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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 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 "mathops.h"

  33. #include "mpegaudiodsp.h"

  34. #include "dsputil.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.     DSPContext dsp;

  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(s->sample_rate_index == 8)

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

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

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

  214.                 exponents as long blocks */

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

  216.                 g->long_end = 8;

  217.             else

  218.                 g->long_end = 6;

  219. 

  220.             g->short_start = 3;

  221.         } else {

  222.             g->long_end    = 0;

  223.             g->short_start = 0;

  224.         }

  225.     } else {

  226.         g->short_start = 13;

  227.         g->long_end    = 22;

  228.     }

  229. }

  230. 

  231. /* layer 1 unscaling */

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

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

  234. {

  235.     int shift, mod;

  236.     int64_t val;

  237. 

  238.     shift   = scale_factor_modshift[scale_factor];

  239.     mod     = shift & 3;

  240.     shift >>= 2;

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

  242.     shift  += n;

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

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

  245. }

  246. 

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

  248. {

  249.     int shift, mod, val;

  250. 

  251.     shift   = scale_factor_modshift[scale_factor];

  252.     mod     = shift & 3;

  253.     shift >>= 2;

  254. 

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

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

  257.     if (shift > 0)

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

  259.     return val;

  260. }

  261. 

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

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

  264. {

  265.     unsigned int m;

  266.     int e;

  267. 

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

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

  270.     e -= exponent >> 2;

  271. #ifdef DEBUG

  272.     if(e < 1)

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

  274. #endif

  275.     if (e > 31)

  276.         return 0;

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

  278. 

  279.     return m;

  280. }

  281. 

  282. static av_cold void decode_init_static(void)

  283. {

  284.     int i, j, k;

  285.     int offset;

  286. 

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

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

  289.         int shift, mod;

  290.         /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */

  291.         shift = i / 3;

  292.         mod   = i % 3;

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

  294.     }

  295. 

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

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

  298.         int n, norm;

  299.         n = i + 2;

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

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

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

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

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

  305.                 scale_factor_mult[i][0],

  306.                 scale_factor_mult[i][1],

  307.                 scale_factor_mult[i][2]);

  308.     }

  309. 

  310.     RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));

  311. 

  312.     /* huffman decode tables */

  313.     offset = 0;

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

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

  316.         int xsize, x, y;

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

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

  319. 

  320.         xsize = h->xsize;

  321. 

  322.         j = 0;

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

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

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

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

  327.             }

  328.         }

  329. 

  330.         /* XXX: fail test */

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

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

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

  334.                  tmp_bits, 1, 1, tmp_codes, 2, 2,

  335.                  INIT_VLC_USE_NEW_STATIC);

  336.         offset += huff_vlc_tables_sizes[i];

  337.     }

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

  339. 

  340.     offset = 0;

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

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

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

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

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

  346.                  INIT_VLC_USE_NEW_STATIC);

  347.         offset += huff_quad_vlc_tables_sizes[i];

  348.     }

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

  350. 

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

  352.         k = 0;

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

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

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

  356.         }

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

  358.     }

  359. 

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

  361. 

  362.     mpegaudio_tableinit();

  363. 

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

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

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

  367.                 int val1, val2, val3, steps;

  368.                 int val = j;

  369.                 steps   = ff_mpa_quant_steps[i];

  370.                 val1    = val % steps;

  371.                 val    /= steps;

  372.                 val2    = val % steps;

  373.                 val3    = val / steps;

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

  375.             }

  376.         }

  377.     }

  378. 

  379. 

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

  381.         float f;

  382.         INTFLOAT v;

  383.         if (i != 6) {

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

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

  386.         } else {

  387.             v = FIXR(1.0);

  388.         }

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

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

  391.     }

  392.     /* invalid values */

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

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

  395. 

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

  397.         double f;

  398.         int e, k;

  399. 

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

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

  402.             f = exp2(e / 4.0);

  403.             k = i & 1;

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

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

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

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

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

  409.         }

  410.     }

  411. 

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

  413.         float ci, cs, ca;

  414.         ci = ci_table[i];

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

  416.         ca = cs * ci;

  417. #if !CONFIG_FLOAT

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

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

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

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

  422. #else

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

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

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

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

  427. #endif

  428.     }

  429. }

  430. 

  431. static av_cold int decode_init(AVCodecContext * avctx)

  432. {

  433.     static int initialized_tables = 0;

  434.     MPADecodeContext *s = avctx->priv_data;

  435. 

  436.     if (!initialized_tables) {

  437.         decode_init_static();

  438.         initialized_tables = 1;

  439.     }

  440. 

  441.     s->avctx = avctx;

  442. 

  443.     ff_mpadsp_init(&s->mpadsp);

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

  445. 

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

  447.         avctx->codec_id != AV_CODEC_ID_MP3ON4)

  448.         avctx->sample_fmt = OUT_FMT;

  449.     else

  450.         avctx->sample_fmt = OUT_FMT_P;

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

  452. 

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

  454.         s->adu_mode = 1;

  455. 

  456.     avcodec_get_frame_defaults(&s->frame);

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

  458. 

  459.     return 0;

  460. }

  461. 

  462. #define C3 FIXHR(0.86602540378443864676/2)

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

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

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

  466. 

  467. /* 12 points IMDCT. We compute it "by hand" by factorizing obvious

  468.    cases. */

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

  470. {

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

  472. 

  473.     in0  = in[0*3];

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

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

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

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

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

  479.     in5 += in3;

  480.     in3 += in1;

  481. 

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

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

  484. 

  485.     t1   = in0 - in4;

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

  487. 

  488.     out[ 7] =

  489.     out[10] = t1 + t2;

  490.     out[ 1] =

  491.     out[ 4] = t1 - t2;

  492. 

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

  494.     in4     = in0 + in2;

  495.     in5    += 2*in1;

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

  497.     out[ 8] =

  498.     out[ 9] = in4 + in1;

  499.     out[ 2] =

  500.     out[ 3] = in4 - in1;

  501. 

  502.     in0    -= in2;

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

  504.     out[ 0] =

  505.     out[ 5] = in0 - in5;

  506.     out[ 6] =

  507.     out[11] = in0 + in5;

  508. }

  509. 

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

  511. static int mp_decode_layer1(MPADecodeContext *s)

  512. {

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

  514.     uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];

  515.     uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];

  516. 

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

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

  519.     else

  520.         bound = SBLIMIT;

  521. 

  522.     /* allocation bits */

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

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

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

  526.         }

  527.     }

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

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

  530. 

  531.     /* scale factors */

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

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

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

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

  536.         }

  537.     }

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

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

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

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

  542.         }

  543.     }

  544. 

  545.     /* compute samples */

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

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

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

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

  550.                 if (n) {

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

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

  553.                 } else {

  554.                     v = 0;

  555.                 }

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

  557.             }

  558.         }

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

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

  561.             if (n) {

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

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

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

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

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

  567.             } else {

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

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

  570.             }

  571.         }

  572.     }

  573.     return 12;

  574. }

  575. 

  576. static int mp_decode_layer2(MPADecodeContext *s)

  577. {

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

  579.     const unsigned char *alloc_table;

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

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

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

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

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

  585. 

  586.     /* select decoding table */

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

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

  589.     sblimit     = ff_mpa_sblimit_table[table];

  590.     alloc_table = ff_mpa_alloc_tables[table];

  591. 

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

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

  594.     else

  595.         bound = sblimit;

  596. 

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

  598. 

  599.     /* sanity check */

  600.     if (bound > sblimit)

  601.         bound = sblimit;

  602. 

  603.     /* parse bit allocation */

  604.     j = 0;

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

  606.         bit_alloc_bits = alloc_table[j];

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

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

  609.         j += 1 << bit_alloc_bits;

  610.     }

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

  612.         bit_alloc_bits = alloc_table[j];

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

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

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

  616.         j += 1 << bit_alloc_bits;

  617.     }

  618. 

  619.     /* scale codes */

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

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

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

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

  624.         }

  625.     }

  626. 

  627.     /* scale factors */

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

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

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

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

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

  633.                 default:

  634.                 case 0:

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

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

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

  638.                     break;

  639.                 case 2:

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

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

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

  643.                     break;

  644.                 case 1:

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

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

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

  648.                     break;

  649.                 case 3:

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

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

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

  653.                     break;

  654.                 }

  655.             }

  656.         }

  657.     }

  658. 

  659.     /* samples */

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

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

  662.             j = 0;

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

  664.                 bit_alloc_bits = alloc_table[j];

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

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

  667.                     if (b) {

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

  669.                         qindex = alloc_table[j+b];

  670.                         bits = ff_mpa_quant_bits[qindex];

  671.                         if (bits < 0) {

  672.                             int v2;

  673.                             /* 3 values at the same time */

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

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

  676.                             steps  = ff_mpa_quant_steps[qindex];

  677. 

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

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

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

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

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

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

  684.                         } else {

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

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

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

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

  689.                             }

  690.                         }

  691.                     } else {

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

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

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

  695.                     }

  696.                 }

  697.                 /* next subband in alloc table */

  698.                 j += 1 << bit_alloc_bits;

  699.             }

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

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

  702.                 bit_alloc_bits = alloc_table[j];

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

  704.                 if (b) {

  705.                     int mant, scale0, scale1;

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

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

  708.                     qindex = alloc_table[j+b];

  709.                     bits = ff_mpa_quant_bits[qindex];

  710.                     if (bits < 0) {

  711.                         /* 3 values at the same time */

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

  713.                         steps = ff_mpa_quant_steps[qindex];

  714.                         mant = v % steps;

  715.                         v = v / steps;

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

  717.                             l2_unscale_group(steps, mant, scale0);

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

  719.                             l2_unscale_group(steps, mant, scale1);

  720.                         mant = v % steps;

  721.                         v = v / steps;

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

  723.                             l2_unscale_group(steps, mant, scale0);

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

  725.                             l2_unscale_group(steps, mant, scale1);

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

  727.                             l2_unscale_group(steps, v, scale0);

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

  729.                             l2_unscale_group(steps, v, scale1);

  730.                     } else {

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

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

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

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

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

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

  737.                         }

  738.                     }

  739.                 } else {

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

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

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

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

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

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

  746.                 }

  747.                 /* next subband in alloc table */

  748.                 j += 1 << bit_alloc_bits;

  749.             }

  750.             /* fill remaining samples to zero */

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

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

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

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

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

  756.                 }

  757.             }

  758.         }

  759.     }

  760.     return 3 * 12;

  761. }

  762. 

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

  764.     if (n == 3) {                   \

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

  766.         dst   = sf - 3 * m;         \

  767.         sf    = m;                  \

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

  769.         dst  = sf & 3;              \

  770.         sf >>= 2;                   \

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

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

  773.         dst   = sf - 5 * m;         \

  774.         sf    = m;                  \

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

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

  777.         dst   = sf - 6 * m;         \

  778.         sf    = m;                  \

  779.     } else {                        \

  780.         dst = 0;                    \

  781.     }

  782. 

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

  784.                                            int n3)

  785. {

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

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

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

  789.     slen[0] = sf;

  790. }

  791. 

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

  793.                                          int16_t *exponents)

  794. {

  795.     const uint8_t *bstab, *pretab;

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

  797.     int16_t *exp_ptr;

  798. 

  799.     exp_ptr = exponents;

  800.     gain    = g->global_gain - 210;

  801.     shift   = g->scalefac_scale + 1;

  802. 

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

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

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

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

  807.         len = bstab[i];

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

  809.             *exp_ptr++ = v0;

  810.     }

  811. 

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

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

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

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

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

  817.         k        = g->long_end;

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

  819.             len = bstab[i];

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

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

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

  823.                     *exp_ptr++ = v0;

  824.             }

  825.         }

  826.     }

  827. }

  828. 

  829. /* handle n = 0 too */

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

  831. {

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

  833. }

  834. 

  835. 

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

  837.                           int *end_pos2)

  838. {

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

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

  841.         s->in_gb.buffer = NULL;

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

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

  844.         *end_pos2 =

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

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

  847.     }

  848. }

  849. 

  850. /* Following is a optimized code for

  851.             INTFLOAT v = *src

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

  853.                 v = -v;

  854.             *dst = v;

  855. */

  856. #if CONFIG_FLOAT

  857. #define READ_FLIP_SIGN(dst,src)                     \

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

  859.     AV_WN32A(dst, v);

  860. #else

  861. #define READ_FLIP_SIGN(dst,src)     \

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

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

  864. #endif

  865. 

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

  867.                           int16_t *exponents, int end_pos2)

  868. {

  869.     int s_index;

  870.     int i;

  871.     int last_pos, bits_left;

  872.     VLC *vlc;

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

  874. 

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

  876.     s_index = 0;

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

  878.         int j, k, l, linbits;

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

  880.         if (j == 0)

  881.             continue;

  882.         /* select vlc table */

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

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

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

  886.         vlc     = &huff_vlc[l];

  887. 

  888.         if (!l) {

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

  890.             s_index += 2 * j;

  891.             continue;

  892.         }

  893. 

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

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

  896.             int exponent, x, y;

  897.             int v;

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

  899. 

  900.             if (pos >= end_pos){

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

  902.                 if (pos >= end_pos)

  903.                     break;

  904.             }

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

  906. 

  907.             if (!y) {

  908.                 g->sb_hybrid[s_index  ] =

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

  910.                 s_index += 2;

  911.                 continue;

  912.             }

  913. 

  914.             exponent= exponents[s_index];

  915. 

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

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

  918.             if (y & 16) {

  919.                 x = y >> 5;

  920.                 y = y & 0x0f;

  921.                 if (x < 15) {

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

  923.                 } else {

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

  925.                     v  = l3_unscale(x, exponent);

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

  927.                         v = -v;

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

  929.                 }

  930.                 if (y < 15) {

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

  932.                 } else {

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

  934.                     v  = l3_unscale(y, exponent);

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

  936.                         v = -v;

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

  938.                 }

  939.             } else {

  940.                 x = y >> 5;

  941.                 y = y & 0x0f;

  942.                 x += y;

  943.                 if (x < 15) {

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

  945.                 } else {

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

  947.                     v  = l3_unscale(x, exponent);

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

  949.                         v = -v;

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

  951.                 }

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

  953.             }

  954.             s_index += 2;

  955.         }

  956.     }

  957. 

  958.     /* high frequencies */

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

  960.     last_pos = 0;

  961.     while (s_index <= 572) {

  962.         int pos, code;

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

  964.         if (pos >= end_pos) {

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

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

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

  968.                 s_index -= 4;

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

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

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

  972.                     s_index=0;

  973.                 break;

  974.             }

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

  976.             if (pos >= end_pos)

  977.                 break;

  978.         }

  979.         last_pos = pos;

  980. 

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

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

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

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

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

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

  987.         while (code) {

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

  989.             int v;

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

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

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

  993.         }

  994.         s_index += 4;

  995.     }

  996.     /* skip extension bits */

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

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

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

  1000.         s_index=0;

  1001.     } else if (bits_left > 0 && (s->err_recognition & (AV_EF_BUFFER|AV_EF_AGGRESSIVE))) {

  1002.         av_log(s->avctx, AV_LOG_ERROR, "bits_left=%d\n", bits_left);

  1003.         s_index = 0;

  1004.     }

  1005.     memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * (576 - s_index));

  1006.     skip_bits_long(&s->gb, bits_left);

  1007. 

  1008.     i = get_bits_count(&s->gb);

  1009.     switch_buffer(s, &i, &end_pos, &end_pos2);

  1010. 

  1011.     return 0;

  1012. }

  1013. 

  1014. /* Reorder short blocks from bitstream order to interleaved order. It

  1015.    would be faster to do it in parsing, but the code would be far more

  1016.    complicated */

  1017. static void reorder_block(MPADecodeContext *s, GranuleDef *g)

  1018. {

  1019.     int i, j, len;

  1020.     INTFLOAT *ptr, *dst, *ptr1;

  1021.     INTFLOAT tmp[576];

  1022. 

  1023.     if (g->block_type != 2)

  1024.         return;

  1025. 

  1026.     if (g->switch_point) {

  1027.         if (s->sample_rate_index != 8)

  1028.             ptr = g->sb_hybrid + 36;

  1029.         else

  1030.             ptr = g->sb_hybrid + 72;

  1031.     } else {

  1032.         ptr = g->sb_hybrid;

  1033.     }

  1034. 

  1035.     for (i = g->short_start; i < 13; i++) {

  1036.         len  = band_size_short[s->sample_rate_index][i];

  1037.         ptr1 = ptr;

  1038.         dst  = tmp;

  1039.         for (j = len; j > 0; j--) {

  1040.             *dst++ = ptr[0*len];

  1041.             *dst++ = ptr[1*len];

  1042.             *dst++ = ptr[2*len];

  1043.             ptr++;

  1044.         }

  1045.         ptr += 2 * len;

  1046.         memcpy(ptr1, tmp, len * 3 * sizeof(*ptr1));

  1047.     }

  1048. }

  1049. 

  1050. #define ISQRT2 FIXR(0.70710678118654752440)

  1051. 

  1052. static void compute_stereo(MPADecodeContext *s, GranuleDef *g0, GranuleDef *g1)

  1053. {

  1054.     int i, j, k, l;

  1055.     int sf_max, sf, len, non_zero_found;

  1056.     INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;

  1057.     int non_zero_found_short[3];

  1058. 

  1059.     /* intensity stereo */

  1060.     if (s->mode_ext & MODE_EXT_I_STEREO) {

  1061.         if (!s->lsf) {

  1062.             is_tab = is_table;

  1063.             sf_max = 7;

  1064.         } else {

  1065.             is_tab = is_table_lsf[g1->scalefac_compress & 1];

  1066.             sf_max = 16;

  1067.         }

  1068. 

  1069.         tab0 = g0->sb_hybrid + 576;

  1070.         tab1 = g1->sb_hybrid + 576;

  1071. 

  1072.         non_zero_found_short[0] = 0;

  1073.         non_zero_found_short[1] = 0;

  1074.         non_zero_found_short[2] = 0;

  1075.         k = (13 - g1->short_start) * 3 + g1->long_end - 3;

  1076.         for (i = 12; i >= g1->short_start; i--) {

  1077.             /* for last band, use previous scale factor */

  1078.             if (i != 11)

  1079.                 k -= 3;

  1080.             len = band_size_short[s->sample_rate_index][i];

  1081.             for (l = 2; l >= 0; l--) {

  1082.                 tab0 -= len;

  1083.                 tab1 -= len;

  1084.                 if (!non_zero_found_short[l]) {

  1085.                     /* test if non zero band. if so, stop doing i-stereo */

  1086.                     for (j = 0; j < len; j++) {

  1087.                         if (tab1[j] != 0) {

  1088.                             non_zero_found_short[l] = 1;

  1089.                             goto found1;

  1090.                         }

  1091.                     }

  1092.                     sf = g1->scale_factors[k + l];

  1093.                     if (sf >= sf_max)

  1094.                         goto found1;

  1095. 

  1096.                     v1 = is_tab[0][sf];

  1097.                     v2 = is_tab[1][sf];

  1098.                     for (j = 0; j < len; j++) {

  1099.                         tmp0    = tab0[j];

  1100.                         tab0[j] = MULLx(tmp0, v1, FRAC_BITS);

  1101.                         tab1[j] = MULLx(tmp0, v2, FRAC_BITS);

  1102.                     }

  1103.                 } else {

  1104. found1:

  1105.                     if (s->mode_ext & MODE_EXT_MS_STEREO) {

  1106.                         /* lower part of the spectrum : do ms stereo

  1107.                            if enabled */

  1108.                         for (j = 0; j < len; j++) {

  1109.                             tmp0    = tab0[j];

  1110.                             tmp1    = tab1[j];

  1111.                             tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);

  1112.                             tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);

  1113.                         }

  1114.                     }

  1115.                 }

  1116.             }

  1117.         }

  1118. 

  1119.         non_zero_found = non_zero_found_short[0] |

  1120.                          non_zero_found_short[1] |

  1121.                          non_zero_found_short[2];

  1122. 

  1123.         for (i = g1->long_end - 1;i >= 0;i--) {

  1124.             len   = band_size_long[s->sample_rate_index][i];

  1125.             tab0 -= len;

  1126.             tab1 -= len;

  1127.             /* test if non zero band. if so, stop doing i-stereo */

  1128.             if (!non_zero_found) {

  1129.                 for (j = 0; j < len; j++) {

  1130.                     if (tab1[j] != 0) {

  1131.                         non_zero_found = 1;

  1132.                         goto found2;

  1133.                     }

  1134.                 }

  1135.                 /* for last band, use previous scale factor */

  1136.                 k  = (i == 21) ? 20 : i;

  1137.                 sf = g1->scale_factors[k];

  1138.                 if (sf >= sf_max)

  1139.                     goto found2;

  1140.                 v1 = is_tab[0][sf];

  1141.                 v2 = is_tab[1][sf];

  1142.                 for (j = 0; j < len; j++) {

  1143.                     tmp0    = tab0[j];

  1144.                     tab0[j] = MULLx(tmp0, v1, FRAC_BITS);

  1145.                     tab1[j] = MULLx(tmp0, v2, FRAC_BITS);

  1146.                 }

  1147.             } else {

  1148. found2:

  1149.                 if (s->mode_ext & MODE_EXT_MS_STEREO) {

  1150.                     /* lower part of the spectrum : do ms stereo

  1151.                        if enabled */

  1152.                     for (j = 0; j < len; j++) {

  1153.                         tmp0    = tab0[j];

  1154.                         tmp1    = tab1[j];

  1155.                         tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);

  1156.                         tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);

  1157.                     }

  1158.                 }

  1159.             }

  1160.         }

  1161.     } else if (s->mode_ext & MODE_EXT_MS_STEREO) {

  1162.         /* ms stereo ONLY */

  1163.         /* NOTE: the 1/sqrt(2) normalization factor is included in the

  1164.            global gain */

  1165. #if CONFIG_FLOAT

  1166.        s-> dsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);

  1167. #else

  1168.         tab0 = g0->sb_hybrid;

  1169.         tab1 = g1->sb_hybrid;

  1170.         for (i = 0; i < 576; i++) {

  1171.             tmp0    = tab0[i];

  1172.             tmp1    = tab1[i];

  1173.             tab0[i] = tmp0 + tmp1;

  1174.             tab1[i] = tmp0 - tmp1;

  1175.         }

  1176. #endif

  1177.     }

  1178. }

  1179. 

  1180. #if CONFIG_FLOAT

  1181. #if HAVE_MIPSFPU

  1182. #   include "mips/compute_antialias_float.h"

  1183. #endif /* HAVE_MIPSFPU */

  1184. #else

  1185. #if HAVE_MIPSDSPR1

  1186. #   include "mips/compute_antialias_fixed.h"

  1187. #endif /* HAVE_MIPSDSPR1 */

  1188. #endif /* CONFIG_FLOAT */

  1189. 

  1190. #ifndef compute_antialias

  1191. #if CONFIG_FLOAT

  1192. #define AA(j) do {                                                      \

  1193.         float tmp0 = ptr[-1-j];                                         \

  1194.         float tmp1 = ptr[   j];                                         \

  1195.         ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1];    \

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

  1197.     } while (0)

  1198. #else

  1199. #define AA(j) do {                                              \

  1200.         int tmp0 = ptr[-1-j];                                   \

  1201.         int tmp1 = ptr[   j];                                   \

  1202.         int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]);          \

  1203.         ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2]));   \

  1204.         ptr[   j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3]));   \

  1205.     } while (0)

  1206. #endif

  1207. 

  1208. static void compute_antialias(MPADecodeContext *s, GranuleDef *g)

  1209. {

  1210.     INTFLOAT *ptr;

  1211.     int n, i;

  1212. 

  1213.     /* we antialias only "long" bands */

  1214.     if (g->block_type == 2) {

  1215.         if (!g->switch_point)

  1216.             return;

  1217.         /* XXX: check this for 8000Hz case */

  1218.         n = 1;

  1219.     } else {

  1220.         n = SBLIMIT - 1;

  1221.     }

  1222. 

  1223.     ptr = g->sb_hybrid + 18;

  1224.     for (i = n; i > 0; i--) {

  1225.         AA(0);

  1226.         AA(1);

  1227.         AA(2);

  1228.         AA(3);

  1229.         AA(4);

  1230.         AA(5);

  1231.         AA(6);

  1232.         AA(7);

  1233. 

  1234.         ptr += 18;

  1235.     }

  1236. }

  1237. #endif /* compute_antialias */

  1238. 

  1239. static void compute_imdct(MPADecodeContext *s, GranuleDef *g,

  1240.                           INTFLOAT *sb_samples, INTFLOAT *mdct_buf)

  1241. {

  1242.     INTFLOAT *win, *out_ptr, *ptr, *buf, *ptr1;

  1243.     INTFLOAT out2[12];

  1244.     int i, j, mdct_long_end, sblimit;

  1245. 

  1246.     /* find last non zero block */

  1247.     ptr  = g->sb_hybrid + 576;

  1248.     ptr1 = g->sb_hybrid + 2 * 18;

  1249.     while (ptr >= ptr1) {

  1250.         int32_t *p;

  1251.         ptr -= 6;

  1252.         p    = (int32_t*)ptr;

  1253.         if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])

  1254.             break;

  1255.     }

  1256.     sblimit = ((ptr - g->sb_hybrid) / 18) + 1;

  1257. 

  1258.     if (g->block_type == 2) {

  1259.         /* XXX: check for 8000 Hz */

  1260.         if (g->switch_point)

  1261.             mdct_long_end = 2;

  1262.         else

  1263.             mdct_long_end = 0;

  1264.     } else {

  1265.         mdct_long_end = sblimit;

  1266.     }

  1267. 

  1268.     s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,

  1269.                                      mdct_long_end, g->switch_point,

  1270.                                      g->block_type);

  1271. 

  1272.     buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);

  1273.     ptr = g->sb_hybrid + 18 * mdct_long_end;

  1274. 

  1275.     for (j = mdct_long_end; j < sblimit; j++) {

  1276.         /* select frequency inversion */

  1277.         win     = RENAME(ff_mdct_win)[2 + (4  & -(j & 1))];

  1278.         out_ptr = sb_samples + j;

  1279. 

  1280.         for (i = 0; i < 6; i++) {

  1281.             *out_ptr = buf[4*i];

  1282.             out_ptr += SBLIMIT;

  1283.         }

  1284.         imdct12(out2, ptr + 0);

  1285.         for (i = 0; i < 6; i++) {

  1286.             *out_ptr     = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*1)];

  1287.             buf[4*(i + 6*2)] = MULH3(out2[i + 6], win[i + 6], 1);

  1288.             out_ptr += SBLIMIT;

  1289.         }

  1290.         imdct12(out2, ptr + 1);

  1291.         for (i = 0; i < 6; i++) {

  1292.             *out_ptr     = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*2)];

  1293.             buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);

  1294.             out_ptr += SBLIMIT;

  1295.         }

  1296.         imdct12(out2, ptr + 2);

  1297.         for (i = 0; i < 6; i++) {

  1298.             buf[4*(i + 6*0)] = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*0)];

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

  1300.             buf[4*(i + 6*2)] = 0;

  1301.         }

  1302.         ptr += 18;

  1303.         buf += (j&3) != 3 ? 1 : (4*18-3);

  1304.     }

  1305.     /* zero bands */

  1306.     for (j = sblimit; j < SBLIMIT; j++) {

  1307.         /* overlap */

  1308.         out_ptr = sb_samples + j;

  1309.         for (i = 0; i < 18; i++) {

  1310.             *out_ptr = buf[4*i];

  1311.             buf[4*i]   = 0;

  1312.             out_ptr += SBLIMIT;

  1313.         }

  1314.         buf += (j&3) != 3 ? 1 : (4*18-3);

  1315.     }

  1316. }

  1317. 

  1318. /* main layer3 decoding function */

  1319. static int mp_decode_layer3(MPADecodeContext *s)

  1320. {

  1321.     int nb_granules, main_data_begin;

  1322.     int gr, ch, blocksplit_flag, i, j, k, n, bits_pos;

  1323.     GranuleDef *g;

  1324.     int16_t exponents[576]; //FIXME try INTFLOAT

  1325. 

  1326.     /* read side info */

  1327.     if (s->lsf) {

  1328.         main_data_begin = get_bits(&s->gb, 8);

  1329.         skip_bits(&s->gb, s->nb_channels);

  1330.         nb_granules = 1;

  1331.     } else {

  1332.         main_data_begin = get_bits(&s->gb, 9);

  1333.         if (s->nb_channels == 2)

  1334.             skip_bits(&s->gb, 3);

  1335.         else

  1336.             skip_bits(&s->gb, 5);

  1337.         nb_granules = 2;

  1338.         for (ch = 0; ch < s->nb_channels; ch++) {

  1339.             s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */

  1340.             s->granules[ch][1].scfsi = get_bits(&s->gb, 4);

  1341.         }

  1342.     }

  1343. 

  1344.     for (gr = 0; gr < nb_granules; gr++) {

  1345.         for (ch = 0; ch < s->nb_channels; ch++) {

  1346.             av_dlog(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);

  1347.             g = &s->granules[ch][gr];

  1348.             g->part2_3_length = get_bits(&s->gb, 12);

  1349.             g->big_values     = get_bits(&s->gb,  9);

  1350.             if (g->big_values > 288) {

  1351.                 av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");

  1352.                 return AVERROR_INVALIDDATA;

  1353.             }

  1354. 

  1355.             g->global_gain = get_bits(&s->gb, 8);

  1356.             /* if MS stereo only is selected, we precompute the

  1357.                1/sqrt(2) renormalization factor */

  1358.             if ((s->mode_ext & (MODE_EXT_MS_STEREO | MODE_EXT_I_STEREO)) ==

  1359.                 MODE_EXT_MS_STEREO)

  1360.                 g->global_gain -= 2;

  1361.             if (s->lsf)

  1362.                 g->scalefac_compress = get_bits(&s->gb, 9);

  1363.             else

  1364.                 g->scalefac_compress = get_bits(&s->gb, 4);

  1365.             blocksplit_flag = get_bits1(&s->gb);

  1366.             if (blocksplit_flag) {

  1367.                 g->block_type = get_bits(&s->gb, 2);

  1368.                 if (g->block_type == 0) {

  1369.                     av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");

  1370.                     return AVERROR_INVALIDDATA;

  1371.                 }

  1372.                 g->switch_point = get_bits1(&s->gb);

  1373.                 for (i = 0; i < 2; i++)

  1374.                     g->table_select[i] = get_bits(&s->gb, 5);

  1375.                 for (i = 0; i < 3; i++)

  1376.                     g->subblock_gain[i] = get_bits(&s->gb, 3);

  1377.                 ff_init_short_region(s, g);

  1378.             } else {

  1379.                 int region_address1, region_address2;

  1380.                 g->block_type = 0;

  1381.                 g->switch_point = 0;

  1382.                 for (i = 0; i < 3; i++)

  1383.                     g->table_select[i] = get_bits(&s->gb, 5);

  1384.                 /* compute huffman coded region sizes */

  1385.                 region_address1 = get_bits(&s->gb, 4);

  1386.                 region_address2 = get_bits(&s->gb, 3);

  1387.                 av_dlog(s->avctx, "region1=%d region2=%d\n",

  1388.                         region_address1, region_address2);

  1389.                 ff_init_long_region(s, g, region_address1, region_address2);

  1390.             }

  1391.             ff_region_offset2size(g);

  1392.             ff_compute_band_indexes(s, g);

  1393. 

  1394.             g->preflag = 0;

  1395.             if (!s->lsf)

  1396.                 g->preflag = get_bits1(&s->gb);

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

  1398.             g->count1table_select = get_bits1(&s->gb);

  1399.             av_dlog(s->avctx, "block_type=%d switch_point=%d\n",

  1400.                     g->block_type, g->switch_point);

  1401.         }

  1402.     }

  1403. 

  1404.     if (!s->adu_mode) {

  1405.         int skip;

  1406.         const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);

  1407.         int extrasize = av_clip(get_bits_left(&s->gb) >> 3, 0, EXTRABYTES);

  1408.         av_assert1((get_bits_count(&s->gb) & 7) == 0);

  1409.         /* now we get bits from the main_data_begin offset */

  1410.         av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",

  1411.                 main_data_begin, s->last_buf_size);

  1412. 

  1413.         memcpy(s->last_buf + s->last_buf_size, ptr, extrasize);

  1414.         s->in_gb = s->gb;

  1415.         init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);

  1416. #if !UNCHECKED_BITSTREAM_READER

  1417.         s->gb.size_in_bits_plus8 += FFMAX(extrasize, LAST_BUF_SIZE - s->last_buf_size) * 8;

  1418. #endif

  1419.         s->last_buf_size <<= 3;

  1420.         for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {

  1421.             for (ch = 0; ch < s->nb_channels; ch++) {

  1422.                 g = &s->granules[ch][gr];

  1423.                 s->last_buf_size += g->part2_3_length;

  1424.                 memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));

  1425.                 compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);

  1426.             }

  1427.         }

  1428.         skip = s->last_buf_size - 8 * main_data_begin;

  1429.         if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {

  1430.             skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);

  1431.             s->gb           = s->in_gb;

  1432.             s->in_gb.buffer = NULL;

  1433.         } else {

  1434.             skip_bits_long(&s->gb, skip);

  1435.         }

  1436.     } else {

  1437.         gr = 0;

  1438.     }

  1439. 

  1440.     for (; gr < nb_granules; gr++) {

  1441.         for (ch = 0; ch < s->nb_channels; ch++) {

  1442.             g = &s->granules[ch][gr];

  1443.             bits_pos = get_bits_count(&s->gb);

  1444. 

  1445.             if (!s->lsf) {

  1446.                 uint8_t *sc;

  1447.                 int slen, slen1, slen2;

  1448. 

  1449.                 /* MPEG1 scale factors */

  1450.                 slen1 = slen_table[0][g->scalefac_compress];

  1451.                 slen2 = slen_table[1][g->scalefac_compress];

  1452.                 av_dlog(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);

  1453.                 if (g->block_type == 2) {

  1454.                     n = g->switch_point ? 17 : 18;

  1455.                     j = 0;

  1456.                     if (slen1) {

  1457.                         for (i = 0; i < n; i++)

  1458.                             g->scale_factors[j++] = get_bits(&s->gb, slen1);

  1459.                     } else {

  1460.                         for (i = 0; i < n; i++)

  1461.                             g->scale_factors[j++] = 0;

  1462.                     }

  1463.                     if (slen2) {

  1464.                         for (i = 0; i < 18; i++)

  1465.                             g->scale_factors[j++] = get_bits(&s->gb, slen2);

  1466.                         for (i = 0; i < 3; i++)

  1467.                             g->scale_factors[j++] = 0;

  1468.                     } else {

  1469.                         for (i = 0; i < 21; i++)

  1470.                             g->scale_factors[j++] = 0;

  1471.                     }

  1472.                 } else {

  1473.                     sc = s->granules[ch][0].scale_factors;

  1474.                     j = 0;

  1475.                     for (k = 0; k < 4; k++) {

  1476.                         n = k == 0 ? 6 : 5;

  1477.                         if ((g->scfsi & (0x8 >> k)) == 0) {

  1478.                             slen = (k < 2) ? slen1 : slen2;

  1479.                             if (slen) {

  1480.                                 for (i = 0; i < n; i++)

  1481.                                     g->scale_factors[j++] = get_bits(&s->gb, slen);

  1482.                             } else {

  1483.                                 for (i = 0; i < n; i++)

  1484.                                     g->scale_factors[j++] = 0;

  1485.                             }

  1486.                         } else {

  1487.                             /* simply copy from last granule */

  1488.                             for (i = 0; i < n; i++) {

  1489.                                 g->scale_factors[j] = sc[j];

  1490.                                 j++;

  1491.                             }

  1492.                         }

  1493.                     }

  1494.                     g->scale_factors[j++] = 0;

  1495.                 }

  1496.             } else {

  1497.                 int tindex, tindex2, slen[4], sl, sf;

  1498. 

  1499.                 /* LSF scale factors */

  1500.                 if (g->block_type == 2)

  1501.                     tindex = g->switch_point ? 2 : 1;

  1502.                 else

  1503.                     tindex = 0;

  1504. 

  1505.                 sf = g->scalefac_compress;

  1506.                 if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {

  1507.                     /* intensity stereo case */

  1508.                     sf >>= 1;

  1509.                     if (sf < 180) {

  1510.                         lsf_sf_expand(slen, sf, 6, 6, 0);

  1511.                         tindex2 = 3;

  1512.                     } else if (sf < 244) {

  1513.                         lsf_sf_expand(slen, sf - 180, 4, 4, 0);

  1514.                         tindex2 = 4;

  1515.                     } else {

  1516.                         lsf_sf_expand(slen, sf - 244, 3, 0, 0);

  1517.                         tindex2 = 5;

  1518.                     }

  1519.                 } else {

  1520.                     /* normal case */

  1521.                     if (sf < 400) {

  1522.                         lsf_sf_expand(slen, sf, 5, 4, 4);

  1523.                         tindex2 = 0;

  1524.                     } else if (sf < 500) {

  1525.                         lsf_sf_expand(slen, sf - 400, 5, 4, 0);

  1526.                         tindex2 = 1;

  1527.                     } else {

  1528.                         lsf_sf_expand(slen, sf - 500, 3, 0, 0);

  1529.                         tindex2 = 2;

  1530.                         g->preflag = 1;

  1531.                     }

  1532.                 }

  1533. 

  1534.                 j = 0;

  1535.                 for (k = 0; k < 4; k++) {

  1536.                     n  = lsf_nsf_table[tindex2][tindex][k];

  1537.                     sl = slen[k];

  1538.                     if (sl) {

  1539.                         for (i = 0; i < n; i++)

  1540.                             g->scale_factors[j++] = get_bits(&s->gb, sl);

  1541.                     } else {

  1542.                         for (i = 0; i < n; i++)

  1543.                             g->scale_factors[j++] = 0;

  1544.                     }

  1545.                 }

  1546.                 /* XXX: should compute exact size */

  1547.                 for (; j < 40; j++)

  1548.                     g->scale_factors[j] = 0;

  1549.             }

  1550. 

  1551.             exponents_from_scale_factors(s, g, exponents);

  1552. 

  1553.             /* read Huffman coded residue */

  1554.             huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);

  1555.         } /* ch */

  1556. 

  1557.         if (s->mode == MPA_JSTEREO)

  1558.             compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]);

  1559. 

  1560.         for (ch = 0; ch < s->nb_channels; ch++) {

  1561.             g = &s->granules[ch][gr];

  1562. 

  1563.             reorder_block(s, g);

  1564.             compute_antialias(s, g);

  1565.             compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);

  1566.         }

  1567.     } /* gr */

  1568.     if (get_bits_count(&s->gb) < 0)

  1569.         skip_bits_long(&s->gb, -get_bits_count(&s->gb));

  1570.     return nb_granules * 18;

  1571. }

  1572. 

  1573. static int mp_decode_frame(MPADecodeContext *s, OUT_INT **samples,

  1574.                            const uint8_t *buf, int buf_size)

  1575. {

  1576.     int i, nb_frames, ch, ret;

  1577.     OUT_INT *samples_ptr;

  1578. 

  1579.     init_get_bits(&s->gb, buf + HEADER_SIZE, (buf_size - HEADER_SIZE) * 8);

  1580. 

  1581.     /* skip error protection field */

  1582.     if (s->error_protection)

  1583.         skip_bits(&s->gb, 16);

  1584. 

  1585.     switch(s->layer) {

  1586.     case 1:

  1587.         s->avctx->frame_size = 384;

  1588.         nb_frames = mp_decode_layer1(s);

  1589.         break;

  1590.     case 2:

  1591.         s->avctx->frame_size = 1152;

  1592.         nb_frames = mp_decode_layer2(s);

  1593.         break;

  1594.     case 3:

  1595.         s->avctx->frame_size = s->lsf ? 576 : 1152;

  1596.     default:

  1597.         nb_frames = mp_decode_layer3(s);

  1598. 

  1599.         s->last_buf_size=0;

  1600.         if (s->in_gb.buffer) {

  1601.             align_get_bits(&s->gb);

  1602.             i = get_bits_left(&s->gb)>>3;

  1603.             if (i >= 0 && i <= BACKSTEP_SIZE) {

  1604.                 memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);

  1605.                 s->last_buf_size=i;

  1606.             } else

  1607.                 av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);

  1608.             s->gb           = s->in_gb;

  1609.             s->in_gb.buffer = NULL;

  1610.         }

  1611. 

  1612.         align_get_bits(&s->gb);

  1613.         av_assert1((get_bits_count(&s->gb) & 7) == 0);

  1614.         i = get_bits_left(&s->gb) >> 3;

  1615. 

  1616.         if (i < 0 || i > BACKSTEP_SIZE || nb_frames < 0) {

  1617.             if (i < 0)

  1618.                 av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);

  1619.             i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);

  1620.         }

  1621.         av_assert1(i <= buf_size - HEADER_SIZE && i >= 0);

  1622.         memcpy(s->last_buf + s->last_buf_size, s->gb.buffer + buf_size - HEADER_SIZE - i, i);

  1623.         s->last_buf_size += i;

  1624.     }

  1625. 

  1626.     if(nb_frames < 0)

  1627.         return nb_frames;

  1628. 

  1629.     /* get output buffer */

  1630.     if (!samples) {

  1631.         s->frame.nb_samples = s->avctx->frame_size;

  1632.         if ((ret = s->avctx->get_buffer(s->avctx, &s->frame)) < 0) {

  1633.             av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1634.             return ret;

  1635.         }

  1636.         samples = (OUT_INT **)s->frame.extended_data;

  1637.     }

  1638. 

  1639.     /* apply the synthesis filter */

  1640.     for (ch = 0; ch < s->nb_channels; ch++) {

  1641.         int sample_stride;

  1642.         if (s->avctx->sample_fmt == OUT_FMT_P) {

  1643.             samples_ptr   = samples[ch];

  1644.             sample_stride = 1;

  1645.         } else {

  1646.             samples_ptr   = samples[0] + ch;

  1647.             sample_stride = s->nb_channels;

  1648.         }

  1649.         for (i = 0; i < nb_frames; i++) {

  1650.             RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],

  1651.                                         &(s->synth_buf_offset[ch]),

  1652.                                         RENAME(ff_mpa_synth_window),

  1653.                                         &s->dither_state, samples_ptr,

  1654.                                         sample_stride, s->sb_samples[ch][i]);

  1655.             samples_ptr += 32 * sample_stride;

  1656.         }

  1657.     }

  1658. 

  1659.     return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;

  1660. }

  1661. 

  1662. static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,

  1663.                         AVPacket *avpkt)

  1664. {

  1665.     const uint8_t *buf  = avpkt->data;

  1666.     int buf_size        = avpkt->size;

  1667.     MPADecodeContext *s = avctx->priv_data;

  1668.     uint32_t header;

  1669.     int ret;

  1670. 

  1671.     while(buf_size && !*buf){

  1672.         buf++;

  1673.         buf_size--;

  1674.     }

  1675. 

  1676.     if (buf_size < HEADER_SIZE)

  1677.         return AVERROR_INVALIDDATA;

  1678. 

  1679.     header = AV_RB32(buf);

  1680.     if (header>>8 == AV_RB32("TAG")>>8) {

  1681.         av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");

  1682.         return buf_size;

  1683.     }

  1684.     if (ff_mpa_check_header(header) < 0) {

  1685.         av_log(avctx, AV_LOG_ERROR, "Header missing\n");

  1686.         return AVERROR_INVALIDDATA;

  1687.     }

  1688. 

  1689.     if (avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {

  1690.         /* free format: prepare to compute frame size */

  1691.         s->frame_size = -1;

  1692.         return AVERROR_INVALIDDATA;

  1693.     }

  1694.     /* update codec info */

  1695.     avctx->channels       = s->nb_channels;

  1696.     avctx->channel_layout = s->nb_channels == 1 ? AV_CH_LAYOUT_MONO : AV_CH_LAYOUT_STEREO;

  1697.     if (!avctx->bit_rate)

  1698.         avctx->bit_rate = s->bit_rate;

  1699. 

  1700.     if (s->frame_size <= 0 || s->frame_size > buf_size) {

  1701.         av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");

  1702.         return AVERROR_INVALIDDATA;

  1703.     } else if (s->frame_size < buf_size) {

  1704.         av_log(avctx, AV_LOG_DEBUG, "incorrect frame size - multiple frames in buffer?\n");

  1705.         buf_size= s->frame_size;

  1706.     }

  1707. 

  1708.     ret = mp_decode_frame(s, NULL, buf, buf_size);

  1709.     if (ret >= 0) {

  1710.         *got_frame_ptr   = 1;

  1711.         *(AVFrame *)data = s->frame;

  1712.         avctx->sample_rate = s->sample_rate;

  1713.         //FIXME maybe move the other codec info stuff from above here too

  1714.     } else {

  1715.         av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");

  1716.         /* Only return an error if the bad frame makes up the whole packet or

  1717.          * the error is related to buffer management.

  1718.          * If there is more data in the packet, just consume the bad frame

  1719.          * instead of returning an error, which would discard the whole

  1720.          * packet. */

  1721.         *got_frame_ptr = 0;

  1722.         if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)

  1723.             return ret;

  1724.     }

  1725.     s->frame_size = 0;

  1726.     return buf_size;

  1727. }

  1728. 

  1729. static void mp_flush(MPADecodeContext *ctx)

  1730. {

  1731.     memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));

  1732.     ctx->last_buf_size = 0;

  1733. }

  1734. 

  1735. static void flush(AVCodecContext *avctx)

  1736. {

  1737.     mp_flush(avctx->priv_data);

  1738. }

  1739. 

  1740. #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER

  1741. static int decode_frame_adu(AVCodecContext *avctx, void *data,

  1742.                             int *got_frame_ptr, AVPacket *avpkt)

  1743. {

  1744.     const uint8_t *buf  = avpkt->data;

  1745.     int buf_size        = avpkt->size;

  1746.     MPADecodeContext *s = avctx->priv_data;

  1747.     uint32_t header;

  1748.     int len, ret;

  1749.     int av_unused out_size;

  1750. 

  1751.     len = buf_size;

  1752. 

  1753.     // Discard too short frames

  1754.     if (buf_size < HEADER_SIZE) {

  1755.         av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");

  1756.         return AVERROR_INVALIDDATA;

  1757.     }

  1758. 

  1759. 

  1760.     if (len > MPA_MAX_CODED_FRAME_SIZE)

  1761.         len = MPA_MAX_CODED_FRAME_SIZE;

  1762. 

  1763.     // Get header and restore sync word

  1764.     header = AV_RB32(buf) | 0xffe00000;

  1765. 

  1766.     if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame

  1767.         av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");

  1768.         return AVERROR_INVALIDDATA;

  1769.     }

  1770. 

  1771.     avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);

  1772.     /* update codec info */

  1773.     avctx->sample_rate = s->sample_rate;

  1774.     avctx->channels    = s->nb_channels;

  1775.     if (!avctx->bit_rate)

  1776.         avctx->bit_rate = s->bit_rate;

  1777. 

  1778.     s->frame_size = len;

  1779. 

  1780.     ret = mp_decode_frame(s, NULL, buf, buf_size);

  1781.     if (ret < 0) {

  1782.         av_log(avctx, AV_LOG_ERROR, "Error while decoding MPEG audio frame.\n");

  1783.         return ret;

  1784.     }

  1785. 

  1786.     *got_frame_ptr   = 1;

  1787.     *(AVFrame *)data = s->frame;

  1788. 

  1789.     return buf_size;

  1790. }

  1791. #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */

  1792. 

  1793. #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER

  1794. 

  1795. /**

  1796.  * Context for MP3On4 decoder

  1797.  */

  1798. typedef struct MP3On4DecodeContext {

  1799.     AVFrame *frame;

  1800.     int frames;                     ///< number of mp3 frames per block (number of mp3 decoder instances)

  1801.     int syncword;                   ///< syncword patch

  1802.     const uint8_t *coff;            ///< channel offsets in output buffer

  1803.     MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance

  1804. } MP3On4DecodeContext;

  1805. 

  1806. #include "mpeg4audio.h"

  1807. 

  1808. /* Next 3 arrays are indexed by channel config number (passed via codecdata) */

  1809. 

  1810. /* number of mp3 decoder instances */

  1811. static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };

  1812. 

  1813. /* offsets into output buffer, assume output order is FL FR C LFE BL BR SL SR */

  1814. static const uint8_t chan_offset[8][5] = {

  1815.     { 0             },

  1816.     { 0             },  // C

  1817.     { 0             },  // FLR

  1818.     { 2, 0          },  // C FLR

  1819.     { 2, 0, 3       },  // C FLR BS

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

  1821.     { 2, 0, 4, 3    },  // C FLR BLRS LFE

  1822.     { 2, 0, 6, 4, 3 },  // C FLR BLRS BLR LFE

  1823. };

  1824. 

  1825. /* mp3on4 channel layouts */

  1826. static const int16_t chan_layout[8] = {

  1827.     0,

  1828.     AV_CH_LAYOUT_MONO,

  1829.     AV_CH_LAYOUT_STEREO,

  1830.     AV_CH_LAYOUT_SURROUND,

  1831.     AV_CH_LAYOUT_4POINT0,

  1832.     AV_CH_LAYOUT_5POINT0,

  1833.     AV_CH_LAYOUT_5POINT1,

  1834.     AV_CH_LAYOUT_7POINT1

  1835. };

  1836. 

  1837. static av_cold int decode_close_mp3on4(AVCodecContext * avctx)

  1838. {

  1839.     MP3On4DecodeContext *s = avctx->priv_data;

  1840.     int i;

  1841. 

  1842.     for (i = 0; i < s->frames; i++)

  1843.         av_free(s->mp3decctx[i]);

  1844. 

  1845.     return 0;

  1846. }

  1847. 

  1848. 

  1849. static int decode_init_mp3on4(AVCodecContext * avctx)

  1850. {

  1851.     MP3On4DecodeContext *s = avctx->priv_data;

  1852.     MPEG4AudioConfig cfg;

  1853.     int i;

  1854. 

  1855.     if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {

  1856.         av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");

  1857.         return AVERROR_INVALIDDATA;

  1858.     }

  1859. 

  1860.     avpriv_mpeg4audio_get_config(&cfg, avctx->extradata,

  1861.                                  avctx->extradata_size * 8, 1);

  1862.     if (!cfg.chan_config || cfg.chan_config > 7) {

  1863.         av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");

  1864.         return AVERROR_INVALIDDATA;

  1865.     }

  1866.     s->frames             = mp3Frames[cfg.chan_config];

  1867.     s->coff               = chan_offset[cfg.chan_config];

  1868.     avctx->channels       = ff_mpeg4audio_channels[cfg.chan_config];

  1869.     avctx->channel_layout = chan_layout[cfg.chan_config];

  1870. 

  1871.     if (cfg.sample_rate < 16000)

  1872.         s->syncword = 0xffe00000;

  1873.     else

  1874.         s->syncword = 0xfff00000;

  1875. 

  1876.     /* Init the first mp3 decoder in standard way, so that all tables get builded

  1877.      * We replace avctx->priv_data with the context of the first decoder so that

  1878.      * decode_init() does not have to be changed.

  1879.      * Other decoders will be initialized here copying data from the first context

  1880.      */

  1881.     // Allocate zeroed memory for the first decoder context

  1882.     s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));

  1883.     if (!s->mp3decctx[0])

  1884.         goto alloc_fail;

  1885.     // Put decoder context in place to make init_decode() happy

  1886.     avctx->priv_data = s->mp3decctx[0];

  1887.     decode_init(avctx);

  1888.     s->frame = avctx->coded_frame;

  1889.     // Restore mp3on4 context pointer

  1890.     avctx->priv_data = s;

  1891.     s->mp3decctx[0]->adu_mode = 1; // Set adu mode

  1892. 

  1893.     /* Create a separate codec/context for each frame (first is already ok).

  1894.      * Each frame is 1 or 2 channels - up to 5 frames allowed

  1895.      */

  1896.     for (i = 1; i < s->frames; i++) {

  1897.         s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));

  1898.         if (!s->mp3decctx[i])

  1899.             goto alloc_fail;

  1900.         s->mp3decctx[i]->adu_mode = 1;

  1901.         s->mp3decctx[i]->avctx = avctx;

  1902.         s->mp3decctx[i]->mpadsp = s->mp3decctx[0]->mpadsp;

  1903.     }

  1904. 

  1905.     return 0;

  1906. alloc_fail:

  1907.     decode_close_mp3on4(avctx);

  1908.     return AVERROR(ENOMEM);

  1909. }

  1910. 

  1911. 

  1912. static void flush_mp3on4(AVCodecContext *avctx)

  1913. {

  1914.     int i;

  1915.     MP3On4DecodeContext *s = avctx->priv_data;

  1916. 

  1917.     for (i = 0; i < s->frames; i++)

  1918.         mp_flush(s->mp3decctx[i]);

  1919. }

  1920. 

  1921. 

  1922. static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,

  1923.                                int *got_frame_ptr, AVPacket *avpkt)

  1924. {

  1925.     const uint8_t *buf     = avpkt->data;

  1926.     int buf_size           = avpkt->size;

  1927.     MP3On4DecodeContext *s = avctx->priv_data;

  1928.     MPADecodeContext *m;

  1929.     int fsize, len = buf_size, out_size = 0;

  1930.     uint32_t header;

  1931.     OUT_INT **out_samples;

  1932.     OUT_INT *outptr[2];

  1933.     int fr, ch, ret;

  1934. 

  1935.     /* get output buffer */

  1936.     s->frame->nb_samples = s->frames * MPA_FRAME_SIZE;

  1937.     if ((ret = avctx->get_buffer(avctx, s->frame)) < 0) {

  1938.         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");

  1939.         return ret;

  1940.     }

  1941.     out_samples = (OUT_INT **)s->frame->extended_data;

  1942. 

  1943.     // Discard too short frames

  1944.     if (buf_size < HEADER_SIZE)

  1945.         return AVERROR_INVALIDDATA;

  1946. 

  1947.     avctx->bit_rate = 0;

  1948. 

  1949.     ch = 0;

  1950.     for (fr = 0; fr < s->frames; fr++) {

  1951.         fsize = AV_RB16(buf) >> 4;

  1952.         fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);

  1953.         m     = s->mp3decctx[fr];

  1954.         av_assert1(m);

  1955. 

  1956.         if (fsize < HEADER_SIZE) {

  1957.             av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");

  1958.             return AVERROR_INVALIDDATA;

  1959.         }

  1960.         header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header

  1961. 

  1962.         if (ff_mpa_check_header(header) < 0) // Bad header, discard block

  1963.             break;

  1964. 

  1965.         avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);

  1966. 

  1967.         if (ch + m->nb_channels > avctx->channels) {

  1968.             av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "

  1969.                                         "channel count\n");

  1970.             return AVERROR_INVALIDDATA;

  1971.         }

  1972.         ch += m->nb_channels;

  1973. 

  1974.         outptr[0] = out_samples[s->coff[fr]];

  1975.         if (m->nb_channels > 1)

  1976.             outptr[1] = out_samples[s->coff[fr] + 1];

  1977. 

  1978.         if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)

  1979.             return ret;

  1980. 

  1981.         out_size += ret;

  1982.         buf      += fsize;

  1983.         len      -= fsize;

  1984. 

  1985.         avctx->bit_rate += m->bit_rate;

  1986.     }

  1987. 

  1988.     /* update codec info */

  1989.     avctx->sample_rate = s->mp3decctx[0]->sample_rate;

  1990. 

  1991.     s->frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));

  1992.     *got_frame_ptr   = 1;

  1993.     *(AVFrame *)data = *s->frame;

  1994. 

  1995.     return buf_size;

  1996. }

  1997. #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */

  1998. 

  1999. #if !CONFIG_FLOAT

  2000. #if CONFIG_MP1_DECODER

  2001. AVCodec ff_mp1_decoder = {

  2002.     .name           = "mp1",

  2003.     .type           = AVMEDIA_TYPE_AUDIO,

  2004.     .id             = AV_CODEC_ID_MP1,

  2005.     .priv_data_size = sizeof(MPADecodeContext),

  2006.     .init           = decode_init,

  2007.     .decode         = decode_frame,

  2008.     .capabilities   = CODEC_CAP_DR1,

  2009.     .flush          = flush,

  2010.     .long_name      = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),

  2011.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2012.                                                       AV_SAMPLE_FMT_S16,

  2013.                                                       AV_SAMPLE_FMT_NONE },

  2014. };

  2015. #endif

  2016. #if CONFIG_MP2_DECODER

  2017. AVCodec ff_mp2_decoder = {

  2018.     .name           = "mp2",

  2019.     .type           = AVMEDIA_TYPE_AUDIO,

  2020.     .id             = AV_CODEC_ID_MP2,

  2021.     .priv_data_size = sizeof(MPADecodeContext),

  2022.     .init           = decode_init,

  2023.     .decode         = decode_frame,

  2024.     .capabilities   = CODEC_CAP_DR1,

  2025.     .flush          = flush,

  2026.     .long_name      = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),

  2027.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2028.                                                       AV_SAMPLE_FMT_S16,

  2029.                                                       AV_SAMPLE_FMT_NONE },

  2030. };

  2031. #endif

  2032. #if CONFIG_MP3_DECODER

  2033. AVCodec ff_mp3_decoder = {

  2034.     .name           = "mp3",

  2035.     .type           = AVMEDIA_TYPE_AUDIO,

  2036.     .id             = AV_CODEC_ID_MP3,

  2037.     .priv_data_size = sizeof(MPADecodeContext),

  2038.     .init           = decode_init,

  2039.     .decode         = decode_frame,

  2040.     .capabilities   = CODEC_CAP_DR1,

  2041.     .flush          = flush,

  2042.     .long_name      = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),

  2043.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2044.                                                       AV_SAMPLE_FMT_S16,

  2045.                                                       AV_SAMPLE_FMT_NONE },

  2046. };

  2047. #endif

  2048. #if CONFIG_MP3ADU_DECODER

  2049. AVCodec ff_mp3adu_decoder = {

  2050.     .name           = "mp3adu",

  2051.     .type           = AVMEDIA_TYPE_AUDIO,

  2052.     .id             = AV_CODEC_ID_MP3ADU,

  2053.     .priv_data_size = sizeof(MPADecodeContext),

  2054.     .init           = decode_init,

  2055.     .decode         = decode_frame_adu,

  2056.     .capabilities   = CODEC_CAP_DR1,

  2057.     .flush          = flush,

  2058.     .long_name      = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),

  2059.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2060.                                                       AV_SAMPLE_FMT_S16,

  2061.                                                       AV_SAMPLE_FMT_NONE },

  2062. };

  2063. #endif

  2064. #if CONFIG_MP3ON4_DECODER

  2065. AVCodec ff_mp3on4_decoder = {

  2066.     .name           = "mp3on4",

  2067.     .type           = AVMEDIA_TYPE_AUDIO,

  2068.     .id             = AV_CODEC_ID_MP3ON4,

  2069.     .priv_data_size = sizeof(MP3On4DecodeContext),

  2070.     .init           = decode_init_mp3on4,

  2071.     .close          = decode_close_mp3on4,

  2072.     .decode         = decode_frame_mp3on4,

  2073.     .capabilities   = CODEC_CAP_DR1,

  2074.     .flush          = flush_mp3on4,

  2075.     .long_name      = NULL_IF_CONFIG_SMALL("MP3onMP4"),

  2076.     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,

  2077.                                                       AV_SAMPLE_FMT_NONE },

  2078. };

  2079. #endif

  2080. #endif