] =
1036 l1_unscale(bits - 1, mant, scale1);
1040 s->sb_samples[0][k * 12 + l + 0][i] = 0;
1041 s->sb_samples[0][k * 12 + l + 1][i] = 0;
1042 s->sb_samples[0][k * 12 + l + 2][i] = 0;
1043 s->sb_samples[1][k * 12 + l + 0][i] = 0;
1044 s->sb_samples[1][k * 12 + l + 1][i] = 0;
1045 s->sb_samples[1][k * 12 + l + 2][i] = 0;
1047 /* next subband in alloc table */
1048 j += 1 << bit_alloc_bits;
1050 /* fill remaining samples to zero */
1051 for(i=sblimit;i<SBLIMIT;i++) {
1052 for(ch=0;ch<s->nb_channels;ch++) {
1053 s->sb_samples[ch][k * 12 + l + 0][i] = 0;
1054 s->sb_samples[ch][k * 12 + l + 1][i] = 0;
1055 s->sb_samples[ch][k * 12 + l + 2][i] = 0;
1063 #define SPLIT(dst,sf,n)\
1065 int m= (sf*171)>>9;\
1072 int m= (sf*205)>>10;\
1076 int m= (sf*171)>>10;\
1083 static av_always_inline void lsf_sf_expand(int *slen,
1084 int sf, int n1, int n2, int n3)
1086 SPLIT(slen[3], sf, n3)
1087 SPLIT(slen[2], sf, n2)
1088 SPLIT(slen[1], sf, n1)
1092 static void exponents_from_scale_factors(MPADecodeContext *s,
1096 const uint8_t *bstab, *pretab;
1097 int len, i, j, k, l, v0, shift, gain, gains[3];
1100 exp_ptr = exponents;
1101 gain = g->global_gain - 210;
1102 shift = g->scalefac_scale + 1;
1104 bstab = band_size_long[s->sample_rate_index];
1105 pretab = mpa_pretab[g->preflag];
1106 for(i=0;i<g->long_end;i++) {
1107 v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
1113 if (g->short_start < 13) {
1114 bstab = band_size_short[s->sample_rate_index];
1115 gains[0] = gain - (g->subblock_gain[0] << 3);
1116 gains[1] = gain - (g->subblock_gain[1] << 3);
1117 gains[2] = gain - (g->subblock_gain[2] << 3);
1119 for(i=g->short_start;i<13;i++) {
1122 v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
1130 /* handle n = 0 too */
1131 static inline int get_bitsz(GetBitContext *s, int n)
1136 return get_bits(s, n);
1140 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_pos2){
1141 if(s->in_gb.buffer && *pos >= s->gb.size_in_bits){
1143 s->in_gb.buffer=NULL;
1144 assert((get_bits_count(&s->gb) & 7) == 0);
1145 skip_bits_long(&s->gb, *pos - *end_pos);
1147 *end_pos= *end_pos2 + get_bits_count(&s->gb) - *pos;
1148 *pos= get_bits_count(&s->gb);
1152 /* Following is a optimized code for
1154 if(get_bits1(&s->gb))
1159 #define READ_FLIP_SIGN(dst,src)\
1160 v = AV_RN32A(src) ^ (get_bits1(&s->gb)<<31);\
1163 #define READ_FLIP_SIGN(dst,src)\
1164 v= -get_bits1(&s->gb);\
1165 *(dst) = (*(src) ^ v) - v;
1168 static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
1169 int16_t *exponents, int end_pos2)
1173 int last_pos, bits_left;
1175 int end_pos= FFMIN(end_pos2, s->gb.size_in_bits);
1177 /* low frequencies (called big values) */
1180 int j, k, l, linbits;
1181 j = g->region_size[i];
1184 /* select vlc table */
1185 k = g->table_select[i];
1186 l = mpa_huff_data[k][0];
1187 linbits = mpa_huff_data[k][1];
1191 memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*2*j);
1196 /* read huffcode and compute each couple */
1200 int pos= get_bits_count(&s->gb);
1202 if (pos >= end_pos){
1203 // av_log(NULL, AV_LOG_ERROR, "pos: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
1204 switch_buffer(s, &pos, &end_pos, &end_pos2);
1205 // av_log(NULL, AV_LOG_ERROR, "new pos: %d %d\n", pos, end_pos);
1209 y = get_vlc2(&s->gb, vlc->table, 7, 3);
1212 g->sb_hybrid[s_index ] =
1213 g->sb_hybrid[s_index+1] = 0;
1218 exponent= exponents[s_index];
1220 av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
1221 i, g->region_size[i] - j, x, y, exponent);
1226 READ_FLIP_SIGN(g->sb_hybrid+s_index, RENAME(expval_table)[ exponent ]+x)
1228 x += get_bitsz(&s->gb, linbits);
1229 v = l3_unscale(x, exponent);
1230 if (get_bits1(&s->gb))
1232 g->sb_hybrid[s_index] = v;
1235 READ_FLIP_SIGN(g->sb_hybrid+s_index+1, RENAME(expval_table)[ exponent ]+y)
1237 y += get_bitsz(&s->gb, linbits);
1238 v = l3_unscale(y, exponent);
1239 if (get_bits1(&s->gb))
1241 g->sb_hybrid[s_index+1] = v;
1248 READ_FLIP_SIGN(g->sb_hybrid+s_index+!!y, RENAME(expval_table)[ exponent ]+x)
1250 x += get_bitsz(&s->gb, linbits);
1251 v = l3_unscale(x, exponent);
1252 if (get_bits1(&s->gb))
1254 g->sb_hybrid[s_index+!!y] = v;
1256 g->sb_hybrid[s_index+ !y] = 0;
1262 /* high frequencies */
1263 vlc = &huff_quad_vlc[g->count1table_select];
1265 while (s_index <= 572) {
1267 pos = get_bits_count(&s->gb);
1268 if (pos >= end_pos) {
1269 if (pos > end_pos2 && last_pos){
1270 /* some encoders generate an incorrect size for this
1271 part. We must go back into the data */
1273 skip_bits_long(&s->gb, last_pos - pos);
1274 av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
1275 if(s->error_recognition >= FF_ER_COMPLIANT)
1279 // av_log(NULL, AV_LOG_ERROR, "pos2: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
1280 switch_buffer(s, &pos, &end_pos, &end_pos2);
1281 // av_log(NULL, AV_LOG_ERROR, "new pos2: %d %d %d\n", pos, end_pos, s_index);
1287 code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
1288 av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
1289 g->sb_hybrid[s_index+0]=
1290 g->sb_hybrid[s_index+1]=
1291 g->sb_hybrid[s_index+2]=
1292 g->sb_hybrid[s_index+3]= 0;
1294 static const int idxtab[16]={3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0};
1296 int pos= s_index+idxtab[code];
1297 code ^= 8>>idxtab[code];
1298 READ_FLIP_SIGN(g->sb_hybrid+pos, RENAME(exp_table)+exponents[pos])
1302 /* skip extension bits */
1303 bits_left = end_pos2 - get_bits_count(&s->gb);
1304 //av_log(NULL, AV_LOG_ERROR, "left:%d buf:%p\n", bits_left, s->in_gb.buffer);
1305 if (bits_left < 0 && s->error_recognition >= FF_ER_COMPLIANT) {
1306 av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
1308 }else if(bits_left > 0 && s->error_recognition >= FF_ER_AGGRESSIVE){
1309 av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
1312 memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*(576 - s_index));
1313 skip_bits_long(&s->gb, bits_left);
1315 i= get_bits_count(&s->gb);
1316 switch_buffer(s, &i, &end_pos, &end_pos2);
1321 /* Reorder short blocks from bitstream order to interleaved order. It
1322 would be faster to do it in parsing, but the code would be far more
1324 static void reorder_block(MPADecodeContext *s, GranuleDef *g)
1327 INTFLOAT *ptr, *dst, *ptr1;
1330 if (g->block_type != 2)
1333 if (g->switch_point) {
1334 if (s->sample_rate_index != 8) {
1335 ptr = g->sb_hybrid + 36;
1337 ptr = g->sb_hybrid + 48;
1343 for(i=g->short_start;i<13;i++) {
1344 len = band_size_short[s->sample_rate_index][i];
1347 for(j=len;j>0;j--) {
1348 *dst++ = ptr[0*len];
1349 *dst++ = ptr[1*len];
1350 *dst++ = ptr[2*len];
1354 memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
1358 #define ISQRT2 FIXR(0.70710678118654752440)
1360 static void compute_stereo(MPADecodeContext *s,
1361 GranuleDef *g0, GranuleDef *g1)
1364 int sf_max, sf, len, non_zero_found;
1365 INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
1366 int non_zero_found_short[3];
1368 /* intensity stereo */
1369 if (s->mode_ext & MODE_EXT_I_STEREO) {
1374 is_tab = is_table_lsf[g1->scalefac_compress & 1];
1378 tab0 = g0->sb_hybrid + 576;
1379 tab1 = g1->sb_hybrid + 576;
1381 non_zero_found_short[0] = 0;
1382 non_zero_found_short[1] = 0;
1383 non_zero_found_short[2] = 0;
1384 k = (13 - g1->short_start) * 3 + g1->long_end - 3;
1385 for(i = 12;i >= g1->short_start;i--) {
1386 /* for last band, use previous scale factor */
1389 len = band_size_short[s->sample_rate_index][i];
1393 if (!non_zero_found_short[l]) {
1394 /* test if non zero band. if so, stop doing i-stereo */
1395 for(j=0;j<len;j++) {
1397 non_zero_found_short[l] = 1;
1401 sf = g1->scale_factors[k + l];
1407 for(j=0;j<len;j++) {
1409 tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1410 tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1414 if (s->mode_ext & MODE_EXT_MS_STEREO) {
1415 /* lower part of the spectrum : do ms stereo
1417 for(j=0;j<len;j++) {
1420 tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1421 tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1428 non_zero_found = non_zero_found_short[0] |
1429 non_zero_found_short[1] |
1430 non_zero_found_short[2];
1432 for(i = g1->long_end - 1;i >= 0;i--) {
1433 len = band_size_long[s->sample_rate_index][i];
1436 /* test if non zero band. if so, stop doing i-stereo */
1437 if (!non_zero_found) {
1438 for(j=0;j<len;j++) {
1444 /* for last band, use previous scale factor */
1445 k = (i == 21) ? 20 : i;
1446 sf = g1->scale_factors[k];
1451 for(j=0;j<len;j++) {
1453 tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1454 tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1458 if (s->mode_ext & MODE_EXT_MS_STEREO) {
1459 /* lower part of the spectrum : do ms stereo
1461 for(j=0;j<len;j++) {
1464 tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1465 tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1470 } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
1471 /* ms stereo ONLY */
1472 /* NOTE: the 1/sqrt(2) normalization factor is included in the
1474 tab0 = g0->sb_hybrid;
1475 tab1 = g1->sb_hybrid;
1476 for(i=0;i<576;i++) {
1479 tab0[i] = tmp0 + tmp1;
1480 tab1[i] = tmp0 - tmp1;
1486 static void compute_antialias_integer(MPADecodeContext *s,
1492 /* we antialias only "long" bands */
1493 if (g->block_type == 2) {
1494 if (!g->switch_point)
1496 /* XXX: check this for 8000Hz case */
1502 ptr = g->sb_hybrid + 18;
1503 for(i = n;i > 0;i--) {
1504 int tmp0, tmp1, tmp2;
1505 csa = &csa_table[0][0];
1509 tmp2= MULH(tmp0 + tmp1, csa[0+4*j]);\
1510 ptr[-1-j] = 4*(tmp2 - MULH(tmp1, csa[2+4*j]));\
1511 ptr[ j] = 4*(tmp2 + MULH(tmp0, csa[3+4*j]));
1527 static void compute_imdct(MPADecodeContext *s,
1529 INTFLOAT *sb_samples,
1532 INTFLOAT *win, *win1, *out_ptr, *ptr, *buf, *ptr1;
1534 int i, j, mdct_long_end, sblimit;
1536 /* find last non zero block */
1537 ptr = g->sb_hybrid + 576;
1538 ptr1 = g->sb_hybrid + 2 * 18;
1539 while (ptr >= ptr1) {
1543 if(p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
1546 sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
1548 if (g->block_type == 2) {
1549 /* XXX: check for 8000 Hz */
1550 if (g->switch_point)
1555 mdct_long_end = sblimit;
1560 for(j=0;j<mdct_long_end;j++) {
1561 /* apply window & overlap with previous buffer */
1562 out_ptr = sb_samples + j;
1564 if (g->switch_point && j < 2)
1567 win1 = mdct_win[g->block_type];
1568 /* select frequency inversion */
1569 win = win1 + ((4 * 36) & -(j & 1));
1570 imdct36(out_ptr, buf, ptr, win);
1571 out_ptr += 18*SBLIMIT;
1575 for(j=mdct_long_end;j<sblimit;j++) {
1576 /* select frequency inversion */
1577 win = mdct_win[2] + ((4 * 36) & -(j & 1));
1578 out_ptr = sb_samples + j;
1584 imdct12(out2, ptr + 0);
1586 *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*1];
1587 buf[i + 6*2] = MULH3(out2[i + 6], win[i + 6], 1);
1590 imdct12(out2, ptr + 1);
1592 *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*2];
1593 buf[i + 6*0] = MULH3(out2[i + 6], win[i + 6], 1);
1596 imdct12(out2, ptr + 2);
1598 buf[i + 6*0] = MULH3(out2[i ], win[i ], 1) + buf[i + 6*0];
1599 buf[i + 6*1] = MULH3(out2[i + 6], win[i + 6], 1);
1606 for(j=sblimit;j<SBLIMIT;j++) {
1608 out_ptr = sb_samples + j;
1618 /* main layer3 decoding function */
1619 static int mp_decode_layer3(MPADecodeContext *s)
1621 int nb_granules, main_data_begin, private_bits;
1622 int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
1624 int16_t exponents[576]; //FIXME try INTFLOAT
1626 /* read side info */
1628 main_data_begin = get_bits(&s->gb, 8);
1629 private_bits = get_bits(&s->gb, s->nb_channels);
1632 main_data_begin = get_bits(&s->gb, 9);
1633 if (s->nb_channels == 2)
1634 private_bits = get_bits(&s->gb, 3);
1636 private_bits = get_bits(&s->gb, 5);
1638 for(ch=0;ch<s->nb_channels;ch++) {
1639 s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
1640 s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
1644 for(gr=0;gr<nb_granules;gr++) {
1645 for(ch=0;ch<s->nb_channels;ch++) {
1646 av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
1647 g = &s->granules[ch][gr];
1648 g->part2_3_length = get_bits(&s->gb, 12);
1649 g->big_values = get_bits(&s->gb, 9);
1650 if(g->big_values > 288){
1651 av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
1655 g->global_gain = get_bits(&s->gb, 8);
1656 /* if MS stereo only is selected, we precompute the
1657 1/sqrt(2) renormalization factor */
1658 if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
1660 g->global_gain -= 2;
1662 g->scalefac_compress = get_bits(&s->gb, 9);
1664 g->scalefac_compress = get_bits(&s->gb, 4);
1665 blocksplit_flag = get_bits1(&s->gb);
1666 if (blocksplit_flag) {
1667 g->block_type = get_bits(&s->gb, 2);
1668 if (g->block_type == 0){
1669 av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
1672 g->switch_point = get_bits1(&s->gb);
1674 g->table_select[i] = get_bits(&s->gb, 5);
1676 g->subblock_gain[i] = get_bits(&s->gb, 3);
1677 ff_init_short_region(s, g);
1679 int region_address1, region_address2;
1681 g->switch_point = 0;
1683 g->table_select[i] = get_bits(&s->gb, 5);
1684 /* compute huffman coded region sizes */
1685 region_address1 = get_bits(&s->gb, 4);
1686 region_address2 = get_bits(&s->gb, 3);
1687 av_dlog(s->avctx, "region1=%d region2=%d\n",
1688 region_address1, region_address2);
1689 ff_init_long_region(s, g, region_address1, region_address2);
1691 ff_region_offset2size(g);
1692 ff_compute_band_indexes(s, g);
1696 g->preflag = get_bits1(&s->gb);
1697 g->scalefac_scale = get_bits1(&s->gb);
1698 g->count1table_select = get_bits1(&s->gb);
1699 av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
1700 g->block_type, g->switch_point);
1705 const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
1706 assert((get_bits_count(&s->gb) & 7) == 0);
1707 /* now we get bits from the main_data_begin offset */
1708 av_dlog(s->avctx, "seekback: %d\n", main_data_begin);
1709 //av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size);
1711 memcpy(s->last_buf + s->last_buf_size, ptr, EXTRABYTES);
1713 init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
1714 skip_bits_long(&s->gb, 8*(s->last_buf_size - main_data_begin));
1717 for(gr=0;gr<nb_granules;gr++) {
1718 for(ch=0;ch<s->nb_channels;ch++) {
1719 g = &s->granules[ch][gr];
1720 if(get_bits_count(&s->gb)<0){
1721 av_log(s->avctx, AV_LOG_DEBUG, "mdb:%d, lastbuf:%d skipping granule %d\n",
1722 main_data_begin, s->last_buf_size, gr);
1723 skip_bits_long(&s->gb, g->part2_3_length);
1724 memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
1725 if(get_bits_count(&s->gb) >= s->gb.size_in_bits && s->in_gb.buffer){
1726 skip_bits_long(&s->in_gb, get_bits_count(&s->gb) - s->gb.size_in_bits);
1728 s->in_gb.buffer=NULL;
1733 bits_pos = get_bits_count(&s->gb);
1737 int slen, slen1, slen2;
1739 /* MPEG1 scale factors */
1740 slen1 = slen_table[0][g->scalefac_compress];
1741 slen2 = slen_table[1][g->scalefac_compress];
1742 av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
1743 if (g->block_type == 2) {
1744 n = g->switch_point ? 17 : 18;
1748 g->scale_factors[j++] = get_bits(&s->gb, slen1);
1751 g->scale_factors[j++] = 0;
1755 g->scale_factors[j++] = get_bits(&s->gb, slen2);
1757 g->scale_factors[j++] = 0;
1760 g->scale_factors[j++] = 0;
1763 sc = s->granules[ch][0].scale_factors;
1766 n = (k == 0 ? 6 : 5);
1767 if ((g->scfsi & (0x8 >> k)) == 0) {
1768 slen = (k < 2) ? slen1 : slen2;
1771 g->scale_factors[j++] = get_bits(&s->gb, slen);
1774 g->scale_factors[j++] = 0;
1777 /* simply copy from last granule */
1779 g->scale_factors[j] = sc[j];
1784 g->scale_factors[j++] = 0;
1787 int tindex, tindex2, slen[4], sl, sf;
1789 /* LSF scale factors */
1790 if (g->block_type == 2) {
1791 tindex = g->switch_point ? 2 : 1;
1795 sf = g->scalefac_compress;
1796 if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
1797 /* intensity stereo case */
1800 lsf_sf_expand(slen, sf, 6, 6, 0);
1802 } else if (sf < 244) {
1803 lsf_sf_expand(slen, sf - 180, 4, 4, 0);
1806 lsf_sf_expand(slen, sf - 244, 3, 0, 0);
1812 lsf_sf_expand(slen, sf, 5, 4, 4);
1814 } else if (sf < 500) {
1815 lsf_sf_expand(slen, sf - 400, 5, 4, 0);
1818 lsf_sf_expand(slen, sf - 500, 3, 0, 0);
1826 n = lsf_nsf_table[tindex2][tindex][k];
1830 g->scale_factors[j++] = get_bits(&s->gb, sl);
1833 g->scale_factors[j++] = 0;
1836 /* XXX: should compute exact size */
1838 g->scale_factors[j] = 0;
1841 exponents_from_scale_factors(s, g, exponents);
1843 /* read Huffman coded residue */
1844 huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
1847 if (s->nb_channels == 2)
1848 compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
1850 for(ch=0;ch<s->nb_channels;ch++) {
1851 g = &s->granules[ch][gr];
1853 reorder_block(s, g);
1854 compute_antialias(s, g);
1855 compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1858 if(get_bits_count(&s->gb)<0)
1859 skip_bits_long(&s->gb, -get_bits_count(&s->gb));
1860 return nb_granules * 18;
1863 static int mp_decode_frame(MPADecodeContext *s,
1864 OUT_INT *samples, const uint8_t *buf, int buf_size)
1866 int i, nb_frames, ch;
1867 OUT_INT *samples_ptr;
1869 init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE)*8);
1871 /* skip error protection field */
1872 if (s->error_protection)
1873 skip_bits(&s->gb, 16);
1875 av_dlog(s->avctx, "frame %d:\n", s->frame_count);
1878 s->avctx->frame_size = 384;
1879 nb_frames = mp_decode_layer1(s);
1882 s->avctx->frame_size = 1152;
1883 nb_frames = mp_decode_layer2(s);
1886 s->avctx->frame_size = s->lsf ? 576 : 1152;
1888 nb_frames = mp_decode_layer3(s);
1891 if(s->in_gb.buffer){
1892 align_get_bits(&s->gb);
1893 i= get_bits_left(&s->gb)>>3;
1894 if(i >= 0 && i <= BACKSTEP_SIZE){
1895 memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
1898 av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
1900 s->in_gb.buffer= NULL;
1903 align_get_bits(&s->gb);
1904 assert((get_bits_count(&s->gb) & 7) == 0);
1905 i= get_bits_left(&s->gb)>>3;
1907 if(i<0 || i > BACKSTEP_SIZE || nb_frames<0){
1909 av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
1910 i= FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
1912 assert(i <= buf_size - HEADER_SIZE && i>= 0);
1913 memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
1914 s->last_buf_size += i;
1919 /* apply the synthesis filter */
1920 for(ch=0;ch<s->nb_channels;ch++) {
1921 samples_ptr = samples + ch;
1922 for(i=0;i<nb_frames;i++) {
1923 RENAME(ff_mpa_synth_filter)(
1927 s->synth_buf[ch], &(s->synth_buf_offset[ch]),
1928 RENAME(ff_mpa_synth_window), &s->dither_state,
1929 samples_ptr, s->nb_channels,
1930 s->sb_samples[ch][i]);
1931 samples_ptr += 32 * s->nb_channels;
1935 return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
1938 static int decode_frame(AVCodecContext * avctx,
1939 void *data, int *data_size,
1942 const uint8_t *buf = avpkt->data;
1943 int buf_size = avpkt->size;
1944 MPADecodeContext *s = avctx->priv_data;
1947 OUT_INT *out_samples = data;
1949 if(buf_size < HEADER_SIZE)
1952 header = AV_RB32(buf);
1953 if(ff_mpa_check_header(header) < 0){
1954 av_log(avctx, AV_LOG_ERROR, "Header missing\n");
1958 if (ff_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
1959 /* free format: prepare to compute frame size */
1963 /* update codec info */
1964 avctx->channels = s->nb_channels;
1965 avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1966 if (!avctx->bit_rate)
1967 avctx->bit_rate = s->bit_rate;
1968 avctx->sub_id = s->layer;
1970 if(*data_size < 1152*avctx->channels*sizeof(OUT_INT))
1974 if(s->frame_size<=0 || s->frame_size > buf_size){
1975 av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
1977 }else if(s->frame_size < buf_size){
1978 av_log(avctx, AV_LOG_ERROR, "incorrect frame size\n");
1979 buf_size= s->frame_size;
1982 out_size = mp_decode_frame(s, out_samples, buf, buf_size);
1984 *data_size = out_size;
1985 avctx->sample_rate = s->sample_rate;
1986 //FIXME maybe move the other codec info stuff from above here too
1988 av_log(avctx, AV_LOG_DEBUG, "Error while decoding MPEG audio frame.\n"); //FIXME return -1 / but also return the number of bytes consumed
1993 static void flush(AVCodecContext *avctx){
1994 MPADecodeContext *s = avctx->priv_data;
1995 memset(s->synth_buf, 0, sizeof(s->synth_buf));
1996 s->last_buf_size= 0;
1999 #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
2000 static int decode_frame_adu(AVCodecContext * avctx,
2001 void *data, int *data_size,
2004 const uint8_t *buf = avpkt->data;
2005 int buf_size = avpkt->size;
2006 MPADecodeContext *s = avctx->priv_data;
2009 OUT_INT *out_samples = data;
2013 // Discard too short frames
2014 if (buf_size < HEADER_SIZE) {
2020 if (len > MPA_MAX_CODED_FRAME_SIZE)
2021 len = MPA_MAX_CODED_FRAME_SIZE;
2023 // Get header and restore sync word
2024 header = AV_RB32(buf) | 0xffe00000;
2026 if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
2031 ff_mpegaudio_decode_header((MPADecodeHeader *)s, header);
2032 /* update codec info */
2033 avctx->sample_rate = s->sample_rate;
2034 avctx->channels = s->nb_channels;
2035 if (!avctx->bit_rate)
2036 avctx->bit_rate = s->bit_rate;
2037 avctx->sub_id = s->layer;
2039 s->frame_size = len;
2041 if (avctx->parse_only) {
2042 out_size = buf_size;
2044 out_size = mp_decode_frame(s, out_samples, buf, buf_size);
2047 *data_size = out_size;
2050 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
2052 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
2055 * Context for MP3On4 decoder
2057 typedef struct MP3On4DecodeContext {
2058 int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
2059 int syncword; ///< syncword patch
2060 const uint8_t *coff; ///< channels offsets in output buffer
2061 MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
2062 } MP3On4DecodeContext;
2064 #include "mpeg4audio.h"
2066 /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
2067 static const uint8_t mp3Frames[8] = {0,1,1,2,3,3,4,5}; /* number of mp3 decoder instances */
2068 /* offsets into output buffer, assume output order is FL FR BL BR C LFE */
2069 static const uint8_t chan_offset[8][5] = {
2074 {2,0,3}, // C FLR BS
2075 {4,0,2}, // C FLR BLRS
2076 {4,0,2,5}, // C FLR BLRS LFE
2077 {4,0,2,6,5}, // C FLR BLRS BLR LFE
2081 static int decode_init_mp3on4(AVCodecContext * avctx)
2083 MP3On4DecodeContext *s = avctx->priv_data;
2084 MPEG4AudioConfig cfg;
2087 if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
2088 av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
2092 ff_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size);
2093 if (!cfg.chan_config || cfg.chan_config > 7) {
2094 av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
2097 s->frames = mp3Frames[cfg.chan_config];
2098 s->coff = chan_offset[cfg.chan_config];
2099 avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
2101 if (cfg.sample_rate < 16000)
2102 s->syncword = 0xffe00000;
2104 s->syncword = 0xfff00000;
2106 /* Init the first mp3 decoder in standard way, so that all tables get builded
2107 * We replace avctx->priv_data with the context of the first decoder so that
2108 * decode_init() does not have to be changed.
2109 * Other decoders will be initialized here copying data from the first context
2111 // Allocate zeroed memory for the first decoder context
2112 s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
2113 // Put decoder context in place to make init_decode() happy
2114 avctx->priv_data = s->mp3decctx[0];
2116 // Restore mp3on4 context pointer
2117 avctx->priv_data = s;
2118 s->mp3decctx[0]->adu_mode = 1; // Set adu mode
2120 /* Create a separate codec/context for each frame (first is already ok).
2121 * Each frame is 1 or 2 channels - up to 5 frames allowed
2123 for (i = 1; i < s->frames; i++) {
2124 s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
2125 s->mp3decctx[i]->adu_mode = 1;
2126 s->mp3decctx[i]->avctx = avctx;
2133 static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
2135 MP3On4DecodeContext *s = avctx->priv_data;
2138 for (i = 0; i < s->frames; i++)
2139 av_free(s->mp3decctx[i]);
2145 static int decode_frame_mp3on4(AVCodecContext * avctx,
2146 void *data, int *data_size,
2149 const uint8_t *buf = avpkt->data;
2150 int buf_size = avpkt->size;
2151 MP3On4DecodeContext *s = avctx->priv_data;
2152 MPADecodeContext *m;
2153 int fsize, len = buf_size, out_size = 0;
2155 OUT_INT *out_samples = data;
2156 OUT_INT decoded_buf[MPA_FRAME_SIZE * MPA_MAX_CHANNELS];
2157 OUT_INT *outptr, *bp;
2160 if(*data_size < MPA_FRAME_SIZE * MPA_MAX_CHANNELS * s->frames * sizeof(OUT_INT))
2164 // Discard too short frames
2165 if (buf_size < HEADER_SIZE)
2168 // If only one decoder interleave is not needed
2169 outptr = s->frames == 1 ? out_samples : decoded_buf;
2171 avctx->bit_rate = 0;
2173 for (fr = 0; fr < s->frames; fr++) {
2174 fsize = AV_RB16(buf) >> 4;
2175 fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
2176 m = s->mp3decctx[fr];
2179 header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
2181 if (ff_mpa_check_header(header) < 0) // Bad header, discard block
2184 ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
2185 out_size += mp_decode_frame(m, outptr, buf, fsize);
2190 n = m->avctx->frame_size*m->nb_channels;
2191 /* interleave output data */
2192 bp = out_samples + s->coff[fr];
2193 if(m->nb_channels == 1) {
2194 for(j = 0; j < n; j++) {
2195 *bp = decoded_buf[j];
2196 bp += avctx->channels;
2199 for(j = 0; j < n; j++) {
2200 bp[0] = decoded_buf[j++];
2201 bp[1] = decoded_buf[j];
2202 bp += avctx->channels;
2206 avctx->bit_rate += m->bit_rate;
2209 /* update codec info */
2210 avctx->sample_rate = s->mp3decctx[0]->sample_rate;
2212 *data_size = out_size;
2215 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
2218 #if CONFIG_MP1_DECODER
2219 AVCodec ff_mp1_decoder =
2224 sizeof(MPADecodeContext),
2229 CODEC_CAP_PARSE_ONLY,
2231 .long_name= NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
2234 #if CONFIG_MP2_DECODER
2235 AVCodec ff_mp2_decoder =
2240 sizeof(MPADecodeContext),
2245 CODEC_CAP_PARSE_ONLY,
2247 .long_name= NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
2250 #if CONFIG_MP3_DECODER
2251 AVCodec ff_mp3_decoder =
2256 sizeof(MPADecodeContext),
2261 CODEC_CAP_PARSE_ONLY,
2263 .long_name= NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
2266 #if CONFIG_MP3ADU_DECODER
2267 AVCodec ff_mp3adu_decoder =
2272 sizeof(MPADecodeContext),
2277 CODEC_CAP_PARSE_ONLY,
2279 .long_name= NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
2282 #if CONFIG_MP3ON4_DECODER
2283 AVCodec ff_mp3on4_decoder =
2288 sizeof(MP3On4DecodeContext),
2291 decode_close_mp3on4,
2292 decode_frame_mp3on4,
2294 .long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"),