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
50 # define OUT_FMT AV_SAMPLE_FMT_FLT
52 # define SHR(a,b) ((a)>>(b))
53 # define compute_antialias compute_antialias_integer
54 /* WARNING: only correct for posititive numbers */
55 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
56 # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
57 # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
58 # define MULH3(x, y, s) MULH((s)*(x), y)
59 # define MULLx(x, y, s) MULL(x,y,s)
61 # define OUT_FMT AV_SAMPLE_FMT_S16
68 #include "mpegaudiodata.h"
69 #include "mpegaudiodectab.h"
77 static void compute_antialias(MPADecodeContext *s, GranuleDef *g);
78 static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
79 int *dither_state, OUT_INT *samples, int incr);
81 /* vlc structure for decoding layer 3 huffman tables */
82 static VLC huff_vlc[16];
83 static VLC_TYPE huff_vlc_tables[
84 0+128+128+128+130+128+154+166+
85 142+204+190+170+542+460+662+414
87 static const int huff_vlc_tables_sizes[16] = {
88 0, 128, 128, 128, 130, 128, 154, 166,
89 142, 204, 190, 170, 542, 460, 662, 414
91 static VLC huff_quad_vlc[2];
92 static VLC_TYPE huff_quad_vlc_tables[128+16][2];
93 static const int huff_quad_vlc_tables_sizes[2] = {
96 /* computed from band_size_long */
97 static uint16_t band_index_long[9][23];
98 #include "mpegaudio_tablegen.h"
99 /* intensity stereo coef table */
100 static INTFLOAT is_table[2][16];
101 static INTFLOAT is_table_lsf[2][2][16];
102 static int32_t csa_table[8][4];
103 static float csa_table_float[8][4];
104 static INTFLOAT mdct_win[8][36];
106 static int16_t division_tab3[1<<6 ];
107 static int16_t division_tab5[1<<8 ];
108 static int16_t division_tab9[1<<11];
110 static int16_t * const division_tabs[4] = {
111 division_tab3, division_tab5, NULL, division_tab9
114 /* lower 2 bits: modulo 3, higher bits: shift */
115 static uint16_t scale_factor_modshift[64];
116 /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
117 static int32_t scale_factor_mult[15][3];
118 /* mult table for layer 2 group quantization */
120 #define SCALE_GEN(v) \
121 { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
123 static const int32_t scale_factor_mult2[3][3] = {
124 SCALE_GEN(4.0 / 3.0), /* 3 steps */
125 SCALE_GEN(4.0 / 5.0), /* 5 steps */
126 SCALE_GEN(4.0 / 9.0), /* 9 steps */
129 DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256];
132 * Convert region offsets to region sizes and truncate
133 * size to big_values.
135 static void ff_region_offset2size(GranuleDef *g){
137 g->region_size[2] = (576 / 2);
139 k = FFMIN(g->region_size[i], g->big_values);
140 g->region_size[i] = k - j;
145 static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g){
146 if (g->block_type == 2)
147 g->region_size[0] = (36 / 2);
149 if (s->sample_rate_index <= 2)
150 g->region_size[0] = (36 / 2);
151 else if (s->sample_rate_index != 8)
152 g->region_size[0] = (54 / 2);
154 g->region_size[0] = (108 / 2);
156 g->region_size[1] = (576 / 2);
159 static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2){
162 band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
163 /* should not overflow */
164 l = FFMIN(ra1 + ra2 + 2, 22);
166 band_index_long[s->sample_rate_index][l] >> 1;
169 static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g){
170 if (g->block_type == 2) {
171 if (g->switch_point) {
172 /* if switched mode, we handle the 36 first samples as
173 long blocks. For 8000Hz, we handle the 48 first
174 exponents as long blocks (XXX: check this!) */
175 if (s->sample_rate_index <= 2)
177 else if (s->sample_rate_index != 8)
180 g->long_end = 4; /* 8000 Hz */
182 g->short_start = 2 + (s->sample_rate_index != 8);
193 /* layer 1 unscaling */
194 /* n = number of bits of the mantissa minus 1 */
195 static inline int l1_unscale(int n, int mant, int scale_factor)
200 shift = scale_factor_modshift[scale_factor];
203 val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
205 /* NOTE: at this point, 1 <= shift >= 21 + 15 */
206 return (int)((val + (1LL << (shift - 1))) >> shift);
209 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
213 shift = scale_factor_modshift[scale_factor];
217 val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
218 /* NOTE: at this point, 0 <= shift <= 21 */
220 val = (val + (1 << (shift - 1))) >> shift;
224 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
225 static inline int l3_unscale(int value, int exponent)
230 e = table_4_3_exp [4*value + (exponent&3)];
231 m = table_4_3_value[4*value + (exponent&3)];
232 e -= (exponent >> 2);
236 m = (m + (1 << (e-1))) >> e;
241 /* all integer n^(4/3) computation code */
244 #define POW_FRAC_BITS 24
245 #define POW_FRAC_ONE (1 << POW_FRAC_BITS)
246 #define POW_FIX(a) ((int)((a) * POW_FRAC_ONE))
247 #define POW_MULL(a,b) (((int64_t)(a) * (int64_t)(b)) >> POW_FRAC_BITS)
249 static int dev_4_3_coefs[DEV_ORDER];
251 static av_cold void int_pow_init(void)
256 for(i=0;i<DEV_ORDER;i++) {
257 a = POW_MULL(a, POW_FIX(4.0 / 3.0) - i * POW_FIX(1.0)) / (i + 1);
258 dev_4_3_coefs[i] = a;
262 static av_cold int decode_init(AVCodecContext * avctx)
264 MPADecodeContext *s = avctx->priv_data;
269 s->apply_window_mp3 = apply_window_mp3_c;
270 #if HAVE_MMX && CONFIG_FLOAT
271 ff_mpegaudiodec_init_mmx(s);
274 ff_dct_init(&s->dct, 5, DCT_II);
276 if (HAVE_ALTIVEC && CONFIG_FLOAT) ff_mpegaudiodec_init_altivec(s);
278 avctx->sample_fmt= OUT_FMT;
279 s->error_recognition= avctx->error_recognition;
281 if (!init && !avctx->parse_only) {
284 /* scale factors table for layer 1/2 */
287 /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
290 scale_factor_modshift[i] = mod | (shift << 2);
293 /* scale factor multiply for layer 1 */
297 norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
298 scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
299 scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
300 scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
301 av_dlog(avctx, "%d: norm=%x s=%x %x %x\n",
303 scale_factor_mult[i][0],
304 scale_factor_mult[i][1],
305 scale_factor_mult[i][2]);
308 RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));
310 /* huffman decode tables */
313 const HuffTable *h = &mpa_huff_tables[i];
315 uint8_t tmp_bits [512];
316 uint16_t tmp_codes[512];
318 memset(tmp_bits , 0, sizeof(tmp_bits ));
319 memset(tmp_codes, 0, sizeof(tmp_codes));
324 for(x=0;x<xsize;x++) {
325 for(y=0;y<xsize;y++){
326 tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
327 tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
332 huff_vlc[i].table = huff_vlc_tables+offset;
333 huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
334 init_vlc(&huff_vlc[i], 7, 512,
335 tmp_bits, 1, 1, tmp_codes, 2, 2,
336 INIT_VLC_USE_NEW_STATIC);
337 offset += huff_vlc_tables_sizes[i];
339 assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
343 huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
344 huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
345 init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
346 mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
347 INIT_VLC_USE_NEW_STATIC);
348 offset += huff_quad_vlc_tables_sizes[i];
350 assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
355 band_index_long[i][j] = k;
356 k += band_size_long[i][j];
358 band_index_long[i][22] = k;
361 /* compute n ^ (4/3) and store it in mantissa/exp format */
364 mpegaudio_tableinit();
366 for (i = 0; i < 4; i++)
367 if (ff_mpa_quant_bits[i] < 0)
368 for (j = 0; j < (1<<(-ff_mpa_quant_bits[i]+1)); j++) {
369 int val1, val2, val3, steps;
371 steps = ff_mpa_quant_steps[i];
376 division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
384 f = tan((double)i * M_PI / 12.0);
385 v = FIXR(f / (1.0 + f));
390 is_table[1][6 - i] = v;
394 is_table[0][i] = is_table[1][i] = 0.0;
401 e = -(j + 1) * ((i + 1) >> 1);
402 f = pow(2.0, e / 4.0);
404 is_table_lsf[j][k ^ 1][i] = FIXR(f);
405 is_table_lsf[j][k][i] = FIXR(1.0);
406 av_dlog(avctx, "is_table_lsf %d %d: %x %x\n",
407 i, j, is_table_lsf[j][0][i], is_table_lsf[j][1][i]);
414 cs = 1.0 / sqrt(1.0 + ci * ci);
416 csa_table[i][0] = FIXHR(cs/4);
417 csa_table[i][1] = FIXHR(ca/4);
418 csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
419 csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);
420 csa_table_float[i][0] = cs;
421 csa_table_float[i][1] = ca;
422 csa_table_float[i][2] = ca + cs;
423 csa_table_float[i][3] = ca - cs;
426 /* compute mdct windows */
434 d= sin(M_PI * (i + 0.5) / 36.0);
437 else if(i>=24) d= sin(M_PI * (i - 18 + 0.5) / 12.0);
441 else if(i< 12) d= sin(M_PI * (i - 6 + 0.5) / 12.0);
444 //merge last stage of imdct into the window coefficients
445 d*= 0.5 / cos(M_PI*(2*i + 19)/72);
448 mdct_win[j][i/3] = FIXHR((d / (1<<5)));
450 mdct_win[j][i ] = FIXHR((d / (1<<5)));
454 /* NOTE: we do frequency inversion adter the MDCT by changing
455 the sign of the right window coefs */
458 mdct_win[j + 4][i] = mdct_win[j][i];
459 mdct_win[j + 4][i + 1] = -mdct_win[j][i + 1];
466 if (avctx->codec_id == CODEC_ID_MP3ADU)
473 static inline float round_sample(float *sum)
480 /* signed 16x16 -> 32 multiply add accumulate */
481 #define MACS(rt, ra, rb) rt+=(ra)*(rb)
483 /* signed 16x16 -> 32 multiply */
484 #define MULS(ra, rb) ((ra)*(rb))
486 #define MLSS(rt, ra, rb) rt-=(ra)*(rb)
490 static inline int round_sample(int64_t *sum)
493 sum1 = (int)((*sum) >> OUT_SHIFT);
494 *sum &= (1<<OUT_SHIFT)-1;
495 return av_clip_int16(sum1);
498 # define MULS(ra, rb) MUL64(ra, rb)
499 # define MACS(rt, ra, rb) MAC64(rt, ra, rb)
500 # define MLSS(rt, ra, rb) MLS64(rt, ra, rb)
503 #define SUM8(op, sum, w, p) \
505 op(sum, (w)[0 * 64], (p)[0 * 64]); \
506 op(sum, (w)[1 * 64], (p)[1 * 64]); \
507 op(sum, (w)[2 * 64], (p)[2 * 64]); \
508 op(sum, (w)[3 * 64], (p)[3 * 64]); \
509 op(sum, (w)[4 * 64], (p)[4 * 64]); \
510 op(sum, (w)[5 * 64], (p)[5 * 64]); \
511 op(sum, (w)[6 * 64], (p)[6 * 64]); \
512 op(sum, (w)[7 * 64], (p)[7 * 64]); \
515 #define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
519 op1(sum1, (w1)[0 * 64], tmp);\
520 op2(sum2, (w2)[0 * 64], tmp);\
522 op1(sum1, (w1)[1 * 64], tmp);\
523 op2(sum2, (w2)[1 * 64], tmp);\
525 op1(sum1, (w1)[2 * 64], tmp);\
526 op2(sum2, (w2)[2 * 64], tmp);\
528 op1(sum1, (w1)[3 * 64], tmp);\
529 op2(sum2, (w2)[3 * 64], tmp);\
531 op1(sum1, (w1)[4 * 64], tmp);\
532 op2(sum2, (w2)[4 * 64], tmp);\
534 op1(sum1, (w1)[5 * 64], tmp);\
535 op2(sum2, (w2)[5 * 64], tmp);\
537 op1(sum1, (w1)[6 * 64], tmp);\
538 op2(sum2, (w2)[6 * 64], tmp);\
540 op1(sum1, (w1)[7 * 64], tmp);\
541 op2(sum2, (w2)[7 * 64], tmp);\
544 void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window)
548 /* max = 18760, max sum over all 16 coefs : 44736 */
551 v = ff_mpa_enwindow[i];
553 v *= 1.0 / (1LL<<(16 + FRAC_BITS));
562 // Needed for avoiding shuffles in ASM implementations
564 for(j=0; j < 16; j++)
565 window[512+16*i+j] = window[64*i+32-j];
568 for(j=0; j < 16; j++)
569 window[512+128+16*i+j] = window[64*i+48-j];
572 static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window,
573 int *dither_state, OUT_INT *samples, int incr)
575 register const MPA_INT *w, *w2, *p;
584 /* copy to avoid wrap */
585 memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf));
587 samples2 = samples + 31 * incr;
593 SUM8(MACS, sum, w, p);
595 SUM8(MLSS, sum, w + 32, p);
596 *samples = round_sample(&sum);
600 /* we calculate two samples at the same time to avoid one memory
601 access per two sample */
604 p = synth_buf + 16 + j;
605 SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
606 p = synth_buf + 48 - j;
607 SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
609 *samples = round_sample(&sum);
612 *samples2 = round_sample(&sum);
619 SUM8(MLSS, sum, w + 32, p);
620 *samples = round_sample(&sum);
625 /* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
627 /* XXX: optimize by avoiding ring buffer usage */
629 void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
630 MPA_INT *window, int *dither_state,
631 OUT_INT *samples, int incr,
632 INTFLOAT sb_samples[SBLIMIT])
634 register MPA_INT *synth_buf;
637 offset = *synth_buf_offset;
638 synth_buf = synth_buf_ptr + offset;
640 dct32(synth_buf, sb_samples);
641 apply_window_mp3_c(synth_buf, window, dither_state, samples, incr);
643 offset = (offset - 32) & 511;
644 *synth_buf_offset = offset;
648 #define C3 FIXHR(0.86602540378443864676/2)
650 /* 0.5 / cos(pi*(2*i+1)/36) */
651 static const INTFLOAT icos36[9] = {
652 FIXR(0.50190991877167369479),
653 FIXR(0.51763809020504152469), //0
654 FIXR(0.55168895948124587824),
655 FIXR(0.61038729438072803416),
656 FIXR(0.70710678118654752439), //1
657 FIXR(0.87172339781054900991),
658 FIXR(1.18310079157624925896),
659 FIXR(1.93185165257813657349), //2
660 FIXR(5.73685662283492756461),
663 /* 0.5 / cos(pi*(2*i+1)/36) */
664 static const INTFLOAT icos36h[9] = {
665 FIXHR(0.50190991877167369479/2),
666 FIXHR(0.51763809020504152469/2), //0
667 FIXHR(0.55168895948124587824/2),
668 FIXHR(0.61038729438072803416/2),
669 FIXHR(0.70710678118654752439/2), //1
670 FIXHR(0.87172339781054900991/2),
671 FIXHR(1.18310079157624925896/4),
672 FIXHR(1.93185165257813657349/4), //2
673 // FIXHR(5.73685662283492756461),
676 /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
678 static void imdct12(INTFLOAT *out, INTFLOAT *in)
680 INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
683 in1= in[1*3] + in[0*3];
684 in2= in[2*3] + in[1*3];
685 in3= in[3*3] + in[2*3];
686 in4= in[4*3] + in[3*3];
687 in5= in[5*3] + in[4*3];
691 in2= MULH3(in2, C3, 2);
692 in3= MULH3(in3, C3, 4);
695 t2 = MULH3(in1 - in5, icos36h[4], 2);
705 in1 = MULH3(in5 + in3, icos36h[1], 1);
712 in5 = MULH3(in5 - in3, icos36h[7], 2);
720 #define C1 FIXHR(0.98480775301220805936/2)
721 #define C2 FIXHR(0.93969262078590838405/2)
722 #define C3 FIXHR(0.86602540378443864676/2)
723 #define C4 FIXHR(0.76604444311897803520/2)
724 #define C5 FIXHR(0.64278760968653932632/2)
725 #define C6 FIXHR(0.5/2)
726 #define C7 FIXHR(0.34202014332566873304/2)
727 #define C8 FIXHR(0.17364817766693034885/2)
730 /* using Lee like decomposition followed by hand coded 9 points DCT */
731 static void imdct36(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in, INTFLOAT *win)
734 INTFLOAT t0, t1, t2, t3, s0, s1, s2, s3;
735 INTFLOAT tmp[18], *tmp1, *in1;
746 t2 = in1[2*4] + in1[2*8] - in1[2*2];
748 t3 = in1[2*0] + SHR(in1[2*6],1);
749 t1 = in1[2*0] - in1[2*6];
750 tmp1[ 6] = t1 - SHR(t2,1);
753 t0 = MULH3(in1[2*2] + in1[2*4] , C2, 2);
754 t1 = MULH3(in1[2*4] - in1[2*8] , -2*C8, 1);
755 t2 = MULH3(in1[2*2] + in1[2*8] , -C4, 2);
757 tmp1[10] = t3 - t0 - t2;
758 tmp1[ 2] = t3 + t0 + t1;
759 tmp1[14] = t3 + t2 - t1;
761 tmp1[ 4] = MULH3(in1[2*5] + in1[2*7] - in1[2*1], -C3, 2);
762 t2 = MULH3(in1[2*1] + in1[2*5], C1, 2);
763 t3 = MULH3(in1[2*5] - in1[2*7], -2*C7, 1);
764 t0 = MULH3(in1[2*3], C3, 2);
766 t1 = MULH3(in1[2*1] + in1[2*7], -C5, 2);
768 tmp1[ 0] = t2 + t3 + t0;
769 tmp1[12] = t2 + t1 - t0;
770 tmp1[ 8] = t3 - t1 - t0;
782 s1 = MULH3(t3 + t2, icos36h[j], 2);
783 s3 = MULLx(t3 - t2, icos36[8 - j], FRAC_BITS);
787 out[(9 + j)*SBLIMIT] = MULH3(t1, win[9 + j], 1) + buf[9 + j];
788 out[(8 - j)*SBLIMIT] = MULH3(t1, win[8 - j], 1) + buf[8 - j];
789 buf[9 + j] = MULH3(t0, win[18 + 9 + j], 1);
790 buf[8 - j] = MULH3(t0, win[18 + 8 - j], 1);
794 out[(9 + 8 - j)*SBLIMIT] = MULH3(t1, win[9 + 8 - j], 1) + buf[9 + 8 - j];
795 out[( j)*SBLIMIT] = MULH3(t1, win[ j], 1) + buf[ j];
796 buf[9 + 8 - j] = MULH3(t0, win[18 + 9 + 8 - j], 1);
797 buf[ + j] = MULH3(t0, win[18 + j], 1);
802 s1 = MULH3(tmp[17], icos36h[4], 2);
805 out[(9 + 4)*SBLIMIT] = MULH3(t1, win[9 + 4], 1) + buf[9 + 4];
806 out[(8 - 4)*SBLIMIT] = MULH3(t1, win[8 - 4], 1) + buf[8 - 4];
807 buf[9 + 4] = MULH3(t0, win[18 + 9 + 4], 1);
808 buf[8 - 4] = MULH3(t0, win[18 + 8 - 4], 1);
811 /* return the number of decoded frames */
812 static int mp_decode_layer1(MPADecodeContext *s)
814 int bound, i, v, n, ch, j, mant;
815 uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
816 uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
818 if (s->mode == MPA_JSTEREO)
819 bound = (s->mode_ext + 1) * 4;
823 /* allocation bits */
824 for(i=0;i<bound;i++) {
825 for(ch=0;ch<s->nb_channels;ch++) {
826 allocation[ch][i] = get_bits(&s->gb, 4);
829 for(i=bound;i<SBLIMIT;i++) {
830 allocation[0][i] = get_bits(&s->gb, 4);
834 for(i=0;i<bound;i++) {
835 for(ch=0;ch<s->nb_channels;ch++) {
836 if (allocation[ch][i])
837 scale_factors[ch][i] = get_bits(&s->gb, 6);
840 for(i=bound;i<SBLIMIT;i++) {
841 if (allocation[0][i]) {
842 scale_factors[0][i] = get_bits(&s->gb, 6);
843 scale_factors[1][i] = get_bits(&s->gb, 6);
847 /* compute samples */
849 for(i=0;i<bound;i++) {
850 for(ch=0;ch<s->nb_channels;ch++) {
851 n = allocation[ch][i];
853 mant = get_bits(&s->gb, n + 1);
854 v = l1_unscale(n, mant, scale_factors[ch][i]);
858 s->sb_samples[ch][j][i] = v;
861 for(i=bound;i<SBLIMIT;i++) {
862 n = allocation[0][i];
864 mant = get_bits(&s->gb, n + 1);
865 v = l1_unscale(n, mant, scale_factors[0][i]);
866 s->sb_samples[0][j][i] = v;
867 v = l1_unscale(n, mant, scale_factors[1][i]);
868 s->sb_samples[1][j][i] = v;
870 s->sb_samples[0][j][i] = 0;
871 s->sb_samples[1][j][i] = 0;
878 static int mp_decode_layer2(MPADecodeContext *s)
880 int sblimit; /* number of used subbands */
881 const unsigned char *alloc_table;
882 int table, bit_alloc_bits, i, j, ch, bound, v;
883 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
884 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
885 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
886 int scale, qindex, bits, steps, k, l, m, b;
888 /* select decoding table */
889 table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
890 s->sample_rate, s->lsf);
891 sblimit = ff_mpa_sblimit_table[table];
892 alloc_table = ff_mpa_alloc_tables[table];
894 if (s->mode == MPA_JSTEREO)
895 bound = (s->mode_ext + 1) * 4;
899 av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
902 if( bound > sblimit ) bound = sblimit;
904 /* parse bit allocation */
906 for(i=0;i<bound;i++) {
907 bit_alloc_bits = alloc_table[j];
908 for(ch=0;ch<s->nb_channels;ch++) {
909 bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
911 j += 1 << bit_alloc_bits;
913 for(i=bound;i<sblimit;i++) {
914 bit_alloc_bits = alloc_table[j];
915 v = get_bits(&s->gb, bit_alloc_bits);
918 j += 1 << bit_alloc_bits;
922 for(i=0;i<sblimit;i++) {
923 for(ch=0;ch<s->nb_channels;ch++) {
924 if (bit_alloc[ch][i])
925 scale_code[ch][i] = get_bits(&s->gb, 2);
930 for(i=0;i<sblimit;i++) {
931 for(ch=0;ch<s->nb_channels;ch++) {
932 if (bit_alloc[ch][i]) {
933 sf = scale_factors[ch][i];
934 switch(scale_code[ch][i]) {
937 sf[0] = get_bits(&s->gb, 6);
938 sf[1] = get_bits(&s->gb, 6);
939 sf[2] = get_bits(&s->gb, 6);
942 sf[0] = get_bits(&s->gb, 6);
947 sf[0] = get_bits(&s->gb, 6);
948 sf[2] = get_bits(&s->gb, 6);
952 sf[0] = get_bits(&s->gb, 6);
953 sf[2] = get_bits(&s->gb, 6);
965 for(i=0;i<bound;i++) {
966 bit_alloc_bits = alloc_table[j];
967 for(ch=0;ch<s->nb_channels;ch++) {
968 b = bit_alloc[ch][i];
970 scale = scale_factors[ch][i][k];
971 qindex = alloc_table[j+b];
972 bits = ff_mpa_quant_bits[qindex];
975 /* 3 values at the same time */
976 v = get_bits(&s->gb, -bits);
977 v2 = division_tabs[qindex][v];
978 steps = ff_mpa_quant_steps[qindex];
980 s->sb_samples[ch][k * 12 + l + 0][i] =
981 l2_unscale_group(steps, v2 & 15, scale);
982 s->sb_samples[ch][k * 12 + l + 1][i] =
983 l2_unscale_group(steps, (v2 >> 4) & 15, scale);
984 s->sb_samples[ch][k * 12 + l + 2][i] =
985 l2_unscale_group(steps, v2 >> 8 , scale);
988 v = get_bits(&s->gb, bits);
989 v = l1_unscale(bits - 1, v, scale);
990 s->sb_samples[ch][k * 12 + l + m][i] = v;
994 s->sb_samples[ch][k * 12 + l + 0][i] = 0;
995 s->sb_samples[ch][k * 12 + l + 1][i] = 0;
996 s->sb_samples[ch][k * 12 + l + 2][i] = 0;
999 /* next subband in alloc table */
1000 j += 1 << bit_alloc_bits;
1002 /* XXX: find a way to avoid this duplication of code */
1003 for(i=bound;i<sblimit;i++) {
1004 bit_alloc_bits = alloc_table[j];
1005 b = bit_alloc[0][i];
1007 int mant, scale0, scale1;
1008 scale0 = scale_factors[0][i][k];
1009 scale1 = scale_factors[1][i][k];
1010 qindex = alloc_table[j+b];
1011 bits = ff_mpa_quant_bits[qindex];
1013 /* 3 values at the same time */
1014 v = get_bits(&s->gb, -bits);
1015 steps = ff_mpa_quant_steps[qindex];
1018 s->sb_samples[0][k * 12 + l + 0][i] =
1019 l2_unscale_group(steps, mant, scale0);
1020 s->sb_samples[1][k * 12 + l + 0][i] =
1021 l2_unscale_group(steps, mant, scale1);
1024 s->sb_samples[0][k * 12 + l + 1][i] =
1025 l2_unscale_group(steps, mant, scale0);
1026 s->sb_samples[1][k * 12 + l + 1][i] =
1027 l2_unscale_group(steps, mant, scale1);
1028 s->sb_samples[0][k * 12 + l + 2][i] =
1029 l2_unscale_group(steps, v, scale0);
1030 s->sb_samples[1][k * 12 + l + 2][i] =
1031 l2_unscale_group(steps, v, scale1);
1034 mant = get_bits(&s->gb, bits);
1035 s->sb_samples[0][k * 12 + l + m][i] =
1036 l1_unscale(bits - 1, mant, scale0);
1037 s->sb_samples[1][k * 12 + l + m][i] =
1038 l1_unscale(bits - 1, mant, scale1);
1042 s->sb_samples[0][k * 12 + l + 0][i] = 0;
1043 s->sb_samples[0][k * 12 + l + 1][i] = 0;
1044 s->sb_samples[0][k * 12 + l + 2][i] = 0;
1045 s->sb_samples[1][k * 12 + l + 0][i] = 0;
1046 s->sb_samples[1][k * 12 + l + 1][i] = 0;
1047 s->sb_samples[1][k * 12 + l + 2][i] = 0;
1049 /* next subband in alloc table */
1050 j += 1 << bit_alloc_bits;
1052 /* fill remaining samples to zero */
1053 for(i=sblimit;i<SBLIMIT;i++) {
1054 for(ch=0;ch<s->nb_channels;ch++) {
1055 s->sb_samples[ch][k * 12 + l + 0][i] = 0;
1056 s->sb_samples[ch][k * 12 + l + 1][i] = 0;
1057 s->sb_samples[ch][k * 12 + l + 2][i] = 0;
1065 #define SPLIT(dst,sf,n)\
1067 int m= (sf*171)>>9;\
1074 int m= (sf*205)>>10;\
1078 int m= (sf*171)>>10;\
1085 static av_always_inline void lsf_sf_expand(int *slen,
1086 int sf, int n1, int n2, int n3)
1088 SPLIT(slen[3], sf, n3)
1089 SPLIT(slen[2], sf, n2)
1090 SPLIT(slen[1], sf, n1)
1094 static void exponents_from_scale_factors(MPADecodeContext *s,
1098 const uint8_t *bstab, *pretab;
1099 int len, i, j, k, l, v0, shift, gain, gains[3];
1102 exp_ptr = exponents;
1103 gain = g->global_gain - 210;
1104 shift = g->scalefac_scale + 1;
1106 bstab = band_size_long[s->sample_rate_index];
1107 pretab = mpa_pretab[g->preflag];
1108 for(i=0;i<g->long_end;i++) {
1109 v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
1115 if (g->short_start < 13) {
1116 bstab = band_size_short[s->sample_rate_index];
1117 gains[0] = gain - (g->subblock_gain[0] << 3);
1118 gains[1] = gain - (g->subblock_gain[1] << 3);
1119 gains[2] = gain - (g->subblock_gain[2] << 3);
1121 for(i=g->short_start;i<13;i++) {
1124 v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
1132 /* handle n = 0 too */
1133 static inline int get_bitsz(GetBitContext *s, int n)
1138 return get_bits(s, n);
1142 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_pos2){
1143 if(s->in_gb.buffer && *pos >= s->gb.size_in_bits){
1145 s->in_gb.buffer=NULL;
1146 assert((get_bits_count(&s->gb) & 7) == 0);
1147 skip_bits_long(&s->gb, *pos - *end_pos);
1149 *end_pos= *end_pos2 + get_bits_count(&s->gb) - *pos;
1150 *pos= get_bits_count(&s->gb);
1154 /* Following is a optimized code for
1156 if(get_bits1(&s->gb))
1161 #define READ_FLIP_SIGN(dst,src)\
1162 v = AV_RN32A(src) ^ (get_bits1(&s->gb)<<31);\
1165 #define READ_FLIP_SIGN(dst,src)\
1166 v= -get_bits1(&s->gb);\
1167 *(dst) = (*(src) ^ v) - v;
1170 static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
1171 int16_t *exponents, int end_pos2)
1175 int last_pos, bits_left;
1177 int end_pos= FFMIN(end_pos2, s->gb.size_in_bits);
1179 /* low frequencies (called big values) */
1182 int j, k, l, linbits;
1183 j = g->region_size[i];
1186 /* select vlc table */
1187 k = g->table_select[i];
1188 l = mpa_huff_data[k][0];
1189 linbits = mpa_huff_data[k][1];
1193 memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*2*j);
1198 /* read huffcode and compute each couple */
1202 int pos= get_bits_count(&s->gb);
1204 if (pos >= end_pos){
1205 // av_log(NULL, AV_LOG_ERROR, "pos: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
1206 switch_buffer(s, &pos, &end_pos, &end_pos2);
1207 // av_log(NULL, AV_LOG_ERROR, "new pos: %d %d\n", pos, end_pos);
1211 y = get_vlc2(&s->gb, vlc->table, 7, 3);
1214 g->sb_hybrid[s_index ] =
1215 g->sb_hybrid[s_index+1] = 0;
1220 exponent= exponents[s_index];
1222 av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
1223 i, g->region_size[i] - j, x, y, exponent);
1228 READ_FLIP_SIGN(g->sb_hybrid+s_index, RENAME(expval_table)[ exponent ]+x)
1230 x += get_bitsz(&s->gb, linbits);
1231 v = l3_unscale(x, exponent);
1232 if (get_bits1(&s->gb))
1234 g->sb_hybrid[s_index] = v;
1237 READ_FLIP_SIGN(g->sb_hybrid+s_index+1, RENAME(expval_table)[ exponent ]+y)
1239 y += get_bitsz(&s->gb, linbits);
1240 v = l3_unscale(y, exponent);
1241 if (get_bits1(&s->gb))
1243 g->sb_hybrid[s_index+1] = v;
1250 READ_FLIP_SIGN(g->sb_hybrid+s_index+!!y, RENAME(expval_table)[ exponent ]+x)
1252 x += get_bitsz(&s->gb, linbits);
1253 v = l3_unscale(x, exponent);
1254 if (get_bits1(&s->gb))
1256 g->sb_hybrid[s_index+!!y] = v;
1258 g->sb_hybrid[s_index+ !y] = 0;
1264 /* high frequencies */
1265 vlc = &huff_quad_vlc[g->count1table_select];
1267 while (s_index <= 572) {
1269 pos = get_bits_count(&s->gb);
1270 if (pos >= end_pos) {
1271 if (pos > end_pos2 && last_pos){
1272 /* some encoders generate an incorrect size for this
1273 part. We must go back into the data */
1275 skip_bits_long(&s->gb, last_pos - pos);
1276 av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
1277 if(s->error_recognition >= FF_ER_COMPLIANT)
1281 // av_log(NULL, AV_LOG_ERROR, "pos2: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
1282 switch_buffer(s, &pos, &end_pos, &end_pos2);
1283 // av_log(NULL, AV_LOG_ERROR, "new pos2: %d %d %d\n", pos, end_pos, s_index);
1289 code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
1290 av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
1291 g->sb_hybrid[s_index+0]=
1292 g->sb_hybrid[s_index+1]=
1293 g->sb_hybrid[s_index+2]=
1294 g->sb_hybrid[s_index+3]= 0;
1296 static const int idxtab[16]={3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0};
1298 int pos= s_index+idxtab[code];
1299 code ^= 8>>idxtab[code];
1300 READ_FLIP_SIGN(g->sb_hybrid+pos, RENAME(exp_table)+exponents[pos])
1304 /* skip extension bits */
1305 bits_left = end_pos2 - get_bits_count(&s->gb);
1306 //av_log(NULL, AV_LOG_ERROR, "left:%d buf:%p\n", bits_left, s->in_gb.buffer);
1307 if (bits_left < 0 && s->error_recognition >= FF_ER_COMPLIANT) {
1308 av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
1310 }else if(bits_left > 0 && s->error_recognition >= FF_ER_AGGRESSIVE){
1311 av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
1314 memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*(576 - s_index));
1315 skip_bits_long(&s->gb, bits_left);
1317 i= get_bits_count(&s->gb);
1318 switch_buffer(s, &i, &end_pos, &end_pos2);
1323 /* Reorder short blocks from bitstream order to interleaved order. It
1324 would be faster to do it in parsing, but the code would be far more
1326 static void reorder_block(MPADecodeContext *s, GranuleDef *g)
1329 INTFLOAT *ptr, *dst, *ptr1;
1332 if (g->block_type != 2)
1335 if (g->switch_point) {
1336 if (s->sample_rate_index != 8) {
1337 ptr = g->sb_hybrid + 36;
1339 ptr = g->sb_hybrid + 48;
1345 for(i=g->short_start;i<13;i++) {
1346 len = band_size_short[s->sample_rate_index][i];
1349 for(j=len;j>0;j--) {
1350 *dst++ = ptr[0*len];
1351 *dst++ = ptr[1*len];
1352 *dst++ = ptr[2*len];
1356 memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
1360 #define ISQRT2 FIXR(0.70710678118654752440)
1362 static void compute_stereo(MPADecodeContext *s,
1363 GranuleDef *g0, GranuleDef *g1)
1366 int sf_max, sf, len, non_zero_found;
1367 INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
1368 int non_zero_found_short[3];
1370 /* intensity stereo */
1371 if (s->mode_ext & MODE_EXT_I_STEREO) {
1376 is_tab = is_table_lsf[g1->scalefac_compress & 1];
1380 tab0 = g0->sb_hybrid + 576;
1381 tab1 = g1->sb_hybrid + 576;
1383 non_zero_found_short[0] = 0;
1384 non_zero_found_short[1] = 0;
1385 non_zero_found_short[2] = 0;
1386 k = (13 - g1->short_start) * 3 + g1->long_end - 3;
1387 for(i = 12;i >= g1->short_start;i--) {
1388 /* for last band, use previous scale factor */
1391 len = band_size_short[s->sample_rate_index][i];
1395 if (!non_zero_found_short[l]) {
1396 /* test if non zero band. if so, stop doing i-stereo */
1397 for(j=0;j<len;j++) {
1399 non_zero_found_short[l] = 1;
1403 sf = g1->scale_factors[k + l];
1409 for(j=0;j<len;j++) {
1411 tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1412 tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1416 if (s->mode_ext & MODE_EXT_MS_STEREO) {
1417 /* lower part of the spectrum : do ms stereo
1419 for(j=0;j<len;j++) {
1422 tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1423 tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1430 non_zero_found = non_zero_found_short[0] |
1431 non_zero_found_short[1] |
1432 non_zero_found_short[2];
1434 for(i = g1->long_end - 1;i >= 0;i--) {
1435 len = band_size_long[s->sample_rate_index][i];
1438 /* test if non zero band. if so, stop doing i-stereo */
1439 if (!non_zero_found) {
1440 for(j=0;j<len;j++) {
1446 /* for last band, use previous scale factor */
1447 k = (i == 21) ? 20 : i;
1448 sf = g1->scale_factors[k];
1453 for(j=0;j<len;j++) {
1455 tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1456 tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1460 if (s->mode_ext & MODE_EXT_MS_STEREO) {
1461 /* lower part of the spectrum : do ms stereo
1463 for(j=0;j<len;j++) {
1466 tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1467 tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1472 } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
1473 /* ms stereo ONLY */
1474 /* NOTE: the 1/sqrt(2) normalization factor is included in the
1476 tab0 = g0->sb_hybrid;
1477 tab1 = g1->sb_hybrid;
1478 for(i=0;i<576;i++) {
1481 tab0[i] = tmp0 + tmp1;
1482 tab1[i] = tmp0 - tmp1;
1488 static void compute_antialias_integer(MPADecodeContext *s,
1494 /* we antialias only "long" bands */
1495 if (g->block_type == 2) {
1496 if (!g->switch_point)
1498 /* XXX: check this for 8000Hz case */
1504 ptr = g->sb_hybrid + 18;
1505 for(i = n;i > 0;i--) {
1506 int tmp0, tmp1, tmp2;
1507 csa = &csa_table[0][0];
1511 tmp2= MULH(tmp0 + tmp1, csa[0+4*j]);\
1512 ptr[-1-j] = 4*(tmp2 - MULH(tmp1, csa[2+4*j]));\
1513 ptr[ j] = 4*(tmp2 + MULH(tmp0, csa[3+4*j]));
1529 static void compute_imdct(MPADecodeContext *s,
1531 INTFLOAT *sb_samples,
1534 INTFLOAT *win, *win1, *out_ptr, *ptr, *buf, *ptr1;
1536 int i, j, mdct_long_end, sblimit;
1538 /* find last non zero block */
1539 ptr = g->sb_hybrid + 576;
1540 ptr1 = g->sb_hybrid + 2 * 18;
1541 while (ptr >= ptr1) {
1545 if(p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
1548 sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
1550 if (g->block_type == 2) {
1551 /* XXX: check for 8000 Hz */
1552 if (g->switch_point)
1557 mdct_long_end = sblimit;
1562 for(j=0;j<mdct_long_end;j++) {
1563 /* apply window & overlap with previous buffer */
1564 out_ptr = sb_samples + j;
1566 if (g->switch_point && j < 2)
1569 win1 = mdct_win[g->block_type];
1570 /* select frequency inversion */
1571 win = win1 + ((4 * 36) & -(j & 1));
1572 imdct36(out_ptr, buf, ptr, win);
1573 out_ptr += 18*SBLIMIT;
1577 for(j=mdct_long_end;j<sblimit;j++) {
1578 /* select frequency inversion */
1579 win = mdct_win[2] + ((4 * 36) & -(j & 1));
1580 out_ptr = sb_samples + j;
1586 imdct12(out2, ptr + 0);
1588 *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*1];
1589 buf[i + 6*2] = MULH3(out2[i + 6], win[i + 6], 1);
1592 imdct12(out2, ptr + 1);
1594 *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[i + 6*2];
1595 buf[i + 6*0] = MULH3(out2[i + 6], win[i + 6], 1);
1598 imdct12(out2, ptr + 2);
1600 buf[i + 6*0] = MULH3(out2[i ], win[i ], 1) + buf[i + 6*0];
1601 buf[i + 6*1] = MULH3(out2[i + 6], win[i + 6], 1);
1608 for(j=sblimit;j<SBLIMIT;j++) {
1610 out_ptr = sb_samples + j;
1620 /* main layer3 decoding function */
1621 static int mp_decode_layer3(MPADecodeContext *s)
1623 int nb_granules, main_data_begin, private_bits;
1624 int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
1626 int16_t exponents[576]; //FIXME try INTFLOAT
1628 /* read side info */
1630 main_data_begin = get_bits(&s->gb, 8);
1631 private_bits = get_bits(&s->gb, s->nb_channels);
1634 main_data_begin = get_bits(&s->gb, 9);
1635 if (s->nb_channels == 2)
1636 private_bits = get_bits(&s->gb, 3);
1638 private_bits = get_bits(&s->gb, 5);
1640 for(ch=0;ch<s->nb_channels;ch++) {
1641 s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
1642 s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
1646 for(gr=0;gr<nb_granules;gr++) {
1647 for(ch=0;ch<s->nb_channels;ch++) {
1648 av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
1649 g = &s->granules[ch][gr];
1650 g->part2_3_length = get_bits(&s->gb, 12);
1651 g->big_values = get_bits(&s->gb, 9);
1652 if(g->big_values > 288){
1653 av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
1657 g->global_gain = get_bits(&s->gb, 8);
1658 /* if MS stereo only is selected, we precompute the
1659 1/sqrt(2) renormalization factor */
1660 if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
1662 g->global_gain -= 2;
1664 g->scalefac_compress = get_bits(&s->gb, 9);
1666 g->scalefac_compress = get_bits(&s->gb, 4);
1667 blocksplit_flag = get_bits1(&s->gb);
1668 if (blocksplit_flag) {
1669 g->block_type = get_bits(&s->gb, 2);
1670 if (g->block_type == 0){
1671 av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
1674 g->switch_point = get_bits1(&s->gb);
1676 g->table_select[i] = get_bits(&s->gb, 5);
1678 g->subblock_gain[i] = get_bits(&s->gb, 3);
1679 ff_init_short_region(s, g);
1681 int region_address1, region_address2;
1683 g->switch_point = 0;
1685 g->table_select[i] = get_bits(&s->gb, 5);
1686 /* compute huffman coded region sizes */
1687 region_address1 = get_bits(&s->gb, 4);
1688 region_address2 = get_bits(&s->gb, 3);
1689 av_dlog(s->avctx, "region1=%d region2=%d\n",
1690 region_address1, region_address2);
1691 ff_init_long_region(s, g, region_address1, region_address2);
1693 ff_region_offset2size(g);
1694 ff_compute_band_indexes(s, g);
1698 g->preflag = get_bits1(&s->gb);
1699 g->scalefac_scale = get_bits1(&s->gb);
1700 g->count1table_select = get_bits1(&s->gb);
1701 av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
1702 g->block_type, g->switch_point);
1707 const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
1708 assert((get_bits_count(&s->gb) & 7) == 0);
1709 /* now we get bits from the main_data_begin offset */
1710 av_dlog(s->avctx, "seekback: %d\n", main_data_begin);
1711 //av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size);
1713 memcpy(s->last_buf + s->last_buf_size, ptr, EXTRABYTES);
1715 init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
1716 skip_bits_long(&s->gb, 8*(s->last_buf_size - main_data_begin));
1719 for(gr=0;gr<nb_granules;gr++) {
1720 for(ch=0;ch<s->nb_channels;ch++) {
1721 g = &s->granules[ch][gr];
1722 if(get_bits_count(&s->gb)<0){
1723 av_log(s->avctx, AV_LOG_DEBUG, "mdb:%d, lastbuf:%d skipping granule %d\n",
1724 main_data_begin, s->last_buf_size, gr);
1725 skip_bits_long(&s->gb, g->part2_3_length);
1726 memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
1727 if(get_bits_count(&s->gb) >= s->gb.size_in_bits && s->in_gb.buffer){
1728 skip_bits_long(&s->in_gb, get_bits_count(&s->gb) - s->gb.size_in_bits);
1730 s->in_gb.buffer=NULL;
1735 bits_pos = get_bits_count(&s->gb);
1739 int slen, slen1, slen2;
1741 /* MPEG1 scale factors */
1742 slen1 = slen_table[0][g->scalefac_compress];
1743 slen2 = slen_table[1][g->scalefac_compress];
1744 av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
1745 if (g->block_type == 2) {
1746 n = g->switch_point ? 17 : 18;
1750 g->scale_factors[j++] = get_bits(&s->gb, slen1);
1753 g->scale_factors[j++] = 0;
1757 g->scale_factors[j++] = get_bits(&s->gb, slen2);
1759 g->scale_factors[j++] = 0;
1762 g->scale_factors[j++] = 0;
1765 sc = s->granules[ch][0].scale_factors;
1768 n = (k == 0 ? 6 : 5);
1769 if ((g->scfsi & (0x8 >> k)) == 0) {
1770 slen = (k < 2) ? slen1 : slen2;
1773 g->scale_factors[j++] = get_bits(&s->gb, slen);
1776 g->scale_factors[j++] = 0;
1779 /* simply copy from last granule */
1781 g->scale_factors[j] = sc[j];
1786 g->scale_factors[j++] = 0;
1789 int tindex, tindex2, slen[4], sl, sf;
1791 /* LSF scale factors */
1792 if (g->block_type == 2) {
1793 tindex = g->switch_point ? 2 : 1;
1797 sf = g->scalefac_compress;
1798 if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
1799 /* intensity stereo case */
1802 lsf_sf_expand(slen, sf, 6, 6, 0);
1804 } else if (sf < 244) {
1805 lsf_sf_expand(slen, sf - 180, 4, 4, 0);
1808 lsf_sf_expand(slen, sf - 244, 3, 0, 0);
1814 lsf_sf_expand(slen, sf, 5, 4, 4);
1816 } else if (sf < 500) {
1817 lsf_sf_expand(slen, sf - 400, 5, 4, 0);
1820 lsf_sf_expand(slen, sf - 500, 3, 0, 0);
1828 n = lsf_nsf_table[tindex2][tindex][k];
1832 g->scale_factors[j++] = get_bits(&s->gb, sl);
1835 g->scale_factors[j++] = 0;
1838 /* XXX: should compute exact size */
1840 g->scale_factors[j] = 0;
1843 exponents_from_scale_factors(s, g, exponents);
1845 /* read Huffman coded residue */
1846 huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
1849 if (s->nb_channels == 2)
1850 compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
1852 for(ch=0;ch<s->nb_channels;ch++) {
1853 g = &s->granules[ch][gr];
1855 reorder_block(s, g);
1856 compute_antialias(s, g);
1857 compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1860 if(get_bits_count(&s->gb)<0)
1861 skip_bits_long(&s->gb, -get_bits_count(&s->gb));
1862 return nb_granules * 18;
1865 static int mp_decode_frame(MPADecodeContext *s,
1866 OUT_INT *samples, const uint8_t *buf, int buf_size)
1868 int i, nb_frames, ch;
1869 OUT_INT *samples_ptr;
1871 init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE)*8);
1873 /* skip error protection field */
1874 if (s->error_protection)
1875 skip_bits(&s->gb, 16);
1877 av_dlog(s->avctx, "frame %d:\n", s->frame_count);
1880 s->avctx->frame_size = 384;
1881 nb_frames = mp_decode_layer1(s);
1884 s->avctx->frame_size = 1152;
1885 nb_frames = mp_decode_layer2(s);
1888 s->avctx->frame_size = s->lsf ? 576 : 1152;
1890 nb_frames = mp_decode_layer3(s);
1893 if(s->in_gb.buffer){
1894 align_get_bits(&s->gb);
1895 i= get_bits_left(&s->gb)>>3;
1896 if(i >= 0 && i <= BACKSTEP_SIZE){
1897 memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
1900 av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
1902 s->in_gb.buffer= NULL;
1905 align_get_bits(&s->gb);
1906 assert((get_bits_count(&s->gb) & 7) == 0);
1907 i= get_bits_left(&s->gb)>>3;
1909 if(i<0 || i > BACKSTEP_SIZE || nb_frames<0){
1911 av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
1912 i= FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
1914 assert(i <= buf_size - HEADER_SIZE && i>= 0);
1915 memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
1916 s->last_buf_size += i;
1921 /* apply the synthesis filter */
1922 for(ch=0;ch<s->nb_channels;ch++) {
1923 samples_ptr = samples + ch;
1924 for(i=0;i<nb_frames;i++) {
1925 RENAME(ff_mpa_synth_filter)(
1929 s->synth_buf[ch], &(s->synth_buf_offset[ch]),
1930 RENAME(ff_mpa_synth_window), &s->dither_state,
1931 samples_ptr, s->nb_channels,
1932 s->sb_samples[ch][i]);
1933 samples_ptr += 32 * s->nb_channels;
1937 return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
1940 static int decode_frame(AVCodecContext * avctx,
1941 void *data, int *data_size,
1944 const uint8_t *buf = avpkt->data;
1945 int buf_size = avpkt->size;
1946 MPADecodeContext *s = avctx->priv_data;
1949 OUT_INT *out_samples = data;
1951 if(buf_size < HEADER_SIZE)
1954 header = AV_RB32(buf);
1955 if(ff_mpa_check_header(header) < 0){
1956 av_log(avctx, AV_LOG_ERROR, "Header missing\n");
1960 if (ff_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
1961 /* free format: prepare to compute frame size */
1965 /* update codec info */
1966 avctx->channels = s->nb_channels;
1967 avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1968 if (!avctx->bit_rate)
1969 avctx->bit_rate = s->bit_rate;
1970 avctx->sub_id = s->layer;
1972 if(*data_size < 1152*avctx->channels*sizeof(OUT_INT))
1976 if(s->frame_size<=0 || s->frame_size > buf_size){
1977 av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
1979 }else if(s->frame_size < buf_size){
1980 av_log(avctx, AV_LOG_ERROR, "incorrect frame size\n");
1981 buf_size= s->frame_size;
1984 out_size = mp_decode_frame(s, out_samples, buf, buf_size);
1986 *data_size = out_size;
1987 avctx->sample_rate = s->sample_rate;
1988 //FIXME maybe move the other codec info stuff from above here too
1990 av_log(avctx, AV_LOG_DEBUG, "Error while decoding MPEG audio frame.\n"); //FIXME return -1 / but also return the number of bytes consumed
1995 static void flush(AVCodecContext *avctx){
1996 MPADecodeContext *s = avctx->priv_data;
1997 memset(s->synth_buf, 0, sizeof(s->synth_buf));
1998 s->last_buf_size= 0;
2001 #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
2002 static int decode_frame_adu(AVCodecContext * avctx,
2003 void *data, int *data_size,
2006 const uint8_t *buf = avpkt->data;
2007 int buf_size = avpkt->size;
2008 MPADecodeContext *s = avctx->priv_data;
2011 OUT_INT *out_samples = data;
2015 // Discard too short frames
2016 if (buf_size < HEADER_SIZE) {
2022 if (len > MPA_MAX_CODED_FRAME_SIZE)
2023 len = MPA_MAX_CODED_FRAME_SIZE;
2025 // Get header and restore sync word
2026 header = AV_RB32(buf) | 0xffe00000;
2028 if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
2033 ff_mpegaudio_decode_header((MPADecodeHeader *)s, header);
2034 /* update codec info */
2035 avctx->sample_rate = s->sample_rate;
2036 avctx->channels = s->nb_channels;
2037 if (!avctx->bit_rate)
2038 avctx->bit_rate = s->bit_rate;
2039 avctx->sub_id = s->layer;
2041 s->frame_size = len;
2043 if (avctx->parse_only) {
2044 out_size = buf_size;
2046 out_size = mp_decode_frame(s, out_samples, buf, buf_size);
2049 *data_size = out_size;
2052 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
2054 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
2057 * Context for MP3On4 decoder
2059 typedef struct MP3On4DecodeContext {
2060 int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
2061 int syncword; ///< syncword patch
2062 const uint8_t *coff; ///< channels offsets in output buffer
2063 MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
2064 } MP3On4DecodeContext;
2066 #include "mpeg4audio.h"
2068 /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
2069 static const uint8_t mp3Frames[8] = {0,1,1,2,3,3,4,5}; /* number of mp3 decoder instances */
2070 /* offsets into output buffer, assume output order is FL FR BL BR C LFE */
2071 static const uint8_t chan_offset[8][5] = {
2076 {2,0,3}, // C FLR BS
2077 {4,0,2}, // C FLR BLRS
2078 {4,0,2,5}, // C FLR BLRS LFE
2079 {4,0,2,6,5}, // C FLR BLRS BLR LFE
2083 static int decode_init_mp3on4(AVCodecContext * avctx)
2085 MP3On4DecodeContext *s = avctx->priv_data;
2086 MPEG4AudioConfig cfg;
2089 if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
2090 av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
2094 ff_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size);
2095 if (!cfg.chan_config || cfg.chan_config > 7) {
2096 av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
2099 s->frames = mp3Frames[cfg.chan_config];
2100 s->coff = chan_offset[cfg.chan_config];
2101 avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
2103 if (cfg.sample_rate < 16000)
2104 s->syncword = 0xffe00000;
2106 s->syncword = 0xfff00000;
2108 /* Init the first mp3 decoder in standard way, so that all tables get builded
2109 * We replace avctx->priv_data with the context of the first decoder so that
2110 * decode_init() does not have to be changed.
2111 * Other decoders will be initialized here copying data from the first context
2113 // Allocate zeroed memory for the first decoder context
2114 s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
2115 // Put decoder context in place to make init_decode() happy
2116 avctx->priv_data = s->mp3decctx[0];
2118 // Restore mp3on4 context pointer
2119 avctx->priv_data = s;
2120 s->mp3decctx[0]->adu_mode = 1; // Set adu mode
2122 /* Create a separate codec/context for each frame (first is already ok).
2123 * Each frame is 1 or 2 channels - up to 5 frames allowed
2125 for (i = 1; i < s->frames; i++) {
2126 s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
2127 s->mp3decctx[i]->adu_mode = 1;
2128 s->mp3decctx[i]->avctx = avctx;
2135 static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
2137 MP3On4DecodeContext *s = avctx->priv_data;
2140 for (i = 0; i < s->frames; i++)
2141 av_free(s->mp3decctx[i]);
2147 static int decode_frame_mp3on4(AVCodecContext * avctx,
2148 void *data, int *data_size,
2151 const uint8_t *buf = avpkt->data;
2152 int buf_size = avpkt->size;
2153 MP3On4DecodeContext *s = avctx->priv_data;
2154 MPADecodeContext *m;
2155 int fsize, len = buf_size, out_size = 0;
2157 OUT_INT *out_samples = data;
2158 OUT_INT decoded_buf[MPA_FRAME_SIZE * MPA_MAX_CHANNELS];
2159 OUT_INT *outptr, *bp;
2162 if(*data_size < MPA_FRAME_SIZE * MPA_MAX_CHANNELS * s->frames * sizeof(OUT_INT))
2166 // Discard too short frames
2167 if (buf_size < HEADER_SIZE)
2170 // If only one decoder interleave is not needed
2171 outptr = s->frames == 1 ? out_samples : decoded_buf;
2173 avctx->bit_rate = 0;
2175 for (fr = 0; fr < s->frames; fr++) {
2176 fsize = AV_RB16(buf) >> 4;
2177 fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
2178 m = s->mp3decctx[fr];
2181 header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
2183 if (ff_mpa_check_header(header) < 0) // Bad header, discard block
2186 ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
2187 out_size += mp_decode_frame(m, outptr, buf, fsize);
2192 n = m->avctx->frame_size*m->nb_channels;
2193 /* interleave output data */
2194 bp = out_samples + s->coff[fr];
2195 if(m->nb_channels == 1) {
2196 for(j = 0; j < n; j++) {
2197 *bp = decoded_buf[j];
2198 bp += avctx->channels;
2201 for(j = 0; j < n; j++) {
2202 bp[0] = decoded_buf[j++];
2203 bp[1] = decoded_buf[j];
2204 bp += avctx->channels;
2208 avctx->bit_rate += m->bit_rate;
2211 /* update codec info */
2212 avctx->sample_rate = s->mp3decctx[0]->sample_rate;
2214 *data_size = out_size;
2217 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
2220 #if CONFIG_MP1_DECODER
2221 AVCodec ff_mp1_decoder =
2226 sizeof(MPADecodeContext),
2231 CODEC_CAP_PARSE_ONLY,
2233 .long_name= NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
2236 #if CONFIG_MP2_DECODER
2237 AVCodec ff_mp2_decoder =
2242 sizeof(MPADecodeContext),
2247 CODEC_CAP_PARSE_ONLY,
2249 .long_name= NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
2252 #if CONFIG_MP3_DECODER
2253 AVCodec ff_mp3_decoder =
2258 sizeof(MPADecodeContext),
2263 CODEC_CAP_PARSE_ONLY,
2265 .long_name= NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
2268 #if CONFIG_MP3ADU_DECODER
2269 AVCodec ff_mp3adu_decoder =
2274 sizeof(MPADecodeContext),
2279 CODEC_CAP_PARSE_ONLY,
2281 .long_name= NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
2284 #if CONFIG_MP3ON4_DECODER
2285 AVCodec ff_mp3on4_decoder =
2290 sizeof(MP3On4DecodeContext),
2293 decode_close_mp3on4,
2294 decode_frame_mp3on4,
2296 .long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"),