falkTX
/
FFmpeg
mirror of https://github.com/falkTX/FFmpeg.git
1
0
Fork 0
Code Issues 0 Releases 338 Wiki Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
47832 Commits
32 Branches
292MB
Tree: b6e7041f90
Branches Tags
${ item.name }
Create branch ${ searchTerm }
from 'b6e7041f90'
${ noResults }
FFmpeg/libavcodec/mpegaudiodec.c

2082 lines
68KB
Raw Blame History 

  1. /*

  2.  * MPEG Audio decoder

  3.  * Copyright (c) 2001, 2002 Fabrice Bellard

  4.  *

  5.  * This file is part of FFmpeg.

  6.  *

  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.

  11.  *

  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.

  16.  *

  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

  20.  */

  21. 

  22. /**

  23.  * @file

  24.  * MPEG Audio decoder

  25.  */

  26. 

  27. #include "libavutil/avassert.h"

  28. #include "libavutil/channel_layout.h"

  29. #include "libavutil/libm.h"

  30. #include "avcodec.h"

  31. #include "get_bits.h"

  32. #include "internal.h"

  33. #include "mathops.h"

  34. #include "mpegaudiodsp.h"

  35. #include "dsputil.h"

  36. 

  37. /*

  38.  * TODO:

  39.  *  - test lsf / mpeg25 extensively.

  40.  */

  41. 

  42. #include "mpegaudio.h"

  43. #include "mpegaudiodecheader.h"

  44. 

  45. #define BACKSTEP_SIZE 512

  46. #define EXTRABYTES 24

  47. #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES

  48. 

  49. /* layer 3 "granule" */

  50. typedef struct GranuleDef {

  51.     uint8_t scfsi;

  52.     int part2_3_length;

  53.     int big_values;

  54.     int global_gain;

  55.     int scalefac_compress;

  56.     uint8_t block_type;

  57.     uint8_t switch_point;

  58.     int table_select[3];

  59.     int subblock_gain[3];

  60.     uint8_t scalefac_scale;

  61.     uint8_t count1table_select;

  62.     int region_size[3]; /* number of huffman codes in each region */

  63.     int preflag;

  64.     int short_start, long_end; /* long/short band indexes */

  65.     uint8_t scale_factors[40];

  66.     DECLARE_ALIGNED(16, INTFLOAT, sb_hybrid)[SBLIMIT * 18]; /* 576 samples */

  67. } GranuleDef;

  68. 

  69. typedef struct MPADecodeContext {

  70.     MPA_DECODE_HEADER

  71.     uint8_t last_buf[LAST_BUF_SIZE];

  72.     int last_buf_size;

  73.     /* next header (used in free format parsing) */

  74.     uint32_t free_format_next_header;

  75.     GetBitContext gb;

  76.     GetBitContext in_gb;

  77.     DECLARE_ALIGNED(32, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512 * 2];

  78.     int synth_buf_offset[MPA_MAX_CHANNELS];

  79.     DECLARE_ALIGNED(32, INTFLOAT, sb_samples)[MPA_MAX_CHANNELS][36][SBLIMIT];

  80.     INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */

  81.     GranuleDef granules[2][2]; /* Used in Layer 3 */

  82.     int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3

  83.     int dither_state;

  84.     int err_recognition;

  85.     AVCodecContext* avctx;

  86.     MPADSPContext mpadsp;

  87.     DSPContext dsp;

  88.     AVFrame frame;

  89. } MPADecodeContext;

  90. 

  91. #if CONFIG_FLOAT

  92. #   define SHR(a,b)       ((a)*(1.0f/(1<<(b))))

  93. #   define FIXR_OLD(a)    ((int)((a) * FRAC_ONE + 0.5))

  94. #   define FIXR(x)        ((float)(x))

  95. #   define FIXHR(x)       ((float)(x))

  96. #   define MULH3(x, y, s) ((s)*(y)*(x))

  97. #   define MULLx(x, y, s) ((y)*(x))

  98. #   define RENAME(a) a ## _float

  99. #   define OUT_FMT   AV_SAMPLE_FMT_FLT

  100. #   define OUT_FMT_P AV_SAMPLE_FMT_FLTP

  101. #else

  102. #   define SHR(a,b)       ((a)>>(b))

  103. /* WARNING: only correct for positive numbers */

  104. #   define FIXR_OLD(a)    ((int)((a) * FRAC_ONE + 0.5))

  105. #   define FIXR(a)        ((int)((a) * FRAC_ONE + 0.5))

  106. #   define FIXHR(a)       ((int)((a) * (1LL<<32) + 0.5))

  107. #   define MULH3(x, y, s) MULH((s)*(x), y)

  108. #   define MULLx(x, y, s) MULL(x,y,s)

  109. #   define RENAME(a)      a ## _fixed

  110. #   define OUT_FMT   AV_SAMPLE_FMT_S16

  111. #   define OUT_FMT_P AV_SAMPLE_FMT_S16P

  112. #endif

  113. 

  114. /****************/

  115. 

  116. #define HEADER_SIZE 4

  117. 

  118. #include "mpegaudiodata.h"

  119. #include "mpegaudiodectab.h"

  120. 

  121. /* vlc structure for decoding layer 3 huffman tables */

  122. static VLC huff_vlc[16];

  123. static VLC_TYPE huff_vlc_tables[

  124.     0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +

  125.   142 + 204 + 190 + 170 + 542 + 460 + 662 + 414

  126.   ][2];

  127. static const int huff_vlc_tables_sizes[16] = {

  128.     0,  128,  128,  128,  130,  128,  154,  166,

  129.   142,  204,  190,  170,  542,  460,  662,  414

  130. };

  131. static VLC huff_quad_vlc[2];

  132. static VLC_TYPE  huff_quad_vlc_tables[128+16][2];

  133. static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };

  134. /* computed from band_size_long */

  135. static uint16_t band_index_long[9][23];

  136. #include "mpegaudio_tablegen.h"

  137. /* intensity stereo coef table */

  138. static INTFLOAT is_table[2][16];

  139. static INTFLOAT is_table_lsf[2][2][16];

  140. static INTFLOAT csa_table[8][4];

  141. 

  142. static int16_t division_tab3[1<<6 ];

  143. static int16_t division_tab5[1<<8 ];

  144. static int16_t division_tab9[1<<11];

  145. 

  146. static int16_t * const division_tabs[4] = {

  147.     division_tab3, division_tab5, NULL, division_tab9

  148. };

  149. 

  150. /* lower 2 bits: modulo 3, higher bits: shift */

  151. static uint16_t scale_factor_modshift[64];

  152. /* [i][j]:  2^(-j/3) * FRAC_ONE * 2^(i+2) / (2^(i+2) - 1) */

  153. static int32_t scale_factor_mult[15][3];

  154. /* mult table for layer 2 group quantization */

  155. 

  156. #define SCALE_GEN(v) \

  157. { FIXR_OLD(1.0 * (v)), FIXR_OLD(0.7937005259 * (v)), FIXR_OLD(0.6299605249 * (v)) }

  158. 

  159. static const int32_t scale_factor_mult2[3][3] = {

  160.     SCALE_GEN(4.0 / 3.0), /* 3 steps */

  161.     SCALE_GEN(4.0 / 5.0), /* 5 steps */

  162.     SCALE_GEN(4.0 / 9.0), /* 9 steps */

  163. };

  164. 

  165. /**

  166.  * Convert region offsets to region sizes and truncate

  167.  * size to big_values.

  168.  */

  169. static void ff_region_offset2size(GranuleDef *g)

  170. {

  171.     int i, k, j = 0;

  172.     g->region_size[2] = 576 / 2;

  173.     for (i = 0; i < 3; i++) {

  174.         k = FFMIN(g->region_size[i], g->big_values);

  175.         g->region_size[i] = k - j;

  176.         j = k;

  177.     }

  178. }

  179. 

  180. static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g)

  181. {

  182.     if (g->block_type == 2) {

  183.         if (s->sample_rate_index != 8)

  184.             g->region_size[0] = (36 / 2);

  185.         else

  186.             g->region_size[0] = (72 / 2);

  187.     } else {

  188.         if (s->sample_rate_index <= 2)

  189.             g->region_size[0] = (36 / 2);

  190.         else if (s->sample_rate_index != 8)

  191.             g->region_size[0] = (54 / 2);

  192.         else

  193.             g->region_size[0] = (108 / 2);

  194.     }

  195.     g->region_size[1] = (576 / 2);

  196. }

  197. 

  198. static void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2)

  199. {

  200.     int l;

  201.     g->region_size[0] = band_index_long[s->sample_rate_index][ra1 + 1] >> 1;

  202.     /* should not overflow */

  203.     l = FFMIN(ra1 + ra2 + 2, 22);

  204.     g->region_size[1] = band_index_long[s->sample_rate_index][      l] >> 1;

  205. }

  206. 

  207. static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g)

  208. {

  209.     if (g->block_type == 2) {

  210.         if (g->switch_point) {

  211.             if(s->sample_rate_index == 8)

  212.                 av_log_ask_for_sample(s->avctx, "switch point in 8khz\n");

  213.             /* if switched mode, we handle the 36 first samples as

  214.                 long blocks.  For 8000Hz, we handle the 72 first

  215.                 exponents as long blocks */

  216.             if (s->sample_rate_index <= 2)

  217.                 g->long_end = 8;

  218.             else

  219.                 g->long_end = 6;

  220. 

  221.             g->short_start = 3;

  222.         } else {

  223.             g->long_end    = 0;

  224.             g->short_start = 0;

  225.         }

  226.     } else {

  227.         g->short_start = 13;

  228.         g->long_end    = 22;

  229.     }

  230. }

  231. 

  232. /* layer 1 unscaling */

  233. /* n = number of bits of the mantissa minus 1 */

  234. static inline int l1_unscale(int n, int mant, int scale_factor)

  235. {

  236.     int shift, mod;

  237.     int64_t val;

  238. 

  239.     shift   = scale_factor_modshift[scale_factor];

  240.     mod     = shift & 3;

  241.     shift >>= 2;

  242.     val     = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);

  243.     shift  += n;

  244.     /* NOTE: at this point, 1 <= shift >= 21 + 15 */

  245.     return (int)((val + (1LL << (shift - 1))) >> shift);

  246. }

  247. 

  248. static inline int l2_unscale_group(int steps, int mant, int scale_factor)

  249. {

  250.     int shift, mod, val;

  251. 

  252.     shift   = scale_factor_modshift[scale_factor];

  253.     mod     = shift & 3;

  254.     shift >>= 2;

  255. 

  256.     val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];

  257.     /* NOTE: at this point, 0 <= shift <= 21 */

  258.     if (shift > 0)

  259.         val = (val + (1 << (shift - 1))) >> shift;

  260.     return val;

  261. }

  262. 

  263. /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */

  264. static inline int l3_unscale(int value, int exponent)

  265. {

  266.     unsigned int m;

  267.     int e;

  268. 

  269.     e  = table_4_3_exp  [4 * value + (exponent & 3)];

  270.     m  = table_4_3_value[4 * value + (exponent & 3)];

  271.     e -= exponent >> 2;

  272. #ifdef DEBUG

  273.     if(e < 1)

  274.         av_log(NULL, AV_LOG_WARNING, "l3_unscale: e is %d\n", e);

  275. #endif

  276.     if (e > 31)

  277.         return 0;

  278.     m = (m + (1 << (e - 1))) >> e;

  279. 

  280.     return m;

  281. }

  282. 

  283. static av_cold void decode_init_static(void)

  284. {

  285.     int i, j, k;

  286.     int offset;

  287. 

  288.     /* scale factors table for layer 1/2 */

  289.     for (i = 0; i < 64; i++) {

  290.         int shift, mod;

  291.         /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */

  292.         shift = i / 3;

  293.         mod   = i % 3;

  294.         scale_factor_modshift[i] = mod | (shift << 2);

  295.     }

  296. 

  297.     /* scale factor multiply for layer 1 */

  298.     for (i = 0; i < 15; i++) {

  299.         int n, norm;

  300.         n = i + 2;

  301.         norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);

  302.         scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0          * 2.0), FRAC_BITS);

  303.         scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);

  304.         scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);

  305.         av_dlog(NULL, "%d: norm=%x s=%x %x %x\n", i, norm,

  306.                 scale_factor_mult[i][0],

  307.                 scale_factor_mult[i][1],

  308.                 scale_factor_mult[i][2]);

  309.     }

  310. 

  311.     RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));

  312. 

  313.     /* huffman decode tables */

  314.     offset = 0;

  315.     for (i = 1; i < 16; i++) {

  316.         const HuffTable *h = &mpa_huff_tables[i];

  317.         int xsize, x, y;

  318.         uint8_t  tmp_bits [512] = { 0 };

  319.         uint16_t tmp_codes[512] = { 0 };

  320. 

  321.         xsize = h->xsize;

  322. 

  323.         j = 0;

  324.         for (x = 0; x < xsize; x++) {

  325.             for (y = 0; y < xsize; y++) {

  326.                 tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j  ];

  327.                 tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];

  328.             }

  329.         }

  330. 

  331.         /* XXX: fail test */

  332.         huff_vlc[i].table = huff_vlc_tables+offset;

  333.         huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];

  334.         init_vlc(&huff_vlc[i], 7, 512,

  335.                  tmp_bits, 1, 1, tmp_codes, 2, 2,

  336.                  INIT_VLC_USE_NEW_STATIC);

  337.         offset += huff_vlc_tables_sizes[i];

  338.     }

  339.     av_assert0(offset == FF_ARRAY_ELEMS(huff_vlc_tables));

  340. 

  341.     offset = 0;

  342.     for (i = 0; i < 2; i++) {

  343.         huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;

  344.         huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];

  345.         init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,

  346.                  mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,

  347.                  INIT_VLC_USE_NEW_STATIC);

  348.         offset += huff_quad_vlc_tables_sizes[i];

  349.     }

  350.     av_assert0(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));

  351. 

  352.     for (i = 0; i < 9; i++) {

  353.         k = 0;

  354.         for (j = 0; j < 22; j++) {

  355.             band_index_long[i][j] = k;

  356.             k += band_size_long[i][j];

  357.         }

  358.         band_index_long[i][22] = k;

  359.     }

  360. 

  361.     /* compute n ^ (4/3) and store it in mantissa/exp format */

  362. 

  363.     mpegaudio_tableinit();

  364. 

  365.     for (i = 0; i < 4; i++) {

  366.         if (ff_mpa_quant_bits[i] < 0) {

  367.             for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {

  368.                 int val1, val2, val3, steps;

  369.                 int val = j;

  370.                 steps   = ff_mpa_quant_steps[i];

  371.                 val1    = val % steps;

  372.                 val    /= steps;

  373.                 val2    = val % steps;

  374.                 val3    = val / steps;

  375.                 division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);

  376.             }

  377.         }

  378.     }

  379. 

  380. 

  381.     for (i = 0; i < 7; i++) {

  382.         float f;

  383.         INTFLOAT v;

  384.         if (i != 6) {

  385.             f = tan((double)i * M_PI / 12.0);

  386.             v = FIXR(f / (1.0 + f));

  387.         } else {

  388.             v = FIXR(1.0);

  389.         }

  390.         is_table[0][    i] = v;

  391.         is_table[1][6 - i] = v;

  392.     }

  393.     /* invalid values */

  394.     for (i = 7; i < 16; i++)

  395.         is_table[0][i] = is_table[1][i] = 0.0;

  396. 

  397.     for (i = 0; i < 16; i++) {

  398.         double f;

  399.         int e, k;

  400. 

  401.         for (j = 0; j < 2; j++) {

  402.             e = -(j + 1) * ((i + 1) >> 1);

  403.             f = exp2(e / 4.0);

  404.             k = i & 1;

  405.             is_table_lsf[j][k ^ 1][i] = FIXR(f);

  406.             is_table_lsf[j][k    ][i] = FIXR(1.0);

  407.             av_dlog(NULL, "is_table_lsf %d %d: %f %f\n",

  408.                     i, j, (float) is_table_lsf[j][0][i],

  409.                     (float) is_table_lsf[j][1][i]);

  410.         }

  411.     }

  412. 

  413.     for (i = 0; i < 8; i++) {

  414.         float ci, cs, ca;

  415.         ci = ci_table[i];

  416.         cs = 1.0 / sqrt(1.0 + ci * ci);

  417.         ca = cs * ci;

  418. #if !CONFIG_FLOAT

  419.         csa_table[i][0] = FIXHR(cs/4);

  420.         csa_table[i][1] = FIXHR(ca/4);

  421.         csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);

  422.         csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);

  423. #else

  424.         csa_table[i][0] = cs;

  425.         csa_table[i][1] = ca;

  426.         csa_table[i][2] = ca + cs;

  427.         csa_table[i][3] = ca - cs;

  428. #endif

  429.     }

  430. }

  431. 

  432. static av_cold int decode_init(AVCodecContext * avctx)

  433. {

  434.     static int initialized_tables = 0;

  435.     MPADecodeContext *s = avctx->priv_data;

  436. 

  437.     if (!initialized_tables) {

  438.         decode_init_static();

  439.         initialized_tables = 1;

  440.     }

  441. 

  442.     s->avctx = avctx;

  443. 

  444.     ff_mpadsp_init(&s->mpadsp);

  445.     ff_dsputil_init(&s->dsp, avctx);

  446. 

  447.     if (avctx->request_sample_fmt == OUT_FMT &&

  448.         avctx->codec_id != AV_CODEC_ID_MP3ON4)

  449.         avctx->sample_fmt = OUT_FMT;

  450.     else

  451.         avctx->sample_fmt = OUT_FMT_P;

  452.     s->err_recognition = avctx->err_recognition;

  453. 

  454.     if (avctx->codec_id == AV_CODEC_ID_MP3ADU)

  455.         s->adu_mode = 1;

  456. 

  457.     avcodec_get_frame_defaults(&s->frame);

  458.     avctx->coded_frame = &s->frame;

  459. 

  460.     return 0;

  461. }

  462. 

  463. #define C3 FIXHR(0.86602540378443864676/2)

  464. #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)

  465. #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)

  466. #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)

  467. 

  468. /* 12 points IMDCT. We compute it "by hand" by factorizing obvious

  469.    cases. */

  470. static void imdct12(INTFLOAT *out, INTFLOAT *in)

  471. {

  472.     INTFLOAT in0, in1, in2, in3, in4, in5, t1, t2;

  473. 

  474.     in0  = in[0*3];

  475.     in1  = in[1*3] + in[0*3];

  476.     in2  = in[2*3] + in[1*3];

  477.     in3  = in[3*3] + in[2*3];

  478.     in4  = in[4*3] + in[3*3];

  479.     in5  = in[5*3] + in[4*3];

  480.     in5 += in3;

  481.     in3 += in1;

  482. 

  483.     in2  = MULH3(in2, C3, 2);

  484.     in3  = MULH3(in3, C3, 4);

  485. 

  486.     t1   = in0 - in4;

  487.     t2   = MULH3(in1 - in5, C4, 2);

  488. 

  489.     out[ 7] =

  490.     out[10] = t1 + t2;

  491.     out[ 1] =

  492.     out[ 4] = t1 - t2;

  493. 

  494.     in0    += SHR(in4, 1);

  495.     in4     = in0 + in2;

  496.     in5    += 2*in1;

  497.     in1     = MULH3(in5 + in3, C5, 1);

  498.     out[ 8] =

  499.     out[ 9] = in4 + in1;

  500.     out[ 2] =

  501.     out[ 3] = in4 - in1;

  502. 

  503.     in0    -= in2;

  504.     in5     = MULH3(in5 - in3, C6, 2);

  505.     out[ 0] =

  506.     out[ 5] = in0 - in5;

  507.     out[ 6] =

  508.     out[11] = in0 + in5;

  509. }

  510. 

  511. /* return the number of decoded frames */

  512. static int mp_decode_layer1(MPADecodeContext *s)

  513. {

  514.     int bound, i, v, n, ch, j, mant;

  515.     uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];

  516.     uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];

  517. 

  518.     if (s->mode == MPA_JSTEREO)

  519.         bound = (s->mode_ext + 1) * 4;

  520.     else

  521.         bound = SBLIMIT;

  522. 

  523.     /* allocation bits */

  524.     for (i = 0; i < bound; i++) {

  525.         for (ch = 0; ch < s->nb_channels; ch++) {

  526.             allocation[ch][i] = get_bits(&s->gb, 4);

  527.         }

  528.     }

  529.     for (i = bound; i < SBLIMIT; i++)

  530.         allocation[0][i] = get_bits(&s->gb, 4);

  531. 

  532.     /* scale factors */

  533.     for (i = 0; i < bound; i++) {

  534.         for (ch = 0; ch < s->nb_channels; ch++) {

  535.             if (allocation[ch][i])

  536.                 scale_factors[ch][i] = get_bits(&s->gb, 6);

  537.         }

  538.     }

  539.     for (i = bound; i < SBLIMIT; i++) {

  540.         if (allocation[0][i]) {

  541.             scale_factors[0][i] = get_bits(&s->gb, 6);

  542.             scale_factors[1][i] = get_bits(&s->gb, 6);

  543.         }

  544.     }

  545. 

  546.     /* compute samples */

  547.     for (j = 0; j < 12; j++) {

  548.         for (i = 0; i < bound; i++) {

  549.             for (ch = 0; ch < s->nb_channels; ch++) {

  550.                 n = allocation[ch][i];

  551.                 if (n) {

  552.                     mant = get_bits(&s->gb, n + 1);

  553.                     v = l1_unscale(n, mant, scale_factors[ch][i]);

  554.                 } else {

  555.                     v = 0;

  556.                 }

  557.                 s->sb_samples[ch][j][i] = v;

  558.             }

  559.         }

  560.         for (i = bound; i < SBLIMIT; i++) {

  561.             n = allocation[0][i];

  562.             if (n) {

  563.                 mant = get_bits(&s->gb, n + 1);

  564.                 v = l1_unscale(n, mant, scale_factors[0][i]);

  565.                 s->sb_samples[0][j][i] = v;

  566.                 v = l1_unscale(n, mant, scale_factors[1][i]);

  567.                 s->sb_samples[1][j][i] = v;

  568.             } else {

  569.                 s->sb_samples[0][j][i] = 0;

  570.                 s->sb_samples[1][j][i] = 0;

  571.             }

  572.         }

  573.     }

  574.     return 12;

  575. }

  576. 

  577. static int mp_decode_layer2(MPADecodeContext *s)

  578. {

  579.     int sblimit; /* number of used subbands */

  580.     const unsigned char *alloc_table;

  581.     int table, bit_alloc_bits, i, j, ch, bound, v;

  582.     unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];

  583.     unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];

  584.     unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;

  585.     int scale, qindex, bits, steps, k, l, m, b;

  586. 

  587.     /* select decoding table */

  588.     table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,

  589.                                    s->sample_rate, s->lsf);

  590.     sblimit     = ff_mpa_sblimit_table[table];

  591.     alloc_table = ff_mpa_alloc_tables[table];

  592. 

  593.     if (s->mode == MPA_JSTEREO)

  594.         bound = (s->mode_ext + 1) * 4;

  595.     else

  596.         bound = sblimit;

  597. 

  598.     av_dlog(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);

  599. 

  600.     /* sanity check */

  601.     if (bound > sblimit)

  602.         bound = sblimit;

  603. 

  604.     /* parse bit allocation */

  605.     j = 0;

  606.     for (i = 0; i < bound; i++) {

  607.         bit_alloc_bits = alloc_table[j];

  608.         for (ch = 0; ch < s->nb_channels; ch++)

  609.             bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);

  610.         j += 1 << bit_alloc_bits;

  611.     }

  612.     for (i = bound; i < sblimit; i++) {

  613.         bit_alloc_bits = alloc_table[j];

  614.         v = get_bits(&s->gb, bit_alloc_bits);

  615.         bit_alloc[0][i] = v;

  616.         bit_alloc[1][i] = v;

  617.         j += 1 << bit_alloc_bits;

  618.     }

  619. 

  620.     /* scale codes */

  621.     for (i = 0; i < sblimit; i++) {

  622.         for (ch = 0; ch < s->nb_channels; ch++) {

  623.             if (bit_alloc[ch][i])

  624.                 scale_code[ch][i] = get_bits(&s->gb, 2);

  625.         }

  626.     }

  627. 

  628.     /* scale factors */

  629.     for (i = 0; i < sblimit; i++) {

  630.         for (ch = 0; ch < s->nb_channels; ch++) {

  631.             if (bit_alloc[ch][i]) {

  632.                 sf = scale_factors[ch][i];

  633.                 switch (scale_code[ch][i]) {

  634.                 default:

  635.                 case 0:

  636.                     sf[0] = get_bits(&s->gb, 6);

  637.                     sf[1] = get_bits(&s->gb, 6);

  638.                     sf[2] = get_bits(&s->gb, 6);

  639.                     break;

  640.                 case 2:

  641.                     sf[0] = get_bits(&s->gb, 6);

  642.                     sf[1] = sf[0];

  643.                     sf[2] = sf[0];

  644.                     break;

  645.                 case 1:

  646.                     sf[0] = get_bits(&s->gb, 6);

  647.                     sf[2] = get_bits(&s->gb, 6);

  648.                     sf[1] = sf[0];

  649.                     break;

  650.                 case 3:

  651.                     sf[0] = get_bits(&s->gb, 6);

  652.                     sf[2] = get_bits(&s->gb, 6);

  653.                     sf[1] = sf[2];

  654.                     break;

  655.                 }

  656.             }

  657.         }

  658.     }

  659. 

  660.     /* samples */

  661.     for (k = 0; k < 3; k++) {

  662.         for (l = 0; l < 12; l += 3) {

  663.             j = 0;

  664.             for (i = 0; i < bound; i++) {

  665.                 bit_alloc_bits = alloc_table[j];

  666.                 for (ch = 0; ch < s->nb_channels; ch++) {

  667.                     b = bit_alloc[ch][i];

  668.                     if (b) {

  669.                         scale = scale_factors[ch][i][k];

  670.                         qindex = alloc_table[j+b];

  671.                         bits = ff_mpa_quant_bits[qindex];

  672.                         if (bits < 0) {

  673.                             int v2;

  674.                             /* 3 values at the same time */

  675.                             v = get_bits(&s->gb, -bits);

  676.                             v2 = division_tabs[qindex][v];

  677.                             steps  = ff_mpa_quant_steps[qindex];

  678. 

  679.                             s->sb_samples[ch][k * 12 + l + 0][i] =

  680.                                 l2_unscale_group(steps,  v2       & 15, scale);

  681.                             s->sb_samples[ch][k * 12 + l + 1][i] =

  682.                                 l2_unscale_group(steps, (v2 >> 4) & 15, scale);

  683.                             s->sb_samples[ch][k * 12 + l + 2][i] =

  684.                                 l2_unscale_group(steps,  v2 >> 8      , scale);

  685.                         } else {

  686.                             for (m = 0; m < 3; m++) {

  687.                                 v = get_bits(&s->gb, bits);

  688.                                 v = l1_unscale(bits - 1, v, scale);

  689.                                 s->sb_samples[ch][k * 12 + l + m][i] = v;

  690.                             }

  691.                         }

  692.                     } else {

  693.                         s->sb_samples[ch][k * 12 + l + 0][i] = 0;

  694.                         s->sb_samples[ch][k * 12 + l + 1][i] = 0;

  695.                         s->sb_samples[ch][k * 12 + l + 2][i] = 0;

  696.                     }

  697.                 }

  698.                 /* next subband in alloc table */

  699.                 j += 1 << bit_alloc_bits;

  700.             }

  701.             /* XXX: find a way to avoid this duplication of code */

  702.             for (i = bound; i < sblimit; i++) {

  703.                 bit_alloc_bits = alloc_table[j];

  704.                 b = bit_alloc[0][i];

  705.                 if (b) {

  706.                     int mant, scale0, scale1;

  707.                     scale0 = scale_factors[0][i][k];

  708.                     scale1 = scale_factors[1][i][k];

  709.                     qindex = alloc_table[j+b];

  710.                     bits = ff_mpa_quant_bits[qindex];

  711.                     if (bits < 0) {

  712.                         /* 3 values at the same time */

  713.                         v = get_bits(&s->gb, -bits);

  714.                         steps = ff_mpa_quant_steps[qindex];

  715.                         mant = v % steps;

  716.                         v = v / steps;

  717.                         s->sb_samples[0][k * 12 + l + 0][i] =

  718.                             l2_unscale_group(steps, mant, scale0);

  719.                         s->sb_samples[1][k * 12 + l + 0][i] =

  720.                             l2_unscale_group(steps, mant, scale1);

  721.                         mant = v % steps;

  722.                         v = v / steps;

  723.                         s->sb_samples[0][k * 12 + l + 1][i] =

  724.                             l2_unscale_group(steps, mant, scale0);

  725.                         s->sb_samples[1][k * 12 + l + 1][i] =

  726.                             l2_unscale_group(steps, mant, scale1);

  727.                         s->sb_samples[0][k * 12 + l + 2][i] =

  728.                             l2_unscale_group(steps, v, scale0);

  729.                         s->sb_samples[1][k * 12 + l + 2][i] =

  730.                             l2_unscale_group(steps, v, scale1);

  731.                     } else {

  732.                         for (m = 0; m < 3; m++) {

  733.                             mant = get_bits(&s->gb, bits);

  734.                             s->sb_samples[0][k * 12 + l + m][i] =

  735.                                 l1_unscale(bits - 1, mant, scale0);

  736.                             s->sb_samples[1][k * 12 + l + m][i] =

  737.                                 l1_unscale(bits - 1, mant, scale1);

  738.                         }

  739.                     }

  740.                 } else {

  741.                     s->sb_samples[0][k * 12 + l + 0][i] = 0;

  742.                     s->sb_samples[0][k * 12 + l + 1][i] = 0;

  743.                     s->sb_samples[0][k * 12 + l + 2][i] = 0;

  744.                     s->sb_samples[1][k * 12 + l + 0][i] = 0;

  745.                     s->sb_samples[1][k * 12 + l + 1][i] = 0;

  746.                     s->sb_samples[1][k * 12 + l + 2][i] = 0;

  747.                 }

  748.                 /* next subband in alloc table */

  749.                 j += 1 << bit_alloc_bits;

  750.             }

  751.             /* fill remaining samples to zero */

  752.             for (i = sblimit; i < SBLIMIT; i++) {

  753.                 for (ch = 0; ch < s->nb_channels; ch++) {

  754.                     s->sb_samples[ch][k * 12 + l + 0][i] = 0;

  755.                     s->sb_samples[ch][k * 12 + l + 1][i] = 0;

  756.                     s->sb_samples[ch][k * 12 + l + 2][i] = 0;

  757.                 }

  758.             }

  759.         }

  760.     }

  761.     return 3 * 12;

  762. }

  763. 

  764. #define SPLIT(dst,sf,n)             \

  765.     if (n == 3) {                   \

  766.         int m = (sf * 171) >> 9;    \

  767.         dst   = sf - 3 * m;         \

  768.         sf    = m;                  \

  769.     } else if (n == 4) {            \

  770.         dst  = sf & 3;              \

  771.         sf >>= 2;                   \

  772.     } else if (n == 5) {            \

  773.         int m = (sf * 205) >> 10;   \

  774.         dst   = sf - 5 * m;         \

  775.         sf    = m;                  \

  776.     } else if (n == 6) {            \

  777.         int m = (sf * 171) >> 10;   \

  778.         dst   = sf - 6 * m;         \

  779.         sf    = m;                  \

  780.     } else {                        \

  781.         dst = 0;                    \

  782.     }

  783. 

  784. static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,

  785.                                            int n3)

  786. {

  787.     SPLIT(slen[3], sf, n3)

  788.     SPLIT(slen[2], sf, n2)

  789.     SPLIT(slen[1], sf, n1)

  790.     slen[0] = sf;

  791. }

  792. 

  793. static void exponents_from_scale_factors(MPADecodeContext *s, GranuleDef *g,

  794.                                          int16_t *exponents)

  795. {

  796.     const uint8_t *bstab, *pretab;

  797.     int len, i, j, k, l, v0, shift, gain, gains[3];

  798.     int16_t *exp_ptr;

  799. 

  800.     exp_ptr = exponents;

  801.     gain    = g->global_gain - 210;

  802.     shift   = g->scalefac_scale + 1;

  803. 

  804.     bstab  = band_size_long[s->sample_rate_index];

  805.     pretab = mpa_pretab[g->preflag];

  806.     for (i = 0; i < g->long_end; i++) {

  807.         v0 = gain - ((g->scale_factors[i] + pretab[i]) << shift) + 400;

  808.         len = bstab[i];

  809.         for (j = len; j > 0; j--)

  810.             *exp_ptr++ = v0;

  811.     }

  812. 

  813.     if (g->short_start < 13) {

  814.         bstab    = band_size_short[s->sample_rate_index];

  815.         gains[0] = gain - (g->subblock_gain[0] << 3);

  816.         gains[1] = gain - (g->subblock_gain[1] << 3);

  817.         gains[2] = gain - (g->subblock_gain[2] << 3);

  818.         k        = g->long_end;

  819.         for (i = g->short_start; i < 13; i++) {

  820.             len = bstab[i];

  821.             for (l = 0; l < 3; l++) {

  822.                 v0 = gains[l] - (g->scale_factors[k++] << shift) + 400;

  823.                 for (j = len; j > 0; j--)

  824.                     *exp_ptr++ = v0;

  825.             }

  826.         }

  827.     }

  828. }

  829. 

  830. /* handle n = 0 too */

  831. static inline int get_bitsz(GetBitContext *s, int n)

  832. {

  833.     return n ? get_bits(s, n) : 0;

  834. }

  835. 

  836. 

  837. static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,

  838.                           int *end_pos2)

  839. {

  840.     if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) {

  841.         s->gb           = s->in_gb;

  842.         s->in_gb.buffer = NULL;

  843.         av_assert2((get_bits_count(&s->gb) & 7) == 0);

  844.         skip_bits_long(&s->gb, *pos - *end_pos);

  845.         *end_pos2 =

  846.         *end_pos  = *end_pos2 + get_bits_count(&s->gb) - *pos;

  847.         *pos      = get_bits_count(&s->gb);

  848.     }

  849. }

  850. 

  851. /* Following is a optimized code for

  852.             INTFLOAT v = *src

  853.             if(get_bits1(&s->gb))

  854.                 v = -v;

  855.             *dst = v;

  856. */

  857. #if CONFIG_FLOAT

  858. #define READ_FLIP_SIGN(dst,src)                     \

  859.     v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31);  \

  860.     AV_WN32A(dst, v);

  861. #else

  862. #define READ_FLIP_SIGN(dst,src)     \

  863.     v      = -get_bits1(&s->gb);    \

  864.     *(dst) = (*(src) ^ v) - v;

  865. #endif

  866. 

  867. static int huffman_decode(MPADecodeContext *s, GranuleDef *g,

  868.                           int16_t *exponents, int end_pos2)

  869. {

  870.     int s_index;

  871.     int i;

  872.     int last_pos, bits_left;

  873.     VLC *vlc;

  874.     int end_pos = FFMIN(end_pos2, s->gb.size_in_bits);

  875. 

  876.     /* low frequencies (called big values) */

  877.     s_index = 0;

  878.     for (i = 0; i < 3; i++) {

  879.         int j, k, l, linbits;

  880.         j = g->region_size[i];

  881.         if (j == 0)

  882.             continue;

  883.         /* select vlc table */

  884.         k       = g->table_select[i];

  885.         l       = mpa_huff_data[k][0];

  886.         linbits = mpa_huff_data[k][1];

  887.         vlc     = &huff_vlc[l];

  888. 

  889.         if (!l) {

  890.             memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);

  891.             s_index += 2 * j;

  892.             continue;

  893.         }

  894. 

  895.         /* read huffcode and compute each couple */

  896.         for (; j > 0; j--) {

  897.             int exponent, x, y;

  898.             int v;

  899.             int pos = get_bits_count(&s->gb);

  900. 

  901.             if (pos >= end_pos){

  902.                 switch_buffer(s, &pos, &end_pos, &end_pos2);

  903.                 if (pos >= end_pos)

  904.                     break;

  905.             }

  906.             y = get_vlc2(&s->gb, vlc->table, 7, 3);

  907. 

  908.             if (!y) {

  909.                 g->sb_hybrid[s_index  ] =

  910.                 g->sb_hybrid[s_index+1] = 0;

  911.                 s_index += 2;

  912.                 continue;

  913.             }

  914. 

  915.             exponent= exponents[s_index];

  916. 

  917.             av_dlog(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",

  918.                     i, g->region_size[i] - j, x, y, exponent);

  919.             if (y & 16) {

  920.                 x = y >> 5;

  921.                 y = y & 0x0f;

  922.                 if (x < 15) {

  923.                     READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)

  924.                 } else {

  925.                     x += get_bitsz(&s->gb, linbits);

  926.                     v  = l3_unscale(x, exponent);

  927.                     if (get_bits1(&s->gb))

  928.                         v = -v;

  929.                     g->sb_hybrid[s_index] = v;

  930.                 }

  931.                 if (y < 15) {

  932.                     READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)

  933.                 } else {

  934.                     y += get_bitsz(&s->gb, linbits);

  935.                     v  = l3_unscale(y, exponent);

  936.                     if (get_bits1(&s->gb))

  937.                         v = -v;

  938.                     g->sb_hybrid[s_index+1] = v;

  939.                 }

  940.             } else {

  941.                 x = y >> 5;

  942.                 y = y & 0x0f;

  943.                 x += y;

  944.                 if (x < 15) {

  945.                     READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)

  946.                 } else {

  947.                     x += get_bitsz(&s->gb, linbits);

  948.                     v  = l3_unscale(x, exponent);

  949.                     if (get_bits1(&s->gb))

  950.                         v = -v;

  951.                     g->sb_hybrid[s_index+!!y] = v;

  952.                 }

  953.                 g->sb_hybrid[s_index + !y] = 0;

  954.             }

  955.             s_index += 2;

  956.         }

  957.     }

  958. 

  959.     /* high frequencies */

  960.     vlc = &huff_quad_vlc[g->count1table_select];

  961.     last_pos = 0;

  962.     while (s_index <= 572) {

  963.         int pos, code;

  964.         pos = get_bits_count(&s->gb);

  965.         if (pos >= end_pos) {

  966.             if (pos > end_pos2 && last_pos) {

  967.                 /* some encoders generate an incorrect size for this

  968.                    part. We must go back into the data */

  969.                 s_index -= 4;

  970.                 skip_bits_long(&s->gb, last_pos - pos);

  971.                 av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);

  972.                 if(s->err_recognition & (AV_EF_BITSTREAM|AV_EF_COMPLIANT))

  973.                     s_index=0;

  974.                 break;

  975.             }

  976.             switch_buffer(s, &pos, &end_pos, &end_pos2);

  977.             if (pos >= end_pos)

  978.                 break;

  979.         }

  980.         last_pos = pos;

  981. 

  982.         code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);

  983.         av_dlog(s->avctx, "t=%d code=%d\n", g->count1table_select, code);

  984.         g->sb_hybrid[s_index+0] =

  985.         g->sb_hybrid[s_index+1] =

  986.         g->sb_hybrid[s_index+2] =

  987.         g->sb_hybrid[s_index+3] = 0;

  988.         while (code) {

  989.             static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };

  990.             int v;

  991.             int pos = s_index + idxtab[code];

  992.             code   ^= 8 >> idxtab[code];

  993.             READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])

  994.         }

  995.         s_index += 4;

  996.     }

  997.     /* skip extension bits */

  998.     bits_left = end_pos2 - get_bits_count(&s->gb);

  999.     if (bits_left < 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_COMPLIANT))) {

  1000.         av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);

  1001.         s_index=0;

  1002.     } else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_AGGRESSIVE))) {

  1003.         av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);

  1004.         s_index = 0;

  1005.     }

  1006.     memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));

  1007.     skip_bits_long(&s->gb, bits_left);

  1008. 

  1009.     i = get_bits_count(&s->gb);

  1010.     switch_buffer(s, &i, &end_pos, &end_pos2);

  1011. 

  1012.     return 0;

  1013. }

  1014. 

  1015. /* Reorder short blocks from bitstream order to interleaved order. It

  1016.    would be faster to do it in parsing, but the code would be far more

  1017.    complicated */

  1018. static void reorder_block(MPADecodeContext *s, GranuleDef *g)

  1019. {

  1020.     int i, j, len;

  1021.     INTFLOAT *ptr, *dst, *ptr1;

  1022.     INTFLOAT tmp[576];

  1023. 

  1024.     if (g->block_type != 2)

  1025.         return;

  1026. 

  1027.     if (g->switch_point) {

  1028.         if (s->sample_rate_index != 8)

  1029.             ptr = g->sb_hybrid + 36;

  1030.         else

  1031.             ptr = g->sb_hybrid + 72;

  1032.     } else {

  1033.         ptr = g->sb_hybrid;

  1034.     }

  1035. 

  1036.     for (i = g->short_start; i < 13; i++) {

  1037.         len  = band_size_short[s->sample_rate_index][i];

  1038.         ptr1 = ptr;

  1039.         dst  = tmp;

  1040.         for (j = len; j > 0; j--) {

  1041.             *dst++ = ptr[0*len];

  1042.             *dst++ = ptr[1*len];

  1043.             *dst++ = ptr[2*len];

  1044.             ptr++;

  1045.         }

  1046.         ptr += 2 * len;

  1047.         memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));

  1048.     }

  1049. }

  1050. 

  1051. #define ISQRT2 FIXR(0.70710678118654752440)

  1052. 

  1053. static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)

  1054. {

  1055.     int i, j, k, l;

  1056.     int sf_max, sf, len, non_zero_found;

  1057.     INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;

  1058.     int non_zero_found_short[3];

  1059. 

  1060.     /* intensity stereo */

  1061.     if (s->mode_ext & MODE_EXT_I_STEREO) {

  1062.         if (!s->lsf) {

  1063.             is_tab = is_table;

  1064.             sf_max = 7;

  1065.         } else {

  1066.             is_tab = is_table_lsf[g1->scalefac_compress & 1];

  1067.             sf_max = 16;

  1068.         }

  1069. 

  1070.         tab0 = g0->sb_hybrid + 576;

  1071.         tab1 = g1->sb_hybrid + 576;

  1072. 

  1073.         non_zero_found_short[0] = 0;

  1074.         non_zero_found_short[1] = 0;

  1075.         non_zero_found_short[2] = 0;

  1076.         k = (13 - g1->short_start) * 3 + g1->long_end - 3;

  1077.         for (i = 12; i >= g1->short_start; i--) {

  1078.             /* for last band, use previous scale factor */

  1079.             if (i != 11)

  1080.                 k -= 3;

  1081.             len = band_size_short[s->sample_rate_index][i];

  1082.             for (l = 2; l >= 0; l--) {

  1083.                 tab0 -= len;

  1084.                 tab1 -= len;

  1085.                 if (!non_zero_found_short[l]) {

  1086.                     /* test if non zero band. if so, stop doing i-stereo */

  1087.                     for (j = 0; j < len; j++) {

  1088.                         if (tab1[j] != 0) {

  1089.                             non_zero_found_short[l] = 1;

  1090.                             goto found1;

  1091.                         }

  1092.                     }

  1093.                     sf = g1->scale_factors[k + l];

  1094.                     if (sf >= sf_max)

  1095.                         goto found1;

  1096. 

  1097.                     v1 = is_tab[0][sf];

  1098.                     v2 = is_tab[1][sf];

  1099.                     for (j = 0; j < len; j++) {

  1100.                         tmp0    = tab0[j];

  1101.                         tab0[j] = MULLx(tmp0, v1, FRAC_BITS);

  1102.                         tab1[j] = MULLx(tmp0, v2, FRAC_BITS);

  1103.                     }

  1104.                 } else {

  1105. found1:

  1106.                     if (s->mode_ext & MODE_EXT_MS_STEREO) {

  1107.                         /* lower part of the spectrum : do ms stereo

  1108.                            if enabled */

  1109.                         for (j = 0; j < len; j++) {

  1110.                             tmp0    = tab0[j];

  1111.                             tmp1    = tab1[j];

  1112.                             tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);

  1113.                             tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);

  1114.                         }

  1115.                     }

  1116.                 }

  1117.             }

  1118.         }

  1119. 

  1120.         non_zero_found = non_zero_found_short[0] |

  1121.                          non_zero_found_short[1] |

  1122.                          non_zero_found_short[2];

  1123. 

  1124.         for (i = g1->long_end - 1;i >= 0;i--) {

  1125.             len   = band_size_long[s->sample_rate_index][i];

  1126.             tab0 -= len;

  1127.             tab1 -= len;

  1128.             /* test if non zero band. if so, stop doing i-stereo */

  1129.             if (!non_zero_found) {

  1130.                 for (j = 0; j < len; j++) {

  1131.                     if (tab1[j] != 0) {

  1132.                         non_zero_found = 1;

  1133.                         goto found2;

  1134.                     }

  1135.                 }

  1136.                 /* for last band, use previous scale factor */

  1137.                 k  = (i == 21) ? 20 : i;

  1138.                 sf = g1->scale_factors[k];

  1139.                 if (sf >= sf_max)

  1140.                     goto found2;

  1141.                 v1 = is_tab[0][sf];

  1142.                 v2 = is_tab[1][sf];

  1143.                 for (j = 0; j < len; j++) {

  1144.                     tmp0    = tab0[j];

  1145.                     tab0[j] = MULLx(tmp0, v1, FRAC_BITS);

  1146.                     tab1[j] = MULLx(tmp0, v2, FRAC_BITS);

  1147.                 }

  1148.             } else {

  1149. found2:

  1150.                 if (s->mode_ext & MODE_EXT_MS_STEREO) {

  1151.                     /* lower part of the spectrum : do ms stereo

  1152.                        if enabled */

  1153.                     for (j = 0; j < len; j++) {

  1154.                         tmp0    = tab0[j];

  1155.                         tmp1    = tab1[j];

  1156.                         tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);

  1157.                         tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);

  1158.                     }

  1159.                 }

  1160.             }

  1161.         }

  1162.     } else if (s->mode_ext & MODE_EXT_MS_STEREO) {

  1163.         /* ms stereo ONLY */

  1164.         /* NOTE: the 1/sqrt(2) normalization factor is included in the

  1165.            global gain */

  1166. #if CONFIG_FLOAT

  1167.        s-> dsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);

  1168. #else

  1169.         tab0 = g0->sb_hybrid;

  1170.         tab1 = g1->sb_hybrid;

  1171.         for (i = 0; i < 576; i++) {

  1172.             tmp0    = tab0[i];

  1173.             tmp1    = tab1[i];

  1174.             tab0[i] = tmp0 + tmp1;

  1175.             tab1[i] = tmp0 - tmp1;

  1176.         }

  1177. #endif

  1178.     }

  1179. }

  1180. 

  1181. #if CONFIG_FLOAT

  1182. #if HAVE_MIPSFPU

  1183. #   include "mips/compute_antialias_float.h"

  1184. #endif /* HAVE_MIPSFPU */

  1185. #else

  1186. #if HAVE_MIPSDSPR1

  1187. #   include "mips/compute_antialias_fixed.h"

  1188. #endif /* HAVE_MIPSDSPR1 */

  1189. #endif /* CONFIG_FLOAT */

  1190. 

  1191. #ifndef compute_antialias

  1192. #if CONFIG_FLOAT

  1193. #define AA(j) do {                                                      \

  1194.         float tmp0 = ptr[-1-j];                                         \

  1195.         float tmp1 = ptr[   j];                                         \

  1196.         ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1];    \

  1197.         ptr[   j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0];    \

  1198.     } while (0)

  1199. #else

  1200. #define AA(j) do {                                              \

  1201.         int tmp0 = ptr[-1-j];                                   \

  1202.         int tmp1 = ptr[   j];                                   \

  1203.         int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]);          \

  1204.         ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2]));   \

  1205.         ptr[   j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3]));   \

  1206.     } while (0)

  1207. #endif

  1208. 

  1209. static void compute_antialias(MPADecodeContext *s, GranuleDef *g)

  1210. {

  1211.     INTFLOAT *ptr;

  1212.     int n, i;

  1213. 

  1214.     /* we antialias only "long" bands */

  1215.     if (g->block_type == 2) {

  1216.         if (!g->switch_point)

  1217.             return;

  1218.         /* XXX: check this for 8000Hz case */

  1219.         n = 1;

  1220.     } else {

  1221.         n = SBLIMIT - 1;

  1222.     }

  1223. 

  1224.     ptr = g->sb_hybrid + 18;

  1225.     for (i = n; i > 0; i--) {

  1226.         AA(0);

  1227.         AA(1);

  1228.         AA(2);

  1229.         AA(3);

  1230.         AA(4);

  1231.         AA(5);

  1232.         AA(6);

  1233.         AA(7);

  1234. 

  1235.         ptr += 18;

  1236.     }

  1237. }

  1238. #endif /* compute_antialias */

  1239. 

  1240. static void compute_imdct(MPADecodeContext *s, GranuleDef *g,

  1241.                           INTFLOAT *sb_samples, INTFLOAT *mdct_buf)

  1242. {

  1243.     INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;

  1244.     INTFLOAT out2[12];

  1245.     int i, j, mdct_long_end, sblimit;

  1246. 

  1247.     /* find last non zero block */

  1248.     ptr  = g->sb_hybrid + 576;

  1249.     ptr1 = g->sb_hybrid + 2 * 18;

  1250.     while (ptr >= ptr1) {

  1251.         int32_t *p;

  1252.         ptr -= 6;

  1253.         p    = (int32_t*)ptr;

  1254.         if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])

  1255.             break;

  1256.     }

  1257.     sblimit = ((ptr - g->sb_hybrid) / 18) + 1;

  1258. 

  1259.     if (g->block_type == 2) {

  1260.         /* XXX: check for 8000 Hz */

  1261.         if (g->switch_point)

  1262.             mdct_long_end = 2;

  1263.         else

  1264.             mdct_long_end = 0;

  1265.     } else {

  1266.         mdct_long_end = sblimit;

  1267.     }

  1268. 

  1269.     s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,

  1270.                                      mdct_long_end, g->switch_point,

  1271.                                      g->block_type);

  1272. 

  1273.     buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);

  1274.     ptr = g->sb_hybrid + 18 * mdct_long_end;

  1275. 

  1276.     for (j = mdct_long_end; j < sblimit; j++) {

  1277.         /* select frequency inversion */

  1278.         win     = RENAME(ff_mdct_win)[2 + (4  & -(j & 1))];

  1279.         out_ptr = sb_samples + j;

  1280. 

  1281.         for (i = 0; i < 6; i++) {

  1282.             *out_ptr = buf[4*i];

  1283.             out_ptr += SBLIMIT;

  1284.         }

  1285.         imdct12(out2, ptr + 0);

  1286.         for (i = 0; i < 6; i++) {

  1287.             *out_ptr     = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*1)];

  1288.             buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);

  1289.             out_ptr += SBLIMIT;

  1290.         }

  1291.         imdct12(out2, ptr + 1);

  1292.         for (i = 0; i < 6; i++) {

  1293.             *out_ptr     = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*2)];

  1294.             buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);

  1295.             out_ptr += SBLIMIT;

  1296.         }

  1297.         imdct12(out2, ptr + 2);

  1298.         for (i = 0; i < 6; i++) {

  1299.             buf[4*(i + 6*0)] = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*0)];

  1300.             buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);

  1301.             buf[4*(i + 6*2)] = 0;

  1302.         }

  1303.         ptr += 18;

  1304.         buf += (j&3) != 3 ? 1 : (4*18-3);

  1305.     }

  1306.     /* zero bands */

  1307.     for (j = sblimit; j < SBLIMIT; j++) {

  1308.         /* overlap */

  1309.         out_ptr = sb_samples + j;

  1310.         for (i = 0; i < 18; i++) {

  1311.             *out_ptr = buf[4*i];

  1312.             buf[4*i]   = 0;

  1313.             out_ptr += SBLIMIT;

  1314.         }

  1315.         buf += (j&3) != 3 ? 1 : (4*18-3);

  1316.     }

  1317. }

  1318. 

  1319. /* main layer3 decoding function */

  1320. static int mp_decode_layer3(MPADecodeContext *s)

  1321. {

  1322.     int nb_granules, main_data_begin;

  1323.     int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;

  1324.     GranuleDef *g;

  1325.     int16_t exponents[576]; //FIXME try INTFLOAT

  1326. 

  1327.     /* read side info */

  1328.     if (s->lsf) {

  1329.         main_data_begin = get_bits(&s->gb, 8);

  1330.         skip_bits(&s->gb, s->nb_channels);

  1331.         nb_granules = 1;

  1332.     } else {

  1333.         main_data_begin = get_bits(&s->gb, 9);

  1334.         if (s->nb_channels == 2)

  1335.             skip_bits(&s->gb, 3);

  1336.         else

  1337.             skip_bits(&s->gb, 5);

  1338.         nb_granules = 2;

  1339.         for (ch = 0; ch < s->nb_channels; ch++) {

  1340.             s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */

  1341.             s->granules[ch][1].scfsi = get_bits(&s->gb, 4);

  1342.         }

  1343.     }

  1344. 

  1345.     for (gr = 0; gr < nb_granules; gr++) {

  1346.         for (ch = 0; ch < s->nb_channels; ch++) {

  1347.             av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);

  1348.             g = &s->granules[ch][gr];

  1349.             g->part2_3_length = get_bits(&s->gb, 12);

  1350.             g->big_values     = get_bits(&s->gb,  9);

  1351.             if (g->big_values > 288) {

  1352.                 av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");

  1353.                 return AVERROR_INVALIDDATA;

  1354.             }

  1355. 

  1356.             g->global_gain = get_bits(&s->gb, 8);

  1357.             /* if MS stereo only is selected, we precompute the

  1358.                1/sqrt(2) renormalization factor */

  1359.             if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==

  1360.                 MODE_EXT_MS_STEREO)

  1361.                 g->global_gain -= 2;

  1362.             if (s->lsf)

  1363.                 g->scalefac_compress = get_bits(&s->gb, 9);

  1364.             else

  1365.                 g->scalefac_compress = get_bits(&s->gb, 4);

  1366.             blocksplit_flag = get_bits1(&s->gb);

  1367.             if (blocksplit_flag) {

  1368.                 g->block_type = get_bits(&s->gb, 2);

  1369.                 if (g->block_type == 0) {

  1370.                     av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");

  1371.                     return AVERROR_INVALIDDATA;

  1372.                 }

  1373.                 g->switch_point = get_bits1(&s->gb);

  1374.                 for (i = 0; i < 2; i++)

  1375.                     g->table_select[i] = get_bits(&s->gb, 5);

  1376.                 for (i = 0; i < 3; i++)

  1377.                     g->subblock_gain[i] = get_bits(&s->gb, 3);

  1378.                 ff_init_short_region(s, g);

  1379.             } else {

  1380.                 int region_address1, region_address2;

  1381.                 g->block_type = 0;

  1382.                 g->switch_point = 0;

  1383.                 for (i = 0; i < 3; i++)

  1384.                     g->table_select[i] = get_bits(&s->gb, 5);

  1385.                 /* compute huffman coded region sizes */

  1386.                 region_address1 = get_bits(&s->gb, 4);

  1387.                 region_address2 = get_bits(&s->gb, 3);

  1388.                 av_dlog(s->avctx, "region1=%d region2=%d\n",

  1389.                         region_address1, region_address2);

  1390.                 ff_init_long_region(s, g, region_address1, region_address2);

  1391.             }

  1392.             ff_region_offset2size(g);

  1393.             ff_compute_band_indexes(s, g);

  1394. 

  1395.             g->preflag = 0;

  1396.             if (!s->lsf)

  1397.                 g->preflag = get_bits1(&s->gb);

  1398.             g->scalefac_scale     = get_bits1(&s->gb);

  1399.             g->count1table_select = get_bits1(&s->gb);

  1400.             av_dlog(s->avctx, "block_type=%d switch_point=%d\n",

  1401.                     g->block_type, g->switch_point);

  1402.         }

  1403.     }

  1404. 

  1405.     if (!s->adu_mode) {

  1406.         int skip;

  1407.         const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);

  1408.         int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0, EXTRABYTES);

  1409.         av_assert1((get_bits_count(&s->gb) & 7) == 0);

  1410.         /* now we get bits from the main_data_begin offset */

  1411.         av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",

  1412.                 main_data_begin, s->last_buf_size);

  1413. 

  1414.         memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);

  1415.         s->in_gb = s->gb;

  1416.         init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);

  1417. #if !UNCHECKED_BITSTREAM_READER

  1418.         s->gb.size_in_bits_plus8 += FFMAX(extrasize, LAST_BUF_SIZE - s->last_buf_size) * 8;

  1419. #endif

  1420.         s->last_buf_size <<= 3;

  1421.         for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {

  1422.             for (ch = 0; ch < s->nb_channels; ch++) {

  1423.                 g = &s->granules[ch][gr];

  1424.                 s->last_buf_size += g->part2_3_length;

  1425.                 memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));

  1426.                 compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);

  1427.             }

  1428.         }

  1429.         skip = s->last_buf_size - 8 * main_data_begin;

  1430.         if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {

  1431.             skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);

  1432.             s->gb           = s->in_gb;

  1433.             s->in_gb.buffer = NULL;

  1434.         } else {

  1435.             skip_bits_long(&s->gb, skip);

  1436.         }

  1437.     } else {

  1438.         gr = 0;

  1439.     }

  1440. 

  1441.     for (; gr < nb_granules; gr++) {

  1442.         for (ch = 0; ch < s->nb_channels; ch++) {

  1443.             g = &s->granules[ch][gr];

  1444.             bits_pos = get_bits_count(&s->gb);

  1445. 

  1446.             if (!s->lsf) {

  1447.                 uint8_t *sc;

  1448.                 int slen, slen1, slen2;

  1449. 

  1450.                 /* MPEG1 scale factors */

  1451.                 slen1 = slen_table[0][g->scalefac_compress];

  1452.                 slen2 = slen_table[1][g->scalefac_compress];

  1453.                 av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);

  1454.                 if (g->block_type == 2) {

  1455.                     n = g->switch_point ? 17 : 18;

  1456.                     j = 0;

  1457.                     if (slen1) {

  1458.                         for (i = 0; i < n; i++)

  1459.                             g->scale_factors[j++] = get_bits(&s->gb, slen1);

  1460.                     } else {

  1461.                         for (i = 0; i < n; i++)

  1462.                             g->scale_factors[j++] = 0;

  1463.                     }

  1464.                     if (slen2) {

  1465.                         for (i = 0; i < 18; i++)

  1466.                             g->scale_factors[j++] = get_bits(&s->gb, slen2);

  1467.                         for (i = 0; i < 3; i++)

  1468.                             g->scale_factors[j++] = 0;

  1469.                     } else {

  1470.                         for (i = 0; i < 21; i++)

  1471.                             g->scale_factors[j++] = 0;

  1472.                     }

  1473.                 } else {

  1474.                     sc = s->granules[ch][0].scale_factors;

  1475.                     j = 0;

  1476.                     for (k = 0; k < 4; k++) {

  1477.                         n = k == 0 ? 6 : 5;

  1478.                         if ((g->scfsi & (0x8 >> k)) == 0) {

  1479.                             slen = (k < 2) ? slen1 : slen2;

  1480.                             if (slen) {

  1481.                                 for (i = 0; i < n; i++)

  1482.                                     g->scale_factors[j++] = get_bits(&s->gb, slen);

  1483.                             } else {

  1484.                                 for (i = 0; i < n; i++)

  1485.                                     g->scale_factors[j++] = 0;

  1486.                             }

  1487.                         } else {

  1488.                             /* simply copy from last granule */

  1489.                             for (i = 0; i < n; i++) {

  1490.                                 g->scale_factors[j] = sc[j];

  1491.                                 j++;

  1492.                             }

  1493.                         }

  1494.                     }

  1495.                     g->scale_factors[j++] = 0;

  1496.                 }

  1497.             } else {

  1498.                 int tindex, tindex2, slen[4], sl, sf;

  1499. 

  1500.                 /* LSF scale factors */

  1501.                 if (g->block_type == 2)

  1502.                     tindex = g->switch_point ? 2 : 1;

  1503.                 else

  1504.                     tindex = 0;

  1505. 

  1506.                 sf = g->scalefac_compress;

  1507.                 if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {

  1508.                     /* intensity stereo case */

  1509.                     sf >>= 1;

  1510.                     if (sf < 180) {

  1511.                         lsf_sf_expand(slen, sf, 6, 6, 0);

  1512.                         tindex2 = 3;

  1513.                     } else if (sf < 244) {

  1514.                         lsf_sf_expand(slen, sf - 180, 4, 4, 0);

  1515.                         tindex2 = 4;

  1516.                     } else {

  1517.                         lsf_sf_expand(slen, sf - 244, 3, 0, 0);

  1518.                         tindex2 = 5;

  1519.                     }

  1520.                 } else {

  1521.                     /* normal case */

  1522.                     if (sf < 400) {

  1523.                         lsf_sf_expand(slen, sf, 5, 4, 4);

  1524.                         tindex2 = 0;

  1525.                     } else if (sf < 500) {

  1526.                         lsf_sf_expand(slen, sf - 400, 5, 4, 0);

  1527.                         tindex2 = 1;

  1528.                     } else {

  1529.                         lsf_sf_expand(slen, sf - 500, 3, 0, 0);

  1530.                         tindex2 = 2;

  1531.                         g->preflag = 1;

  1532.                     }

  1533.                 }

  1534. 

  1535.                 j = 0;

  1536.                 for (k = 0; k < 4; k++) {

  1537.                     n  = lsf_nsf_table[tindex2][tindex][k];

  1538.                     sl = slen[k];

  1539.                     if (sl) {

  1540.                         for (i = 0; i < n; i++)

  1541.                             g->scale_factors[j++] = get_bits(&s->gb, sl);

  1542.                     } else {

  1543.                         for (i = 0; i < n; i++)

  1544.                             g->scale_factors[j++] = 0;

  1545.                     }

  1546.                 }

  1547.                 /* XXX: should compute exact size */

  1548.                 for (; j < 40; j++)

  1549.                     g->scale_factors[j] = 0;

  1550.             }

  1551. 

  1552.             exponents_from_scale_factors(s, g, exponents);

  1553. 

  1554.             /* read Huffman coded residue */

  1555.             huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);

  1556.         } /* ch */

  1557. 

  1558.         if (s->mode == MPA_JSTEREO)

  1559.             compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);

  1560. 

  1561.         for (ch = 0; ch < s->nb_channels; ch++) {

  1562.             g = &s->granules[ch][gr];

  1563. 

  1564.             reorder_block(s, g);

  1565.             compute_antialias(s, g);

  1566.             compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);

  1567.         }

  1568.     } /* gr */

  1569.     if (get_bits_count(&s->gb) < 0)

  1570.         skip_bits_long(&s->gb, -get_bits_count(&s->gb));

  1571.     return nb_granules * 18;

  1572. }

  1573. 

  1574. static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,

  1575.                            const uint8_t *buf, int buf_size)

  1576. {

  1577.     int i, nb_frames, ch, ret;

  1578.     OUT_INT *samples_ptr;

  1579. 

  1580.     init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);

  1581. 

  1582.     /* skip error protection field */

  1583.     if (s->error_protection)

  1584.         skip_bits(&s->gb, 16);

  1585. 

  1586.     switch(s->layer) {

  1587.     case 1:

  1588.         s->avctx->frame_size = 384;

  1589.         nb_frames = mp_decode_layer1(s);

  1590.         break;

  1591.     case 2:

  1592.         s->avctx->frame_size = 1152;

  1593.         nb_frames = mp_decode_layer2(s);

  1594.         break;

  1595.     case 3:

  1596.         s->avctx->frame_size = s->lsf ? 576 : 1152;

  1597.     default:

  1598.         nb_frames = mp_decode_layer3(s);

  1599. 

  1600.         s->last_buf_size=0;

  1601.         if (s->in_gb.buffer) {

  1602.             align_get_bits(&s->gb);

  1603.             i = get_bits_left(&s->gb)>>3;

  1604.             if (i >= 0 && i <= BACKSTEP_SIZE) {

  1605.                 memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);

  1606.                 s->last_buf_size=i;

  1607.             } else

  1608.                 av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);

  1609.             s->gb           = s->in_gb;

  1610.             s->in_gb.buffer = NULL;

  1611.         }

  1612. 

  1613.         align_get_bits(&s->gb);

  1614.         av_assert1((get_bits_count(&s->gb) & 7) == 0);

  1615.         i = get_bits_left(&s->gb) >> 3;

  1616. 

  1617.         if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {

  1618.             if (i < 0)

  1619.                 av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);

  1620.             i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);

  1621.         }

  1622.         av_assert1(i <= buf_size - HEADER_SIZE && i >= 0);

  1623.         memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);

  1624.         s->last_buf_size += i;

  1625.     }

  1626. 

  1627.     if(nb_frames < 0)

  1628.         return nb_frames;

  1629. 

  1630.     /* get output buffer */

  1631.     if (!samples) {

  1632.         s->frame.nb_samples = s->avctx->frame_size;

  1633.         if ((ret = ff_get_buffer(s->avctx, &s->frame)) < 0) {

  1634.             av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1635.             return ret;

  1636.         }

  1637.         samples = (OUT_INT **)s->frame.extended_data;

  1638.     }

  1639. 

  1640.     /* apply the synthesis filter */

  1641.     for (ch = 0; ch < s->nb_channels; ch++) {

  1642.         int sample_stride;

  1643.         if (s->avctx->sample_fmt == OUT_FMT_P) {

  1644.             samples_ptr   = samples[ch];

  1645.             sample_stride = 1;

  1646.         } else {

  1647.             samples_ptr   = samples[0] + ch;

  1648.             sample_stride = s->nb_channels;

  1649.         }

  1650.         for (i = 0; i < nb_frames; i++) {

  1651.             RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],

  1652.                                         &(s->synth_buf_offset[ch]),

  1653.                                         RENAME(ff_mpa_synth_window),

  1654.                                         &s->dither_state, samples_ptr,

  1655.                                         sample_stride, s->sb_samples[ch][i]);

  1656.             samples_ptr += 32 * sample_stride;

  1657.         }

  1658.     }

  1659. 

  1660.     return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;

  1661. }

  1662. 

  1663. static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,

  1664.                         AVPacket *avpkt)

  1665. {

  1666.     const uint8_t *buf  = avpkt->data;

  1667.     int buf_size        = avpkt->size;

  1668.     MPADecodeContext *s = avctx->priv_data;

  1669.     uint32_t header;

  1670.     int ret;

  1671. 

  1672.     while(buf_size && !*buf){

  1673.         buf++;

  1674.         buf_size--;

  1675.     }

  1676. 

  1677.     if (buf_size < HEADER_SIZE)

  1678.         return AVERROR_INVALIDDATA;

  1679. 

  1680.     header = AV_RB32(buf);

  1681.     if (header>>8 == AV_RB32("TAG")>>8) {

  1682.         av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");

  1683.         return buf_size;

  1684.     }

  1685.     if (ff_mpa_check_header(header) < 0) {

  1686.         av_log(avctx, AV_LOG_ERROR, "Header missing\n");

  1687.         return AVERROR_INVALIDDATA;

  1688.     }

  1689. 

  1690.     if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {

  1691.         /* free format: prepare to compute frame size */

  1692.         s->frame_size = -1;

  1693.         return AVERROR_INVALIDDATA;

  1694.     }

  1695.     /* update codec info */

  1696.     avctx->channels       = s->nb_channels;

  1697.     avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;

  1698.     if (!avctx->bit_rate)

  1699.         avctx->bit_rate = s->bit_rate;

  1700. 

  1701.     if (s->frame_size <= 0 || s->frame_size > buf_size) {

  1702.         av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");

  1703.         return AVERROR_INVALIDDATA;

  1704.     } else if (s->frame_size < buf_size) {

  1705.         av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");

  1706.         buf_size= s->frame_size;

  1707.     }

  1708. 

  1709.     ret = mp_decode_frame(s, NULL, buf, buf_size);

  1710.     if (ret >= 0) {

  1711.         *got_frame_ptr   = 1;

  1712.         *(AVFrame *)data = s->frame;

  1713.         avctx->sample_rate = s->sample_rate;

  1714.         //FIXME maybe move the other codec info stuff from above here too

  1715.     } else {

  1716.         av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");

  1717.         /* Only return an error if the bad frame makes up the whole packet or

  1718.          * the error is related to buffer management.

  1719.          * If there is more data in the packet, just consume the bad frame

  1720.          * instead of returning an error, which would discard the whole

  1721.          * packet. */

  1722.         *got_frame_ptr = 0;

  1723.         if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)

  1724.             return ret;

  1725.     }

  1726.     s->frame_size = 0;

  1727.     return buf_size;

  1728. }

  1729. 

  1730. static void mp_flush(MPADecodeContext *ctx)

  1731. {

  1732.     memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));

  1733.     ctx->last_buf_size = 0;

  1734. }

  1735. 

  1736. static void flush(AVCodecContext *avctx)

  1737. {

  1738.     mp_flush(avctx->priv_data);

  1739. }

  1740. 

  1741. #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER

  1742. static int decode_frame_adu(AVCodecContext *avctx, void *data,

  1743.                             int *got_frame_ptr, AVPacket *avpkt)

  1744. {

  1745.     const uint8_t *buf  = avpkt->data;

  1746.     int buf_size        = avpkt->size;

  1747.     MPADecodeContext *s = avctx->priv_data;

  1748.     uint32_t header;

  1749.     int len, ret;

  1750.     int av_unused out_size;

  1751. 

  1752.     len = buf_size;

  1753. 

  1754.     // Discard too short frames

  1755.     if (buf_size < HEADER_SIZE) {

  1756.         av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");

  1757.         return AVERROR_INVALIDDATA;

  1758.     }

  1759. 

  1760. 

  1761.     if (len > MPA_MAX_CODED_FRAME_SIZE)

  1762.         len = MPA_MAX_CODED_FRAME_SIZE;

  1763. 

  1764.     // Get header and restore sync word

  1765.     header = AV_RB32(buf) | 0xffe00000;

  1766. 

  1767.     if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame

  1768.         av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");

  1769.         return AVERROR_INVALIDDATA;

  1770.     }

  1771. 

  1772.     avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);

  1773.     /* update codec info */

  1774.     avctx->sample_rate = s->sample_rate;

  1775.     avctx->channels    = s->nb_channels;

  1776.     if (!avctx->bit_rate)

  1777.         avctx->bit_rate = s->bit_rate;

  1778. 

  1779.     s->frame_size = len;

  1780. 

  1781.     ret = mp_decode_frame(s, NULL, buf, buf_size);

  1782.     if (ret < 0) {

  1783.         av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");

  1784.         return ret;

  1785.     }

  1786. 

  1787.     *got_frame_ptr   = 1;

  1788.     *(AVFrame *)data = s->frame;

  1789. 

  1790.     return buf_size;

  1791. }

  1792. #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */

  1793. 

  1794. #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER

  1795. 

  1796. /**

  1797.  * Context for MP3On4 decoder

  1798.  */

  1799. typedef struct MP3On4DecodeContext {

  1800.     AVFrame *frame;

  1801.     int frames;                     ///< number of mp3 frames per block (number of mp3 decoder instances)

  1802.     int syncword;                   ///< syncword patch

  1803.     const uint8_t *coff;            ///< channel offsets in output buffer

  1804.     MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance

  1805. } MP3On4DecodeContext;

  1806. 

  1807. #include "mpeg4audio.h"

  1808. 

  1809. /* Next 3 arrays are indexed by channel config number (passed via codecdata) */

  1810. 

  1811. /* number of mp3 decoder instances */

  1812. static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };

  1813. 

  1814. /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */

  1815. static const uint8_t chan_offset[8][5] = {

  1816.     { 0             },

  1817.     { 0             },  // C

  1818.     { 0             },  // FLR

  1819.     { 2, 0          },  // C FLR

  1820.     { 2, 0, 3       },  // C FLR BS

  1821.     { 2, 0, 3       },  // C FLR BLRS

  1822.     { 2, 0, 4, 3    },  // C FLR BLRS LFE

  1823.     { 2, 0, 6, 4, 3 },  // C FLR BLRS BLR LFE

  1824. };

  1825. 

  1826. /* mp3on4 channel layouts */

  1827. static const int16_t chan_layout[8] = {

  1828.     0,

  1829.     AV_CH_LAYOUT_MONO,

  1830.     AV_CH_LAYOUT_STEREO,

  1831.     AV_CH_LAYOUT_SURROUND,

  1832.     AV_CH_LAYOUT_4POINT0,

  1833.     AV_CH_LAYOUT_5POINT0,

  1834.     AV_CH_LAYOUT_5POINT1,

  1835.     AV_CH_LAYOUT_7POINT1

  1836. };

  1837. 

  1838. static av_cold int decode_close_mp3on4(AVCodecContext * avctx)

  1839. {

  1840.     MP3On4DecodeContext *s = avctx->priv_data;

  1841.     int i;

  1842. 

  1843.     for (i = 0; i < s->frames; i++)

  1844.         av_free(s->mp3decctx[i]);

  1845. 

  1846.     return 0;

  1847. }

  1848. 

  1849. 

  1850. static int decode_init_mp3on4(AVCodecContext * avctx)

  1851. {

  1852.     MP3On4DecodeContext *s = avctx->priv_data;

  1853.     MPEG4AudioConfig cfg;

  1854.     int i;

  1855. 

  1856.     if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {

  1857.         av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");

  1858.         return AVERROR_INVALIDDATA;

  1859.     }

  1860. 

  1861.     avpriv_mpeg4audio_get_config(&cfg, avctx->extradata,

  1862.                                  avctx->extradata_size * 8, 1);

  1863.     if (!cfg.chan_config || cfg.chan_config > 7) {

  1864.         av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");

  1865.         return AVERROR_INVALIDDATA;

  1866.     }

  1867.     s->frames             = mp3Frames[cfg.chan_config];

  1868.     s->coff               = chan_offset[cfg.chan_config];

  1869.     avctx->channels       = ff_mpeg4audio_channels[cfg.chan_config];

  1870.     avctx->channel_layout = chan_layout[cfg.chan_config];

  1871. 

  1872.     if (cfg.sample_rate < 16000)

  1873.         s->syncword = 0xffe00000;

  1874.     else

  1875.         s->syncword = 0xfff00000;

  1876. 

  1877.     /* Init the first mp3 decoder in standard way, so that all tables get builded

  1878.      * We replace avctx->priv_data with the context of the first decoder so that

  1879.      * decode_init() does not have to be changed.

  1880.      * Other decoders will be initialized here copying data from the first context

  1881.      */

  1882.     // Allocate zeroed memory for the first decoder context

  1883.     s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));

  1884.     if (!s->mp3decctx[0])

  1885.         goto alloc_fail;

  1886.     // Put decoder context in place to make init_decode() happy

  1887.     avctx->priv_data = s->mp3decctx[0];

  1888.     decode_init(avctx);

  1889.     s->frame = avctx->coded_frame;

  1890.     // Restore mp3on4 context pointer

  1891.     avctx->priv_data = s;

  1892.     s->mp3decctx[0]->adu_mode = 1; // Set adu mode

  1893. 

  1894.     /* Create a separate codec/context for each frame (first is already ok).

  1895.      * Each frame is 1 or 2 channels - up to 5 frames allowed

  1896.      */

  1897.     for (i = 1; i < s->frames; i++) {

  1898.         s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));

  1899.         if (!s->mp3decctx[i])

  1900.             goto alloc_fail;

  1901.         s->mp3decctx[i]->adu_mode = 1;

  1902.         s->mp3decctx[i]->avctx = avctx;

  1903.         s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;

  1904.     }

  1905. 

  1906.     return 0;

  1907. alloc_fail:

  1908.     decode_close_mp3on4(avctx);

  1909.     return AVERROR(ENOMEM);

  1910. }

  1911. 

  1912. 

  1913. static void flush_mp3on4(AVCodecContext *avctx)

  1914. {

  1915.     int i;

  1916.     MP3On4DecodeContext *s = avctx->priv_data;

  1917. 

  1918.     for (i = 0; i < s->frames; i++)

  1919.         mp_flush(s->mp3decctx[i]);

  1920. }

  1921. 

  1922. 

  1923. static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,

  1924.                                int *got_frame_ptr, AVPacket *avpkt)

  1925. {

  1926.     const uint8_t *buf     = avpkt->data;

  1927.     int buf_size           = avpkt->size;

  1928.     MP3On4DecodeContext *s = avctx->priv_data;

  1929.     MPADecodeContext *m;

  1930.     int fsize, len = buf_size, out_size = 0;

  1931.     uint32_t header;

  1932.     OUT_INT **out_samples;

  1933.     OUT_INT *outptr[2];

  1934.     int fr, ch, ret;

  1935. 

  1936.     /* get output buffer */

  1937.     s->frame->nb_samples = s->frames * MPA_FRAME_SIZE;

  1938.     if ((ret = ff_get_buffer(avctx, s->frame)) < 0) {

  1939.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1940.         return ret;

  1941.     }

  1942.     out_samples = (OUT_INT **)s->frame->extended_data;

  1943. 

  1944.     // Discard too short frames

  1945.     if (buf_size < HEADER_SIZE)

  1946.         return AVERROR_INVALIDDATA;

  1947. 

  1948.     avctx->bit_rate = 0;

  1949. 

  1950.     ch = 0;

  1951.     for (fr = 0; fr < s->frames; fr++) {

  1952.         fsize = AV_RB16(buf) >> 4;

  1953.         fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);

  1954.         m     = s->mp3decctx[fr];

  1955.         av_assert1(m);

  1956. 

  1957.         if (fsize < HEADER_SIZE) {

  1958.             av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");

  1959.             return AVERROR_INVALIDDATA;

  1960.         }

  1961.         header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header

  1962. 

  1963.         if (ff_mpa_check_header(header) < 0) // Bad header, discard block

  1964.             break;

  1965. 

  1966.         avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);

  1967. 

  1968.         if (ch + m->nb_channels > avctx->channels) {

  1969.             av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "

  1970.                                         "channel count\n");

  1971.             return AVERROR_INVALIDDATA;

  1972.         }

  1973.         ch += m->nb_channels;

  1974. 

  1975.         outptr[0] = out_samples[s->coff[fr]];

  1976.         if (m->nb_channels > 1)

  1977.             outptr[1] = out_samples[s->coff[fr] + 1];

  1978. 

  1979.         if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)

  1980.             return ret;

  1981. 

  1982.         out_size += ret;

  1983.         buf      += fsize;

  1984.         len      -= fsize;

  1985. 

  1986.         avctx->bit_rate += m->bit_rate;

  1987.     }

  1988. 

  1989.     /* update codec info */

  1990.     avctx->sample_rate = s->mp3decctx[0]->sample_rate;

  1991. 

  1992.     s->frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));

  1993.     *got_frame_ptr   = 1;

  1994.     *(AVFrame *)data = *s->frame;

  1995. 

  1996.     return buf_size;

  1997. }

  1998. #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */

  1999. 

  2000. #if !CONFIG_FLOAT

  2001. #if CONFIG_MP1_DECODER

  2002. AVCodec ff_mp1_decoder = {

  2003.     .name           = "mp1",

  2004.     .type           = AVMEDIA_TYPE_AUDIO,

  2005.     .id             = AV_CODEC_ID_MP1,

  2006.     .priv_data_size = sizeof(MPADecodeContext),

  2007.     .init           = decode_init,

  2008.     .decode         = decode_frame,

  2009.     .capabilities   = CODEC_CAP_DR1,

  2010.     .flush          = flush,

  2011.     .long_name      = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),

  2012.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2013.                                                       AV_SAMPLE_FMT_S16,

  2014.                                                       AV_SAMPLE_FMT_NONE },

  2015. };

  2016. #endif

  2017. #if CONFIG_MP2_DECODER

  2018. AVCodec ff_mp2_decoder = {

  2019.     .name           = "mp2",

  2020.     .type           = AVMEDIA_TYPE_AUDIO,

  2021.     .id             = AV_CODEC_ID_MP2,

  2022.     .priv_data_size = sizeof(MPADecodeContext),

  2023.     .init           = decode_init,

  2024.     .decode         = decode_frame,

  2025.     .capabilities   = CODEC_CAP_DR1,

  2026.     .flush          = flush,

  2027.     .long_name      = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),

  2028.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2029.                                                       AV_SAMPLE_FMT_S16,

  2030.                                                       AV_SAMPLE_FMT_NONE },

  2031. };

  2032. #endif

  2033. #if CONFIG_MP3_DECODER

  2034. AVCodec ff_mp3_decoder = {

  2035.     .name           = "mp3",

  2036.     .type           = AVMEDIA_TYPE_AUDIO,

  2037.     .id             = AV_CODEC_ID_MP3,

  2038.     .priv_data_size = sizeof(MPADecodeContext),

  2039.     .init           = decode_init,

  2040.     .decode         = decode_frame,

  2041.     .capabilities   = CODEC_CAP_DR1,

  2042.     .flush          = flush,

  2043.     .long_name      = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),

  2044.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2045.                                                       AV_SAMPLE_FMT_S16,

  2046.                                                       AV_SAMPLE_FMT_NONE },

  2047. };

  2048. #endif

  2049. #if CONFIG_MP3ADU_DECODER

  2050. AVCodec ff_mp3adu_decoder = {

  2051.     .name           = "mp3adu",

  2052.     .type           = AVMEDIA_TYPE_AUDIO,

  2053.     .id             = AV_CODEC_ID_MP3ADU,

  2054.     .priv_data_size = sizeof(MPADecodeContext),

  2055.     .init           = decode_init,

  2056.     .decode         = decode_frame_adu,

  2057.     .capabilities   = CODEC_CAP_DR1,

  2058.     .flush          = flush,

  2059.     .long_name      = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),

  2060.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2061.                                                       AV_SAMPLE_FMT_S16,

  2062.                                                       AV_SAMPLE_FMT_NONE },

  2063. };

  2064. #endif

  2065. #if CONFIG_MP3ON4_DECODER

  2066. AVCodec ff_mp3on4_decoder = {

  2067.     .name           = "mp3on4",

  2068.     .type           = AVMEDIA_TYPE_AUDIO,

  2069.     .id             = AV_CODEC_ID_MP3ON4,

  2070.     .priv_data_size = sizeof(MP3On4DecodeContext),

  2071.     .init           = decode_init_mp3on4,

  2072.     .close          = decode_close_mp3on4,

  2073.     .decode         = decode_frame_mp3on4,

  2074.     .capabilities   = CODEC_CAP_DR1,

  2075.     .flush          = flush_mp3on4,

  2076.     .long_name      = NULL_IF_CONFIG_SMALL("MP3onMP4"),

  2077.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2078.                                                       AV_SAMPLE_FMT_NONE },

  2079. };

  2080. #endif

  2081. #endif