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FFmpeg/libavcodec/mpegaudiodec.c

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  1. /*

  2.  * MPEG Audio decoder

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

  4.  *

  5.  * This file is part of Libav.

  6.  *

  7.  * Libav is free software; you can redistribute it and/or

  8.  * modify it under the terms of the GNU Lesser General Public

  9.  * License as published by the Free Software Foundation; either

  10.  * version 2.1 of the License, or (at your option) any later version.

  11.  *

  12.  * Libav is distributed in the hope that it will be useful,

  13.  * but WITHOUT ANY WARRANTY; without even the implied warranty of

  14.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

  15.  * Lesser General Public License for more details.

  16.  *

  17.  * You should have received a copy of the GNU Lesser General Public

  18.  * License along with Libav; if not, write to the Free Software

  19.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

  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/float_dsp.h"

  30. #include "avcodec.h"

  31. #include "get_bits.h"

  32. #include "internal.h"

  33. #include "mathops.h"

  34. #include "mpegaudiodsp.h"

  35. 

  36. /*

  37.  * TODO:

  38.  *  - test lsf / mpeg25 extensively.

  39.  */

  40. 

  41. #include "mpegaudio.h"

  42. #include "mpegaudiodecheader.h"

  43. 

  44. #define BACKSTEP_SIZE 512

  45. #define EXTRABYTES 24

  46. #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES

  47. 

  48. /* layer 3 "granule" */

  49. typedef struct GranuleDef {

  50.     uint8_t scfsi;

  51.     int part2_3_length;

  52.     int big_values;

  53.     int global_gain;

  54.     int scalefac_compress;

  55.     uint8_t block_type;

  56.     uint8_t switch_point;

  57.     int table_select[3];

  58.     int subblock_gain[3];

  59.     uint8_t scalefac_scale;

  60.     uint8_t count1table_select;

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

  62.     int preflag;

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

  64.     uint8_t scale_factors[40];

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

  66. } GranuleDef;

  67. 

  68. typedef struct MPADecodeContext {

  69.     MPA_DECODE_HEADER

  70.     uint8_t last_buf[LAST_BUF_SIZE];

  71.     int last_buf_size;

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

  73.     uint32_t free_format_next_header;

  74.     GetBitContext gb;

  75.     GetBitContext in_gb;

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

  77.     int synth_buf_offset[MPA_MAX_CHANNELS];

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

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

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

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

  82.     int dither_state;

  83.     int err_recognition;

  84.     AVCodecContext* avctx;

  85.     MPADSPContext mpadsp;

  86.     AVFloatDSPContext fdsp;

  87.     AVFrame *frame;

  88. } MPADecodeContext;

  89. 

  90. #if CONFIG_FLOAT

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

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

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

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

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

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

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

  98. #   define OUT_FMT   AV_SAMPLE_FMT_FLT

  99. #   define OUT_FMT_P AV_SAMPLE_FMT_FLTP

  100. #else

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

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

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

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

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

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

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

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

  109. #   define OUT_FMT   AV_SAMPLE_FMT_S16

  110. #   define OUT_FMT_P AV_SAMPLE_FMT_S16P

  111. #endif

  112. 

  113. /****************/

  114. 

  115. #define HEADER_SIZE 4

  116. 

  117. #include "mpegaudiodata.h"

  118. #include "mpegaudiodectab.h"

  119. 

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

  121. static VLC huff_vlc[16];

  122. static VLC_TYPE huff_vlc_tables[

  123.     0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +

  124.   142 + 204 + 190 + 170 + 542 + 460 + 662 + 414

  125.   ][2];

  126. static const int huff_vlc_tables_sizes[16] = {

  127.     0,  128,  128,  128,  130,  128,  154,  166,

  128.   142,  204,  190,  170,  542,  460,  662,  414

  129. };

  130. static VLC huff_quad_vlc[2];

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

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

  133. /* computed from band_size_long */

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

  135. #include "mpegaudio_tablegen.h"

  136. /* intensity stereo coef table */

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

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

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

  140. 

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

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

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

  144. 

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

  146.     division_tab3, division_tab5, NULL, division_tab9

  147. };

  148. 

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

  150. static uint16_t scale_factor_modshift[64];

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

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

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

  154. 

  155. #define SCALE_GEN(v) \

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

  157. 

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

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

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

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

  162. };

  163. 

  164. /**

  165.  * Convert region offsets to region sizes and truncate

  166.  * size to big_values.

  167.  */

  168. static void ff_region_offset2size(GranuleDef *g)

  169. {

  170.     int i, k, j = 0;

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

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

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

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

  175.         j = k;

  176.     }

  177. }

  178. 

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

  180. {

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

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

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

  184.         else

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

  186.     } else {

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

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

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

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

  191.         else

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

  193.     }

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

  195. }

  196. 

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

  198. {

  199.     int l;

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

  201.     /* should not overflow */

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

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

  204. }

  205. 

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

  207. {

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

  209.         if (g->switch_point) {

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

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

  212.                 exponents as long blocks */

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

  214.                 g->long_end = 8;

  215.             else

  216.                 g->long_end = 6;

  217. 

  218.             g->short_start = 3;

  219.         } else {

  220.             g->long_end    = 0;

  221.             g->short_start = 0;

  222.         }

  223.     } else {

  224.         g->short_start = 13;

  225.         g->long_end    = 22;

  226.     }

  227. }

  228. 

  229. /* layer 1 unscaling */

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

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

  232. {

  233.     int shift, mod;

  234.     int64_t val;

  235. 

  236.     shift   = scale_factor_modshift[scale_factor];

  237.     mod     = shift & 3;

  238.     shift >>= 2;

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

  240.     shift  += n;

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

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

  243. }

  244. 

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

  246. {

  247.     int shift, mod, val;

  248. 

  249.     shift   = scale_factor_modshift[scale_factor];

  250.     mod     = shift & 3;

  251.     shift >>= 2;

  252. 

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

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

  255.     if (shift > 0)

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

  257.     return val;

  258. }

  259. 

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

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

  262. {

  263.     unsigned int m;

  264.     int e;

  265. 

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

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

  268.     e -= exponent >> 2;

  269.     assert(e >= 1);

  270.     if (e > 31)

  271.         return 0;

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

  273. 

  274.     return m;

  275. }

  276. 

  277. static av_cold void decode_init_static(void)

  278. {

  279.     int i, j, k;

  280.     int offset;

  281. 

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

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

  284.         int shift, mod;

  285.         /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */

  286.         shift = i / 3;

  287.         mod   = i % 3;

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

  289.     }

  290. 

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

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

  293.         int n, norm;

  294.         n = i + 2;

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

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

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

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

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

  300.                 scale_factor_mult[i][0],

  301.                 scale_factor_mult[i][1],

  302.                 scale_factor_mult[i][2]);

  303.     }

  304. 

  305.     RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));

  306. 

  307.     /* huffman decode tables */

  308.     offset = 0;

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

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

  311.         int xsize, x, y;

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

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

  314. 

  315.         xsize = h->xsize;

  316. 

  317.         j = 0;

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

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

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

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

  322.             }

  323.         }

  324. 

  325.         /* XXX: fail test */

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

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

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

  329.                  tmp_bits, 1, 1, tmp_codes, 2, 2,

  330.                  INIT_VLC_USE_NEW_STATIC);

  331.         offset += huff_vlc_tables_sizes[i];

  332.     }

  333.     assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));

  334. 

  335.     offset = 0;

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

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

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

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

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

  341.                  INIT_VLC_USE_NEW_STATIC);

  342.         offset += huff_quad_vlc_tables_sizes[i];

  343.     }

  344.     assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));

  345. 

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

  347.         k = 0;

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

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

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

  351.         }

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

  353.     }

  354. 

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

  356. 

  357.     mpegaudio_tableinit();

  358. 

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

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

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

  362.                 int val1, val2, val3, steps;

  363.                 int val = j;

  364.                 steps   = ff_mpa_quant_steps[i];

  365.                 val1    = val % steps;

  366.                 val    /= steps;

  367.                 val2    = val % steps;

  368.                 val3    = val / steps;

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

  370.             }

  371.         }

  372.     }

  373. 

  374. 

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

  376.         float f;

  377.         INTFLOAT v;

  378.         if (i != 6) {

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

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

  381.         } else {

  382.             v = FIXR(1.0);

  383.         }

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

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

  386.     }

  387.     /* invalid values */

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

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

  390. 

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

  392.         double f;

  393.         int e, k;

  394. 

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

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

  397.             f = pow(2.0, e / 4.0);

  398.             k = i & 1;

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

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

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

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

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

  404.         }

  405.     }

  406. 

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

  408.         float ci, cs, ca;

  409.         ci = ci_table[i];

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

  411.         ca = cs * ci;

  412. #if !CONFIG_FLOAT

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

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

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

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

  417. #else

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

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

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

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

  422. #endif

  423.     }

  424. }

  425. 

  426. static av_cold int decode_init(AVCodecContext * avctx)

  427. {

  428.     static int initialized_tables = 0;

  429.     MPADecodeContext *s = avctx->priv_data;

  430. 

  431.     if (!initialized_tables) {

  432.         decode_init_static();

  433.         initialized_tables = 1;

  434.     }

  435. 

  436.     s->avctx = avctx;

  437. 

  438.     avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

  439.     ff_mpadsp_init(&s->mpadsp);

  440. 

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

  442.         avctx->codec_id != AV_CODEC_ID_MP3ON4)

  443.         avctx->sample_fmt = OUT_FMT;

  444.     else

  445.         avctx->sample_fmt = OUT_FMT_P;

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

  447. 

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

  449.         s->adu_mode = 1;

  450. 

  451.     return 0;

  452. }

  453. 

  454. #define C3 FIXHR(0.86602540378443864676/2)

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

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

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

  458. 

  459. /* 12 points IMDCT. We compute it "by hand" by factorizing obvious

  460.    cases. */

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

  462. {

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

  464. 

  465.     in0  = in[0*3];

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

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

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

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

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

  471.     in5 += in3;

  472.     in3 += in1;

  473. 

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

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

  476. 

  477.     t1   = in0 - in4;

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

  479. 

  480.     out[ 7] =

  481.     out[10] = t1 + t2;

  482.     out[ 1] =

  483.     out[ 4] = t1 - t2;

  484. 

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

  486.     in4     = in0 + in2;

  487.     in5    += 2*in1;

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

  489.     out[ 8] =

  490.     out[ 9] = in4 + in1;

  491.     out[ 2] =

  492.     out[ 3] = in4 - in1;

  493. 

  494.     in0    -= in2;

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

  496.     out[ 0] =

  497.     out[ 5] = in0 - in5;

  498.     out[ 6] =

  499.     out[11] = in0 + in5;

  500. }

  501. 

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

  503. static int mp_decode_layer1(MPADecodeContext *s)

  504. {

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

  506.     uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];

  507.     uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];

  508. 

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

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

  511.     else

  512.         bound = SBLIMIT;

  513. 

  514.     /* allocation bits */

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

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

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

  518.         }

  519.     }

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

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

  522. 

  523.     /* scale factors */

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

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

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

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

  528.         }

  529.     }

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

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

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

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

  534.         }

  535.     }

  536. 

  537.     /* compute samples */

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

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

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

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

  542.                 if (n) {

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

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

  545.                 } else {

  546.                     v = 0;

  547.                 }

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

  549.             }

  550.         }

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

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

  553.             if (n) {

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

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

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

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

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

  559.             } else {

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

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

  562.             }

  563.         }

  564.     }

  565.     return 12;

  566. }

  567. 

  568. static int mp_decode_layer2(MPADecodeContext *s)

  569. {

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

  571.     const unsigned char *alloc_table;

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

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

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

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

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

  577. 

  578.     /* select decoding table */

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

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

  581.     sblimit     = ff_mpa_sblimit_table[table];

  582.     alloc_table = ff_mpa_alloc_tables[table];

  583. 

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

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

  586.     else

  587.         bound = sblimit;

  588. 

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

  590. 

  591.     /* sanity check */

  592.     if (bound > sblimit)

  593.         bound = sblimit;

  594. 

  595.     /* parse bit allocation */

  596.     j = 0;

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

  598.         bit_alloc_bits = alloc_table[j];

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

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

  601.         j += 1 << bit_alloc_bits;

  602.     }

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

  604.         bit_alloc_bits = alloc_table[j];

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

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

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

  608.         j += 1 << bit_alloc_bits;

  609.     }

  610. 

  611.     /* scale codes */

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

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

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

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

  616.         }

  617.     }

  618. 

  619.     /* scale factors */

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

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

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

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

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

  625.                 default:

  626.                 case 0:

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

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

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

  630.                     break;

  631.                 case 2:

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

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

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

  635.                     break;

  636.                 case 1:

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

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

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

  640.                     break;

  641.                 case 3:

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

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

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

  645.                     break;

  646.                 }

  647.             }

  648.         }

  649.     }

  650. 

  651.     /* samples */

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

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

  654.             j = 0;

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

  656.                 bit_alloc_bits = alloc_table[j];

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

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

  659.                     if (b) {

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

  661.                         qindex = alloc_table[j+b];

  662.                         bits = ff_mpa_quant_bits[qindex];

  663.                         if (bits < 0) {

  664.                             int v2;

  665.                             /* 3 values at the same time */

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

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

  668.                             steps  = ff_mpa_quant_steps[qindex];

  669. 

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

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

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

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

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

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

  676.                         } else {

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

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

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

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

  681.                             }

  682.                         }

  683.                     } else {

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

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

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

  687.                     }

  688.                 }

  689.                 /* next subband in alloc table */

  690.                 j += 1 << bit_alloc_bits;

  691.             }

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

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

  694.                 bit_alloc_bits = alloc_table[j];

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

  696.                 if (b) {

  697.                     int mant, scale0, scale1;

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

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

  700.                     qindex = alloc_table[j+b];

  701.                     bits = ff_mpa_quant_bits[qindex];

  702.                     if (bits < 0) {

  703.                         /* 3 values at the same time */

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

  705.                         steps = ff_mpa_quant_steps[qindex];

  706.                         mant = v % steps;

  707.                         v = v / steps;

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

  709.                             l2_unscale_group(steps, mant, scale0);

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

  711.                             l2_unscale_group(steps, mant, scale1);

  712.                         mant = v % steps;

  713.                         v = v / steps;

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

  715.                             l2_unscale_group(steps, mant, scale0);

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

  717.                             l2_unscale_group(steps, mant, scale1);

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

  719.                             l2_unscale_group(steps, v, scale0);

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

  721.                             l2_unscale_group(steps, v, scale1);

  722.                     } else {

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

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

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

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

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

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

  729.                         }

  730.                     }

  731.                 } else {

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

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

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

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

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

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

  738.                 }

  739.                 /* next subband in alloc table */

  740.                 j += 1 << bit_alloc_bits;

  741.             }

  742.             /* fill remaining samples to zero */

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

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

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

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

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

  748.                 }

  749.             }

  750.         }

  751.     }

  752.     return 3 * 12;

  753. }

  754. 

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

  756.     if (n == 3) {                   \

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

  758.         dst   = sf - 3 * m;         \

  759.         sf    = m;                  \

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

  761.         dst  = sf & 3;              \

  762.         sf >>= 2;                   \

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

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

  765.         dst   = sf - 5 * m;         \

  766.         sf    = m;                  \

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

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

  769.         dst   = sf - 6 * m;         \

  770.         sf    = m;                  \

  771.     } else {                        \

  772.         dst = 0;                    \

  773.     }

  774. 

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

  776.                                            int n3)

  777. {

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

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

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

  781.     slen[0] = sf;

  782. }

  783. 

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

  785.                                          int16_t *exponents)

  786. {

  787.     const uint8_t *bstab, *pretab;

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

  789.     int16_t *exp_ptr;

  790. 

  791.     exp_ptr = exponents;

  792.     gain    = g->global_gain - 210;

  793.     shift   = g->scalefac_scale + 1;

  794. 

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

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

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

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

  799.         len = bstab[i];

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

  801.             *exp_ptr++ = v0;

  802.     }

  803. 

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

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

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

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

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

  809.         k        = g->long_end;

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

  811.             len = bstab[i];

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

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

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

  815.                     *exp_ptr++ = v0;

  816.             }

  817.         }

  818.     }

  819. }

  820. 

  821. /* handle n = 0 too */

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

  823. {

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

  825. }

  826. 

  827. 

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

  829.                           int *end_pos2)

  830. {

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

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

  833.         s->in_gb.buffer = NULL;

  834.         assert((get_bits_count(&s->gb) & 7) == 0);

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

  836.         *end_pos2 =

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

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

  839.     }

  840. }

  841. 

  842. /* Following is a optimized code for

  843.             INTFLOAT v = *src

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

  845.                 v = -v;

  846.             *dst = v;

  847. */

  848. #if CONFIG_FLOAT

  849. #define READ_FLIP_SIGN(dst,src)                     \

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

  851.     AV_WN32A(dst, v);

  852. #else

  853. #define READ_FLIP_SIGN(dst,src)     \

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

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

  856. #endif

  857. 

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

  859.                           int16_t *exponents, int end_pos2)

  860. {

  861.     int s_index;

  862.     int i;

  863.     int last_pos, bits_left;

  864.     VLC *vlc;

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

  866. 

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

  868.     s_index = 0;

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

  870.         int j, k, l, linbits;

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

  872.         if (j == 0)

  873.             continue;

  874.         /* select vlc table */

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

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

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

  878.         vlc     = &huff_vlc[l];

  879. 

  880.         if (!l) {

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

  882.             s_index += 2 * j;

  883.             continue;

  884.         }

  885. 

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

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

  888.             int exponent, x, y;

  889.             int v;

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

  891. 

  892.             if (pos >= end_pos){

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

  894.                 if (pos >= end_pos)

  895.                     break;

  896.             }

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

  898. 

  899.             if (!y) {

  900.                 g->sb_hybrid[s_index  ] =

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

  902.                 s_index += 2;

  903.                 continue;

  904.             }

  905. 

  906.             exponent= exponents[s_index];

  907. 

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

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

  910.             if (y & 16) {

  911.                 x = y >> 5;

  912.                 y = y & 0x0f;

  913.                 if (x < 15) {

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

  915.                 } else {

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

  917.                     v  = l3_unscale(x, exponent);

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

  919.                         v = -v;

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

  921.                 }

  922.                 if (y < 15) {

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

  924.                 } else {

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

  926.                     v  = l3_unscale(y, exponent);

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

  928.                         v = -v;

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

  930.                 }

  931.             } else {

  932.                 x = y >> 5;

  933.                 y = y & 0x0f;

  934.                 x += y;

  935.                 if (x < 15) {

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

  937.                 } else {

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

  939.                     v  = l3_unscale(x, exponent);

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

  941.                         v = -v;

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

  943.                 }

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

  945.             }

  946.             s_index += 2;

  947.         }

  948.     }

  949. 

  950.     /* high frequencies */

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

  952.     last_pos = 0;

  953.     while (s_index <= 572) {

  954.         int pos, code;

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

  956.         if (pos >= end_pos) {

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

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

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

  960.                 s_index -= 4;

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

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

  963.                 if(s->err_recognition & AV_EF_BITSTREAM)

  964.                     s_index=0;

  965.                 break;

  966.             }

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

  968.             if (pos >= end_pos)

  969.                 break;

  970.         }

  971.         last_pos = pos;

  972. 

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

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

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

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

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

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

  979.         while (code) {

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

  981.             int v;

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

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

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

  985.         }

  986.         s_index += 4;

  987.     }

  988.     /* skip extension bits */

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

  990.     if (bits_left < 0 && (s->err_recognition & AV_EF_BUFFER)) {

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

  992.         s_index=0;

  993.     } else if (bits_left > 0 && (s->err_recognition & AV_EF_BUFFER)) {

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

  995.         s_index = 0;

  996.     }

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

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

  999. 

  1000.     i = get_bits_count(&s->gb);

  1001.     switch_buffer(s, &i, &end_pos, &end_pos2);

  1002. 

  1003.     return 0;

  1004. }

  1005. 

  1006. /* Reorder short blocks from bitstream order to interleaved order. It

  1007.    would be faster to do it in parsing, but the code would be far more

  1008.    complicated */

  1009. static void reorder_block(MPADecodeContext *s, GranuleDef *g)

  1010. {

  1011.     int i, j, len;

  1012.     INTFLOAT *ptr, *dst, *ptr1;

  1013.     INTFLOAT tmp[576];

  1014. 

  1015.     if (g->block_type != 2)

  1016.         return;

  1017. 

  1018.     if (g->switch_point) {

  1019.         if (s->sample_rate_index != 8)

  1020.             ptr = g->sb_hybrid + 36;

  1021.         else

  1022.             ptr = g->sb_hybrid + 72;

  1023.     } else {

  1024.         ptr = g->sb_hybrid;

  1025.     }

  1026. 

  1027.     for (i = g->short_start; i < 13; i++) {

  1028.         len  = band_size_short[s->sample_rate_index][i];

  1029.         ptr1 = ptr;

  1030.         dst  = tmp;

  1031.         for (j = len; j > 0; j--) {

  1032.             *dst++ = ptr[0*len];

  1033.             *dst++ = ptr[1*len];

  1034.             *dst++ = ptr[2*len];

  1035.             ptr++;

  1036.         }

  1037.         ptr += 2 * len;

  1038.         memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));

  1039.     }

  1040. }

  1041. 

  1042. #define ISQRT2 FIXR(0.70710678118654752440)

  1043. 

  1044. static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)

  1045. {

  1046.     int i, j, k, l;

  1047.     int sf_max, sf, len, non_zero_found;

  1048.     INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;

  1049.     int non_zero_found_short[3];

  1050. 

  1051.     /* intensity stereo */

  1052.     if (s->mode_ext & MODE_EXT_I_STEREO) {

  1053.         if (!s->lsf) {

  1054.             is_tab = is_table;

  1055.             sf_max = 7;

  1056.         } else {

  1057.             is_tab = is_table_lsf[g1->scalefac_compress & 1];

  1058.             sf_max = 16;

  1059.         }

  1060. 

  1061.         tab0 = g0->sb_hybrid + 576;

  1062.         tab1 = g1->sb_hybrid + 576;

  1063. 

  1064.         non_zero_found_short[0] = 0;

  1065.         non_zero_found_short[1] = 0;

  1066.         non_zero_found_short[2] = 0;

  1067.         k = (13 - g1->short_start) * 3 + g1->long_end - 3;

  1068.         for (i = 12; i >= g1->short_start; i--) {

  1069.             /* for last band, use previous scale factor */

  1070.             if (i != 11)

  1071.                 k -= 3;

  1072.             len = band_size_short[s->sample_rate_index][i];

  1073.             for (l = 2; l >= 0; l--) {

  1074.                 tab0 -= len;

  1075.                 tab1 -= len;

  1076.                 if (!non_zero_found_short[l]) {

  1077.                     /* test if non zero band. if so, stop doing i-stereo */

  1078.                     for (j = 0; j < len; j++) {

  1079.                         if (tab1[j] != 0) {

  1080.                             non_zero_found_short[l] = 1;

  1081.                             goto found1;

  1082.                         }

  1083.                     }

  1084.                     sf = g1->scale_factors[k + l];

  1085.                     if (sf >= sf_max)

  1086.                         goto found1;

  1087. 

  1088.                     v1 = is_tab[0][sf];

  1089.                     v2 = is_tab[1][sf];

  1090.                     for (j = 0; j < len; j++) {

  1091.                         tmp0    = tab0[j];

  1092.                         tab0[j] = MULLx(tmp0, v1, FRAC_BITS);

  1093.                         tab1[j] = MULLx(tmp0, v2, FRAC_BITS);

  1094.                     }

  1095.                 } else {

  1096. found1:

  1097.                     if (s->mode_ext & MODE_EXT_MS_STEREO) {

  1098.                         /* lower part of the spectrum : do ms stereo

  1099.                            if enabled */

  1100.                         for (j = 0; j < len; j++) {

  1101.                             tmp0    = tab0[j];

  1102.                             tmp1    = tab1[j];

  1103.                             tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);

  1104.                             tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);

  1105.                         }

  1106.                     }

  1107.                 }

  1108.             }

  1109.         }

  1110. 

  1111.         non_zero_found = non_zero_found_short[0] |

  1112.                          non_zero_found_short[1] |

  1113.                          non_zero_found_short[2];

  1114. 

  1115.         for (i = g1->long_end - 1;i >= 0;i--) {

  1116.             len   = band_size_long[s->sample_rate_index][i];

  1117.             tab0 -= len;

  1118.             tab1 -= len;

  1119.             /* test if non zero band. if so, stop doing i-stereo */

  1120.             if (!non_zero_found) {

  1121.                 for (j = 0; j < len; j++) {

  1122.                     if (tab1[j] != 0) {

  1123.                         non_zero_found = 1;

  1124.                         goto found2;

  1125.                     }

  1126.                 }

  1127.                 /* for last band, use previous scale factor */

  1128.                 k  = (i == 21) ? 20 : i;

  1129.                 sf = g1->scale_factors[k];

  1130.                 if (sf >= sf_max)

  1131.                     goto found2;

  1132.                 v1 = is_tab[0][sf];

  1133.                 v2 = is_tab[1][sf];

  1134.                 for (j = 0; j < len; j++) {

  1135.                     tmp0    = tab0[j];

  1136.                     tab0[j] = MULLx(tmp0, v1, FRAC_BITS);

  1137.                     tab1[j] = MULLx(tmp0, v2, FRAC_BITS);

  1138.                 }

  1139.             } else {

  1140. found2:

  1141.                 if (s->mode_ext & MODE_EXT_MS_STEREO) {

  1142.                     /* lower part of the spectrum : do ms stereo

  1143.                        if enabled */

  1144.                     for (j = 0; j < len; j++) {

  1145.                         tmp0    = tab0[j];

  1146.                         tmp1    = tab1[j];

  1147.                         tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);

  1148.                         tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);

  1149.                     }

  1150.                 }

  1151.             }

  1152.         }

  1153.     } else if (s->mode_ext & MODE_EXT_MS_STEREO) {

  1154.         /* ms stereo ONLY */

  1155.         /* NOTE: the 1/sqrt(2) normalization factor is included in the

  1156.            global gain */

  1157. #if CONFIG_FLOAT

  1158.        s->fdsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);

  1159. #else

  1160.         tab0 = g0->sb_hybrid;

  1161.         tab1 = g1->sb_hybrid;

  1162.         for (i = 0; i < 576; i++) {

  1163.             tmp0    = tab0[i];

  1164.             tmp1    = tab1[i];

  1165.             tab0[i] = tmp0 + tmp1;

  1166.             tab1[i] = tmp0 - tmp1;

  1167.         }

  1168. #endif

  1169.     }

  1170. }

  1171. 

  1172. #if CONFIG_FLOAT

  1173. #define AA(j) do {                                                      \

  1174.         float tmp0 = ptr[-1-j];                                         \

  1175.         float tmp1 = ptr[   j];                                         \

  1176.         ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1];    \

  1177.         ptr[   j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0];    \

  1178.     } while (0)

  1179. #else

  1180. #define AA(j) do {                                              \

  1181.         int tmp0 = ptr[-1-j];                                   \

  1182.         int tmp1 = ptr[   j];                                   \

  1183.         int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]);          \

  1184.         ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2]));   \

  1185.         ptr[   j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3]));   \

  1186.     } while (0)

  1187. #endif

  1188. 

  1189. static void compute_antialias(MPADecodeContext *s, GranuleDef *g)

  1190. {

  1191.     INTFLOAT *ptr;

  1192.     int n, i;

  1193. 

  1194.     /* we antialias only "long" bands */

  1195.     if (g->block_type == 2) {

  1196.         if (!g->switch_point)

  1197.             return;

  1198.         /* XXX: check this for 8000Hz case */

  1199.         n = 1;

  1200.     } else {

  1201.         n = SBLIMIT - 1;

  1202.     }

  1203. 

  1204.     ptr = g->sb_hybrid + 18;

  1205.     for (i = n; i > 0; i--) {

  1206.         AA(0);

  1207.         AA(1);

  1208.         AA(2);

  1209.         AA(3);

  1210.         AA(4);

  1211.         AA(5);

  1212.         AA(6);

  1213.         AA(7);

  1214. 

  1215.         ptr += 18;

  1216.     }

  1217. }

  1218. 

  1219. static void compute_imdct(MPADecodeContext *s, GranuleDef *g,

  1220.                           INTFLOAT *sb_samples, INTFLOAT *mdct_buf)

  1221. {

  1222.     INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;

  1223.     INTFLOAT out2[12];

  1224.     int i, j, mdct_long_end, sblimit;

  1225. 

  1226.     /* find last non zero block */

  1227.     ptr  = g->sb_hybrid + 576;

  1228.     ptr1 = g->sb_hybrid + 2 * 18;

  1229.     while (ptr >= ptr1) {

  1230.         int32_t *p;

  1231.         ptr -= 6;

  1232.         p    = (int32_t*)ptr;

  1233.         if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])

  1234.             break;

  1235.     }

  1236.     sblimit = ((ptr - g->sb_hybrid) / 18) + 1;

  1237. 

  1238.     if (g->block_type == 2) {

  1239.         /* XXX: check for 8000 Hz */

  1240.         if (g->switch_point)

  1241.             mdct_long_end = 2;

  1242.         else

  1243.             mdct_long_end = 0;

  1244.     } else {

  1245.         mdct_long_end = sblimit;

  1246.     }

  1247. 

  1248.     s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,

  1249.                                      mdct_long_end, g->switch_point,

  1250.                                      g->block_type);

  1251. 

  1252.     buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);

  1253.     ptr = g->sb_hybrid + 18 * mdct_long_end;

  1254. 

  1255.     for (j = mdct_long_end; j < sblimit; j++) {

  1256.         /* select frequency inversion */

  1257.         win     = RENAME(ff_mdct_win)[2 + (4  & -(j & 1))];

  1258.         out_ptr = sb_samples + j;

  1259. 

  1260.         for (i = 0; i < 6; i++) {

  1261.             *out_ptr = buf[4*i];

  1262.             out_ptr += SBLIMIT;

  1263.         }

  1264.         imdct12(out2, ptr + 0);

  1265.         for (i = 0; i < 6; i++) {

  1266.             *out_ptr     = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*1)];

  1267.             buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);

  1268.             out_ptr += SBLIMIT;

  1269.         }

  1270.         imdct12(out2, ptr + 1);

  1271.         for (i = 0; i < 6; i++) {

  1272.             *out_ptr     = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*2)];

  1273.             buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);

  1274.             out_ptr += SBLIMIT;

  1275.         }

  1276.         imdct12(out2, ptr + 2);

  1277.         for (i = 0; i < 6; i++) {

  1278.             buf[4*(i + 6*0)] = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*0)];

  1279.             buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);

  1280.             buf[4*(i + 6*2)] = 0;

  1281.         }

  1282.         ptr += 18;

  1283.         buf += (j&3) != 3 ? 1 : (4*18-3);

  1284.     }

  1285.     /* zero bands */

  1286.     for (j = sblimit; j < SBLIMIT; j++) {

  1287.         /* overlap */

  1288.         out_ptr = sb_samples + j;

  1289.         for (i = 0; i < 18; i++) {

  1290.             *out_ptr = buf[4*i];

  1291.             buf[4*i]   = 0;

  1292.             out_ptr += SBLIMIT;

  1293.         }

  1294.         buf += (j&3) != 3 ? 1 : (4*18-3);

  1295.     }

  1296. }

  1297. 

  1298. /* main layer3 decoding function */

  1299. static int mp_decode_layer3(MPADecodeContext *s)

  1300. {

  1301.     int nb_granules, main_data_begin;

  1302.     int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;

  1303.     GranuleDef *g;

  1304.     int16_t exponents[576]; //FIXME try INTFLOAT

  1305. 

  1306.     /* read side info */

  1307.     if (s->lsf) {

  1308.         main_data_begin = get_bits(&s->gb, 8);

  1309.         skip_bits(&s->gb, s->nb_channels);

  1310.         nb_granules = 1;

  1311.     } else {

  1312.         main_data_begin = get_bits(&s->gb, 9);

  1313.         if (s->nb_channels == 2)

  1314.             skip_bits(&s->gb, 3);

  1315.         else

  1316.             skip_bits(&s->gb, 5);

  1317.         nb_granules = 2;

  1318.         for (ch = 0; ch < s->nb_channels; ch++) {

  1319.             s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */

  1320.             s->granules[ch][1].scfsi = get_bits(&s->gb, 4);

  1321.         }

  1322.     }

  1323. 

  1324.     for (gr = 0; gr < nb_granules; gr++) {

  1325.         for (ch = 0; ch < s->nb_channels; ch++) {

  1326.             av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);

  1327.             g = &s->granules[ch][gr];

  1328.             g->part2_3_length = get_bits(&s->gb, 12);

  1329.             g->big_values     = get_bits(&s->gb,  9);

  1330.             if (g->big_values > 288) {

  1331.                 av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");

  1332.                 return AVERROR_INVALIDDATA;

  1333.             }

  1334. 

  1335.             g->global_gain = get_bits(&s->gb, 8);

  1336.             /* if MS stereo only is selected, we precompute the

  1337.                1/sqrt(2) renormalization factor */

  1338.             if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==

  1339.                 MODE_EXT_MS_STEREO)

  1340.                 g->global_gain -= 2;

  1341.             if (s->lsf)

  1342.                 g->scalefac_compress = get_bits(&s->gb, 9);

  1343.             else

  1344.                 g->scalefac_compress = get_bits(&s->gb, 4);

  1345.             blocksplit_flag = get_bits1(&s->gb);

  1346.             if (blocksplit_flag) {

  1347.                 g->block_type = get_bits(&s->gb, 2);

  1348.                 if (g->block_type == 0) {

  1349.                     av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");

  1350.                     return AVERROR_INVALIDDATA;

  1351.                 }

  1352.                 g->switch_point = get_bits1(&s->gb);

  1353.                 for (i = 0; i < 2; i++)

  1354.                     g->table_select[i] = get_bits(&s->gb, 5);

  1355.                 for (i = 0; i < 3; i++)

  1356.                     g->subblock_gain[i] = get_bits(&s->gb, 3);

  1357.                 ff_init_short_region(s, g);

  1358.             } else {

  1359.                 int region_address1, region_address2;

  1360.                 g->block_type = 0;

  1361.                 g->switch_point = 0;

  1362.                 for (i = 0; i < 3; i++)

  1363.                     g->table_select[i] = get_bits(&s->gb, 5);

  1364.                 /* compute huffman coded region sizes */

  1365.                 region_address1 = get_bits(&s->gb, 4);

  1366.                 region_address2 = get_bits(&s->gb, 3);

  1367.                 av_dlog(s->avctx, "region1=%d region2=%d\n",

  1368.                         region_address1, region_address2);

  1369.                 ff_init_long_region(s, g, region_address1, region_address2);

  1370.             }

  1371.             ff_region_offset2size(g);

  1372.             ff_compute_band_indexes(s, g);

  1373. 

  1374.             g->preflag = 0;

  1375.             if (!s->lsf)

  1376.                 g->preflag = get_bits1(&s->gb);

  1377.             g->scalefac_scale     = get_bits1(&s->gb);

  1378.             g->count1table_select = get_bits1(&s->gb);

  1379.             av_dlog(s->avctx, "block_type=%d switch_point=%d\n",

  1380.                     g->block_type, g->switch_point);

  1381.         }

  1382.     }

  1383. 

  1384.     if (!s->adu_mode) {

  1385.         int skip;

  1386.         const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);

  1387.         int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0,

  1388.                                 FFMAX(0, LAST_BUF_SIZE - s->last_buf_size));

  1389.         assert((get_bits_count(&s->gb) & 7) == 0);

  1390.         /* now we get bits from the main_data_begin offset */

  1391.         av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",

  1392.                 main_data_begin, s->last_buf_size);

  1393. 

  1394.         memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);

  1395.         s->in_gb = s->gb;

  1396.         init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);

  1397. #if !UNCHECKED_BITSTREAM_READER

  1398.         s->gb.size_in_bits_plus8 += extrasize * 8;

  1399. #endif

  1400.         s->last_buf_size <<= 3;

  1401.         for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {

  1402.             for (ch = 0; ch < s->nb_channels; ch++) {

  1403.                 g = &s->granules[ch][gr];

  1404.                 s->last_buf_size += g->part2_3_length;

  1405.                 memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));

  1406.                 compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);

  1407.             }

  1408.         }

  1409.         skip = s->last_buf_size - 8 * main_data_begin;

  1410.         if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {

  1411.             skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);

  1412.             s->gb           = s->in_gb;

  1413.             s->in_gb.buffer = NULL;

  1414.         } else {

  1415.             skip_bits_long(&s->gb, skip);

  1416.         }

  1417.     } else {

  1418.         gr = 0;

  1419.     }

  1420. 

  1421.     for (; gr < nb_granules; gr++) {

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

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

  1424.             bits_pos = get_bits_count(&s->gb);

  1425. 

  1426.             if (!s->lsf) {

  1427.                 uint8_t *sc;

  1428.                 int slen, slen1, slen2;

  1429. 

  1430.                 /* MPEG1 scale factors */

  1431.                 slen1 = slen_table[0][g->scalefac_compress];

  1432.                 slen2 = slen_table[1][g->scalefac_compress];

  1433.                 av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);

  1434.                 if (g->block_type == 2) {

  1435.                     n = g->switch_point ? 17 : 18;

  1436.                     j = 0;

  1437.                     if (slen1) {

  1438.                         for (i = 0; i < n; i++)

  1439.                             g->scale_factors[j++] = get_bits(&s->gb, slen1);

  1440.                     } else {

  1441.                         for (i = 0; i < n; i++)

  1442.                             g->scale_factors[j++] = 0;

  1443.                     }

  1444.                     if (slen2) {

  1445.                         for (i = 0; i < 18; i++)

  1446.                             g->scale_factors[j++] = get_bits(&s->gb, slen2);

  1447.                         for (i = 0; i < 3; i++)

  1448.                             g->scale_factors[j++] = 0;

  1449.                     } else {

  1450.                         for (i = 0; i < 21; i++)

  1451.                             g->scale_factors[j++] = 0;

  1452.                     }

  1453.                 } else {

  1454.                     sc = s->granules[ch][0].scale_factors;

  1455.                     j = 0;

  1456.                     for (k = 0; k < 4; k++) {

  1457.                         n = k == 0 ? 6 : 5;

  1458.                         if ((g->scfsi & (0x8 >> k)) == 0) {

  1459.                             slen = (k < 2) ? slen1 : slen2;

  1460.                             if (slen) {

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

  1462.                                     g->scale_factors[j++] = get_bits(&s->gb, slen);

  1463.                             } else {

  1464.                                 for (i = 0; i < n; i++)

  1465.                                     g->scale_factors[j++] = 0;

  1466.                             }

  1467.                         } else {

  1468.                             /* simply copy from last granule */

  1469.                             for (i = 0; i < n; i++) {

  1470.                                 g->scale_factors[j] = sc[j];

  1471.                                 j++;

  1472.                             }

  1473.                         }

  1474.                     }

  1475.                     g->scale_factors[j++] = 0;

  1476.                 }

  1477.             } else {

  1478.                 int tindex, tindex2, slen[4], sl, sf;

  1479. 

  1480.                 /* LSF scale factors */

  1481.                 if (g->block_type == 2)

  1482.                     tindex = g->switch_point ? 2 : 1;

  1483.                 else

  1484.                     tindex = 0;

  1485. 

  1486.                 sf = g->scalefac_compress;

  1487.                 if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {

  1488.                     /* intensity stereo case */

  1489.                     sf >>= 1;

  1490.                     if (sf < 180) {

  1491.                         lsf_sf_expand(slen, sf, 6, 6, 0);

  1492.                         tindex2 = 3;

  1493.                     } else if (sf < 244) {

  1494.                         lsf_sf_expand(slen, sf - 180, 4, 4, 0);

  1495.                         tindex2 = 4;

  1496.                     } else {

  1497.                         lsf_sf_expand(slen, sf - 244, 3, 0, 0);

  1498.                         tindex2 = 5;

  1499.                     }

  1500.                 } else {

  1501.                     /* normal case */

  1502.                     if (sf < 400) {

  1503.                         lsf_sf_expand(slen, sf, 5, 4, 4);

  1504.                         tindex2 = 0;

  1505.                     } else if (sf < 500) {

  1506.                         lsf_sf_expand(slen, sf - 400, 5, 4, 0);

  1507.                         tindex2 = 1;

  1508.                     } else {

  1509.                         lsf_sf_expand(slen, sf - 500, 3, 0, 0);

  1510.                         tindex2 = 2;

  1511.                         g->preflag = 1;

  1512.                     }

  1513.                 }

  1514. 

  1515.                 j = 0;

  1516.                 for (k = 0; k < 4; k++) {

  1517.                     n  = lsf_nsf_table[tindex2][tindex][k];

  1518.                     sl = slen[k];

  1519.                     if (sl) {

  1520.                         for (i = 0; i < n; i++)

  1521.                             g->scale_factors[j++] = get_bits(&s->gb, sl);

  1522.                     } else {

  1523.                         for (i = 0; i < n; i++)

  1524.                             g->scale_factors[j++] = 0;

  1525.                     }

  1526.                 }

  1527.                 /* XXX: should compute exact size */

  1528.                 for (; j < 40; j++)

  1529.                     g->scale_factors[j] = 0;

  1530.             }

  1531. 

  1532.             exponents_from_scale_factors(s, g, exponents);

  1533. 

  1534.             /* read Huffman coded residue */

  1535.             huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);

  1536.         } /* ch */

  1537. 

  1538.         if (s->mode == MPA_JSTEREO)

  1539.             compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);

  1540. 

  1541.         for (ch = 0; ch < s->nb_channels; ch++) {

  1542.             g = &s->granules[ch][gr];

  1543. 

  1544.             reorder_block(s, g);

  1545.             compute_antialias(s, g);

  1546.             compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);

  1547.         }

  1548.     } /* gr */

  1549.     if (get_bits_count(&s->gb) < 0)

  1550.         skip_bits_long(&s->gb, -get_bits_count(&s->gb));

  1551.     return nb_granules * 18;

  1552. }

  1553. 

  1554. static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,

  1555.                            const uint8_t *buf, int buf_size)

  1556. {

  1557.     int i, nb_frames, ch, ret;

  1558.     OUT_INT *samples_ptr;

  1559. 

  1560.     init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);

  1561. 

  1562.     /* skip error protection field */

  1563.     if (s->error_protection)

  1564.         skip_bits(&s->gb, 16);

  1565. 

  1566.     switch(s->layer) {

  1567.     case 1:

  1568.         s->avctx->frame_size = 384;

  1569.         nb_frames = mp_decode_layer1(s);

  1570.         break;

  1571.     case 2:

  1572.         s->avctx->frame_size = 1152;

  1573.         nb_frames = mp_decode_layer2(s);

  1574.         break;

  1575.     case 3:

  1576.         s->avctx->frame_size = s->lsf ? 576 : 1152;

  1577.     default:

  1578.         nb_frames = mp_decode_layer3(s);

  1579. 

  1580.         if (nb_frames < 0)

  1581.             return nb_frames;

  1582. 

  1583.         s->last_buf_size=0;

  1584.         if (s->in_gb.buffer) {

  1585.             align_get_bits(&s->gb);

  1586.             i = get_bits_left(&s->gb)>>3;

  1587.             if (i >= 0 && i <= BACKSTEP_SIZE) {

  1588.                 memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);

  1589.                 s->last_buf_size=i;

  1590.             } else

  1591.                 av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);

  1592.             s->gb           = s->in_gb;

  1593.             s->in_gb.buffer = NULL;

  1594.         }

  1595. 

  1596.         align_get_bits(&s->gb);

  1597.         assert((get_bits_count(&s->gb) & 7) == 0);

  1598.         i = get_bits_left(&s->gb) >> 3;

  1599. 

  1600.         if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {

  1601.             if (i < 0)

  1602.                 av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);

  1603.             i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);

  1604.         }

  1605.         assert(i <= buf_size - HEADER_SIZE && i >= 0);

  1606.         memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);

  1607.         s->last_buf_size += i;

  1608.     }

  1609. 

  1610.     /* get output buffer */

  1611.     if (!samples) {

  1612.         av_assert0(s->frame != NULL);

  1613.         s->frame->nb_samples = s->avctx->frame_size;

  1614.         if ((ret = ff_get_buffer(s->avctx, s->frame, 0)) < 0) {

  1615.             av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1616.             return ret;

  1617.         }

  1618.         samples = (OUT_INT **)s->frame->extended_data;

  1619.     }

  1620. 

  1621.     /* apply the synthesis filter */

  1622.     for (ch = 0; ch < s->nb_channels; ch++) {

  1623.         int sample_stride;

  1624.         if (s->avctx->sample_fmt == OUT_FMT_P) {

  1625.             samples_ptr   = samples[ch];

  1626.             sample_stride = 1;

  1627.         } else {

  1628.             samples_ptr   = samples[0] + ch;

  1629.             sample_stride = s->nb_channels;

  1630.         }

  1631.         for (i = 0; i < nb_frames; i++) {

  1632.             RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],

  1633.                                         &(s->synth_buf_offset[ch]),

  1634.                                         RENAME(ff_mpa_synth_window),

  1635.                                         &s->dither_state, samples_ptr,

  1636.                                         sample_stride, s->sb_samples[ch][i]);

  1637.             samples_ptr += 32 * sample_stride;

  1638.         }

  1639.     }

  1640. 

  1641.     return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;

  1642. }

  1643. 

  1644. static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,

  1645.                         AVPacket *avpkt)

  1646. {

  1647.     const uint8_t *buf  = avpkt->data;

  1648.     int buf_size        = avpkt->size;

  1649.     MPADecodeContext *s = avctx->priv_data;

  1650.     uint32_t header;

  1651.     int ret;

  1652. 

  1653.     if (buf_size < HEADER_SIZE)

  1654.         return AVERROR_INVALIDDATA;

  1655. 

  1656.     header = AV_RB32(buf);

  1657.     if (ff_mpa_check_header(header) < 0) {

  1658.         av_log(avctx, AV_LOG_ERROR, "Header missing\n");

  1659.         return AVERROR_INVALIDDATA;

  1660.     }

  1661. 

  1662.     if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {

  1663.         /* free format: prepare to compute frame size */

  1664.         s->frame_size = -1;

  1665.         return AVERROR_INVALIDDATA;

  1666.     }

  1667.     /* update codec info */

  1668.     avctx->channels       = s->nb_channels;

  1669.     avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;

  1670.     if (!avctx->bit_rate)

  1671.         avctx->bit_rate = s->bit_rate;

  1672. 

  1673.     if (s->frame_size <= 0 || s->frame_size > buf_size) {

  1674.         av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");

  1675.         return AVERROR_INVALIDDATA;

  1676.     } else if (s->frame_size < buf_size) {

  1677.         buf_size= s->frame_size;

  1678.     }

  1679. 

  1680.     s->frame = data;

  1681. 

  1682.     ret = mp_decode_frame(s, NULL, buf, buf_size);

  1683.     if (ret >= 0) {

  1684.         s->frame->nb_samples = avctx->frame_size;

  1685.         *got_frame_ptr       = 1;

  1686.         avctx->sample_rate   = s->sample_rate;

  1687.         //FIXME maybe move the other codec info stuff from above here too

  1688.     } else {

  1689.         av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");

  1690.         /* Only return an error if the bad frame makes up the whole packet or

  1691.          * the error is related to buffer management.

  1692.          * If there is more data in the packet, just consume the bad frame

  1693.          * instead of returning an error, which would discard the whole

  1694.          * packet. */

  1695.         *got_frame_ptr = 0;

  1696.         if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)

  1697.             return ret;

  1698.     }

  1699.     s->frame_size = 0;

  1700.     return buf_size;

  1701. }

  1702. 

  1703. static void mp_flush(MPADecodeContext *ctx)

  1704. {

  1705.     memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));

  1706.     ctx->last_buf_size = 0;

  1707. }

  1708. 

  1709. static void flush(AVCodecContext *avctx)

  1710. {

  1711.     mp_flush(avctx->priv_data);

  1712. }

  1713. 

  1714. #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER

  1715. static int decode_frame_adu(AVCodecContext *avctx, void *data,

  1716.                             int *got_frame_ptr, AVPacket *avpkt)

  1717. {

  1718.     const uint8_t *buf  = avpkt->data;

  1719.     int buf_size        = avpkt->size;

  1720.     MPADecodeContext *s = avctx->priv_data;

  1721.     uint32_t header;

  1722.     int len, ret;

  1723. 

  1724.     len = buf_size;

  1725. 

  1726.     // Discard too short frames

  1727.     if (buf_size < HEADER_SIZE) {

  1728.         av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");

  1729.         return AVERROR_INVALIDDATA;

  1730.     }

  1731. 

  1732. 

  1733.     if (len > MPA_MAX_CODED_FRAME_SIZE)

  1734.         len = MPA_MAX_CODED_FRAME_SIZE;

  1735. 

  1736.     // Get header and restore sync word

  1737.     header = AV_RB32(buf) | 0xffe00000;

  1738. 

  1739.     if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame

  1740.         av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");

  1741.         return AVERROR_INVALIDDATA;

  1742.     }

  1743. 

  1744.     avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);

  1745.     /* update codec info */

  1746.     avctx->sample_rate = s->sample_rate;

  1747.     avctx->channels    = s->nb_channels;

  1748.     if (!avctx->bit_rate)

  1749.         avctx->bit_rate = s->bit_rate;

  1750. 

  1751.     s->frame_size = len;

  1752. 

  1753.     s->frame = data;

  1754. 

  1755.     ret = mp_decode_frame(s, NULL, buf, buf_size);

  1756.     if (ret < 0) {

  1757.         av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");

  1758.         return ret;

  1759.     }

  1760. 

  1761.     *got_frame_ptr = 1;

  1762. 

  1763.     return buf_size;

  1764. }

  1765. #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */

  1766. 

  1767. #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER

  1768. 

  1769. /**

  1770.  * Context for MP3On4 decoder

  1771.  */

  1772. typedef struct MP3On4DecodeContext {

  1773.     int frames;                     ///< number of mp3 frames per block (number of mp3 decoder instances)

  1774.     int syncword;                   ///< syncword patch

  1775.     const uint8_t *coff;            ///< channel offsets in output buffer

  1776.     MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance

  1777. } MP3On4DecodeContext;

  1778. 

  1779. #include "mpeg4audio.h"

  1780. 

  1781. /* Next 3 arrays are indexed by channel config number (passed via codecdata) */

  1782. 

  1783. /* number of mp3 decoder instances */

  1784. static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };

  1785. 

  1786. /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */

  1787. static const uint8_t chan_offset[8][5] = {

  1788.     { 0             },

  1789.     { 0             },  // C

  1790.     { 0             },  // FLR

  1791.     { 2, 0          },  // C FLR

  1792.     { 2, 0, 3       },  // C FLR BS

  1793.     { 2, 0, 3       },  // C FLR BLRS

  1794.     { 2, 0, 4, 3    },  // C FLR BLRS LFE

  1795.     { 2, 0, 6, 4, 3 },  // C FLR BLRS BLR LFE

  1796. };

  1797. 

  1798. /* mp3on4 channel layouts */

  1799. static const int16_t chan_layout[8] = {

  1800.     0,

  1801.     AV_CH_LAYOUT_MONO,

  1802.     AV_CH_LAYOUT_STEREO,

  1803.     AV_CH_LAYOUT_SURROUND,

  1804.     AV_CH_LAYOUT_4POINT0,

  1805.     AV_CH_LAYOUT_5POINT0,

  1806.     AV_CH_LAYOUT_5POINT1,

  1807.     AV_CH_LAYOUT_7POINT1

  1808. };

  1809. 

  1810. static av_cold int decode_close_mp3on4(AVCodecContext * avctx)

  1811. {

  1812.     MP3On4DecodeContext *s = avctx->priv_data;

  1813.     int i;

  1814. 

  1815.     for (i = 0; i < s->frames; i++)

  1816.         av_free(s->mp3decctx[i]);

  1817. 

  1818.     return 0;

  1819. }

  1820. 

  1821. 

  1822. static int decode_init_mp3on4(AVCodecContext * avctx)

  1823. {

  1824.     MP3On4DecodeContext *s = avctx->priv_data;

  1825.     MPEG4AudioConfig cfg;

  1826.     int i;

  1827. 

  1828.     if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {

  1829.         av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");

  1830.         return AVERROR_INVALIDDATA;

  1831.     }

  1832. 

  1833.     avpriv_mpeg4audio_get_config(&cfg, avctx->extradata,

  1834.                                  avctx->extradata_size * 8, 1);

  1835.     if (!cfg.chan_config || cfg.chan_config > 7) {

  1836.         av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");

  1837.         return AVERROR_INVALIDDATA;

  1838.     }

  1839.     s->frames             = mp3Frames[cfg.chan_config];

  1840.     s->coff               = chan_offset[cfg.chan_config];

  1841.     avctx->channels       = ff_mpeg4audio_channels[cfg.chan_config];

  1842.     avctx->channel_layout = chan_layout[cfg.chan_config];

  1843. 

  1844.     if (cfg.sample_rate < 16000)

  1845.         s->syncword = 0xffe00000;

  1846.     else

  1847.         s->syncword = 0xfff00000;

  1848. 

  1849.     /* Init the first mp3 decoder in standard way, so that all tables get builded

  1850.      * We replace avctx->priv_data with the context of the first decoder so that

  1851.      * decode_init() does not have to be changed.

  1852.      * Other decoders will be initialized here copying data from the first context

  1853.      */

  1854.     // Allocate zeroed memory for the first decoder context

  1855.     s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));

  1856.     if (!s->mp3decctx[0])

  1857.         goto alloc_fail;

  1858.     // Put decoder context in place to make init_decode() happy

  1859.     avctx->priv_data = s->mp3decctx[0];

  1860.     decode_init(avctx);

  1861.     // Restore mp3on4 context pointer

  1862.     avctx->priv_data = s;

  1863.     s->mp3decctx[0]->adu_mode = 1; // Set adu mode

  1864. 

  1865.     /* Create a separate codec/context for each frame (first is already ok).

  1866.      * Each frame is 1 or 2 channels - up to 5 frames allowed

  1867.      */

  1868.     for (i = 1; i < s->frames; i++) {

  1869.         s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));

  1870.         if (!s->mp3decctx[i])

  1871.             goto alloc_fail;

  1872.         s->mp3decctx[i]->adu_mode = 1;

  1873.         s->mp3decctx[i]->avctx = avctx;

  1874.         s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;

  1875.     }

  1876. 

  1877.     return 0;

  1878. alloc_fail:

  1879.     decode_close_mp3on4(avctx);

  1880.     return AVERROR(ENOMEM);

  1881. }

  1882. 

  1883. 

  1884. static void flush_mp3on4(AVCodecContext *avctx)

  1885. {

  1886.     int i;

  1887.     MP3On4DecodeContext *s = avctx->priv_data;

  1888. 

  1889.     for (i = 0; i < s->frames; i++)

  1890.         mp_flush(s->mp3decctx[i]);

  1891. }

  1892. 

  1893. 

  1894. static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,

  1895.                                int *got_frame_ptr, AVPacket *avpkt)

  1896. {

  1897.     AVFrame *frame         = data;

  1898.     const uint8_t *buf     = avpkt->data;

  1899.     int buf_size           = avpkt->size;

  1900.     MP3On4DecodeContext *s = avctx->priv_data;

  1901.     MPADecodeContext *m;

  1902.     int fsize, len = buf_size, out_size = 0;

  1903.     uint32_t header;

  1904.     OUT_INT **out_samples;

  1905.     OUT_INT *outptr[2];

  1906.     int fr, ch, ret;

  1907. 

  1908.     /* get output buffer */

  1909.     frame->nb_samples = MPA_FRAME_SIZE;

  1910.     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {

  1911.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1912.         return ret;

  1913.     }

  1914.     out_samples = (OUT_INT **)frame->extended_data;

  1915. 

  1916.     // Discard too short frames

  1917.     if (buf_size < HEADER_SIZE)

  1918.         return AVERROR_INVALIDDATA;

  1919. 

  1920.     avctx->bit_rate = 0;

  1921. 

  1922.     ch = 0;

  1923.     for (fr = 0; fr < s->frames; fr++) {

  1924.         fsize = AV_RB16(buf) >> 4;

  1925.         fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);

  1926.         m     = s->mp3decctx[fr];

  1927.         assert(m != NULL);

  1928. 

  1929.         if (fsize < HEADER_SIZE) {

  1930.             av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");

  1931.             return AVERROR_INVALIDDATA;

  1932.         }

  1933.         header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header

  1934. 

  1935.         if (ff_mpa_check_header(header) < 0) // Bad header, discard block

  1936.             break;

  1937. 

  1938.         avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);

  1939. 

  1940.         if (ch + m->nb_channels > avctx->channels) {

  1941.             av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "

  1942.                                         "channel count\n");

  1943.             return AVERROR_INVALIDDATA;

  1944.         }

  1945.         ch += m->nb_channels;

  1946. 

  1947.         outptr[0] = out_samples[s->coff[fr]];

  1948.         if (m->nb_channels > 1)

  1949.             outptr[1] = out_samples[s->coff[fr] + 1];

  1950. 

  1951.         if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)

  1952.             return ret;

  1953. 

  1954.         out_size += ret;

  1955.         buf      += fsize;

  1956.         len      -= fsize;

  1957. 

  1958.         avctx->bit_rate += m->bit_rate;

  1959.     }

  1960. 

  1961.     /* update codec info */

  1962.     avctx->sample_rate = s->mp3decctx[0]->sample_rate;

  1963. 

  1964.     frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));

  1965.     *got_frame_ptr    = 1;

  1966. 

  1967.     return buf_size;

  1968. }

  1969. #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */

  1970. 

  1971. #if !CONFIG_FLOAT

  1972. #if CONFIG_MP1_DECODER

  1973. AVCodec ff_mp1_decoder = {

  1974.     .name           = "mp1",

  1975.     .type           = AVMEDIA_TYPE_AUDIO,

  1976.     .id             = AV_CODEC_ID_MP1,

  1977.     .priv_data_size = sizeof(MPADecodeContext),

  1978.     .init           = decode_init,

  1979.     .decode         = decode_frame,

  1980.     .capabilities   = CODEC_CAP_DR1,

  1981.     .flush          = flush,

  1982.     .long_name      = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),

  1983.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  1984.                                                       AV_SAMPLE_FMT_S16,

  1985.                                                       AV_SAMPLE_FMT_NONE },

  1986. };

  1987. #endif

  1988. #if CONFIG_MP2_DECODER

  1989. AVCodec ff_mp2_decoder = {

  1990.     .name           = "mp2",

  1991.     .type           = AVMEDIA_TYPE_AUDIO,

  1992.     .id             = AV_CODEC_ID_MP2,

  1993.     .priv_data_size = sizeof(MPADecodeContext),

  1994.     .init           = decode_init,

  1995.     .decode         = decode_frame,

  1996.     .capabilities   = CODEC_CAP_DR1,

  1997.     .flush          = flush,

  1998.     .long_name      = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),

  1999.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2000.                                                       AV_SAMPLE_FMT_S16,

  2001.                                                       AV_SAMPLE_FMT_NONE },

  2002. };

  2003. #endif

  2004. #if CONFIG_MP3_DECODER

  2005. AVCodec ff_mp3_decoder = {

  2006.     .name           = "mp3",

  2007.     .type           = AVMEDIA_TYPE_AUDIO,

  2008.     .id             = AV_CODEC_ID_MP3,

  2009.     .priv_data_size = sizeof(MPADecodeContext),

  2010.     .init           = decode_init,

  2011.     .decode         = decode_frame,

  2012.     .capabilities   = CODEC_CAP_DR1,

  2013.     .flush          = flush,

  2014.     .long_name      = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),

  2015.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2016.                                                       AV_SAMPLE_FMT_S16,

  2017.                                                       AV_SAMPLE_FMT_NONE },

  2018. };

  2019. #endif

  2020. #if CONFIG_MP3ADU_DECODER

  2021. AVCodec ff_mp3adu_decoder = {

  2022.     .name           = "mp3adu",

  2023.     .type           = AVMEDIA_TYPE_AUDIO,

  2024.     .id             = AV_CODEC_ID_MP3ADU,

  2025.     .priv_data_size = sizeof(MPADecodeContext),

  2026.     .init           = decode_init,

  2027.     .decode         = decode_frame_adu,

  2028.     .capabilities   = CODEC_CAP_DR1,

  2029.     .flush          = flush,

  2030.     .long_name      = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),

  2031.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2032.                                                       AV_SAMPLE_FMT_S16,

  2033.                                                       AV_SAMPLE_FMT_NONE },

  2034. };

  2035. #endif

  2036. #if CONFIG_MP3ON4_DECODER

  2037. AVCodec ff_mp3on4_decoder = {

  2038.     .name           = "mp3on4",

  2039.     .type           = AVMEDIA_TYPE_AUDIO,

  2040.     .id             = AV_CODEC_ID_MP3ON4,

  2041.     .priv_data_size = sizeof(MP3On4DecodeContext),

  2042.     .init           = decode_init_mp3on4,

  2043.     .close          = decode_close_mp3on4,

  2044.     .decode         = decode_frame_mp3on4,

  2045.     .capabilities   = CODEC_CAP_DR1,

  2046.     .flush          = flush_mp3on4,

  2047.     .long_name      = NULL_IF_CONFIG_SMALL("MP3onMP4"),

  2048.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2049.                                                       AV_SAMPLE_FMT_NONE },

  2050. };

  2051. #endif

  2052. #endif