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/avassert.h"
28 #include "libavutil/channel_layout.h"
29 #include "libavutil/float_dsp.h"
34 #include "mpegaudiodsp.h"
39 * - test lsf / mpeg25 extensively.
42 #include "mpegaudio.h"
43 #include "mpegaudiodecheader.h"
45 #define BACKSTEP_SIZE 512
47 #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
49 /* layer 3 "granule" */
50 typedef struct GranuleDef {
55 int scalefac_compress;
60 uint8_t scalefac_scale;
61 uint8_t count1table_select;
62 int region_size[3]; /* number of huffman codes in each region */
64 int short_start, long_end; /* long/short band indexes */
65 uint8_t scale_factors[40];
66 DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
69 typedef struct MPADecodeContext {
71 uint8_t last_buf[LAST_BUF_SIZE];
73 /* next header (used in free format parsing) */
74 uint32_t free_format_next_header;
77 DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
78 int synth_buf_offset[MPA_MAX_CHANNELS];
79 DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
80 INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
81 GranuleDef granules[2][2]; /* Used in Layer 3 */
82 int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
85 AVCodecContext* avctx;
87 AVFloatDSPContext fdsp;
92 # define SHR(a,b) ((a)*(1.0f/(1<<(b))))
93 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
94 # define FIXR(x) ((float)(x))
95 # define FIXHR(x) ((float)(x))
96 # define MULH3(x, y, s) ((s)*(y)*(x))
97 # define MULLx(x, y, s) ((y)*(x))
98 # define RENAME(a) a ## _float
99 # define OUT_FMT AV_SAMPLE_FMT_FLT
100 # define OUT_FMT_P AV_SAMPLE_FMT_FLTP
102 # define SHR(a,b) ((a)>>(b))
103 /* WARNING: only correct for positive numbers */
104 # define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
105 # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
106 # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
107 # define MULH3(x, y, s) MULH((s)*(x), y)
108 # define MULLx(x, y, s) MULL(x,y,s)
109 # define RENAME(a) a ## _fixed
110 # define OUT_FMT AV_SAMPLE_FMT_S16
111 # define OUT_FMT_P AV_SAMPLE_FMT_S16P
116 #define HEADER_SIZE 4
118 #include "mpegaudiodata.h"
119 #include "mpegaudiodectab.h"
121 /* vlc structure for decoding layer 3 huffman tables */
122 static VLC huff_vlc[16];
123 static VLC_TYPE huff_vlc_tables[
124 0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
125 142 + 204 + 190 + 170 + 542 + 460 + 662 + 414
127 static const int huff_vlc_tables_sizes[16] = {
128 0, 128, 128, 128, 130, 128, 154, 166,
129 142, 204, 190, 170, 542, 460, 662, 414
131 static VLC huff_quad_vlc[2];
132 static VLC_TYPE huff_quad_vlc_tables[128+16][2];
133 static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };
134 /* computed from band_size_long */
135 static uint16_t band_index_long[9][23];
136 #include "mpegaudio_tablegen.h"
137 /* intensity stereo coef table */
138 static INTFLOAT is_table[2][16];
139 static INTFLOAT is_table_lsf[2][2][16];
140 static INTFLOAT csa_table[8][4];
142 static int16_t division_tab3[1<<6 ];
143 static int16_t division_tab5[1<<8 ];
144 static int16_t division_tab9[1<<11];
146 static int16_t * const division_tabs[4] = {
147 division_tab3, division_tab5, NULL, division_tab9
150 /* lower 2 bits: modulo 3, higher bits: shift */
151 static uint16_t scale_factor_modshift[64];
152 /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
153 static int32_t scale_factor_mult[15][3];
154 /* mult table for layer 2 group quantization */
156 #define SCALE_GEN(v) \
157 { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
159 static const int32_t scale_factor_mult2[3][3] = {
160 SCALE_GEN(4.0 / 3.0), /* 3 steps */
161 SCALE_GEN(4.0 / 5.0), /* 5 steps */
162 SCALE_GEN(4.0 / 9.0), /* 9 steps */
166 * Convert region offsets to region sizes and truncate
167 * size to big_values.
169 static void ff_region_offset2size(GranuleDef *g)
172 g->region_size[2] = 576 / 2;
173 for (i = 0; i < 3; i++) {
174 k = FFMIN(g->region_size[i], g->big_values);
175 g->region_size[i] = k - j;
180 static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g)
182 if (g->block_type == 2) {
183 if (s->sample_rate_index != 8)
184 g->region_size[0] = (36 / 2);
186 g->region_size[0] = (72 / 2);
188 if (s->sample_rate_index <= 2)
189 g->region_size[0] = (36 / 2);
190 else if (s->sample_rate_index != 8)
191 g->region_size[0] = (54 / 2);
193 g->region_size[0] = (108 / 2);
195 g->region_size[1] = (576 / 2);
198 static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2)
201 g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
202 /* should not overflow */
203 l = FFMIN(ra1 + ra2 + 2, 22);
204 g->region_size[1] = band_index_long[s->sample_rate_index][ l] >> 1;
207 static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
209 if (g->block_type == 2) {
210 if (g->switch_point) {
211 /* if switched mode, we handle the 36 first samples as
212 long blocks. For 8000Hz, we handle the 72 first
213 exponents as long blocks */
214 if (s->sample_rate_index <= 2)
230 /* layer 1 unscaling */
231 /* n = number of bits of the mantissa minus 1 */
232 static inline int l1_unscale(int n, int mant, int scale_factor)
237 shift = scale_factor_modshift[scale_factor];
240 val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
242 /* NOTE: at this point, 1 <= shift >= 21 + 15 */
243 return (int)((val + (1LL << (shift - 1))) >> shift);
246 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
250 shift = scale_factor_modshift[scale_factor];
254 val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
255 /* NOTE: at this point, 0 <= shift <= 21 */
257 val = (val + (1 << (shift - 1))) >> shift;
261 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
262 static inline int l3_unscale(int value, int exponent)
267 e = table_4_3_exp [4 * value + (exponent & 3)];
268 m = table_4_3_value[4 * value + (exponent & 3)];
273 m = (m + (1 << (e - 1))) >> e;
278 static av_cold void decode_init_static(void)
283 /* scale factors table for layer 1/2 */
284 for (i = 0; i < 64; i++) {
286 /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
289 scale_factor_modshift[i] = mod | (shift << 2);
292 /* scale factor multiply for layer 1 */
293 for (i = 0; i < 15; i++) {
296 norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
297 scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
298 scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
299 scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
300 av_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm,
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 */
310 for (i = 1; i < 16; i++) {
311 const HuffTable *h = &mpa_huff_tables[i];
313 uint8_t tmp_bits [512] = { 0 };
314 uint16_t tmp_codes[512] = { 0 };
319 for (x = 0; x < xsize; x++) {
320 for (y = 0; y < xsize; y++) {
321 tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j ];
322 tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];
327 huff_vlc[i].table = huff_vlc_tables+offset;
328 huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
329 init_vlc(&huff_vlc[i], 7, 512,
330 tmp_bits, 1, 1, tmp_codes, 2, 2,
331 INIT_VLC_USE_NEW_STATIC);
332 offset += huff_vlc_tables_sizes[i];
334 assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
337 for (i = 0; i < 2; i++) {
338 huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
339 huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
340 init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
341 mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
342 INIT_VLC_USE_NEW_STATIC);
343 offset += huff_quad_vlc_tables_sizes[i];
345 assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
347 for (i = 0; i < 9; i++) {
349 for (j = 0; j < 22; j++) {
350 band_index_long[i][j] = k;
351 k += band_size_long[i][j];
353 band_index_long[i][22] = k;
356 /* compute n ^ (4/3) and store it in mantissa/exp format */
358 mpegaudio_tableinit();
360 for (i = 0; i < 4; i++) {
361 if (ff_mpa_quant_bits[i] < 0) {
362 for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
363 int val1, val2, val3, steps;
365 steps = ff_mpa_quant_steps[i];
370 division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);
376 for (i = 0; i < 7; i++) {
380 f = tan((double)i * M_PI / 12.0);
381 v = FIXR(f / (1.0 + f));
386 is_table[1][6 - i] = v;
389 for (i = 7; i < 16; i++)
390 is_table[0][i] = is_table[1][i] = 0.0;
392 for (i = 0; i < 16; i++) {
396 for (j = 0; j < 2; j++) {
397 e = -(j + 1) * ((i + 1) >> 1);
398 f = pow(2.0, e / 4.0);
400 is_table_lsf[j][k ^ 1][i] = FIXR(f);
401 is_table_lsf[j][k ][i] = FIXR(1.0);
402 av_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
403 i, j, (float) is_table_lsf[j][0][i],
404 (float) is_table_lsf[j][1][i]);
408 for (i = 0; i < 8; i++) {
411 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);
419 csa_table[i][0] = cs;
420 csa_table[i][1] = ca;
421 csa_table[i][2] = ca + cs;
422 csa_table[i][3] = ca - cs;
427 static av_cold int decode_init(AVCodecContext * avctx)
429 static int initialized_tables = 0;
430 MPADecodeContext *s = avctx->priv_data;
432 if (!initialized_tables) {
433 decode_init_static();
434 initialized_tables = 1;
439 avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
440 ff_mpadsp_init(&s->mpadsp);
442 if (avctx->request_sample_fmt == OUT_FMT &&
443 avctx->codec_id != AV_CODEC_ID_MP3ON4)
444 avctx->sample_fmt = OUT_FMT;
446 avctx->sample_fmt = OUT_FMT_P;
447 s->err_recognition = avctx->err_recognition;
449 if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
455 #define C3 FIXHR(0.86602540378443864676/2)
456 #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
457 #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
458 #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
460 /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
462 static void imdct12(INTFLOAT *out, INTFLOAT *in)
464 INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
467 in1 = in[1*3] + in[0*3];
468 in2 = in[2*3] + in[1*3];
469 in3 = in[3*3] + in[2*3];
470 in4 = in[4*3] + in[3*3];
471 in5 = in[5*3] + in[4*3];
475 in2 = MULH3(in2, C3, 2);
476 in3 = MULH3(in3, C3, 4);
479 t2 = MULH3(in1 - in5, C4, 2);
489 in1 = MULH3(in5 + in3, C5, 1);
496 in5 = MULH3(in5 - in3, C6, 2);
503 /* return the number of decoded frames */
504 static int mp_decode_layer1(MPADecodeContext *s)
506 int bound, i, v, n, ch, j, mant;
507 uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
508 uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
510 if (s->mode == MPA_JSTEREO)
511 bound = (s->mode_ext + 1) * 4;
515 /* allocation bits */
516 for (i = 0; i < bound; i++) {
517 for (ch = 0; ch < s->nb_channels; ch++) {
518 allocation[ch][i] = get_bits(&s->gb, 4);
521 for (i = bound; i < SBLIMIT; i++)
522 allocation[0][i] = get_bits(&s->gb, 4);
525 for (i = 0; i < bound; i++) {
526 for (ch = 0; ch < s->nb_channels; ch++) {
527 if (allocation[ch][i])
528 scale_factors[ch][i] = get_bits(&s->gb, 6);
531 for (i = bound; i < SBLIMIT; i++) {
532 if (allocation[0][i]) {
533 scale_factors[0][i] = get_bits(&s->gb, 6);
534 scale_factors[1][i] = get_bits(&s->gb, 6);
538 /* compute samples */
539 for (j = 0; j < 12; j++) {
540 for (i = 0; i < bound; i++) {
541 for (ch = 0; ch < s->nb_channels; ch++) {
542 n = allocation[ch][i];
544 mant = get_bits(&s->gb, n + 1);
545 v = l1_unscale(n, mant, scale_factors[ch][i]);
549 s->sb_samples[ch][j][i] = v;
552 for (i = bound; i < SBLIMIT; i++) {
553 n = allocation[0][i];
555 mant = get_bits(&s->gb, n + 1);
556 v = l1_unscale(n, mant, scale_factors[0][i]);
557 s->sb_samples[0][j][i] = v;
558 v = l1_unscale(n, mant, scale_factors[1][i]);
559 s->sb_samples[1][j][i] = v;
561 s->sb_samples[0][j][i] = 0;
562 s->sb_samples[1][j][i] = 0;
569 static int mp_decode_layer2(MPADecodeContext *s)
571 int sblimit; /* number of used subbands */
572 const unsigned char *alloc_table;
573 int table, bit_alloc_bits, i, j, ch, bound, v;
574 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
575 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
576 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
577 int scale, qindex, bits, steps, k, l, m, b;
579 /* select decoding table */
580 table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
581 s->sample_rate, s->lsf);
582 sblimit = ff_mpa_sblimit_table[table];
583 alloc_table = ff_mpa_alloc_tables[table];
585 if (s->mode == MPA_JSTEREO)
586 bound = (s->mode_ext + 1) * 4;
590 av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
596 /* parse bit allocation */
598 for (i = 0; i < bound; i++) {
599 bit_alloc_bits = alloc_table[j];
600 for (ch = 0; ch < s->nb_channels; ch++)
601 bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
602 j += 1 << bit_alloc_bits;
604 for (i = bound; i < sblimit; i++) {
605 bit_alloc_bits = alloc_table[j];
606 v = get_bits(&s->gb, bit_alloc_bits);
609 j += 1 << bit_alloc_bits;
613 for (i = 0; i < sblimit; i++) {
614 for (ch = 0; ch < s->nb_channels; ch++) {
615 if (bit_alloc[ch][i])
616 scale_code[ch][i] = get_bits(&s->gb, 2);
621 for (i = 0; i < sblimit; i++) {
622 for (ch = 0; ch < s->nb_channels; ch++) {
623 if (bit_alloc[ch][i]) {
624 sf = scale_factors[ch][i];
625 switch (scale_code[ch][i]) {
628 sf[0] = get_bits(&s->gb, 6);
629 sf[1] = get_bits(&s->gb, 6);
630 sf[2] = get_bits(&s->gb, 6);
633 sf[0] = get_bits(&s->gb, 6);
638 sf[0] = get_bits(&s->gb, 6);
639 sf[2] = get_bits(&s->gb, 6);
643 sf[0] = get_bits(&s->gb, 6);
644 sf[2] = get_bits(&s->gb, 6);
653 for (k = 0; k < 3; k++) {
654 for (l = 0; l < 12; l += 3) {
656 for (i = 0; i < bound; i++) {
657 bit_alloc_bits = alloc_table[j];
658 for (ch = 0; ch < s->nb_channels; ch++) {
659 b = bit_alloc[ch][i];
661 scale = scale_factors[ch][i][k];
662 qindex = alloc_table[j+b];
663 bits = ff_mpa_quant_bits[qindex];
666 /* 3 values at the same time */
667 v = get_bits(&s->gb, -bits);
668 v2 = division_tabs[qindex][v];
669 steps = ff_mpa_quant_steps[qindex];
671 s->sb_samples[ch][k * 12 + l + 0][i] =
672 l2_unscale_group(steps, v2 & 15, scale);
673 s->sb_samples[ch][k * 12 + l + 1][i] =
674 l2_unscale_group(steps, (v2 >> 4) & 15, scale);
675 s->sb_samples[ch][k * 12 + l + 2][i] =
676 l2_unscale_group(steps, v2 >> 8 , scale);
678 for (m = 0; m < 3; m++) {
679 v = get_bits(&s->gb, bits);
680 v = l1_unscale(bits - 1, v, scale);
681 s->sb_samples[ch][k * 12 + l + m][i] = v;
685 s->sb_samples[ch][k * 12 + l + 0][i] = 0;
686 s->sb_samples[ch][k * 12 + l + 1][i] = 0;
687 s->sb_samples[ch][k * 12 + l + 2][i] = 0;
690 /* next subband in alloc table */
691 j += 1 << bit_alloc_bits;
693 /* XXX: find a way to avoid this duplication of code */
694 for (i = bound; i < sblimit; i++) {
695 bit_alloc_bits = alloc_table[j];
698 int mant, scale0, scale1;
699 scale0 = scale_factors[0][i][k];
700 scale1 = scale_factors[1][i][k];
701 qindex = alloc_table[j+b];
702 bits = ff_mpa_quant_bits[qindex];
704 /* 3 values at the same time */
705 v = get_bits(&s->gb, -bits);
706 steps = ff_mpa_quant_steps[qindex];
709 s->sb_samples[0][k * 12 + l + 0][i] =
710 l2_unscale_group(steps, mant, scale0);
711 s->sb_samples[1][k * 12 + l + 0][i] =
712 l2_unscale_group(steps, mant, scale1);
715 s->sb_samples[0][k * 12 + l + 1][i] =
716 l2_unscale_group(steps, mant, scale0);
717 s->sb_samples[1][k * 12 + l + 1][i] =
718 l2_unscale_group(steps, mant, scale1);
719 s->sb_samples[0][k * 12 + l + 2][i] =
720 l2_unscale_group(steps, v, scale0);
721 s->sb_samples[1][k * 12 + l + 2][i] =
722 l2_unscale_group(steps, v, scale1);
724 for (m = 0; m < 3; m++) {
725 mant = get_bits(&s->gb, bits);
726 s->sb_samples[0][k * 12 + l + m][i] =
727 l1_unscale(bits - 1, mant, scale0);
728 s->sb_samples[1][k * 12 + l + m][i] =
729 l1_unscale(bits - 1, mant, scale1);
733 s->sb_samples[0][k * 12 + l + 0][i] = 0;
734 s->sb_samples[0][k * 12 + l + 1][i] = 0;
735 s->sb_samples[0][k * 12 + l + 2][i] = 0;
736 s->sb_samples[1][k * 12 + l + 0][i] = 0;
737 s->sb_samples[1][k * 12 + l + 1][i] = 0;
738 s->sb_samples[1][k * 12 + l + 2][i] = 0;
740 /* next subband in alloc table */
741 j += 1 << bit_alloc_bits;
743 /* fill remaining samples to zero */
744 for (i = sblimit; i < SBLIMIT; i++) {
745 for (ch = 0; ch < s->nb_channels; ch++) {
746 s->sb_samples[ch][k * 12 + l + 0][i] = 0;
747 s->sb_samples[ch][k * 12 + l + 1][i] = 0;
748 s->sb_samples[ch][k * 12 + l + 2][i] = 0;
756 #define SPLIT(dst,sf,n) \
758 int m = (sf * 171) >> 9; \
761 } else if (n == 4) { \
764 } else if (n == 5) { \
765 int m = (sf * 205) >> 10; \
768 } else if (n == 6) { \
769 int m = (sf * 171) >> 10; \
776 static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
779 SPLIT(slen[3], sf, n3)
780 SPLIT(slen[2], sf, n2)
781 SPLIT(slen[1], sf, n1)
785 static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g,
788 const uint8_t *bstab, *pretab;
789 int len, i, j, k, l, v0, shift, gain, gains[3];
793 gain = g->global_gain - 210;
794 shift = g->scalefac_scale + 1;
796 bstab = band_size_long[s->sample_rate_index];
797 pretab = mpa_pretab[g->preflag];
798 for (i = 0; i < g->long_end; i++) {
799 v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
801 for (j = len; j > 0; j--)
805 if (g->short_start < 13) {
806 bstab = band_size_short[s->sample_rate_index];
807 gains[0] = gain - (g->subblock_gain[0] << 3);
808 gains[1] = gain - (g->subblock_gain[1] << 3);
809 gains[2] = gain - (g->subblock_gain[2] << 3);
811 for (i = g->short_start; i < 13; i++) {
813 for (l = 0; l < 3; l++) {
814 v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
815 for (j = len; j > 0; j--)
822 /* handle n = 0 too */
823 static inline int get_bitsz(GetBitContext *s, int n)
825 return n ? get_bits(s, n) : 0;
829 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
832 if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) {
834 s->in_gb.buffer = NULL;
835 assert((get_bits_count(&s->gb) & 7) == 0);
836 skip_bits_long(&s->gb, *pos - *end_pos);
838 *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
839 *pos = get_bits_count(&s->gb);
843 /* Following is a optimized code for
845 if(get_bits1(&s->gb))
850 #define READ_FLIP_SIGN(dst,src) \
851 v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
854 #define READ_FLIP_SIGN(dst,src) \
855 v = -get_bits1(&s->gb); \
856 *(dst) = (*(src) ^ v) - v;
859 static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
860 int16_t *exponents, int end_pos2)
864 int last_pos, bits_left;
866 int end_pos = FFMIN(end_pos2, s->gb.size_in_bits);
868 /* low frequencies (called big values) */
870 for (i = 0; i < 3; i++) {
871 int j, k, l, linbits;
872 j = g->region_size[i];
875 /* select vlc table */
876 k = g->table_select[i];
877 l = mpa_huff_data[k][0];
878 linbits = mpa_huff_data[k][1];
882 memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
887 /* read huffcode and compute each couple */
891 int pos = get_bits_count(&s->gb);
894 switch_buffer(s, &pos, &end_pos, &end_pos2);
898 y = get_vlc2(&s->gb, vlc->table, 7, 3);
901 g->sb_hybrid[s_index ] =
902 g->sb_hybrid[s_index+1] = 0;
907 exponent= exponents[s_index];
909 av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
910 i, g->region_size[i] - j, x, y, exponent);
915 READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
917 x += get_bitsz(&s->gb, linbits);
918 v = l3_unscale(x, exponent);
919 if (get_bits1(&s->gb))
921 g->sb_hybrid[s_index] = v;
924 READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
926 y += get_bitsz(&s->gb, linbits);
927 v = l3_unscale(y, exponent);
928 if (get_bits1(&s->gb))
930 g->sb_hybrid[s_index+1] = v;
937 READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
939 x += get_bitsz(&s->gb, linbits);
940 v = l3_unscale(x, exponent);
941 if (get_bits1(&s->gb))
943 g->sb_hybrid[s_index+!!y] = v;
945 g->sb_hybrid[s_index + !y] = 0;
951 /* high frequencies */
952 vlc = &huff_quad_vlc[g->count1table_select];
954 while (s_index <= 572) {
956 pos = get_bits_count(&s->gb);
957 if (pos >= end_pos) {
958 if (pos > end_pos2 && last_pos) {
959 /* some encoders generate an incorrect size for this
960 part. We must go back into the data */
962 skip_bits_long(&s->gb, last_pos - pos);
963 av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
964 if(s->err_recognition & AV_EF_BITSTREAM)
968 switch_buffer(s, &pos, &end_pos, &end_pos2);
974 code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
975 av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
976 g->sb_hybrid[s_index+0] =
977 g->sb_hybrid[s_index+1] =
978 g->sb_hybrid[s_index+2] =
979 g->sb_hybrid[s_index+3] = 0;
981 static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
983 int pos = s_index + idxtab[code];
984 code ^= 8 >> idxtab[code];
985 READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
989 /* skip extension bits */
990 bits_left = end_pos2 - get_bits_count(&s->gb);
991 if (bits_left < 0 && (s->err_recognition & AV_EF_BUFFER)) {
992 av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
994 } else if (bits_left > 0 && (s->err_recognition & AV_EF_BUFFER)) {
995 av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
998 memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
999 skip_bits_long(&s->gb, bits_left);
1001 i = get_bits_count(&s->gb);
1002 switch_buffer(s, &i, &end_pos, &end_pos2);
1007 /* Reorder short blocks from bitstream order to interleaved order. It
1008 would be faster to do it in parsing, but the code would be far more
1010 static void reorder_block(MPADecodeContext *s, GranuleDef *g)
1013 INTFLOAT *ptr, *dst, *ptr1;
1016 if (g->block_type != 2)
1019 if (g->switch_point) {
1020 if (s->sample_rate_index != 8)
1021 ptr = g->sb_hybrid + 36;
1023 ptr = g->sb_hybrid + 72;
1028 for (i = g->short_start; i < 13; i++) {
1029 len = band_size_short[s->sample_rate_index][i];
1032 for (j = len; j > 0; j--) {
1033 *dst++ = ptr[0*len];
1034 *dst++ = ptr[1*len];
1035 *dst++ = ptr[2*len];
1039 memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
1043 #define ISQRT2 FIXR(0.70710678118654752440)
1045 static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
1048 int sf_max, sf, len, non_zero_found;
1049 INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;
1050 int non_zero_found_short[3];
1052 /* intensity stereo */
1053 if (s->mode_ext & MODE_EXT_I_STEREO) {
1058 is_tab = is_table_lsf[g1->scalefac_compress & 1];
1062 tab0 = g0->sb_hybrid + 576;
1063 tab1 = g1->sb_hybrid + 576;
1065 non_zero_found_short[0] = 0;
1066 non_zero_found_short[1] = 0;
1067 non_zero_found_short[2] = 0;
1068 k = (13 - g1->short_start) * 3 + g1->long_end - 3;
1069 for (i = 12; i >= g1->short_start; i--) {
1070 /* for last band, use previous scale factor */
1073 len = band_size_short[s->sample_rate_index][i];
1074 for (l = 2; l >= 0; l--) {
1077 if (!non_zero_found_short[l]) {
1078 /* test if non zero band. if so, stop doing i-stereo */
1079 for (j = 0; j < len; j++) {
1081 non_zero_found_short[l] = 1;
1085 sf = g1->scale_factors[k + l];
1091 for (j = 0; j < len; j++) {
1093 tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1094 tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1098 if (s->mode_ext & MODE_EXT_MS_STEREO) {
1099 /* lower part of the spectrum : do ms stereo
1101 for (j = 0; j < len; j++) {
1104 tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1105 tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1112 non_zero_found = non_zero_found_short[0] |
1113 non_zero_found_short[1] |
1114 non_zero_found_short[2];
1116 for (i = g1->long_end - 1;i >= 0;i--) {
1117 len = band_size_long[s->sample_rate_index][i];
1120 /* test if non zero band. if so, stop doing i-stereo */
1121 if (!non_zero_found) {
1122 for (j = 0; j < len; j++) {
1128 /* for last band, use previous scale factor */
1129 k = (i == 21) ? 20 : i;
1130 sf = g1->scale_factors[k];
1135 for (j = 0; j < len; j++) {
1137 tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1138 tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1142 if (s->mode_ext & MODE_EXT_MS_STEREO) {
1143 /* lower part of the spectrum : do ms stereo
1145 for (j = 0; j < len; j++) {
1148 tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1149 tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1154 } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
1155 /* ms stereo ONLY */
1156 /* NOTE: the 1/sqrt(2) normalization factor is included in the
1159 s->fdsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
1161 tab0 = g0->sb_hybrid;
1162 tab1 = g1->sb_hybrid;
1163 for (i = 0; i < 576; i++) {
1166 tab0[i] = tmp0 + tmp1;
1167 tab1[i] = tmp0 - tmp1;
1174 #define AA(j) do { \
1175 float tmp0 = ptr[-1-j]; \
1176 float tmp1 = ptr[ j]; \
1177 ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
1178 ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
1181 #define AA(j) do { \
1182 int tmp0 = ptr[-1-j]; \
1183 int tmp1 = ptr[ j]; \
1184 int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
1185 ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
1186 ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
1190 static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
1195 /* we antialias only "long" bands */
1196 if (g->block_type == 2) {
1197 if (!g->switch_point)
1199 /* XXX: check this for 8000Hz case */
1205 ptr = g->sb_hybrid + 18;
1206 for (i = n; i > 0; i--) {
1220 static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
1221 INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
1223 INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
1225 int i, j, mdct_long_end, sblimit;
1227 /* find last non zero block */
1228 ptr = g->sb_hybrid + 576;
1229 ptr1 = g->sb_hybrid + 2 * 18;
1230 while (ptr >= ptr1) {
1234 if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
1237 sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
1239 if (g->block_type == 2) {
1240 /* XXX: check for 8000 Hz */
1241 if (g->switch_point)
1246 mdct_long_end = sblimit;
1249 s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
1250 mdct_long_end, g->switch_point,
1253 buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
1254 ptr = g->sb_hybrid + 18 * mdct_long_end;
1256 for (j = mdct_long_end; j < sblimit; j++) {
1257 /* select frequency inversion */
1258 win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
1259 out_ptr = sb_samples + j;
1261 for (i = 0; i < 6; i++) {
1262 *out_ptr = buf[4*i];
1265 imdct12(out2, ptr + 0);
1266 for (i = 0; i < 6; i++) {
1267 *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
1268 buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
1271 imdct12(out2, ptr + 1);
1272 for (i = 0; i < 6; i++) {
1273 *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
1274 buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
1277 imdct12(out2, ptr + 2);
1278 for (i = 0; i < 6; i++) {
1279 buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
1280 buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
1281 buf[4*(i + 6*2)] = 0;
1284 buf += (j&3) != 3 ? 1 : (4*18-3);
1287 for (j = sblimit; j < SBLIMIT; j++) {
1289 out_ptr = sb_samples + j;
1290 for (i = 0; i < 18; i++) {
1291 *out_ptr = buf[4*i];
1295 buf += (j&3) != 3 ? 1 : (4*18-3);
1299 /* main layer3 decoding function */
1300 static int mp_decode_layer3(MPADecodeContext *s)
1302 int nb_granules, main_data_begin;
1303 int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
1305 int16_t exponents[576]; //FIXME try INTFLOAT
1307 /* read side info */
1309 main_data_begin = get_bits(&s->gb, 8);
1310 skip_bits(&s->gb, s->nb_channels);
1313 main_data_begin = get_bits(&s->gb, 9);
1314 if (s->nb_channels == 2)
1315 skip_bits(&s->gb, 3);
1317 skip_bits(&s->gb, 5);
1319 for (ch = 0; ch < s->nb_channels; ch++) {
1320 s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
1321 s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
1325 for (gr = 0; gr < nb_granules; gr++) {
1326 for (ch = 0; ch < s->nb_channels; ch++) {
1327 av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
1328 g = &s->granules[ch][gr];
1329 g->part2_3_length = get_bits(&s->gb, 12);
1330 g->big_values = get_bits(&s->gb, 9);
1331 if (g->big_values > 288) {
1332 av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
1333 return AVERROR_INVALIDDATA;
1336 g->global_gain = get_bits(&s->gb, 8);
1337 /* if MS stereo only is selected, we precompute the
1338 1/sqrt(2) renormalization factor */
1339 if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
1341 g->global_gain -= 2;
1343 g->scalefac_compress = get_bits(&s->gb, 9);
1345 g->scalefac_compress = get_bits(&s->gb, 4);
1346 blocksplit_flag = get_bits1(&s->gb);
1347 if (blocksplit_flag) {
1348 g->block_type = get_bits(&s->gb, 2);
1349 if (g->block_type == 0) {
1350 av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
1351 return AVERROR_INVALIDDATA;
1353 g->switch_point = get_bits1(&s->gb);
1354 for (i = 0; i < 2; i++)
1355 g->table_select[i] = get_bits(&s->gb, 5);
1356 for (i = 0; i < 3; i++)
1357 g->subblock_gain[i] = get_bits(&s->gb, 3);
1358 ff_init_short_region(s, g);
1360 int region_address1, region_address2;
1362 g->switch_point = 0;
1363 for (i = 0; i < 3; i++)
1364 g->table_select[i] = get_bits(&s->gb, 5);
1365 /* compute huffman coded region sizes */
1366 region_address1 = get_bits(&s->gb, 4);
1367 region_address2 = get_bits(&s->gb, 3);
1368 av_dlog(s->avctx, "region1=%d region2=%d\n",
1369 region_address1, region_address2);
1370 ff_init_long_region(s, g, region_address1, region_address2);
1372 ff_region_offset2size(g);
1373 ff_compute_band_indexes(s, g);
1377 g->preflag = get_bits1(&s->gb);
1378 g->scalefac_scale = get_bits1(&s->gb);
1379 g->count1table_select = get_bits1(&s->gb);
1380 av_dlog(s->avctx, "block_type=%d switch_point=%d\n",
1381 g->block_type, g->switch_point);
1387 const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
1388 int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0,
1389 FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
1390 assert((get_bits_count(&s->gb) & 7) == 0);
1391 /* now we get bits from the main_data_begin offset */
1392 av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
1393 main_data_begin, s->last_buf_size);
1395 memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);
1397 init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
1398 #if !UNCHECKED_BITSTREAM_READER
1399 s->gb.size_in_bits_plus8 += extrasize * 8;
1401 s->last_buf_size <<= 3;
1402 for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
1403 for (ch = 0; ch < s->nb_channels; ch++) {
1404 g = &s->granules[ch][gr];
1405 s->last_buf_size += g->part2_3_length;
1406 memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
1407 compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1410 skip = s->last_buf_size - 8 * main_data_begin;
1411 if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {
1412 skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);
1414 s->in_gb.buffer = NULL;
1416 skip_bits_long(&s->gb, skip);
1422 for (; gr < nb_granules; gr++) {
1423 for (ch = 0; ch < s->nb_channels; ch++) {
1424 g = &s->granules[ch][gr];
1425 bits_pos = get_bits_count(&s->gb);
1429 int slen, slen1, slen2;
1431 /* MPEG1 scale factors */
1432 slen1 = slen_table[0][g->scalefac_compress];
1433 slen2 = slen_table[1][g->scalefac_compress];
1434 av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
1435 if (g->block_type == 2) {
1436 n = g->switch_point ? 17 : 18;
1439 for (i = 0; i < n; i++)
1440 g->scale_factors[j++] = get_bits(&s->gb, slen1);
1442 for (i = 0; i < n; i++)
1443 g->scale_factors[j++] = 0;
1446 for (i = 0; i < 18; i++)
1447 g->scale_factors[j++] = get_bits(&s->gb, slen2);
1448 for (i = 0; i < 3; i++)
1449 g->scale_factors[j++] = 0;
1451 for (i = 0; i < 21; i++)
1452 g->scale_factors[j++] = 0;
1455 sc = s->granules[ch][0].scale_factors;
1457 for (k = 0; k < 4; k++) {
1459 if ((g->scfsi & (0x8 >> k)) == 0) {
1460 slen = (k < 2) ? slen1 : slen2;
1462 for (i = 0; i < n; i++)
1463 g->scale_factors[j++] = get_bits(&s->gb, slen);
1465 for (i = 0; i < n; i++)
1466 g->scale_factors[j++] = 0;
1469 /* simply copy from last granule */
1470 for (i = 0; i < n; i++) {
1471 g->scale_factors[j] = sc[j];
1476 g->scale_factors[j++] = 0;
1479 int tindex, tindex2, slen[4], sl, sf;
1481 /* LSF scale factors */
1482 if (g->block_type == 2)
1483 tindex = g->switch_point ? 2 : 1;
1487 sf = g->scalefac_compress;
1488 if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
1489 /* intensity stereo case */
1492 lsf_sf_expand(slen, sf, 6, 6, 0);
1494 } else if (sf < 244) {
1495 lsf_sf_expand(slen, sf - 180, 4, 4, 0);
1498 lsf_sf_expand(slen, sf - 244, 3, 0, 0);
1504 lsf_sf_expand(slen, sf, 5, 4, 4);
1506 } else if (sf < 500) {
1507 lsf_sf_expand(slen, sf - 400, 5, 4, 0);
1510 lsf_sf_expand(slen, sf - 500, 3, 0, 0);
1517 for (k = 0; k < 4; k++) {
1518 n = lsf_nsf_table[tindex2][tindex][k];
1521 for (i = 0; i < n; i++)
1522 g->scale_factors[j++] = get_bits(&s->gb, sl);
1524 for (i = 0; i < n; i++)
1525 g->scale_factors[j++] = 0;
1528 /* XXX: should compute exact size */
1530 g->scale_factors[j] = 0;
1533 exponents_from_scale_factors(s, g, exponents);
1535 /* read Huffman coded residue */
1536 huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
1539 if (s->mode == MPA_JSTEREO)
1540 compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
1542 for (ch = 0; ch < s->nb_channels; ch++) {
1543 g = &s->granules[ch][gr];
1545 reorder_block(s, g);
1546 compute_antialias(s, g);
1547 compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1550 if (get_bits_count(&s->gb) < 0)
1551 skip_bits_long(&s->gb, -get_bits_count(&s->gb));
1552 return nb_granules * 18;
1555 static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,
1556 const uint8_t *buf, int buf_size)
1558 int i, nb_frames, ch, ret;
1559 OUT_INT *samples_ptr;
1561 init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
1563 /* skip error protection field */
1564 if (s->error_protection)
1565 skip_bits(&s->gb, 16);
1569 s->avctx->frame_size = 384;
1570 nb_frames = mp_decode_layer1(s);
1573 s->avctx->frame_size = 1152;
1574 nb_frames = mp_decode_layer2(s);
1577 s->avctx->frame_size = s->lsf ? 576 : 1152;
1579 nb_frames = mp_decode_layer3(s);
1585 if (s->in_gb.buffer) {
1586 align_get_bits(&s->gb);
1587 i = get_bits_left(&s->gb)>>3;
1588 if (i >= 0 && i <= BACKSTEP_SIZE) {
1589 memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
1592 av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
1594 s->in_gb.buffer = NULL;
1597 align_get_bits(&s->gb);
1598 assert((get_bits_count(&s->gb) & 7) == 0);
1599 i = get_bits_left(&s->gb) >> 3;
1601 if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
1603 av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
1604 i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
1606 assert(i <= buf_size - HEADER_SIZE && i >= 0);
1607 memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
1608 s->last_buf_size += i;
1611 /* get output buffer */
1613 av_assert0(s->frame != NULL);
1614 s->frame->nb_samples = s->avctx->frame_size;
1615 if ((ret = ff_get_buffer(s->avctx, s->frame)) < 0) {
1616 av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1619 samples = (OUT_INT **)s->frame->extended_data;
1622 /* apply the synthesis filter */
1623 for (ch = 0; ch < s->nb_channels; ch++) {
1625 if (s->avctx->sample_fmt == OUT_FMT_P) {
1626 samples_ptr = samples[ch];
1629 samples_ptr = samples[0] + ch;
1630 sample_stride = s->nb_channels;
1632 for (i = 0; i < nb_frames; i++) {
1633 RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
1634 &(s->synth_buf_offset[ch]),
1635 RENAME(ff_mpa_synth_window),
1636 &s->dither_state, samples_ptr,
1637 sample_stride, s->sb_samples[ch][i]);
1638 samples_ptr += 32 * sample_stride;
1642 return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
1645 static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
1648 const uint8_t *buf = avpkt->data;
1649 int buf_size = avpkt->size;
1650 MPADecodeContext *s = avctx->priv_data;
1654 if (buf_size < HEADER_SIZE)
1655 return AVERROR_INVALIDDATA;
1657 header = AV_RB32(buf);
1658 if (ff_mpa_check_header(header) < 0) {
1659 av_log(avctx, AV_LOG_ERROR, "Header missing\n");
1660 return AVERROR_INVALIDDATA;
1663 if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
1664 /* free format: prepare to compute frame size */
1666 return AVERROR_INVALIDDATA;
1668 /* update codec info */
1669 avctx->channels = s->nb_channels;
1670 avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;
1671 if (!avctx->bit_rate)
1672 avctx->bit_rate = s->bit_rate;
1674 if (s->frame_size <= 0 || s->frame_size > buf_size) {
1675 av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
1676 return AVERROR_INVALIDDATA;
1677 } else if (s->frame_size < buf_size) {
1678 buf_size= s->frame_size;
1683 ret = mp_decode_frame(s, NULL, buf, buf_size);
1685 s->frame->nb_samples = avctx->frame_size;
1687 avctx->sample_rate = s->sample_rate;
1688 //FIXME maybe move the other codec info stuff from above here too
1690 av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1691 /* Only return an error if the bad frame makes up the whole packet or
1692 * the error is related to buffer management.
1693 * If there is more data in the packet, just consume the bad frame
1694 * instead of returning an error, which would discard the whole
1697 if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
1704 static void mp_flush(MPADecodeContext *ctx)
1706 memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
1707 ctx->last_buf_size = 0;
1710 static void flush(AVCodecContext *avctx)
1712 mp_flush(avctx->priv_data);
1715 #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
1716 static int decode_frame_adu(AVCodecContext *avctx, void *data,
1717 int *got_frame_ptr, AVPacket *avpkt)
1719 const uint8_t *buf = avpkt->data;
1720 int buf_size = avpkt->size;
1721 MPADecodeContext *s = avctx->priv_data;
1727 // Discard too short frames
1728 if (buf_size < HEADER_SIZE) {
1729 av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
1730 return AVERROR_INVALIDDATA;
1734 if (len > MPA_MAX_CODED_FRAME_SIZE)
1735 len = MPA_MAX_CODED_FRAME_SIZE;
1737 // Get header and restore sync word
1738 header = AV_RB32(buf) | 0xffe00000;
1740 if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
1741 av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
1742 return AVERROR_INVALIDDATA;
1745 avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
1746 /* update codec info */
1747 avctx->sample_rate = s->sample_rate;
1748 avctx->channels = s->nb_channels;
1749 if (!avctx->bit_rate)
1750 avctx->bit_rate = s->bit_rate;
1752 s->frame_size = len;
1756 ret = mp_decode_frame(s, NULL, buf, buf_size);
1758 av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1766 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
1768 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
1771 * Context for MP3On4 decoder
1773 typedef struct MP3On4DecodeContext {
1774 int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
1775 int syncword; ///< syncword patch
1776 const uint8_t *coff; ///< channel offsets in output buffer
1777 MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
1778 } MP3On4DecodeContext;
1780 #include "mpeg4audio.h"
1782 /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
1784 /* number of mp3 decoder instances */
1785 static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
1787 /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
1788 static const uint8_t chan_offset[8][5] = {
1793 { 2, 0, 3 }, // C FLR BS
1794 { 2, 0, 3 }, // C FLR BLRS
1795 { 2, 0, 4, 3 }, // C FLR BLRS LFE
1796 { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
1799 /* mp3on4 channel layouts */
1800 static const int16_t chan_layout[8] = {
1803 AV_CH_LAYOUT_STEREO,
1804 AV_CH_LAYOUT_SURROUND,
1805 AV_CH_LAYOUT_4POINT0,
1806 AV_CH_LAYOUT_5POINT0,
1807 AV_CH_LAYOUT_5POINT1,
1808 AV_CH_LAYOUT_7POINT1
1811 static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
1813 MP3On4DecodeContext *s = avctx->priv_data;
1816 for (i = 0; i < s->frames; i++)
1817 av_free(s->mp3decctx[i]);
1823 static int decode_init_mp3on4(AVCodecContext * avctx)
1825 MP3On4DecodeContext *s = avctx->priv_data;
1826 MPEG4AudioConfig cfg;
1829 if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
1830 av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
1831 return AVERROR_INVALIDDATA;
1834 avpriv_mpeg4audio_get_config(&cfg, avctx->extradata,
1835 avctx->extradata_size * 8, 1);
1836 if (!cfg.chan_config || cfg.chan_config > 7) {
1837 av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
1838 return AVERROR_INVALIDDATA;
1840 s->frames = mp3Frames[cfg.chan_config];
1841 s->coff = chan_offset[cfg.chan_config];
1842 avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
1843 avctx->channel_layout = chan_layout[cfg.chan_config];
1845 if (cfg.sample_rate < 16000)
1846 s->syncword = 0xffe00000;
1848 s->syncword = 0xfff00000;
1850 /* Init the first mp3 decoder in standard way, so that all tables get builded
1851 * We replace avctx->priv_data with the context of the first decoder so that
1852 * decode_init() does not have to be changed.
1853 * Other decoders will be initialized here copying data from the first context
1855 // Allocate zeroed memory for the first decoder context
1856 s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
1857 if (!s->mp3decctx[0])
1859 // Put decoder context in place to make init_decode() happy
1860 avctx->priv_data = s->mp3decctx[0];
1862 // Restore mp3on4 context pointer
1863 avctx->priv_data = s;
1864 s->mp3decctx[0]->adu_mode = 1; // Set adu mode
1866 /* Create a separate codec/context for each frame (first is already ok).
1867 * Each frame is 1 or 2 channels - up to 5 frames allowed
1869 for (i = 1; i < s->frames; i++) {
1870 s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
1871 if (!s->mp3decctx[i])
1873 s->mp3decctx[i]->adu_mode = 1;
1874 s->mp3decctx[i]->avctx = avctx;
1875 s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
1880 decode_close_mp3on4(avctx);
1881 return AVERROR(ENOMEM);
1885 static void flush_mp3on4(AVCodecContext *avctx)
1888 MP3On4DecodeContext *s = avctx->priv_data;
1890 for (i = 0; i < s->frames; i++)
1891 mp_flush(s->mp3decctx[i]);
1895 static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
1896 int *got_frame_ptr, AVPacket *avpkt)
1898 AVFrame *frame = data;
1899 const uint8_t *buf = avpkt->data;
1900 int buf_size = avpkt->size;
1901 MP3On4DecodeContext *s = avctx->priv_data;
1902 MPADecodeContext *m;
1903 int fsize, len = buf_size, out_size = 0;
1905 OUT_INT **out_samples;
1909 /* get output buffer */
1910 frame->nb_samples = MPA_FRAME_SIZE;
1911 if ((ret = ff_get_buffer(avctx, frame)) < 0) {
1912 av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1915 out_samples = (OUT_INT **)frame->extended_data;
1917 // Discard too short frames
1918 if (buf_size < HEADER_SIZE)
1919 return AVERROR_INVALIDDATA;
1921 avctx->bit_rate = 0;
1924 for (fr = 0; fr < s->frames; fr++) {
1925 fsize = AV_RB16(buf) >> 4;
1926 fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
1927 m = s->mp3decctx[fr];
1930 if (fsize < HEADER_SIZE) {
1931 av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
1932 return AVERROR_INVALIDDATA;
1934 header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
1936 if (ff_mpa_check_header(header) < 0) // Bad header, discard block
1939 avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);
1941 if (ch + m->nb_channels > avctx->channels) {
1942 av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
1944 return AVERROR_INVALIDDATA;
1946 ch += m->nb_channels;
1948 outptr[0] = out_samples[s->coff[fr]];
1949 if (m->nb_channels > 1)
1950 outptr[1] = out_samples[s->coff[fr] + 1];
1952 if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
1959 avctx->bit_rate += m->bit_rate;
1962 /* update codec info */
1963 avctx->sample_rate = s->mp3decctx[0]->sample_rate;
1965 frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
1970 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */
1973 #if CONFIG_MP1_DECODER
1974 AVCodec ff_mp1_decoder = {
1976 .type = AVMEDIA_TYPE_AUDIO,
1977 .id = AV_CODEC_ID_MP1,
1978 .priv_data_size = sizeof(MPADecodeContext),
1979 .init = decode_init,
1980 .decode = decode_frame,
1981 .capabilities = CODEC_CAP_DR1,
1983 .long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
1984 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
1986 AV_SAMPLE_FMT_NONE },
1989 #if CONFIG_MP2_DECODER
1990 AVCodec ff_mp2_decoder = {
1992 .type = AVMEDIA_TYPE_AUDIO,
1993 .id = AV_CODEC_ID_MP2,
1994 .priv_data_size = sizeof(MPADecodeContext),
1995 .init = decode_init,
1996 .decode = decode_frame,
1997 .capabilities = CODEC_CAP_DR1,
1999 .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
2000 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2002 AV_SAMPLE_FMT_NONE },
2005 #if CONFIG_MP3_DECODER
2006 AVCodec ff_mp3_decoder = {
2008 .type = AVMEDIA_TYPE_AUDIO,
2009 .id = AV_CODEC_ID_MP3,
2010 .priv_data_size = sizeof(MPADecodeContext),
2011 .init = decode_init,
2012 .decode = decode_frame,
2013 .capabilities = CODEC_CAP_DR1,
2015 .long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
2016 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2018 AV_SAMPLE_FMT_NONE },
2021 #if CONFIG_MP3ADU_DECODER
2022 AVCodec ff_mp3adu_decoder = {
2024 .type = AVMEDIA_TYPE_AUDIO,
2025 .id = AV_CODEC_ID_MP3ADU,
2026 .priv_data_size = sizeof(MPADecodeContext),
2027 .init = decode_init,
2028 .decode = decode_frame_adu,
2029 .capabilities = CODEC_CAP_DR1,
2031 .long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
2032 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2034 AV_SAMPLE_FMT_NONE },
2037 #if CONFIG_MP3ON4_DECODER
2038 AVCodec ff_mp3on4_decoder = {
2040 .type = AVMEDIA_TYPE_AUDIO,
2041 .id = AV_CODEC_ID_MP3ON4,
2042 .priv_data_size = sizeof(MP3On4DecodeContext),
2043 .init = decode_init_mp3on4,
2044 .close = decode_close_mp3on4,
2045 .decode = decode_frame_mp3on4,
2046 .capabilities = CODEC_CAP_DR1,
2047 .flush = flush_mp3on4,
2048 .long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
2049 .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
2050 AV_SAMPLE_FMT_NONE },