Bug 1878930 - s/RawBuffer/Span/: Remove RawBuffer and unused utils. r=gfx-reviewers...
[gecko.git] / media / ffvpx / libavcodec / mpegaudiodec_template.c
blobc227604107906377327818ae0852291ceb528952
1 /*
2 * MPEG Audio decoder
3 * Copyright (c) 2001, 2002 Fabrice Bellard
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 /**
23 * @file
24 * MPEG Audio decoder
27 #include "config_components.h"
29 #include "libavutil/attributes.h"
30 #include "libavutil/avassert.h"
31 #include "libavutil/channel_layout.h"
32 #include "libavutil/crc.h"
33 #include "libavutil/float_dsp.h"
34 #include "libavutil/libm.h"
35 #include "libavutil/mem_internal.h"
36 #include "libavutil/thread.h"
38 #include "avcodec.h"
39 #include "decode.h"
40 #include "get_bits.h"
41 #include "mathops.h"
42 #include "mpegaudiodsp.h"
45 * TODO:
46 * - test lsf / mpeg25 extensively.
49 #include "mpegaudio.h"
50 #include "mpegaudiodecheader.h"
52 #define BACKSTEP_SIZE 512
53 #define EXTRABYTES 24
54 #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
56 /* layer 3 "granule" */
57 typedef struct GranuleDef {
58 uint8_t scfsi;
59 int part2_3_length;
60 int big_values;
61 int global_gain;
62 int scalefac_compress;
63 uint8_t block_type;
64 uint8_t switch_point;
65 int table_select[3];
66 int subblock_gain[3];
67 uint8_t scalefac_scale;
68 uint8_t count1table_select;
69 int region_size[3]; /* number of huffman codes in each region */
70 int preflag;
71 int short_start, long_end; /* long/short band indexes */
72 uint8_t scale_factors[40];
73 DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */
74 } GranuleDef;
76 typedef struct MPADecodeContext {
77 MPA_DECODE_HEADER
78 uint8_t last_buf[LAST_BUF_SIZE];
79 int last_buf_size;
80 int extrasize;
81 /* next header (used in free format parsing) */
82 uint32_t free_format_next_header;
83 GetBitContext gb;
84 GetBitContext in_gb;
85 DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];
86 int synth_buf_offset[MPA_MAX_CHANNELS];
87 DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];
88 INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
89 GranuleDef granules[2][2]; /* Used in Layer 3 */
90 int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
91 int dither_state;
92 int err_recognition;
93 AVCodecContext* avctx;
94 MPADSPContext mpadsp;
95 void (*butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len);
96 AVFrame *frame;
97 uint32_t crc;
98 } MPADecodeContext;
100 #define HEADER_SIZE 4
102 #include "mpegaudiodata.h"
104 #include "mpegaudio_tablegen.h"
105 /* intensity stereo coef table */
106 static INTFLOAT is_table_lsf[2][2][16];
108 /* [i][j]: 2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */
109 static int32_t scale_factor_mult[15][3];
110 /* mult table for layer 2 group quantization */
112 #define SCALE_GEN(v) \
113 { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }
115 static const int32_t scale_factor_mult2[3][3] = {
116 SCALE_GEN(4.0 / 3.0), /* 3 steps */
117 SCALE_GEN(4.0 / 5.0), /* 5 steps */
118 SCALE_GEN(4.0 / 9.0), /* 9 steps */
122 * Convert region offsets to region sizes and truncate
123 * size to big_values.
125 static void region_offset2size(GranuleDef *g)
127 int i, k, j = 0;
128 g->region_size[2] = 576 / 2;
129 for (i = 0; i < 3; i++) {
130 k = FFMIN(g->region_size[i], g->big_values);
131 g->region_size[i] = k - j;
132 j = k;
136 static void init_short_region(MPADecodeContext *s, GranuleDef *g)
138 if (g->block_type == 2) {
139 if (s->sample_rate_index != 8)
140 g->region_size[0] = (36 / 2);
141 else
142 g->region_size[0] = (72 / 2);
143 } else {
144 if (s->sample_rate_index <= 2)
145 g->region_size[0] = (36 / 2);
146 else if (s->sample_rate_index != 8)
147 g->region_size[0] = (54 / 2);
148 else
149 g->region_size[0] = (108 / 2);
151 g->region_size[1] = (576 / 2);
154 static void init_long_region(MPADecodeContext *s, GranuleDef *g,
155 int ra1, int ra2)
157 int l;
158 g->region_size[0] = ff_band_index_long[s->sample_rate_index][ra1 + 1];
159 /* should not overflow */
160 l = FFMIN(ra1 + ra2 + 2, 22);
161 g->region_size[1] = ff_band_index_long[s->sample_rate_index][ l];
164 static void compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
166 if (g->block_type == 2) {
167 if (g->switch_point) {
168 if(s->sample_rate_index == 8)
169 avpriv_request_sample(s->avctx, "switch point in 8khz");
170 /* if switched mode, we handle the 36 first samples as
171 long blocks. For 8000Hz, we handle the 72 first
172 exponents as long blocks */
173 if (s->sample_rate_index <= 2)
174 g->long_end = 8;
175 else
176 g->long_end = 6;
178 g->short_start = 3;
179 } else {
180 g->long_end = 0;
181 g->short_start = 0;
183 } else {
184 g->short_start = 13;
185 g->long_end = 22;
189 /* layer 1 unscaling */
190 /* n = number of bits of the mantissa minus 1 */
191 static inline int l1_unscale(int n, int mant, int scale_factor)
193 int shift, mod;
194 int64_t val;
196 shift = ff_scale_factor_modshift[scale_factor];
197 mod = shift & 3;
198 shift >>= 2;
199 val = MUL64((int)(mant + (-1U << n) + 1), scale_factor_mult[n-1][mod]);
200 shift += n;
201 /* NOTE: at this point, 1 <= shift >= 21 + 15 */
202 return (int)((val + (1LL << (shift - 1))) >> shift);
205 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
207 int shift, mod, val;
209 shift = ff_scale_factor_modshift[scale_factor];
210 mod = shift & 3;
211 shift >>= 2;
213 val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
214 /* NOTE: at this point, 0 <= shift <= 21 */
215 if (shift > 0)
216 val = (val + (1 << (shift - 1))) >> shift;
217 return val;
220 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
221 static inline int l3_unscale(int value, int exponent)
223 unsigned int m;
224 int e;
226 e = ff_table_4_3_exp [4 * value + (exponent & 3)];
227 m = ff_table_4_3_value[4 * value + (exponent & 3)];
228 e -= exponent >> 2;
229 #ifdef DEBUG
230 if(e < 1)
231 av_log(NULL, AV_LOG_WARNING, "l3_unscale: e is %d\n", e);
232 #endif
233 if (e > (SUINT)31)
234 return 0;
235 m = (m + ((1U << e) >> 1)) >> e;
237 return m;
240 static av_cold void decode_init_static(void)
242 int i, j;
244 /* scale factor multiply for layer 1 */
245 for (i = 0; i < 15; i++) {
246 int n, norm;
247 n = i + 2;
248 norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
249 scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0 * 2.0), FRAC_BITS);
250 scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
251 scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);
252 ff_dlog(NULL, "%d: norm=%x s=%"PRIx32" %"PRIx32" %"PRIx32"\n", i,
253 (unsigned)norm,
254 scale_factor_mult[i][0],
255 scale_factor_mult[i][1],
256 scale_factor_mult[i][2]);
259 /* compute n ^ (4/3) and store it in mantissa/exp format */
261 mpegaudio_tableinit();
263 for (i = 0; i < 16; i++) {
264 double f;
265 int e, k;
267 for (j = 0; j < 2; j++) {
268 e = -(j + 1) * ((i + 1) >> 1);
269 f = exp2(e / 4.0);
270 k = i & 1;
271 is_table_lsf[j][k ^ 1][i] = FIXR(f);
272 is_table_lsf[j][k ][i] = FIXR(1.0);
273 ff_dlog(NULL, "is_table_lsf %d %d: %f %f\n",
274 i, j, (float) is_table_lsf[j][0][i],
275 (float) is_table_lsf[j][1][i]);
278 RENAME(ff_mpa_synth_init)();
279 ff_mpegaudiodec_common_init_static();
282 static av_cold int decode_init(AVCodecContext * avctx)
284 static AVOnce init_static_once = AV_ONCE_INIT;
285 MPADecodeContext *s = avctx->priv_data;
287 s->avctx = avctx;
289 #if USE_FLOATS
291 AVFloatDSPContext *fdsp;
292 fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
293 if (!fdsp)
294 return AVERROR(ENOMEM);
295 s->butterflies_float = fdsp->butterflies_float;
296 av_free(fdsp);
298 #endif
300 ff_mpadsp_init(&s->mpadsp);
302 if (avctx->request_sample_fmt == OUT_FMT &&
303 avctx->codec_id != AV_CODEC_ID_MP3ON4)
304 avctx->sample_fmt = OUT_FMT;
305 else
306 avctx->sample_fmt = OUT_FMT_P;
307 s->err_recognition = avctx->err_recognition;
309 if (avctx->codec_id == AV_CODEC_ID_MP3ADU)
310 s->adu_mode = 1;
312 ff_thread_once(&init_static_once, decode_init_static);
314 return 0;
317 #define C3 FIXHR(0.86602540378443864676/2)
318 #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
319 #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
320 #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)
322 /* 12 points IMDCT. We compute it "by hand" by factorizing obvious
323 cases. */
324 static void imdct12(INTFLOAT *out, SUINTFLOAT *in)
326 SUINTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;
328 in0 = in[0*3];
329 in1 = in[1*3] + in[0*3];
330 in2 = in[2*3] + in[1*3];
331 in3 = in[3*3] + in[2*3];
332 in4 = in[4*3] + in[3*3];
333 in5 = in[5*3] + in[4*3];
334 in5 += in3;
335 in3 += in1;
337 in2 = MULH3(in2, C3, 2);
338 in3 = MULH3(in3, C3, 4);
340 t1 = in0 - in4;
341 t2 = MULH3(in1 - in5, C4, 2);
343 out[ 7] =
344 out[10] = t1 + t2;
345 out[ 1] =
346 out[ 4] = t1 - t2;
348 in0 += SHR(in4, 1);
349 in4 = in0 + in2;
350 in5 += 2*in1;
351 in1 = MULH3(in5 + in3, C5, 1);
352 out[ 8] =
353 out[ 9] = in4 + in1;
354 out[ 2] =
355 out[ 3] = in4 - in1;
357 in0 -= in2;
358 in5 = MULH3(in5 - in3, C6, 2);
359 out[ 0] =
360 out[ 5] = in0 - in5;
361 out[ 6] =
362 out[11] = in0 + in5;
365 static int handle_crc(MPADecodeContext *s, int sec_len)
367 if (s->error_protection && (s->err_recognition & AV_EF_CRCCHECK)) {
368 const uint8_t *buf = s->gb.buffer - HEADER_SIZE;
369 int sec_byte_len = sec_len >> 3;
370 int sec_rem_bits = sec_len & 7;
371 const AVCRC *crc_tab = av_crc_get_table(AV_CRC_16_ANSI);
372 uint8_t tmp_buf[4];
373 uint32_t crc_val = av_crc(crc_tab, UINT16_MAX, &buf[2], 2);
374 crc_val = av_crc(crc_tab, crc_val, &buf[6], sec_byte_len);
376 AV_WB32(tmp_buf,
377 ((buf[6 + sec_byte_len] & (0xFF00U >> sec_rem_bits)) << 24) +
378 ((s->crc << 16) >> sec_rem_bits));
380 crc_val = av_crc(crc_tab, crc_val, tmp_buf, 3);
382 if (crc_val) {
383 av_log(s->avctx, AV_LOG_ERROR, "CRC mismatch %X!\n", crc_val);
384 if (s->err_recognition & AV_EF_EXPLODE)
385 return AVERROR_INVALIDDATA;
388 return 0;
391 /* return the number of decoded frames */
392 static int mp_decode_layer1(MPADecodeContext *s)
394 int bound, i, v, n, ch, j, mant;
395 uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
396 uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];
397 int ret;
399 ret = handle_crc(s, (s->nb_channels == 1) ? 8*16 : 8*32);
400 if (ret < 0)
401 return ret;
403 if (s->mode == MPA_JSTEREO)
404 bound = (s->mode_ext + 1) * 4;
405 else
406 bound = SBLIMIT;
408 /* allocation bits */
409 for (i = 0; i < bound; i++) {
410 for (ch = 0; ch < s->nb_channels; ch++) {
411 allocation[ch][i] = get_bits(&s->gb, 4);
414 for (i = bound; i < SBLIMIT; i++)
415 allocation[0][i] = get_bits(&s->gb, 4);
417 /* scale factors */
418 for (i = 0; i < bound; i++) {
419 for (ch = 0; ch < s->nb_channels; ch++) {
420 if (allocation[ch][i])
421 scale_factors[ch][i] = get_bits(&s->gb, 6);
424 for (i = bound; i < SBLIMIT; i++) {
425 if (allocation[0][i]) {
426 scale_factors[0][i] = get_bits(&s->gb, 6);
427 scale_factors[1][i] = get_bits(&s->gb, 6);
431 /* compute samples */
432 for (j = 0; j < 12; j++) {
433 for (i = 0; i < bound; i++) {
434 for (ch = 0; ch < s->nb_channels; ch++) {
435 n = allocation[ch][i];
436 if (n) {
437 mant = get_bits(&s->gb, n + 1);
438 v = l1_unscale(n, mant, scale_factors[ch][i]);
439 } else {
440 v = 0;
442 s->sb_samples[ch][j][i] = v;
445 for (i = bound; i < SBLIMIT; i++) {
446 n = allocation[0][i];
447 if (n) {
448 mant = get_bits(&s->gb, n + 1);
449 v = l1_unscale(n, mant, scale_factors[0][i]);
450 s->sb_samples[0][j][i] = v;
451 v = l1_unscale(n, mant, scale_factors[1][i]);
452 s->sb_samples[1][j][i] = v;
453 } else {
454 s->sb_samples[0][j][i] = 0;
455 s->sb_samples[1][j][i] = 0;
459 return 12;
462 static int mp_decode_layer2(MPADecodeContext *s)
464 int sblimit; /* number of used subbands */
465 const unsigned char *alloc_table;
466 int table, bit_alloc_bits, i, j, ch, bound, v;
467 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
468 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
469 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
470 int scale, qindex, bits, steps, k, l, m, b;
471 int ret;
473 /* select decoding table */
474 table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
475 s->sample_rate, s->lsf);
476 sblimit = ff_mpa_sblimit_table[table];
477 alloc_table = ff_mpa_alloc_tables[table];
479 if (s->mode == MPA_JSTEREO)
480 bound = (s->mode_ext + 1) * 4;
481 else
482 bound = sblimit;
484 ff_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
486 /* sanity check */
487 if (bound > sblimit)
488 bound = sblimit;
490 /* parse bit allocation */
491 j = 0;
492 for (i = 0; i < bound; i++) {
493 bit_alloc_bits = alloc_table[j];
494 for (ch = 0; ch < s->nb_channels; ch++)
495 bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
496 j += 1 << bit_alloc_bits;
498 for (i = bound; i < sblimit; i++) {
499 bit_alloc_bits = alloc_table[j];
500 v = get_bits(&s->gb, bit_alloc_bits);
501 bit_alloc[0][i] = v;
502 bit_alloc[1][i] = v;
503 j += 1 << bit_alloc_bits;
506 /* scale codes */
507 for (i = 0; i < sblimit; i++) {
508 for (ch = 0; ch < s->nb_channels; ch++) {
509 if (bit_alloc[ch][i])
510 scale_code[ch][i] = get_bits(&s->gb, 2);
514 ret = handle_crc(s, get_bits_count(&s->gb) - 16);
515 if (ret < 0)
516 return ret;
518 /* scale factors */
519 for (i = 0; i < sblimit; i++) {
520 for (ch = 0; ch < s->nb_channels; ch++) {
521 if (bit_alloc[ch][i]) {
522 sf = scale_factors[ch][i];
523 switch (scale_code[ch][i]) {
524 default:
525 case 0:
526 sf[0] = get_bits(&s->gb, 6);
527 sf[1] = get_bits(&s->gb, 6);
528 sf[2] = get_bits(&s->gb, 6);
529 break;
530 case 2:
531 sf[0] = get_bits(&s->gb, 6);
532 sf[1] = sf[0];
533 sf[2] = sf[0];
534 break;
535 case 1:
536 sf[0] = get_bits(&s->gb, 6);
537 sf[2] = get_bits(&s->gb, 6);
538 sf[1] = sf[0];
539 break;
540 case 3:
541 sf[0] = get_bits(&s->gb, 6);
542 sf[2] = get_bits(&s->gb, 6);
543 sf[1] = sf[2];
544 break;
550 /* samples */
551 for (k = 0; k < 3; k++) {
552 for (l = 0; l < 12; l += 3) {
553 j = 0;
554 for (i = 0; i < bound; i++) {
555 bit_alloc_bits = alloc_table[j];
556 for (ch = 0; ch < s->nb_channels; ch++) {
557 b = bit_alloc[ch][i];
558 if (b) {
559 scale = scale_factors[ch][i][k];
560 qindex = alloc_table[j+b];
561 bits = ff_mpa_quant_bits[qindex];
562 if (bits < 0) {
563 int v2;
564 /* 3 values at the same time */
565 v = get_bits(&s->gb, -bits);
566 v2 = ff_division_tabs[qindex][v];
567 steps = ff_mpa_quant_steps[qindex];
569 s->sb_samples[ch][k * 12 + l + 0][i] =
570 l2_unscale_group(steps, v2 & 15, scale);
571 s->sb_samples[ch][k * 12 + l + 1][i] =
572 l2_unscale_group(steps, (v2 >> 4) & 15, scale);
573 s->sb_samples[ch][k * 12 + l + 2][i] =
574 l2_unscale_group(steps, v2 >> 8 , scale);
575 } else {
576 for (m = 0; m < 3; m++) {
577 v = get_bits(&s->gb, bits);
578 v = l1_unscale(bits - 1, v, scale);
579 s->sb_samples[ch][k * 12 + l + m][i] = v;
582 } else {
583 s->sb_samples[ch][k * 12 + l + 0][i] = 0;
584 s->sb_samples[ch][k * 12 + l + 1][i] = 0;
585 s->sb_samples[ch][k * 12 + l + 2][i] = 0;
588 /* next subband in alloc table */
589 j += 1 << bit_alloc_bits;
591 /* XXX: find a way to avoid this duplication of code */
592 for (i = bound; i < sblimit; i++) {
593 bit_alloc_bits = alloc_table[j];
594 b = bit_alloc[0][i];
595 if (b) {
596 int mant, scale0, scale1;
597 scale0 = scale_factors[0][i][k];
598 scale1 = scale_factors[1][i][k];
599 qindex = alloc_table[j + b];
600 bits = ff_mpa_quant_bits[qindex];
601 if (bits < 0) {
602 /* 3 values at the same time */
603 v = get_bits(&s->gb, -bits);
604 steps = ff_mpa_quant_steps[qindex];
605 mant = v % steps;
606 v = v / steps;
607 s->sb_samples[0][k * 12 + l + 0][i] =
608 l2_unscale_group(steps, mant, scale0);
609 s->sb_samples[1][k * 12 + l + 0][i] =
610 l2_unscale_group(steps, mant, scale1);
611 mant = v % steps;
612 v = v / steps;
613 s->sb_samples[0][k * 12 + l + 1][i] =
614 l2_unscale_group(steps, mant, scale0);
615 s->sb_samples[1][k * 12 + l + 1][i] =
616 l2_unscale_group(steps, mant, scale1);
617 s->sb_samples[0][k * 12 + l + 2][i] =
618 l2_unscale_group(steps, v, scale0);
619 s->sb_samples[1][k * 12 + l + 2][i] =
620 l2_unscale_group(steps, v, scale1);
621 } else {
622 for (m = 0; m < 3; m++) {
623 mant = get_bits(&s->gb, bits);
624 s->sb_samples[0][k * 12 + l + m][i] =
625 l1_unscale(bits - 1, mant, scale0);
626 s->sb_samples[1][k * 12 + l + m][i] =
627 l1_unscale(bits - 1, mant, scale1);
630 } else {
631 s->sb_samples[0][k * 12 + l + 0][i] = 0;
632 s->sb_samples[0][k * 12 + l + 1][i] = 0;
633 s->sb_samples[0][k * 12 + l + 2][i] = 0;
634 s->sb_samples[1][k * 12 + l + 0][i] = 0;
635 s->sb_samples[1][k * 12 + l + 1][i] = 0;
636 s->sb_samples[1][k * 12 + l + 2][i] = 0;
638 /* next subband in alloc table */
639 j += 1 << bit_alloc_bits;
641 /* fill remaining samples to zero */
642 for (i = sblimit; i < SBLIMIT; i++) {
643 for (ch = 0; ch < s->nb_channels; ch++) {
644 s->sb_samples[ch][k * 12 + l + 0][i] = 0;
645 s->sb_samples[ch][k * 12 + l + 1][i] = 0;
646 s->sb_samples[ch][k * 12 + l + 2][i] = 0;
651 return 3 * 12;
654 #define SPLIT(dst,sf,n) \
655 if (n == 3) { \
656 int m = (sf * 171) >> 9; \
657 dst = sf - 3 * m; \
658 sf = m; \
659 } else if (n == 4) { \
660 dst = sf & 3; \
661 sf >>= 2; \
662 } else if (n == 5) { \
663 int m = (sf * 205) >> 10; \
664 dst = sf - 5 * m; \
665 sf = m; \
666 } else if (n == 6) { \
667 int m = (sf * 171) >> 10; \
668 dst = sf - 6 * m; \
669 sf = m; \
670 } else { \
671 dst = 0; \
674 static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
675 int n3)
677 SPLIT(slen[3], sf, n3)
678 SPLIT(slen[2], sf, n2)
679 SPLIT(slen[1], sf, n1)
680 slen[0] = sf;
683 static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g,
684 int16_t *exponents)
686 const uint8_t *bstab, *pretab;
687 int len, i, j, k, l, v0, shift, gain, gains[3];
688 int16_t *exp_ptr;
690 exp_ptr = exponents;
691 gain = g->global_gain - 210;
692 shift = g->scalefac_scale + 1;
694 bstab = ff_band_size_long[s->sample_rate_index];
695 pretab = ff_mpa_pretab[g->preflag];
696 for (i = 0; i < g->long_end; i++) {
697 v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;
698 len = bstab[i];
699 for (j = len; j > 0; j--)
700 *exp_ptr++ = v0;
703 if (g->short_start < 13) {
704 bstab = ff_band_size_short[s->sample_rate_index];
705 gains[0] = gain - (g->subblock_gain[0] << 3);
706 gains[1] = gain - (g->subblock_gain[1] << 3);
707 gains[2] = gain - (g->subblock_gain[2] << 3);
708 k = g->long_end;
709 for (i = g->short_start; i < 13; i++) {
710 len = bstab[i];
711 for (l = 0; l < 3; l++) {
712 v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;
713 for (j = len; j > 0; j--)
714 *exp_ptr++ = v0;
720 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
721 int *end_pos2)
723 if (s->in_gb.buffer && *pos >= s->gb.size_in_bits - s->extrasize * 8) {
724 s->gb = s->in_gb;
725 s->in_gb.buffer = NULL;
726 s->extrasize = 0;
727 av_assert2((get_bits_count(&s->gb) & 7) == 0);
728 skip_bits_long(&s->gb, *pos - *end_pos);
729 *end_pos2 =
730 *end_pos = *end_pos2 + get_bits_count(&s->gb) - *pos;
731 *pos = get_bits_count(&s->gb);
735 /* Following is an optimized code for
736 INTFLOAT v = *src
737 if(get_bits1(&s->gb))
738 v = -v;
739 *dst = v;
741 #if USE_FLOATS
742 #define READ_FLIP_SIGN(dst,src) \
743 v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
744 AV_WN32A(dst, v);
745 #else
746 #define READ_FLIP_SIGN(dst,src) \
747 v = -get_bits1(&s->gb); \
748 *(dst) = (*(src) ^ v) - v;
749 #endif
751 static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
752 int16_t *exponents, int end_pos2)
754 int s_index;
755 int i;
756 int last_pos, bits_left;
757 VLC *vlc;
758 int end_pos = FFMIN(end_pos2, s->gb.size_in_bits - s->extrasize * 8);
760 /* low frequencies (called big values) */
761 s_index = 0;
762 for (i = 0; i < 3; i++) {
763 const VLCElem *vlctab;
764 int j, k, l, linbits;
765 j = g->region_size[i];
766 if (j == 0)
767 continue;
768 /* select vlc table */
769 k = g->table_select[i];
770 l = ff_mpa_huff_data[k][0];
771 linbits = ff_mpa_huff_data[k][1];
773 if (!l) {
774 memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
775 s_index += 2 * j;
776 continue;
778 vlctab = ff_huff_vlc[l];
780 /* read huffcode and compute each couple */
781 for (; j > 0; j--) {
782 int exponent, x, y;
783 int v;
784 int pos = get_bits_count(&s->gb);
786 if (pos >= end_pos){
787 switch_buffer(s, &pos, &end_pos, &end_pos2);
788 if (pos >= end_pos)
789 break;
791 y = get_vlc2(&s->gb, vlctab, 7, 3);
793 if (!y) {
794 g->sb_hybrid[s_index ] =
795 g->sb_hybrid[s_index + 1] = 0;
796 s_index += 2;
797 continue;
800 exponent= exponents[s_index];
802 ff_dlog(s->avctx, "region=%d n=%d y=%d exp=%d\n",
803 i, g->region_size[i] - j, y, exponent);
804 if (y & 16) {
805 x = y >> 5;
806 y = y & 0x0f;
807 if (x < 15) {
808 READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
809 } else {
810 x += get_bitsz(&s->gb, linbits);
811 v = l3_unscale(x, exponent);
812 if (get_bits1(&s->gb))
813 v = -v;
814 g->sb_hybrid[s_index] = v;
816 if (y < 15) {
817 READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
818 } else {
819 y += get_bitsz(&s->gb, linbits);
820 v = l3_unscale(y, exponent);
821 if (get_bits1(&s->gb))
822 v = -v;
823 g->sb_hybrid[s_index + 1] = v;
825 } else {
826 x = y >> 5;
827 y = y & 0x0f;
828 x += y;
829 if (x < 15) {
830 READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
831 } else {
832 x += get_bitsz(&s->gb, linbits);
833 v = l3_unscale(x, exponent);
834 if (get_bits1(&s->gb))
835 v = -v;
836 g->sb_hybrid[s_index+!!y] = v;
838 g->sb_hybrid[s_index + !y] = 0;
840 s_index += 2;
844 /* high frequencies */
845 vlc = &ff_huff_quad_vlc[g->count1table_select];
846 last_pos = 0;
847 while (s_index <= 572) {
848 int pos, code;
849 pos = get_bits_count(&s->gb);
850 if (pos >= end_pos) {
851 if (pos > end_pos2 && last_pos) {
852 /* some encoders generate an incorrect size for this
853 part. We must go back into the data */
854 s_index -= 4;
855 skip_bits_long(&s->gb, last_pos - pos);
856 av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
857 if(s->err_recognition & (AV_EF_BITSTREAM|AV_EF_COMPLIANT))
858 s_index=0;
859 break;
861 switch_buffer(s, &pos, &end_pos, &end_pos2);
862 if (pos >= end_pos)
863 break;
865 last_pos = pos;
867 code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
868 ff_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
869 g->sb_hybrid[s_index + 0] =
870 g->sb_hybrid[s_index + 1] =
871 g->sb_hybrid[s_index + 2] =
872 g->sb_hybrid[s_index + 3] = 0;
873 while (code) {
874 static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
875 int v;
876 int pos = s_index + idxtab[code];
877 code ^= 8 >> idxtab[code];
878 READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])
880 s_index += 4;
882 /* skip extension bits */
883 bits_left = end_pos2 - get_bits_count(&s->gb);
884 if (bits_left < 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_COMPLIANT))) {
885 av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
886 s_index=0;
887 } else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_AGGRESSIVE))) {
888 av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
889 s_index = 0;
891 memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));
892 skip_bits_long(&s->gb, bits_left);
894 i = get_bits_count(&s->gb);
895 switch_buffer(s, &i, &end_pos, &end_pos2);
897 return 0;
900 /* Reorder short blocks from bitstream order to interleaved order. It
901 would be faster to do it in parsing, but the code would be far more
902 complicated */
903 static void reorder_block(MPADecodeContext *s, GranuleDef *g)
905 int i, j, len;
906 INTFLOAT *ptr, *dst, *ptr1;
907 INTFLOAT tmp[576];
909 if (g->block_type != 2)
910 return;
912 if (g->switch_point) {
913 if (s->sample_rate_index != 8)
914 ptr = g->sb_hybrid + 36;
915 else
916 ptr = g->sb_hybrid + 72;
917 } else {
918 ptr = g->sb_hybrid;
921 for (i = g->short_start; i < 13; i++) {
922 len = ff_band_size_short[s->sample_rate_index][i];
923 ptr1 = ptr;
924 dst = tmp;
925 for (j = len; j > 0; j--) {
926 *dst++ = ptr[0*len];
927 *dst++ = ptr[1*len];
928 *dst++ = ptr[2*len];
929 ptr++;
931 ptr += 2 * len;
932 memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));
936 #define ISQRT2 FIXR(0.70710678118654752440)
938 static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)
940 int i, j, k, l;
941 int sf_max, sf, len, non_zero_found;
942 INTFLOAT *tab0, *tab1, v1, v2;
943 const INTFLOAT (*is_tab)[16];
944 SUINTFLOAT tmp0, tmp1;
945 int non_zero_found_short[3];
947 /* intensity stereo */
948 if (s->mode_ext & MODE_EXT_I_STEREO) {
949 if (!s->lsf) {
950 is_tab = is_table;
951 sf_max = 7;
952 } else {
953 is_tab = is_table_lsf[g1->scalefac_compress & 1];
954 sf_max = 16;
957 tab0 = g0->sb_hybrid + 576;
958 tab1 = g1->sb_hybrid + 576;
960 non_zero_found_short[0] = 0;
961 non_zero_found_short[1] = 0;
962 non_zero_found_short[2] = 0;
963 k = (13 - g1->short_start) * 3 + g1->long_end - 3;
964 for (i = 12; i >= g1->short_start; i--) {
965 /* for last band, use previous scale factor */
966 if (i != 11)
967 k -= 3;
968 len = ff_band_size_short[s->sample_rate_index][i];
969 for (l = 2; l >= 0; l--) {
970 tab0 -= len;
971 tab1 -= len;
972 if (!non_zero_found_short[l]) {
973 /* test if non zero band. if so, stop doing i-stereo */
974 for (j = 0; j < len; j++) {
975 if (tab1[j] != 0) {
976 non_zero_found_short[l] = 1;
977 goto found1;
980 sf = g1->scale_factors[k + l];
981 if (sf >= sf_max)
982 goto found1;
984 v1 = is_tab[0][sf];
985 v2 = is_tab[1][sf];
986 for (j = 0; j < len; j++) {
987 tmp0 = tab0[j];
988 tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
989 tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
991 } else {
992 found1:
993 if (s->mode_ext & MODE_EXT_MS_STEREO) {
994 /* lower part of the spectrum : do ms stereo
995 if enabled */
996 for (j = 0; j < len; j++) {
997 tmp0 = tab0[j];
998 tmp1 = tab1[j];
999 tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1000 tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1007 non_zero_found = non_zero_found_short[0] |
1008 non_zero_found_short[1] |
1009 non_zero_found_short[2];
1011 for (i = g1->long_end - 1;i >= 0;i--) {
1012 len = ff_band_size_long[s->sample_rate_index][i];
1013 tab0 -= len;
1014 tab1 -= len;
1015 /* test if non zero band. if so, stop doing i-stereo */
1016 if (!non_zero_found) {
1017 for (j = 0; j < len; j++) {
1018 if (tab1[j] != 0) {
1019 non_zero_found = 1;
1020 goto found2;
1023 /* for last band, use previous scale factor */
1024 k = (i == 21) ? 20 : i;
1025 sf = g1->scale_factors[k];
1026 if (sf >= sf_max)
1027 goto found2;
1028 v1 = is_tab[0][sf];
1029 v2 = is_tab[1][sf];
1030 for (j = 0; j < len; j++) {
1031 tmp0 = tab0[j];
1032 tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
1033 tab1[j] = MULLx(tmp0, v2, FRAC_BITS);
1035 } else {
1036 found2:
1037 if (s->mode_ext & MODE_EXT_MS_STEREO) {
1038 /* lower part of the spectrum : do ms stereo
1039 if enabled */
1040 for (j = 0; j < len; j++) {
1041 tmp0 = tab0[j];
1042 tmp1 = tab1[j];
1043 tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
1044 tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);
1049 } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
1050 /* ms stereo ONLY */
1051 /* NOTE: the 1/sqrt(2) normalization factor is included in the
1052 global gain */
1053 #if USE_FLOATS
1054 s->butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
1055 #else
1056 tab0 = g0->sb_hybrid;
1057 tab1 = g1->sb_hybrid;
1058 for (i = 0; i < 576; i++) {
1059 tmp0 = tab0[i];
1060 tmp1 = tab1[i];
1061 tab0[i] = tmp0 + tmp1;
1062 tab1[i] = tmp0 - tmp1;
1064 #endif
1068 #if USE_FLOATS
1069 #if HAVE_MIPSFPU
1070 # include "mips/compute_antialias_float.h"
1071 #endif /* HAVE_MIPSFPU */
1072 #else
1073 #if HAVE_MIPSDSP
1074 # include "mips/compute_antialias_fixed.h"
1075 #endif /* HAVE_MIPSDSP */
1076 #endif /* USE_FLOATS */
1078 #ifndef compute_antialias
1079 #if USE_FLOATS
1080 #define AA(j) do { \
1081 float tmp0 = ptr[-1-j]; \
1082 float tmp1 = ptr[ j]; \
1083 ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1]; \
1084 ptr[ j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0]; \
1085 } while (0)
1086 #else
1087 #define AA(j) do { \
1088 SUINT tmp0 = ptr[-1-j]; \
1089 SUINT tmp1 = ptr[ j]; \
1090 SUINT tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]); \
1091 ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2])); \
1092 ptr[ j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3])); \
1093 } while (0)
1094 #endif
1096 static void compute_antialias(MPADecodeContext *s, GranuleDef *g)
1098 INTFLOAT *ptr;
1099 int n, i;
1101 /* we antialias only "long" bands */
1102 if (g->block_type == 2) {
1103 if (!g->switch_point)
1104 return;
1105 /* XXX: check this for 8000Hz case */
1106 n = 1;
1107 } else {
1108 n = SBLIMIT - 1;
1111 ptr = g->sb_hybrid + 18;
1112 for (i = n; i > 0; i--) {
1113 AA(0);
1114 AA(1);
1115 AA(2);
1116 AA(3);
1117 AA(4);
1118 AA(5);
1119 AA(6);
1120 AA(7);
1122 ptr += 18;
1125 #endif /* compute_antialias */
1127 static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
1128 INTFLOAT *sb_samples, INTFLOAT *mdct_buf)
1130 INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;
1131 INTFLOAT out2[12];
1132 int i, j, mdct_long_end, sblimit;
1134 /* find last non zero block */
1135 ptr = g->sb_hybrid + 576;
1136 ptr1 = g->sb_hybrid + 2 * 18;
1137 while (ptr >= ptr1) {
1138 int32_t *p;
1139 ptr -= 6;
1140 p = (int32_t*)ptr;
1141 if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])
1142 break;
1144 sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
1146 if (g->block_type == 2) {
1147 /* XXX: check for 8000 Hz */
1148 if (g->switch_point)
1149 mdct_long_end = 2;
1150 else
1151 mdct_long_end = 0;
1152 } else {
1153 mdct_long_end = sblimit;
1156 s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
1157 mdct_long_end, g->switch_point,
1158 g->block_type);
1160 buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
1161 ptr = g->sb_hybrid + 18 * mdct_long_end;
1163 for (j = mdct_long_end; j < sblimit; j++) {
1164 /* select frequency inversion */
1165 win = RENAME(ff_mdct_win)[2 + (4 & -(j & 1))];
1166 out_ptr = sb_samples + j;
1168 for (i = 0; i < 6; i++) {
1169 *out_ptr = buf[4*i];
1170 out_ptr += SBLIMIT;
1172 imdct12(out2, ptr + 0);
1173 for (i = 0; i < 6; i++) {
1174 *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*1)];
1175 buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);
1176 out_ptr += SBLIMIT;
1178 imdct12(out2, ptr + 1);
1179 for (i = 0; i < 6; i++) {
1180 *out_ptr = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*2)];
1181 buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);
1182 out_ptr += SBLIMIT;
1184 imdct12(out2, ptr + 2);
1185 for (i = 0; i < 6; i++) {
1186 buf[4*(i + 6*0)] = MULH3(out2[i ], win[i ], 1) + buf[4*(i + 6*0)];
1187 buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
1188 buf[4*(i + 6*2)] = 0;
1190 ptr += 18;
1191 buf += (j&3) != 3 ? 1 : (4*18-3);
1193 /* zero bands */
1194 for (j = sblimit; j < SBLIMIT; j++) {
1195 /* overlap */
1196 out_ptr = sb_samples + j;
1197 for (i = 0; i < 18; i++) {
1198 *out_ptr = buf[4*i];
1199 buf[4*i] = 0;
1200 out_ptr += SBLIMIT;
1202 buf += (j&3) != 3 ? 1 : (4*18-3);
1206 /* main layer3 decoding function */
1207 static int mp_decode_layer3(MPADecodeContext *s)
1209 int nb_granules, main_data_begin;
1210 int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;
1211 GranuleDef *g;
1212 int16_t exponents[576]; //FIXME try INTFLOAT
1213 int ret;
1215 /* read side info */
1216 if (s->lsf) {
1217 ret = handle_crc(s, ((s->nb_channels == 1) ? 8*9 : 8*17));
1218 main_data_begin = get_bits(&s->gb, 8);
1219 skip_bits(&s->gb, s->nb_channels);
1220 nb_granules = 1;
1221 } else {
1222 ret = handle_crc(s, ((s->nb_channels == 1) ? 8*17 : 8*32));
1223 main_data_begin = get_bits(&s->gb, 9);
1224 if (s->nb_channels == 2)
1225 skip_bits(&s->gb, 3);
1226 else
1227 skip_bits(&s->gb, 5);
1228 nb_granules = 2;
1229 for (ch = 0; ch < s->nb_channels; ch++) {
1230 s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
1231 s->granules[ch][1].scfsi = get_bits(&s->gb, 4);
1234 if (ret < 0)
1235 return ret;
1237 for (gr = 0; gr < nb_granules; gr++) {
1238 for (ch = 0; ch < s->nb_channels; ch++) {
1239 ff_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
1240 g = &s->granules[ch][gr];
1241 g->part2_3_length = get_bits(&s->gb, 12);
1242 g->big_values = get_bits(&s->gb, 9);
1243 if (g->big_values > 288) {
1244 av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
1245 return AVERROR_INVALIDDATA;
1248 g->global_gain = get_bits(&s->gb, 8);
1249 /* if MS stereo only is selected, we precompute the
1250 1/sqrt(2) renormalization factor */
1251 if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==
1252 MODE_EXT_MS_STEREO)
1253 g->global_gain -= 2;
1254 if (s->lsf)
1255 g->scalefac_compress = get_bits(&s->gb, 9);
1256 else
1257 g->scalefac_compress = get_bits(&s->gb, 4);
1258 blocksplit_flag = get_bits1(&s->gb);
1259 if (blocksplit_flag) {
1260 g->block_type = get_bits(&s->gb, 2);
1261 if (g->block_type == 0) {
1262 av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
1263 return AVERROR_INVALIDDATA;
1265 g->switch_point = get_bits1(&s->gb);
1266 for (i = 0; i < 2; i++)
1267 g->table_select[i] = get_bits(&s->gb, 5);
1268 for (i = 0; i < 3; i++)
1269 g->subblock_gain[i] = get_bits(&s->gb, 3);
1270 init_short_region(s, g);
1271 } else {
1272 int region_address1, region_address2;
1273 g->block_type = 0;
1274 g->switch_point = 0;
1275 for (i = 0; i < 3; i++)
1276 g->table_select[i] = get_bits(&s->gb, 5);
1277 /* compute huffman coded region sizes */
1278 region_address1 = get_bits(&s->gb, 4);
1279 region_address2 = get_bits(&s->gb, 3);
1280 ff_dlog(s->avctx, "region1=%d region2=%d\n",
1281 region_address1, region_address2);
1282 init_long_region(s, g, region_address1, region_address2);
1284 region_offset2size(g);
1285 compute_band_indexes(s, g);
1287 g->preflag = 0;
1288 if (!s->lsf)
1289 g->preflag = get_bits1(&s->gb);
1290 g->scalefac_scale = get_bits1(&s->gb);
1291 g->count1table_select = get_bits1(&s->gb);
1292 ff_dlog(s->avctx, "block_type=%d switch_point=%d\n",
1293 g->block_type, g->switch_point);
1297 if (!s->adu_mode) {
1298 int skip;
1299 const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb) >> 3);
1300 s->extrasize = av_clip((get_bits_left(&s->gb) >> 3) - s->extrasize, 0,
1301 FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));
1302 av_assert1((get_bits_count(&s->gb) & 7) == 0);
1303 /* now we get bits from the main_data_begin offset */
1304 ff_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
1305 main_data_begin, s->last_buf_size);
1307 memcpy(s->last_buf + s->last_buf_size, ptr, s->extrasize);
1308 s->in_gb = s->gb;
1309 init_get_bits(&s->gb, s->last_buf, (s->last_buf_size + s->extrasize) * 8);
1310 s->last_buf_size <<= 3;
1311 for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
1312 for (ch = 0; ch < s->nb_channels; ch++) {
1313 g = &s->granules[ch][gr];
1314 s->last_buf_size += g->part2_3_length;
1315 memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
1316 compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1319 skip = s->last_buf_size - 8 * main_data_begin;
1320 if (skip >= s->gb.size_in_bits - s->extrasize * 8 && s->in_gb.buffer) {
1321 skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits + s->extrasize * 8);
1322 s->gb = s->in_gb;
1323 s->in_gb.buffer = NULL;
1324 s->extrasize = 0;
1325 } else {
1326 skip_bits_long(&s->gb, skip);
1328 } else {
1329 gr = 0;
1330 s->extrasize = 0;
1333 for (; gr < nb_granules; gr++) {
1334 for (ch = 0; ch < s->nb_channels; ch++) {
1335 g = &s->granules[ch][gr];
1336 bits_pos = get_bits_count(&s->gb);
1338 if (!s->lsf) {
1339 uint8_t *sc;
1340 int slen, slen1, slen2;
1342 /* MPEG-1 scale factors */
1343 slen1 = ff_slen_table[0][g->scalefac_compress];
1344 slen2 = ff_slen_table[1][g->scalefac_compress];
1345 ff_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
1346 if (g->block_type == 2) {
1347 n = g->switch_point ? 17 : 18;
1348 j = 0;
1349 if (slen1) {
1350 for (i = 0; i < n; i++)
1351 g->scale_factors[j++] = get_bits(&s->gb, slen1);
1352 } else {
1353 for (i = 0; i < n; i++)
1354 g->scale_factors[j++] = 0;
1356 if (slen2) {
1357 for (i = 0; i < 18; i++)
1358 g->scale_factors[j++] = get_bits(&s->gb, slen2);
1359 for (i = 0; i < 3; i++)
1360 g->scale_factors[j++] = 0;
1361 } else {
1362 for (i = 0; i < 21; i++)
1363 g->scale_factors[j++] = 0;
1365 } else {
1366 sc = s->granules[ch][0].scale_factors;
1367 j = 0;
1368 for (k = 0; k < 4; k++) {
1369 n = k == 0 ? 6 : 5;
1370 if ((g->scfsi & (0x8 >> k)) == 0) {
1371 slen = (k < 2) ? slen1 : slen2;
1372 if (slen) {
1373 for (i = 0; i < n; i++)
1374 g->scale_factors[j++] = get_bits(&s->gb, slen);
1375 } else {
1376 for (i = 0; i < n; i++)
1377 g->scale_factors[j++] = 0;
1379 } else {
1380 /* simply copy from last granule */
1381 for (i = 0; i < n; i++) {
1382 g->scale_factors[j] = sc[j];
1383 j++;
1387 g->scale_factors[j++] = 0;
1389 } else {
1390 int tindex, tindex2, slen[4], sl, sf;
1392 /* LSF scale factors */
1393 if (g->block_type == 2)
1394 tindex = g->switch_point ? 2 : 1;
1395 else
1396 tindex = 0;
1398 sf = g->scalefac_compress;
1399 if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
1400 /* intensity stereo case */
1401 sf >>= 1;
1402 if (sf < 180) {
1403 lsf_sf_expand(slen, sf, 6, 6, 0);
1404 tindex2 = 3;
1405 } else if (sf < 244) {
1406 lsf_sf_expand(slen, sf - 180, 4, 4, 0);
1407 tindex2 = 4;
1408 } else {
1409 lsf_sf_expand(slen, sf - 244, 3, 0, 0);
1410 tindex2 = 5;
1412 } else {
1413 /* normal case */
1414 if (sf < 400) {
1415 lsf_sf_expand(slen, sf, 5, 4, 4);
1416 tindex2 = 0;
1417 } else if (sf < 500) {
1418 lsf_sf_expand(slen, sf - 400, 5, 4, 0);
1419 tindex2 = 1;
1420 } else {
1421 lsf_sf_expand(slen, sf - 500, 3, 0, 0);
1422 tindex2 = 2;
1423 g->preflag = 1;
1427 j = 0;
1428 for (k = 0; k < 4; k++) {
1429 n = ff_lsf_nsf_table[tindex2][tindex][k];
1430 sl = slen[k];
1431 if (sl) {
1432 for (i = 0; i < n; i++)
1433 g->scale_factors[j++] = get_bits(&s->gb, sl);
1434 } else {
1435 for (i = 0; i < n; i++)
1436 g->scale_factors[j++] = 0;
1439 /* XXX: should compute exact size */
1440 for (; j < 40; j++)
1441 g->scale_factors[j] = 0;
1444 exponents_from_scale_factors(s, g, exponents);
1446 /* read Huffman coded residue */
1447 huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
1448 } /* ch */
1450 if (s->mode == MPA_JSTEREO)
1451 compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);
1453 for (ch = 0; ch < s->nb_channels; ch++) {
1454 g = &s->granules[ch][gr];
1456 reorder_block(s, g);
1457 compute_antialias(s, g);
1458 compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
1460 } /* gr */
1461 if (get_bits_count(&s->gb) < 0)
1462 skip_bits_long(&s->gb, -get_bits_count(&s->gb));
1463 return nb_granules * 18;
1466 static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,
1467 const uint8_t *buf, int buf_size)
1469 int i, nb_frames, ch, ret;
1470 OUT_INT *samples_ptr;
1472 init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);
1473 if (s->error_protection)
1474 s->crc = get_bits(&s->gb, 16);
1476 switch(s->layer) {
1477 case 1:
1478 s->avctx->frame_size = 384;
1479 nb_frames = mp_decode_layer1(s);
1480 break;
1481 case 2:
1482 s->avctx->frame_size = 1152;
1483 nb_frames = mp_decode_layer2(s);
1484 break;
1485 case 3:
1486 s->avctx->frame_size = s->lsf ? 576 : 1152;
1487 default:
1488 nb_frames = mp_decode_layer3(s);
1490 s->last_buf_size=0;
1491 if (s->in_gb.buffer) {
1492 align_get_bits(&s->gb);
1493 i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
1494 if (i >= 0 && i <= BACKSTEP_SIZE) {
1495 memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb) >> 3), i);
1496 s->last_buf_size=i;
1497 } else
1498 av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
1499 s->gb = s->in_gb;
1500 s->in_gb.buffer = NULL;
1501 s->extrasize = 0;
1504 align_get_bits(&s->gb);
1505 av_assert1((get_bits_count(&s->gb) & 7) == 0);
1506 i = (get_bits_left(&s->gb) >> 3) - s->extrasize;
1507 if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {
1508 if (i < 0)
1509 av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
1510 i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
1512 av_assert1(i <= buf_size - HEADER_SIZE && i >= 0);
1513 memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);
1514 s->last_buf_size += i;
1517 if(nb_frames < 0)
1518 return nb_frames;
1520 /* get output buffer */
1521 if (!samples) {
1522 av_assert0(s->frame);
1523 s->frame->nb_samples = s->avctx->frame_size;
1524 if ((ret = ff_get_buffer(s->avctx, s->frame, 0)) < 0)
1525 return ret;
1526 samples = (OUT_INT **)s->frame->extended_data;
1529 /* apply the synthesis filter */
1530 for (ch = 0; ch < s->nb_channels; ch++) {
1531 int sample_stride;
1532 if (s->avctx->sample_fmt == OUT_FMT_P) {
1533 samples_ptr = samples[ch];
1534 sample_stride = 1;
1535 } else {
1536 samples_ptr = samples[0] + ch;
1537 sample_stride = s->nb_channels;
1539 for (i = 0; i < nb_frames; i++) {
1540 RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
1541 &(s->synth_buf_offset[ch]),
1542 RENAME(ff_mpa_synth_window),
1543 &s->dither_state, samples_ptr,
1544 sample_stride, s->sb_samples[ch][i]);
1545 samples_ptr += 32 * sample_stride;
1549 return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
1552 static int decode_frame(AVCodecContext *avctx, AVFrame *frame,
1553 int *got_frame_ptr, AVPacket *avpkt)
1555 const uint8_t *buf = avpkt->data;
1556 int buf_size = avpkt->size;
1557 MPADecodeContext *s = avctx->priv_data;
1558 uint32_t header;
1559 int ret;
1561 int skipped = 0;
1562 while(buf_size && !*buf){
1563 buf++;
1564 buf_size--;
1565 skipped++;
1568 if (buf_size < HEADER_SIZE)
1569 return AVERROR_INVALIDDATA;
1571 header = AV_RB32(buf);
1572 if (header >> 8 == AV_RB32("TAG") >> 8) {
1573 av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");
1574 return buf_size + skipped;
1576 ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
1577 if (ret < 0) {
1578 av_log(avctx, AV_LOG_ERROR, "Header missing\n");
1579 return AVERROR_INVALIDDATA;
1580 } else if (ret == 1) {
1581 /* free format: prepare to compute frame size */
1582 s->frame_size = -1;
1583 return AVERROR_INVALIDDATA;
1585 /* update codec info */
1586 av_channel_layout_uninit(&avctx->ch_layout);
1587 avctx->ch_layout = s->nb_channels == 1 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO :
1588 (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO;
1589 if (!avctx->bit_rate)
1590 avctx->bit_rate = s->bit_rate;
1592 if (s->frame_size <= 0) {
1593 av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
1594 return AVERROR_INVALIDDATA;
1595 } else if (s->frame_size < buf_size) {
1596 av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");
1597 buf_size= s->frame_size;
1600 s->frame = frame;
1602 ret = mp_decode_frame(s, NULL, buf, buf_size);
1603 if (ret >= 0) {
1604 s->frame->nb_samples = avctx->frame_size;
1605 *got_frame_ptr = 1;
1606 avctx->sample_rate = s->sample_rate;
1607 //FIXME maybe move the other codec info stuff from above here too
1608 } else {
1609 av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1610 /* Only return an error if the bad frame makes up the whole packet or
1611 * the error is related to buffer management.
1612 * If there is more data in the packet, just consume the bad frame
1613 * instead of returning an error, which would discard the whole
1614 * packet. */
1615 *got_frame_ptr = 0;
1616 if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
1617 return ret;
1619 s->frame_size = 0;
1620 return buf_size + skipped;
1623 static void mp_flush(MPADecodeContext *ctx)
1625 memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
1626 memset(ctx->mdct_buf, 0, sizeof(ctx->mdct_buf));
1627 ctx->last_buf_size = 0;
1628 ctx->dither_state = 0;
1631 static void flush(AVCodecContext *avctx)
1633 mp_flush(avctx->priv_data);
1636 #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER
1637 static int decode_frame_adu(AVCodecContext *avctx, AVFrame *frame,
1638 int *got_frame_ptr, AVPacket *avpkt)
1640 const uint8_t *buf = avpkt->data;
1641 int buf_size = avpkt->size;
1642 MPADecodeContext *s = avctx->priv_data;
1643 uint32_t header;
1644 int len, ret;
1646 len = buf_size;
1648 // Discard too short frames
1649 if (buf_size < HEADER_SIZE) {
1650 av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
1651 return AVERROR_INVALIDDATA;
1655 if (len > MPA_MAX_CODED_FRAME_SIZE)
1656 len = MPA_MAX_CODED_FRAME_SIZE;
1658 // Get header and restore sync word
1659 header = AV_RB32(buf) | 0xffe00000;
1661 ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);
1662 if (ret < 0) {
1663 av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
1664 return ret;
1666 /* update codec info */
1667 avctx->sample_rate = s->sample_rate;
1668 av_channel_layout_uninit(&avctx->ch_layout);
1669 avctx->ch_layout = s->nb_channels == 1 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO :
1670 (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO;
1671 if (!avctx->bit_rate)
1672 avctx->bit_rate = s->bit_rate;
1674 s->frame_size = len;
1676 s->frame = frame;
1678 ret = mp_decode_frame(s, NULL, buf, buf_size);
1679 if (ret < 0) {
1680 av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");
1681 return ret;
1684 *got_frame_ptr = 1;
1686 return buf_size;
1688 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */
1690 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER
1693 * Context for MP3On4 decoder
1695 typedef struct MP3On4DecodeContext {
1696 int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
1697 int syncword; ///< syncword patch
1698 const uint8_t *coff; ///< channel offsets in output buffer
1699 MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
1700 } MP3On4DecodeContext;
1702 #include "mpeg4audio.h"
1704 /* Next 3 arrays are indexed by channel config number (passed via codecdata) */
1706 /* number of mp3 decoder instances */
1707 static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
1709 /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */
1710 static const uint8_t chan_offset[8][5] = {
1711 { 0 },
1712 { 0 }, // C
1713 { 0 }, // FLR
1714 { 2, 0 }, // C FLR
1715 { 2, 0, 3 }, // C FLR BS
1716 { 2, 0, 3 }, // C FLR BLRS
1717 { 2, 0, 4, 3 }, // C FLR BLRS LFE
1718 { 2, 0, 6, 4, 3 }, // C FLR BLRS BLR LFE
1721 /* mp3on4 channel layouts */
1722 static const int16_t chan_layout[8] = {
1724 AV_CH_LAYOUT_MONO,
1725 AV_CH_LAYOUT_STEREO,
1726 AV_CH_LAYOUT_SURROUND,
1727 AV_CH_LAYOUT_4POINT0,
1728 AV_CH_LAYOUT_5POINT0,
1729 AV_CH_LAYOUT_5POINT1,
1730 AV_CH_LAYOUT_7POINT1
1733 static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
1735 MP3On4DecodeContext *s = avctx->priv_data;
1736 int i;
1738 for (i = 0; i < s->frames; i++)
1739 av_freep(&s->mp3decctx[i]);
1741 return 0;
1745 static av_cold int decode_init_mp3on4(AVCodecContext * avctx)
1747 MP3On4DecodeContext *s = avctx->priv_data;
1748 MPEG4AudioConfig cfg;
1749 int i, ret;
1751 if ((avctx->extradata_size < 2) || !avctx->extradata) {
1752 av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");
1753 return AVERROR_INVALIDDATA;
1756 avpriv_mpeg4audio_get_config2(&cfg, avctx->extradata,
1757 avctx->extradata_size, 1, avctx);
1758 if (!cfg.chan_config || cfg.chan_config > 7) {
1759 av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
1760 return AVERROR_INVALIDDATA;
1762 s->frames = mp3Frames[cfg.chan_config];
1763 s->coff = chan_offset[cfg.chan_config];
1764 av_channel_layout_uninit(&avctx->ch_layout);
1765 av_channel_layout_from_mask(&avctx->ch_layout, chan_layout[cfg.chan_config]);
1767 if (cfg.sample_rate < 16000)
1768 s->syncword = 0xffe00000;
1769 else
1770 s->syncword = 0xfff00000;
1772 /* Init the first mp3 decoder in standard way, so that all tables get builded
1773 * We replace avctx->priv_data with the context of the first decoder so that
1774 * decode_init() does not have to be changed.
1775 * Other decoders will be initialized here copying data from the first context
1777 // Allocate zeroed memory for the first decoder context
1778 s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
1779 if (!s->mp3decctx[0])
1780 return AVERROR(ENOMEM);
1781 // Put decoder context in place to make init_decode() happy
1782 avctx->priv_data = s->mp3decctx[0];
1783 ret = decode_init(avctx);
1784 // Restore mp3on4 context pointer
1785 avctx->priv_data = s;
1786 if (ret < 0)
1787 return ret;
1788 s->mp3decctx[0]->adu_mode = 1; // Set adu mode
1790 /* Create a separate codec/context for each frame (first is already ok).
1791 * Each frame is 1 or 2 channels - up to 5 frames allowed
1793 for (i = 1; i < s->frames; i++) {
1794 s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
1795 if (!s->mp3decctx[i])
1796 return AVERROR(ENOMEM);
1797 s->mp3decctx[i]->adu_mode = 1;
1798 s->mp3decctx[i]->avctx = avctx;
1799 s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;
1800 s->mp3decctx[i]->butterflies_float = s->mp3decctx[0]->butterflies_float;
1803 return 0;
1807 static void flush_mp3on4(AVCodecContext *avctx)
1809 int i;
1810 MP3On4DecodeContext *s = avctx->priv_data;
1812 for (i = 0; i < s->frames; i++)
1813 mp_flush(s->mp3decctx[i]);
1817 static int decode_frame_mp3on4(AVCodecContext *avctx, AVFrame *frame,
1818 int *got_frame_ptr, AVPacket *avpkt)
1820 const uint8_t *buf = avpkt->data;
1821 int buf_size = avpkt->size;
1822 MP3On4DecodeContext *s = avctx->priv_data;
1823 MPADecodeContext *m;
1824 int fsize, len = buf_size, out_size = 0;
1825 uint32_t header;
1826 OUT_INT **out_samples;
1827 OUT_INT *outptr[2];
1828 int fr, ch, ret;
1830 /* get output buffer */
1831 frame->nb_samples = MPA_FRAME_SIZE;
1832 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1833 return ret;
1834 out_samples = (OUT_INT **)frame->extended_data;
1836 // Discard too short frames
1837 if (buf_size < HEADER_SIZE)
1838 return AVERROR_INVALIDDATA;
1840 avctx->bit_rate = 0;
1842 ch = 0;
1843 for (fr = 0; fr < s->frames; fr++) {
1844 fsize = AV_RB16(buf) >> 4;
1845 fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
1846 m = s->mp3decctx[fr];
1847 av_assert1(m);
1849 if (fsize < HEADER_SIZE) {
1850 av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
1851 return AVERROR_INVALIDDATA;
1853 header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
1855 ret = avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);
1856 if (ret < 0) {
1857 av_log(avctx, AV_LOG_ERROR, "Bad header, discard block\n");
1858 return AVERROR_INVALIDDATA;
1861 if (ch + m->nb_channels > avctx->ch_layout.nb_channels ||
1862 s->coff[fr] + m->nb_channels > avctx->ch_layout.nb_channels) {
1863 av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
1864 "channel count\n");
1865 return AVERROR_INVALIDDATA;
1867 ch += m->nb_channels;
1869 outptr[0] = out_samples[s->coff[fr]];
1870 if (m->nb_channels > 1)
1871 outptr[1] = out_samples[s->coff[fr] + 1];
1873 if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0) {
1874 av_log(avctx, AV_LOG_ERROR, "failed to decode channel %d\n", ch);
1875 memset(outptr[0], 0, MPA_FRAME_SIZE*sizeof(OUT_INT));
1876 if (m->nb_channels > 1)
1877 memset(outptr[1], 0, MPA_FRAME_SIZE*sizeof(OUT_INT));
1878 ret = m->nb_channels * MPA_FRAME_SIZE*sizeof(OUT_INT);
1881 out_size += ret;
1882 buf += fsize;
1883 len -= fsize;
1885 avctx->bit_rate += m->bit_rate;
1887 if (ch != avctx->ch_layout.nb_channels) {
1888 av_log(avctx, AV_LOG_ERROR, "failed to decode all channels\n");
1889 return AVERROR_INVALIDDATA;
1892 /* update codec info */
1893 avctx->sample_rate = s->mp3decctx[0]->sample_rate;
1895 frame->nb_samples = out_size / (avctx->ch_layout.nb_channels * sizeof(OUT_INT));
1896 *got_frame_ptr = 1;
1898 return buf_size;
1900 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */