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
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"
42 #include "mpegaudiodsp.h"
46 * - test lsf / mpeg25 extensively.
49 #include "mpegaudio.h"
50 #include "mpegaudiodecheader.h"
52 #define BACKSTEP_SIZE 512
54 #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES
56 /* layer 3 "granule" */
57 typedef struct GranuleDef
{
62 int scalefac_compress
;
67 uint8_t scalefac_scale
;
68 uint8_t count1table_select
;
69 int region_size
[3]; /* number of huffman codes in each region */
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 */
76 typedef struct MPADecodeContext
{
78 uint8_t last_buf
[LAST_BUF_SIZE
];
81 /* next header (used in free format parsing) */
82 uint32_t free_format_next_header
;
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
93 AVCodecContext
* avctx
;
95 void (*butterflies_float
)(float *av_restrict v1
, float *av_restrict v2
, int len
);
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
)
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
;
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);
142 g
->region_size
[0] = (72 / 2);
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);
149 g
->region_size
[0] = (108 / 2);
151 g
->region_size
[1] = (576 / 2);
154 static void init_long_region(MPADecodeContext
*s
, GranuleDef
*g
,
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)
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
)
196 shift
= ff_scale_factor_modshift
[scale_factor
];
199 val
= MUL64((int)(mant
+ (-1U << n
) + 1), scale_factor_mult
[n
-1][mod
]);
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
)
209 shift
= ff_scale_factor_modshift
[scale_factor
];
213 val
= (mant
- (steps
>> 1)) * scale_factor_mult2
[steps
>> 2][mod
];
214 /* NOTE: at this point, 0 <= shift <= 21 */
216 val
= (val
+ (1 << (shift
- 1))) >> shift
;
220 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
221 static inline int l3_unscale(int value
, int exponent
)
226 e
= ff_table_4_3_exp
[4 * value
+ (exponent
& 3)];
227 m
= ff_table_4_3_value
[4 * value
+ (exponent
& 3)];
231 av_log(NULL
, AV_LOG_WARNING
, "l3_unscale: e is %d\n", e
);
235 m
= (m
+ ((1U << e
) >> 1)) >> e
;
240 static av_cold
void decode_init_static(void)
244 /* scale factor multiply for layer 1 */
245 for (i
= 0; i
< 15; i
++) {
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
,
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
++) {
267 for (j
= 0; j
< 2; j
++) {
268 e
= -(j
+ 1) * ((i
+ 1) >> 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
;
291 AVFloatDSPContext
*fdsp
;
292 fdsp
= avpriv_float_dsp_alloc(avctx
->flags
& AV_CODEC_FLAG_BITEXACT
);
294 return AVERROR(ENOMEM
);
295 s
->butterflies_float
= fdsp
->butterflies_float
;
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
;
306 avctx
->sample_fmt
= OUT_FMT_P
;
307 s
->err_recognition
= avctx
->err_recognition
;
309 if (avctx
->codec_id
== AV_CODEC_ID_MP3ADU
)
312 ff_thread_once(&init_static_once
, decode_init_static
);
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
324 static void imdct12(INTFLOAT
*out
, SUINTFLOAT
*in
)
326 SUINTFLOAT in0
, in1
, in2
, in3
, in4
, in5
, t1
, t2
;
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];
337 in2
= MULH3(in2
, C3
, 2);
338 in3
= MULH3(in3
, C3
, 4);
341 t2
= MULH3(in1
- in5
, C4
, 2);
351 in1
= MULH3(in5
+ in3
, C5
, 1);
358 in5
= MULH3(in5
- in3
, C6
, 2);
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
);
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
);
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);
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
;
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
];
399 ret
= handle_crc(s
, (s
->nb_channels
== 1) ? 8*16 : 8*32);
403 if (s
->mode
== MPA_JSTEREO
)
404 bound
= (s
->mode_ext
+ 1) * 4;
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);
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
];
437 mant
= get_bits(&s
->gb
, n
+ 1);
438 v
= l1_unscale(n
, mant
, scale_factors
[ch
][i
]);
442 s
->sb_samples
[ch
][j
][i
] = v
;
445 for (i
= bound
; i
< SBLIMIT
; i
++) {
446 n
= allocation
[0][i
];
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
;
454 s
->sb_samples
[0][j
][i
] = 0;
455 s
->sb_samples
[1][j
][i
] = 0;
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
;
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;
484 ff_dlog(s
->avctx
, "bound=%d sblimit=%d\n", bound
, sblimit
);
490 /* parse bit allocation */
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
);
503 j
+= 1 << bit_alloc_bits
;
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);
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
]) {
526 sf
[0] = get_bits(&s
->gb
, 6);
527 sf
[1] = get_bits(&s
->gb
, 6);
528 sf
[2] = get_bits(&s
->gb
, 6);
531 sf
[0] = get_bits(&s
->gb
, 6);
536 sf
[0] = get_bits(&s
->gb
, 6);
537 sf
[2] = get_bits(&s
->gb
, 6);
541 sf
[0] = get_bits(&s
->gb
, 6);
542 sf
[2] = get_bits(&s
->gb
, 6);
551 for (k
= 0; k
< 3; k
++) {
552 for (l
= 0; l
< 12; l
+= 3) {
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
];
559 scale
= scale_factors
[ch
][i
][k
];
560 qindex
= alloc_table
[j
+b
];
561 bits
= ff_mpa_quant_bits
[qindex
];
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
);
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
;
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
];
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
];
602 /* 3 values at the same time */
603 v
= get_bits(&s
->gb
, -bits
);
604 steps
= ff_mpa_quant_steps
[qindex
];
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
);
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
);
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
);
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;
654 #define SPLIT(dst,sf,n) \
656 int m = (sf * 171) >> 9; \
659 } else if (n == 4) { \
662 } else if (n == 5) { \
663 int m = (sf * 205) >> 10; \
666 } else if (n == 6) { \
667 int m = (sf * 171) >> 10; \
674 static av_always_inline
void lsf_sf_expand(int *slen
, int sf
, int n1
, int n2
,
677 SPLIT(slen
[3], sf
, n3
)
678 SPLIT(slen
[2], sf
, n2
)
679 SPLIT(slen
[1], sf
, n1
)
683 static void exponents_from_scale_factors(MPADecodeContext
*s
, GranuleDef
*g
,
686 const uint8_t *bstab
, *pretab
;
687 int len
, i
, j
, k
, l
, v0
, shift
, gain
, gains
[3];
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;
699 for (j
= len
; j
> 0; j
--)
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);
709 for (i
= g
->short_start
; i
< 13; 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
--)
720 static void switch_buffer(MPADecodeContext
*s
, int *pos
, int *end_pos
,
723 if (s
->in_gb
.buffer
&& *pos
>= s
->gb
.size_in_bits
- s
->extrasize
* 8) {
725 s
->in_gb
.buffer
= NULL
;
727 av_assert2((get_bits_count(&s
->gb
) & 7) == 0);
728 skip_bits_long(&s
->gb
, *pos
- *end_pos
);
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
737 if(get_bits1(&s->gb))
742 #define READ_FLIP_SIGN(dst,src) \
743 v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31); \
746 #define READ_FLIP_SIGN(dst,src) \
747 v = -get_bits1(&s->gb); \
748 *(dst) = (*(src) ^ v) - v;
751 static int huffman_decode(MPADecodeContext
*s
, GranuleDef
*g
,
752 int16_t *exponents
, int end_pos2
)
756 int last_pos
, bits_left
;
758 int end_pos
= FFMIN(end_pos2
, s
->gb
.size_in_bits
- s
->extrasize
* 8);
760 /* low frequencies (called big values) */
762 for (i
= 0; i
< 3; i
++) {
763 const VLCElem
*vlctab
;
764 int j
, k
, l
, linbits
;
765 j
= g
->region_size
[i
];
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];
774 memset(&g
->sb_hybrid
[s_index
], 0, sizeof(*g
->sb_hybrid
) * 2 * j
);
778 vlctab
= ff_huff_vlc
[l
];
780 /* read huffcode and compute each couple */
784 int pos
= get_bits_count(&s
->gb
);
787 switch_buffer(s
, &pos
, &end_pos
, &end_pos2
);
791 y
= get_vlc2(&s
->gb
, vlctab
, 7, 3);
794 g
->sb_hybrid
[s_index
] =
795 g
->sb_hybrid
[s_index
+ 1] = 0;
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
);
808 READ_FLIP_SIGN(g
->sb_hybrid
+ s_index
, RENAME(expval_table
)[exponent
] + x
)
810 x
+= get_bitsz(&s
->gb
, linbits
);
811 v
= l3_unscale(x
, exponent
);
812 if (get_bits1(&s
->gb
))
814 g
->sb_hybrid
[s_index
] = v
;
817 READ_FLIP_SIGN(g
->sb_hybrid
+ s_index
+ 1, RENAME(expval_table
)[exponent
] + y
)
819 y
+= get_bitsz(&s
->gb
, linbits
);
820 v
= l3_unscale(y
, exponent
);
821 if (get_bits1(&s
->gb
))
823 g
->sb_hybrid
[s_index
+ 1] = v
;
830 READ_FLIP_SIGN(g
->sb_hybrid
+ s_index
+ !!y
, RENAME(expval_table
)[exponent
] + x
)
832 x
+= get_bitsz(&s
->gb
, linbits
);
833 v
= l3_unscale(x
, exponent
);
834 if (get_bits1(&s
->gb
))
836 g
->sb_hybrid
[s_index
+!!y
] = v
;
838 g
->sb_hybrid
[s_index
+ !y
] = 0;
844 /* high frequencies */
845 vlc
= &ff_huff_quad_vlc
[g
->count1table_select
];
847 while (s_index
<= 572) {
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 */
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
))
861 switch_buffer(s
, &pos
, &end_pos
, &end_pos2
);
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;
874 static const int idxtab
[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };
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
])
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
);
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
);
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
);
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
903 static void reorder_block(MPADecodeContext
*s
, GranuleDef
*g
)
906 INTFLOAT
*ptr
, *dst
, *ptr1
;
909 if (g
->block_type
!= 2)
912 if (g
->switch_point
) {
913 if (s
->sample_rate_index
!= 8)
914 ptr
= g
->sb_hybrid
+ 36;
916 ptr
= g
->sb_hybrid
+ 72;
921 for (i
= g
->short_start
; i
< 13; i
++) {
922 len
= ff_band_size_short
[s
->sample_rate_index
][i
];
925 for (j
= len
; j
> 0; j
--) {
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
)
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
) {
953 is_tab
= is_table_lsf
[g1
->scalefac_compress
& 1];
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 */
968 len
= ff_band_size_short
[s
->sample_rate_index
][i
];
969 for (l
= 2; l
>= 0; l
--) {
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
++) {
976 non_zero_found_short
[l
] = 1;
980 sf
= g1
->scale_factors
[k
+ l
];
986 for (j
= 0; j
< len
; j
++) {
988 tab0
[j
] = MULLx(tmp0
, v1
, FRAC_BITS
);
989 tab1
[j
] = MULLx(tmp0
, v2
, FRAC_BITS
);
993 if (s
->mode_ext
& MODE_EXT_MS_STEREO
) {
994 /* lower part of the spectrum : do ms stereo
996 for (j
= 0; j
< len
; 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
];
1015 /* test if non zero band. if so, stop doing i-stereo */
1016 if (!non_zero_found
) {
1017 for (j
= 0; j
< len
; j
++) {
1023 /* for last band, use previous scale factor */
1024 k
= (i
== 21) ? 20 : i
;
1025 sf
= g1
->scale_factors
[k
];
1030 for (j
= 0; j
< len
; j
++) {
1032 tab0
[j
] = MULLx(tmp0
, v1
, FRAC_BITS
);
1033 tab1
[j
] = MULLx(tmp0
, v2
, FRAC_BITS
);
1037 if (s
->mode_ext
& MODE_EXT_MS_STEREO
) {
1038 /* lower part of the spectrum : do ms stereo
1040 for (j
= 0; j
< len
; 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
1054 s
->butterflies_float(g0
->sb_hybrid
, g1
->sb_hybrid
, 576);
1056 tab0
= g0
->sb_hybrid
;
1057 tab1
= g1
->sb_hybrid
;
1058 for (i
= 0; i
< 576; i
++) {
1061 tab0
[i
] = tmp0
+ tmp1
;
1062 tab1
[i
] = tmp0
- tmp1
;
1070 # include "mips/compute_antialias_float.h"
1071 #endif /* HAVE_MIPSFPU */
1074 # include "mips/compute_antialias_fixed.h"
1075 #endif /* HAVE_MIPSDSP */
1076 #endif /* USE_FLOATS */
1078 #ifndef compute_antialias
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]; \
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])); \
1096 static void compute_antialias(MPADecodeContext
*s
, GranuleDef
*g
)
1101 /* we antialias only "long" bands */
1102 if (g
->block_type
== 2) {
1103 if (!g
->switch_point
)
1105 /* XXX: check this for 8000Hz case */
1111 ptr
= g
->sb_hybrid
+ 18;
1112 for (i
= n
; i
> 0; i
--) {
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
;
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
) {
1141 if (p
[0] | p
[1] | p
[2] | p
[3] | p
[4] | p
[5])
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
)
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
,
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
];
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);
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);
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;
1191 buf
+= (j
&3) != 3 ? 1 : (4*18-3);
1194 for (j
= sblimit
; j
< SBLIMIT
; j
++) {
1196 out_ptr
= sb_samples
+ j
;
1197 for (i
= 0; i
< 18; i
++) {
1198 *out_ptr
= buf
[4*i
];
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
;
1212 int16_t exponents
[576]; //FIXME try INTFLOAT
1215 /* read side info */
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
);
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);
1227 skip_bits(&s
->gb
, 5);
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);
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
)) ==
1253 g
->global_gain
-= 2;
1255 g
->scalefac_compress
= get_bits(&s
->gb
, 9);
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
);
1272 int region_address1
, region_address2
;
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
);
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
);
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
);
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);
1323 s
->in_gb
.buffer
= NULL
;
1326 skip_bits_long(&s
->gb
, skip
);
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
);
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;
1350 for (i
= 0; i
< n
; i
++)
1351 g
->scale_factors
[j
++] = get_bits(&s
->gb
, slen1
);
1353 for (i
= 0; i
< n
; i
++)
1354 g
->scale_factors
[j
++] = 0;
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;
1362 for (i
= 0; i
< 21; i
++)
1363 g
->scale_factors
[j
++] = 0;
1366 sc
= s
->granules
[ch
][0].scale_factors
;
1368 for (k
= 0; k
< 4; k
++) {
1370 if ((g
->scfsi
& (0x8 >> k
)) == 0) {
1371 slen
= (k
< 2) ? slen1
: slen2
;
1373 for (i
= 0; i
< n
; i
++)
1374 g
->scale_factors
[j
++] = get_bits(&s
->gb
, slen
);
1376 for (i
= 0; i
< n
; i
++)
1377 g
->scale_factors
[j
++] = 0;
1380 /* simply copy from last granule */
1381 for (i
= 0; i
< n
; i
++) {
1382 g
->scale_factors
[j
] = sc
[j
];
1387 g
->scale_factors
[j
++] = 0;
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;
1398 sf
= g
->scalefac_compress
;
1399 if ((s
->mode_ext
& MODE_EXT_I_STEREO
) && ch
== 1) {
1400 /* intensity stereo case */
1403 lsf_sf_expand(slen
, sf
, 6, 6, 0);
1405 } else if (sf
< 244) {
1406 lsf_sf_expand(slen
, sf
- 180, 4, 4, 0);
1409 lsf_sf_expand(slen
, sf
- 244, 3, 0, 0);
1415 lsf_sf_expand(slen
, sf
, 5, 4, 4);
1417 } else if (sf
< 500) {
1418 lsf_sf_expand(slen
, sf
- 400, 5, 4, 0);
1421 lsf_sf_expand(slen
, sf
- 500, 3, 0, 0);
1428 for (k
= 0; k
< 4; k
++) {
1429 n
= ff_lsf_nsf_table
[tindex2
][tindex
][k
];
1432 for (i
= 0; i
< n
; i
++)
1433 g
->scale_factors
[j
++] = get_bits(&s
->gb
, sl
);
1435 for (i
= 0; i
< n
; i
++)
1436 g
->scale_factors
[j
++] = 0;
1439 /* XXX: should compute exact size */
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
);
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
]);
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);
1478 s
->avctx
->frame_size
= 384;
1479 nb_frames
= mp_decode_layer1(s
);
1482 s
->avctx
->frame_size
= 1152;
1483 nb_frames
= mp_decode_layer2(s
);
1486 s
->avctx
->frame_size
= s
->lsf
? 576 : 1152;
1488 nb_frames
= mp_decode_layer3(s
);
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
);
1498 av_log(s
->avctx
, AV_LOG_ERROR
, "invalid old backstep %d\n", i
);
1500 s
->in_gb
.buffer
= NULL
;
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) {
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
;
1520 /* get output buffer */
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)
1526 samples
= (OUT_INT
**)s
->frame
->extended_data
;
1529 /* apply the synthesis filter */
1530 for (ch
= 0; ch
< s
->nb_channels
; ch
++) {
1532 if (s
->avctx
->sample_fmt
== OUT_FMT_P
) {
1533 samples_ptr
= samples
[ch
];
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
;
1562 while(buf_size
&& !*buf
){
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
);
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 */
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
;
1602 ret
= mp_decode_frame(s
, NULL
, buf
, buf_size
);
1604 s
->frame
->nb_samples
= avctx
->frame_size
;
1606 avctx
->sample_rate
= s
->sample_rate
;
1607 //FIXME maybe move the other codec info stuff from above here too
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
1616 if (buf_size
== avpkt
->size
|| ret
!= AVERROR_INVALIDDATA
)
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
;
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
);
1663 av_log(avctx
, AV_LOG_ERROR
, "Invalid frame header\n");
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
;
1678 ret
= mp_decode_frame(s
, NULL
, buf
, buf_size
);
1680 av_log(avctx
, AV_LOG_ERROR
, "Error while decoding MPEG audio frame.\n");
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] = {
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] = {
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
;
1738 for (i
= 0; i
< s
->frames
; i
++)
1739 av_freep(&s
->mp3decctx
[i
]);
1745 static av_cold
int decode_init_mp3on4(AVCodecContext
* avctx
)
1747 MP3On4DecodeContext
*s
= avctx
->priv_data
;
1748 MPEG4AudioConfig cfg
;
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;
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
;
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
;
1807 static void flush_mp3on4(AVCodecContext
*avctx
)
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;
1826 OUT_INT
**out_samples
;
1830 /* get output buffer */
1831 frame
->nb_samples
= MPA_FRAME_SIZE
;
1832 if ((ret
= ff_get_buffer(avctx
, frame
, 0)) < 0)
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;
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
];
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
);
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 "
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
);
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
));
1900 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */