/*
  * MPEG Audio decoder

  * Copyright (c) 2001, 2002 Fabrice Bellard

  *

  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or

  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either

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

  *

  * FFmpeg is distributed in the hope that it will be useful,

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

  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.

  *

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

  * License along with FFmpeg; if not, write to the Free Software

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

  */

 
 /**

  * @file

  * MPEG Audio decoder

  */

 #include "libavutil/avassert.h"

 #include "libavutil/channel_layout.h"

 #include "libavutil/libm.h"

 #include "avcodec.h"

 #include "get_bits.h"

 #include "internal.h"

 #include "mathops.h"

 #include "mpegaudiodsp.h"

 #include "dsputil.h"

 
 /*

  * TODO:
  *  - test lsf / mpeg25 extensively.

  */
 

 #include "mpegaudio.h"

 #include "mpegaudiodecheader.h"

 #define BACKSTEP_SIZE 512
 #define EXTRABYTES 24

 #define LAST_BUF_SIZE 2 * BACKSTEP_SIZE + EXTRABYTES

 
 /* layer 3 "granule" */
 typedef struct GranuleDef {
     uint8_t scfsi;
     int part2_3_length;
     int big_values;
     int global_gain;
     int scalefac_compress;
     uint8_t block_type;
     uint8_t switch_point;
     int table_select[3];
     int subblock_gain[3];
     uint8_t scalefac_scale;
     uint8_t count1table_select;
     int region_size[3]; /* number of huffman codes in each region */
     int preflag;
     int short_start, long_end; /* long/short band indexes */
     uint8_t scale_factors[40];

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

 } GranuleDef;
 
 typedef struct MPADecodeContext {
     MPA_DECODE_HEADER

     uint8_t last_buf[LAST_BUF_SIZE];

     int last_buf_size;
     /* next header (used in free format parsing) */
     uint32_t free_format_next_header;
     GetBitContext gb;
     GetBitContext in_gb;

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

     int synth_buf_offset[MPA_MAX_CHANNELS];

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

     INTFLOAT mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
     GranuleDef granules[2][2]; /* Used in Layer 3 */
     int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
     int dither_state;

     int err_recognition;

     AVCodecContext* avctx;
     MPADSPContext mpadsp;

     DSPContext dsp;

     AVFrame frame;

 } MPADecodeContext;
 

 #if CONFIG_FLOAT

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

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

 #   define FIXR(x)        ((float)(x))
 #   define FIXHR(x)       ((float)(x))

 #   define MULH3(x, y, s) ((s)*(y)*(x))
 #   define MULLx(x, y, s) ((y)*(x))
 #   define RENAME(a) a ## _float

 #   define OUT_FMT   AV_SAMPLE_FMT_FLT
 #   define OUT_FMT_P AV_SAMPLE_FMT_FLTP

 #else
 #   define SHR(a,b)       ((a)>>(b))

 /* WARNING: only correct for positive numbers */

 #   define FIXR_OLD(a)    ((int)((a) * FRAC_ONE + 0.5))
 #   define FIXR(a)        ((int)((a) * FRAC_ONE + 0.5))
 #   define FIXHR(a)       ((int)((a) * (1LL<<32) + 0.5))
 #   define MULH3(x, y, s) MULH((s)*(x), y)
 #   define MULLx(x, y, s) MULL(x,y,s)

 #   define RENAME(a)      a ## _fixed

 #   define OUT_FMT   AV_SAMPLE_FMT_S16
 #   define OUT_FMT_P AV_SAMPLE_FMT_S16P

 #endif

 /****************/
 

 #define HEADER_SIZE 4
 

 #include "mpegaudiodata.h"

 #include "mpegaudiodectab.h"
 
 /* vlc structure for decoding layer 3 huffman tables */

 static VLC huff_vlc[16];

 static VLC_TYPE huff_vlc_tables[

     0 + 128 + 128 + 128 + 130 + 128 + 154 + 166 +
   142 + 204 + 190 + 170 + 542 + 460 + 662 + 414

   ][2];
 static const int huff_vlc_tables_sizes[16] = {

     0,  128,  128,  128,  130,  128,  154,  166,
   142,  204,  190,  170,  542,  460,  662,  414

 };

 static VLC huff_quad_vlc[2];

 static VLC_TYPE  huff_quad_vlc_tables[128+16][2];
 static const int huff_quad_vlc_tables_sizes[2] = { 128, 16 };

 /* computed from band_size_long */

 static uint16_t band_index_long[9][23];

 #include "mpegaudio_tablegen.h"

 /* intensity stereo coef table */

 static INTFLOAT is_table[2][16];
 static INTFLOAT is_table_lsf[2][2][16];

 static INTFLOAT csa_table[8][4];

 static int16_t division_tab3[1<<6 ];
 static int16_t division_tab5[1<<8 ];
 static int16_t division_tab9[1<<11];
 
 static int16_t * const division_tabs[4] = {
     division_tab3, division_tab5, NULL, division_tab9
 };
 

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

 static uint16_t scale_factor_modshift[64];

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

 static int32_t scale_factor_mult[15][3];

 /* mult table for layer 2 group quantization */
 
 #define SCALE_GEN(v) \

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

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

     SCALE_GEN(4.0 / 3.0), /* 3 steps */
     SCALE_GEN(4.0 / 5.0), /* 5 steps */
     SCALE_GEN(4.0 / 9.0), /* 9 steps */

 };
 

 /**
  * Convert region offsets to region sizes and truncate
  * size to big_values.
  */

 static void ff_region_offset2size(GranuleDef *g)
 {
     int i, k, j = 0;
     g->region_size[2] = 576 / 2;
     for (i = 0; i < 3; i++) {

         k = FFMIN(g->region_size[i], g->big_values);
         g->region_size[i] = k - j;
         j = k;
     }
 }
 

 static void ff_init_short_region(MPADecodeContext *s, GranuleDef *g)
 {

     if (g->block_type == 2) {
         if (s->sample_rate_index != 8)
             g->region_size[0] = (36 / 2);
         else
             g->region_size[0] = (72 / 2);
     } else {

         if (s->sample_rate_index <= 2)
             g->region_size[0] = (36 / 2);
         else if (s->sample_rate_index != 8)
             g->region_size[0] = (54 / 2);
         else
             g->region_size[0] = (108 / 2);
     }
     g->region_size[1] = (576 / 2);
 }
 

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

     int l;

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

     /* should not overflow */
     l = FFMIN(ra1 + ra2 + 2, 22);

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

 }
 

 static void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g)
 {

     if (g->block_type == 2) {
         if (g->switch_point) {

             if(s->sample_rate_index == 8)
                 av_log_ask_for_sample(s->avctx, "switch point in 8khz\n");

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

                 long blocks.  For 8000Hz, we handle the 72 first
                 exponents as long blocks */

             if (s->sample_rate_index <= 2)
                 g->long_end = 8;
             else

                 g->long_end = 6;

             g->short_start = 3;

         } else {

             g->long_end    = 0;

             g->short_start = 0;
         }
     } else {
         g->short_start = 13;

         g->long_end    = 22;

     }
 }
 

 /* layer 1 unscaling */
 /* n = number of bits of the mantissa minus 1 */
 static inline int l1_unscale(int n, int mant, int scale_factor)
 {
     int shift, mod;

     int64_t val;

     shift   = scale_factor_modshift[scale_factor];
     mod     = shift & 3;

     shift >>= 2;

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

     /* NOTE: at this point, 1 <= shift >= 21 + 15 */
     return (int)((val + (1LL << (shift - 1))) >> shift);

 }
 
 static inline int l2_unscale_group(int steps, int mant, int scale_factor)
 {
     int shift, mod, val;
 

     shift   = scale_factor_modshift[scale_factor];
     mod     = shift & 3;

     shift >>= 2;

 
     val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
     /* NOTE: at this point, 0 <= shift <= 21 */
     if (shift > 0)
         val = (val + (1 << (shift - 1))) >> shift;
     return val;

 }
 
 /* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
 static inline int l3_unscale(int value, int exponent)
 {
     unsigned int m;
     int e;
 

     e  = table_4_3_exp  [4 * value + (exponent & 3)];
     m  = table_4_3_value[4 * value + (exponent & 3)];
     e -= exponent >> 2;

 #ifdef DEBUG
     if(e < 1)

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

 #endif

     if (e > 31)

         return 0;

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

     return m;
 }
 

 static av_cold void decode_init_static(void)

 {

     int i, j, k;

     int offset;
 
     /* scale factors table for layer 1/2 */
     for (i = 0; i < 64; i++) {
         int shift, mod;
         /* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
         shift = i / 3;
         mod   = i % 3;
         scale_factor_modshift[i] = mod | (shift << 2);
     }

     /* scale factor multiply for layer 1 */
     for (i = 0; i < 15; i++) {
         int n, norm;
         n = i + 2;
         norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
         scale_factor_mult[i][0] = MULLx(norm, FIXR(1.0          * 2.0), FRAC_BITS);
         scale_factor_mult[i][1] = MULLx(norm, FIXR(0.7937005259 * 2.0), FRAC_BITS);
         scale_factor_mult[i][2] = MULLx(norm, FIXR(0.6299605249 * 2.0), FRAC_BITS);

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

                 scale_factor_mult[i][0],
                 scale_factor_mult[i][1],
                 scale_factor_mult[i][2]);
     }

     RENAME(ff_mpa_synth_init)(RENAME(ff_mpa_synth_window));

     /* huffman decode tables */
     offset = 0;
     for (i = 1; i < 16; i++) {
         const HuffTable *h = &mpa_huff_tables[i];
         int xsize, x, y;

         uint8_t  tmp_bits [512] = { 0 };
         uint16_t tmp_codes[512] = { 0 };

         xsize = h->xsize;

         j = 0;
         for (x = 0; x < xsize; x++) {
             for (y = 0; y < xsize; y++) {
                 tmp_bits [(x << 5) | y | ((x&&y)<<4)]= h->bits [j  ];
                 tmp_codes[(x << 5) | y | ((x&&y)<<4)]= h->codes[j++];

             }
         }
 

         /* XXX: fail test */
         huff_vlc[i].table = huff_vlc_tables+offset;
         huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
         init_vlc(&huff_vlc[i], 7, 512,
                  tmp_bits, 1, 1, tmp_codes, 2, 2,
                  INIT_VLC_USE_NEW_STATIC);
         offset += huff_vlc_tables_sizes[i];
     }

     av_assert0(offset == FF_ARRAY_ELEMS(huff_vlc_tables));

 
     offset = 0;
     for (i = 0; i < 2; i++) {
         huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
         huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
         init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
                  mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
                  INIT_VLC_USE_NEW_STATIC);
         offset += huff_quad_vlc_tables_sizes[i];
     }

     av_assert0(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));

 
     for (i = 0; i < 9; i++) {
         k = 0;
         for (j = 0; j < 22; j++) {
             band_index_long[i][j] = k;
             k += band_size_long[i][j];

         }

         band_index_long[i][22] = k;
     }

     /* compute n ^ (4/3) and store it in mantissa/exp format */
 
     mpegaudio_tableinit();
 
     for (i = 0; i < 4; i++) {
         if (ff_mpa_quant_bits[i] < 0) {
             for (j = 0; j < (1 << (-ff_mpa_quant_bits[i]+1)); j++) {
                 int val1, val2, val3, steps;
                 int val = j;
                 steps   = ff_mpa_quant_steps[i];
                 val1    = val % steps;
                 val    /= steps;
                 val2    = val % steps;
                 val3    = val / steps;
                 division_tabs[i][j] = val1 + (val2 << 4) + (val3 << 8);

             }
         }

     }

     for (i = 0; i < 7; i++) {
         float f;
         INTFLOAT v;
         if (i != 6) {
             f = tan((double)i * M_PI / 12.0);
             v = FIXR(f / (1.0 + f));
         } else {
             v = FIXR(1.0);

         }

         is_table[0][    i] = v;
         is_table[1][6 - i] = v;
     }
     /* invalid values */
     for (i = 7; i < 16; i++)
         is_table[0][i] = is_table[1][i] = 0.0;
 
     for (i = 0; i < 16; i++) {
         double f;
         int e, k;
 
         for (j = 0; j < 2; j++) {
             e = -(j + 1) * ((i + 1) >> 1);

             f = exp2(e / 4.0);

             k = i & 1;
             is_table_lsf[j][k ^ 1][i] = FIXR(f);
             is_table_lsf[j][k    ][i] = FIXR(1.0);

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

                     i, j, (float) is_table_lsf[j][0][i],
                     (float) is_table_lsf[j][1][i]);

         }

     }

     for (i = 0; i < 8; i++) {
         float ci, cs, ca;
         ci = ci_table[i];
         cs = 1.0 / sqrt(1.0 + ci * ci);
         ca = cs * ci;

 #if !CONFIG_FLOAT

         csa_table[i][0] = FIXHR(cs/4);
         csa_table[i][1] = FIXHR(ca/4);
         csa_table[i][2] = FIXHR(ca/4) + FIXHR(cs/4);
         csa_table[i][3] = FIXHR(ca/4) - FIXHR(cs/4);

 #else

         csa_table[i][0] = cs;
         csa_table[i][1] = ca;
         csa_table[i][2] = ca + cs;
         csa_table[i][3] = ca - cs;

 #endif

     }

 }

 static av_cold int decode_init(AVCodecContext * avctx)
 {

     static int initialized_tables = 0;

     MPADecodeContext *s = avctx->priv_data;
 

     if (!initialized_tables) {
         decode_init_static();
         initialized_tables = 1;
     }
 

     s->avctx = avctx;
 
     ff_mpadsp_init(&s->mpadsp);

     ff_dsputil_init(&s->dsp, avctx);

     if (avctx->request_sample_fmt == OUT_FMT &&
         avctx->codec_id != AV_CODEC_ID_MP3ON4)
         avctx->sample_fmt = OUT_FMT;
     else
         avctx->sample_fmt = OUT_FMT_P;

     s->err_recognition = avctx->err_recognition;

     if (avctx->codec_id == AV_CODEC_ID_MP3ADU)

         s->adu_mode = 1;

 
     avcodec_get_frame_defaults(&s->frame);
     avctx->coded_frame = &s->frame;
 

     return 0;
 }
 

 #define C3 FIXHR(0.86602540378443864676/2)

 #define C4 FIXHR(0.70710678118654752439/2) //0.5 / cos(pi*(9)/36)
 #define C5 FIXHR(0.51763809020504152469/2) //0.5 / cos(pi*(5)/36)
 #define C6 FIXHR(1.93185165257813657349/4) //0.5 / cos(pi*(15)/36)

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

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

 {

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

     in0  = in[0*3];
     in1  = in[1*3] + in[0*3];
     in2  = in[2*3] + in[1*3];
     in3  = in[3*3] + in[2*3];
     in4  = in[4*3] + in[3*3];
     in5  = in[5*3] + in[4*3];

     in5 += in3;
     in3 += in1;
 

     in2  = MULH3(in2, C3, 2);
     in3  = MULH3(in3, C3, 4);
 
     t1   = in0 - in4;

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

 
     out[ 7] =
     out[10] = t1 + t2;
     out[ 1] =
     out[ 4] = t1 - t2;
 
     in0    += SHR(in4, 1);
     in4     = in0 + in2;
     in5    += 2*in1;

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

     out[ 8] =
     out[ 9] = in4 + in1;
     out[ 2] =
     out[ 3] = in4 - in1;
 
     in0    -= in2;

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

     out[ 0] =
     out[ 5] = in0 - in5;
     out[ 6] =
     out[11] = in0 + in5;

 }
 
 /* return the number of decoded frames */
 static int mp_decode_layer1(MPADecodeContext *s)

 {

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

     uint8_t allocation[MPA_MAX_CHANNELS][SBLIMIT];
     uint8_t scale_factors[MPA_MAX_CHANNELS][SBLIMIT];

     if (s->mode == MPA_JSTEREO)

         bound = (s->mode_ext + 1) * 4;
     else
         bound = SBLIMIT;
 
     /* allocation bits */

     for (i = 0; i < bound; i++) {
         for (ch = 0; ch < s->nb_channels; ch++) {

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

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

         allocation[0][i] = get_bits(&s->gb, 4);
 
     /* scale factors */

     for (i = 0; i < bound; i++) {
         for (ch = 0; ch < s->nb_channels; ch++) {

             if (allocation[ch][i])
                 scale_factors[ch][i] = get_bits(&s->gb, 6);
         }
     }

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

         if (allocation[0][i]) {
             scale_factors[0][i] = get_bits(&s->gb, 6);
             scale_factors[1][i] = get_bits(&s->gb, 6);
         }
     }

     /* compute samples */

     for (j = 0; j < 12; j++) {
         for (i = 0; i < bound; i++) {
             for (ch = 0; ch < s->nb_channels; ch++) {

                 n = allocation[ch][i];
                 if (n) {
                     mant = get_bits(&s->gb, n + 1);
                     v = l1_unscale(n, mant, scale_factors[ch][i]);
                 } else {
                     v = 0;
                 }
                 s->sb_samples[ch][j][i] = v;
             }
         }

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

             n = allocation[0][i];
             if (n) {
                 mant = get_bits(&s->gb, n + 1);
                 v = l1_unscale(n, mant, scale_factors[0][i]);
                 s->sb_samples[0][j][i] = v;
                 v = l1_unscale(n, mant, scale_factors[1][i]);
                 s->sb_samples[1][j][i] = v;
             } else {
                 s->sb_samples[0][j][i] = 0;
                 s->sb_samples[1][j][i] = 0;
             }
         }
     }
     return 12;
 }
 
 static int mp_decode_layer2(MPADecodeContext *s)
 {
     int sblimit; /* number of used subbands */
     const unsigned char *alloc_table;
     int table, bit_alloc_bits, i, j, ch, bound, v;
     unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
     unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
     unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3], *sf;
     int scale, qindex, bits, steps, k, l, m, b;

     /* select decoding table */

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

                                    s->sample_rate, s->lsf);
     sblimit     = ff_mpa_sblimit_table[table];

     alloc_table = ff_mpa_alloc_tables[table];

     if (s->mode == MPA_JSTEREO)

         bound = (s->mode_ext + 1) * 4;
     else
         bound = sblimit;
 

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

 
     /* sanity check */

     if (bound > sblimit)
         bound = sblimit;

     /* parse bit allocation */
     j = 0;

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

         bit_alloc_bits = alloc_table[j];

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

             bit_alloc[ch][i] = get_bits(&s->gb, bit_alloc_bits);
         j += 1 << bit_alloc_bits;
     }

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

         bit_alloc_bits = alloc_table[j];
         v = get_bits(&s->gb, bit_alloc_bits);
         bit_alloc[0][i] = v;
         bit_alloc[1][i] = v;
         j += 1 << bit_alloc_bits;

     }

 
     /* scale codes */

     for (i = 0; i < sblimit; i++) {
         for (ch = 0; ch < s->nb_channels; ch++) {

             if (bit_alloc[ch][i])

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

     /* scale factors */

     for (i = 0; i < sblimit; i++) {
         for (ch = 0; ch < s->nb_channels; ch++) {

             if (bit_alloc[ch][i]) {
                 sf = scale_factors[ch][i];

                 switch (scale_code[ch][i]) {

                 default:
                 case 0:
                     sf[0] = get_bits(&s->gb, 6);
                     sf[1] = get_bits(&s->gb, 6);
                     sf[2] = get_bits(&s->gb, 6);
                     break;
                 case 2:
                     sf[0] = get_bits(&s->gb, 6);
                     sf[1] = sf[0];
                     sf[2] = sf[0];
                     break;
                 case 1:
                     sf[0] = get_bits(&s->gb, 6);
                     sf[2] = get_bits(&s->gb, 6);
                     sf[1] = sf[0];
                     break;
                 case 3:
                     sf[0] = get_bits(&s->gb, 6);
                     sf[2] = get_bits(&s->gb, 6);
                     sf[1] = sf[2];
                     break;
                 }
             }
         }
     }
 
     /* samples */

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

             j = 0;

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

                 bit_alloc_bits = alloc_table[j];

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

                     b = bit_alloc[ch][i];
                     if (b) {
                         scale = scale_factors[ch][i][k];
                         qindex = alloc_table[j+b];

                         bits = ff_mpa_quant_bits[qindex];

                         if (bits < 0) {

                             int v2;

                             /* 3 values at the same time */
                             v = get_bits(&s->gb, -bits);

                             v2 = division_tabs[qindex][v];
                             steps  = ff_mpa_quant_steps[qindex];
 

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

                                 l2_unscale_group(steps,  v2       & 15, scale);

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

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

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

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

                         } else {

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

                                 v = get_bits(&s->gb, bits);
                                 v = l1_unscale(bits - 1, v, scale);
                                 s->sb_samples[ch][k * 12 + l + m][i] = v;
                             }
                         }
                     } else {
                         s->sb_samples[ch][k * 12 + l + 0][i] = 0;
                         s->sb_samples[ch][k * 12 + l + 1][i] = 0;
                         s->sb_samples[ch][k * 12 + l + 2][i] = 0;
                     }
                 }
                 /* next subband in alloc table */

                 j += 1 << bit_alloc_bits;

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

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

                 bit_alloc_bits = alloc_table[j];
                 b = bit_alloc[0][i];
                 if (b) {
                     int mant, scale0, scale1;
                     scale0 = scale_factors[0][i][k];
                     scale1 = scale_factors[1][i][k];
                     qindex = alloc_table[j+b];

                     bits = ff_mpa_quant_bits[qindex];

                     if (bits < 0) {
                         /* 3 values at the same time */
                         v = get_bits(&s->gb, -bits);

                         steps = ff_mpa_quant_steps[qindex];

                         mant = v % steps;
                         v = v / steps;

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

                             l2_unscale_group(steps, mant, scale0);

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

                             l2_unscale_group(steps, mant, scale1);
                         mant = v % steps;
                         v = v / steps;

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

                             l2_unscale_group(steps, mant, scale0);

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

                             l2_unscale_group(steps, mant, scale1);

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

                             l2_unscale_group(steps, v, scale0);

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

                             l2_unscale_group(steps, v, scale1);
                     } else {

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

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

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

                                 l1_unscale(bits - 1, mant, scale0);

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

                                 l1_unscale(bits - 1, mant, scale1);
                         }
                     }
                 } else {
                     s->sb_samples[0][k * 12 + l + 0][i] = 0;
                     s->sb_samples[0][k * 12 + l + 1][i] = 0;
                     s->sb_samples[0][k * 12 + l + 2][i] = 0;
                     s->sb_samples[1][k * 12 + l + 0][i] = 0;
                     s->sb_samples[1][k * 12 + l + 1][i] = 0;
                     s->sb_samples[1][k * 12 + l + 2][i] = 0;
                 }
                 /* next subband in alloc table */

                 j += 1 << bit_alloc_bits;

             }
             /* fill remaining samples to zero */

             for (i = sblimit; i < SBLIMIT; i++) {
                 for (ch = 0; ch < s->nb_channels; ch++) {

                     s->sb_samples[ch][k * 12 + l + 0][i] = 0;
                     s->sb_samples[ch][k * 12 + l + 1][i] = 0;
                     s->sb_samples[ch][k * 12 + l + 2][i] = 0;
                 }
             }
         }
     }
     return 3 * 12;

 }
 

 #define SPLIT(dst,sf,n)             \
     if (n == 3) {                   \
         int m = (sf * 171) >> 9;    \
         dst   = sf - 3 * m;         \
         sf    = m;                  \
     } else if (n == 4) {            \
         dst  = sf & 3;              \
         sf >>= 2;                   \
     } else if (n == 5) {            \
         int m = (sf * 205) >> 10;   \
         dst   = sf - 5 * m;         \
         sf    = m;                  \
     } else if (n == 6) {            \
         int m = (sf * 171) >> 10;   \
         dst   = sf - 6 * m;         \
         sf    = m;                  \
     } else {                        \
         dst = 0;                    \

     }
 

 static av_always_inline void lsf_sf_expand(int *slen, int sf, int n1, int n2,
                                            int n3)

 {

     SPLIT(slen[3], sf, n3)
     SPLIT(slen[2], sf, n2)
     SPLIT(slen[1], sf, n1)

     slen[0] = sf;
 }
 

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

                                          int16_t *exponents)

 {

     const uint8_t *bstab, *pretab;

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

     int16_t *exp_ptr;

 
     exp_ptr = exponents;

     gain    = g->global_gain - 210;
     shift   = g->scalefac_scale + 1;

     bstab  = band_size_long[s->sample_rate_index];

     pretab = mpa_pretab[g->preflag];

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

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

         len = bstab[i];

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

             *exp_ptr++ = v0;
     }
 
     if (g->short_start < 13) {

         bstab    = band_size_short[s->sample_rate_index];

         gains[0] = gain - (g->subblock_gain[0] << 3);
         gains[1] = gain - (g->subblock_gain[1] << 3);
         gains[2] = gain - (g->subblock_gain[2] << 3);

         k        = g->long_end;
         for (i = g->short_start; i < 13; i++) {

             len = bstab[i];

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

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

                 for (j = len; j > 0; j--)
                     *exp_ptr++ = v0;

             }
         }
     }
 }
 
 /* handle n = 0 too */
 static inline int get_bitsz(GetBitContext *s, int n)
 {

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

 }
 

 static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos,
                           int *end_pos2)
 {
     if (s->in_gb.buffer && *pos >= s->gb.size_in_bits) {
         s->gb           = s->in_gb;
         s->in_gb.buffer = NULL;

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

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

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

     }
 }
 

 /* Following is a optimized code for
             INTFLOAT v = *src
             if(get_bits1(&s->gb))
                 v = -v;
             *dst = v;
 */
 #if CONFIG_FLOAT

 #define READ_FLIP_SIGN(dst,src)                     \
     v = AV_RN32A(src) ^ (get_bits1(&s->gb) << 31);  \
     AV_WN32A(dst, v);

 #else

 #define READ_FLIP_SIGN(dst,src)     \
     v      = -get_bits1(&s->gb);    \
     *(dst) = (*(src) ^ v) - v;

 #endif
 

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

                           int16_t *exponents, int end_pos2)

 {
     int s_index;

     int i;

     int last_pos, bits_left;

     VLC *vlc;

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

 
     /* low frequencies (called big values) */
     s_index = 0;

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

         int j, k, l, linbits;

         j = g->region_size[i];
         if (j == 0)
             continue;
         /* select vlc table */

         k       = g->table_select[i];
         l       = mpa_huff_data[k][0];

         linbits = mpa_huff_data[k][1];

         vlc     = &huff_vlc[l];

         if (!l) {
             memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid) * 2 * j);
             s_index += 2 * j;

             continue;
         }
 

         /* read huffcode and compute each couple */

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

             int exponent, x, y;

             int v;

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

 
             if (pos >= end_pos){

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

                 if (pos >= end_pos)

                     break;
             }

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

             if (!y) {

                 g->sb_hybrid[s_index  ] =
                 g->sb_hybrid[s_index+1] = 0;
                 s_index += 2;
                 continue;
             }
 

             exponent= exponents[s_index];

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

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

             if (y & 16) {

                 x = y >> 5;
                 y = y & 0x0f;

                 if (x < 15) {
                     READ_FLIP_SIGN(g->sb_hybrid + s_index, RENAME(expval_table)[exponent] + x)
                 } else {

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

                     v  = l3_unscale(x, exponent);

                     if (get_bits1(&s->gb))
                         v = -v;
                     g->sb_hybrid[s_index] = v;

                 }

                 if (y < 15) {
                     READ_FLIP_SIGN(g->sb_hybrid + s_index + 1, RENAME(expval_table)[exponent] + y)
                 } else {

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

                     v  = l3_unscale(y, exponent);

                     if (get_bits1(&s->gb))
                         v = -v;
                     g->sb_hybrid[s_index+1] = v;

                 }

             } else {

                 x = y >> 5;
                 y = y & 0x0f;
                 x += y;

                 if (x < 15) {
                     READ_FLIP_SIGN(g->sb_hybrid + s_index + !!y, RENAME(expval_table)[exponent] + x)
                 } else {

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

                     v  = l3_unscale(x, exponent);

                     if (get_bits1(&s->gb))
                         v = -v;
                     g->sb_hybrid[s_index+!!y] = v;

                 }

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

             }

             s_index += 2;

         }
     }

     /* high frequencies */
     vlc = &huff_quad_vlc[g->count1table_select];

     last_pos = 0;

     while (s_index <= 572) {

         int pos, code;

         pos = get_bits_count(&s->gb);
         if (pos >= end_pos) {

             if (pos > end_pos2 && last_pos) {

                 /* some encoders generate an incorrect size for this
                    part. We must go back into the data */
                 s_index -= 4;
                 skip_bits_long(&s->gb, last_pos - pos);

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

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

                     s_index=0;

                 break;
             }

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

             if (pos >= end_pos)

                 break;

         }

         last_pos = pos;

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

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

         g->sb_hybrid[s_index+0] =
         g->sb_hybrid[s_index+1] =
         g->sb_hybrid[s_index+2] =
         g->sb_hybrid[s_index+3] = 0;
         while (code) {
             static const int idxtab[16] = { 3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0 };

             int v;

             int pos = s_index + idxtab[code];
             code   ^= 8 >> idxtab[code];
             READ_FLIP_SIGN(g->sb_hybrid + pos, RENAME(exp_table)+exponents[pos])

         }

         s_index += 4;

     }

     /* skip extension bits */

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

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

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

         s_index=0;

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

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

         s_index = 0;

     }

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

     skip_bits_long(&s->gb, bits_left);
 

     i = get_bits_count(&s->gb);

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

     return 0;
 }
 

 /* Reorder short blocks from bitstream order to interleaved order. It
    would be faster to do it in parsing, but the code would be far more
    complicated */
 static void reorder_block(MPADecodeContext *s, GranuleDef *g)
 {

     int i, j, len;

     INTFLOAT *ptr, *dst, *ptr1;
     INTFLOAT tmp[576];

 
     if (g->block_type != 2)
         return;
 
     if (g->switch_point) {

         if (s->sample_rate_index != 8)

             ptr = g->sb_hybrid + 36;

         else

             ptr = g->sb_hybrid + 72;

     } else {
         ptr = g->sb_hybrid;
     }

     for (i = g->short_start; i < 13; i++) {
         len  = band_size_short[s->sample_rate_index][i];

         ptr1 = ptr;

         dst  = tmp;
         for (j = len; j > 0; j--) {

             *dst++ = ptr[0*len];
             *dst++ = ptr[1*len];
             *dst++ = ptr[2*len];
             ptr++;

         }

         ptr += 2 * len;

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

     }
 }
 
 #define ISQRT2 FIXR(0.70710678118654752440)
 

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

 {
     int i, j, k, l;

     int sf_max, sf, len, non_zero_found;
     INTFLOAT (*is_tab)[16], *tab0, *tab1, tmp0, tmp1, v1, v2;

     int non_zero_found_short[3];
 
     /* intensity stereo */
     if (s->mode_ext & MODE_EXT_I_STEREO) {
         if (!s->lsf) {
             is_tab = is_table;
             sf_max = 7;
         } else {
             is_tab = is_table_lsf[g1->scalefac_compress & 1];
             sf_max = 16;
         }

         tab0 = g0->sb_hybrid + 576;
         tab1 = g1->sb_hybrid + 576;
 
         non_zero_found_short[0] = 0;
         non_zero_found_short[1] = 0;
         non_zero_found_short[2] = 0;
         k = (13 - g1->short_start) * 3 + g1->long_end - 3;

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

             /* for last band, use previous scale factor */
             if (i != 11)
                 k -= 3;
             len = band_size_short[s->sample_rate_index][i];

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

                 tab0 -= len;
                 tab1 -= len;
                 if (!non_zero_found_short[l]) {
                     /* test if non zero band. if so, stop doing i-stereo */

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

                         if (tab1[j] != 0) {
                             non_zero_found_short[l] = 1;
                             goto found1;
                         }
                     }
                     sf = g1->scale_factors[k + l];
                     if (sf >= sf_max)
                         goto found1;
 
                     v1 = is_tab[0][sf];
                     v2 = is_tab[1][sf];

                     for (j = 0; j < len; j++) {
                         tmp0    = tab0[j];

                         tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
                         tab1[j] = MULLx(tmp0, v2, FRAC_BITS);

                     }
                 } else {

 found1:

                     if (s->mode_ext & MODE_EXT_MS_STEREO) {
                         /* lower part of the spectrum : do ms stereo
                            if enabled */

                         for (j = 0; j < len; j++) {
                             tmp0    = tab0[j];
                             tmp1    = tab1[j];

                             tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
                             tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);

                         }
                     }
                 }
             }
         }
 

         non_zero_found = non_zero_found_short[0] |

                          non_zero_found_short[1] |
                          non_zero_found_short[2];

         for (i = g1->long_end - 1;i >= 0;i--) {
             len   = band_size_long[s->sample_rate_index][i];

             tab0 -= len;
             tab1 -= len;
             /* test if non zero band. if so, stop doing i-stereo */
             if (!non_zero_found) {

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

                     if (tab1[j] != 0) {
                         non_zero_found = 1;
                         goto found2;
                     }
                 }
                 /* for last band, use previous scale factor */

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

                 sf = g1->scale_factors[k];
                 if (sf >= sf_max)
                     goto found2;
                 v1 = is_tab[0][sf];
                 v2 = is_tab[1][sf];

                 for (j = 0; j < len; j++) {
                     tmp0    = tab0[j];

                     tab0[j] = MULLx(tmp0, v1, FRAC_BITS);
                     tab1[j] = MULLx(tmp0, v2, FRAC_BITS);

                 }
             } else {

 found2:

                 if (s->mode_ext & MODE_EXT_MS_STEREO) {
                     /* lower part of the spectrum : do ms stereo
                        if enabled */

                     for (j = 0; j < len; j++) {
                         tmp0    = tab0[j];
                         tmp1    = tab1[j];

                         tab0[j] = MULLx(tmp0 + tmp1, ISQRT2, FRAC_BITS);
                         tab1[j] = MULLx(tmp0 - tmp1, ISQRT2, FRAC_BITS);

                     }
                 }
             }
         }
     } else if (s->mode_ext & MODE_EXT_MS_STEREO) {
         /* ms stereo ONLY */
         /* NOTE: the 1/sqrt(2) normalization factor is included in the
            global gain */

 #if CONFIG_FLOAT
        s-> dsp.butterflies_float(g0->sb_hybrid, g1->sb_hybrid, 576);
 #else

         tab0 = g0->sb_hybrid;
         tab1 = g1->sb_hybrid;

         for (i = 0; i < 576; i++) {
             tmp0    = tab0[i];
             tmp1    = tab1[i];

             tab0[i] = tmp0 + tmp1;
             tab1[i] = tmp0 - tmp1;
         }

 #endif

     }
 }
 

 #if CONFIG_FLOAT

 #if HAVE_MIPSFPU
 #   include "mips/compute_antialias_float.h"
 #endif /* HAVE_MIPSFPU */
 #else
 #if HAVE_MIPSDSPR1
 #   include "mips/compute_antialias_fixed.h"
 #endif /* HAVE_MIPSDSPR1 */
 #endif /* CONFIG_FLOAT */
 
 #ifndef compute_antialias
 #if CONFIG_FLOAT

 #define AA(j) do {                                                      \
         float tmp0 = ptr[-1-j];                                         \
         float tmp1 = ptr[   j];                                         \
         ptr[-1-j] = tmp0 * csa_table[j][0] - tmp1 * csa_table[j][1];    \
         ptr[   j] = tmp0 * csa_table[j][1] + tmp1 * csa_table[j][0];    \
     } while (0)
 #else
 #define AA(j) do {                                              \
         int tmp0 = ptr[-1-j];                                   \
         int tmp1 = ptr[   j];                                   \
         int tmp2 = MULH(tmp0 + tmp1, csa_table[j][0]);          \

         ptr[-1-j] = 4 * (tmp2 - MULH(tmp1, csa_table[j][2]));   \
         ptr[   j] = 4 * (tmp2 + MULH(tmp0, csa_table[j][3]));   \

     } while (0)
 #endif
 
 static void compute_antialias(MPADecodeContext *s, GranuleDef *g)

 {

     INTFLOAT *ptr;

     int n, i;

 
     /* we antialias only "long" bands */
     if (g->block_type == 2) {
         if (!g->switch_point)
             return;
         /* XXX: check this for 8000Hz case */
         n = 1;
     } else {
         n = SBLIMIT - 1;
     }

     ptr = g->sb_hybrid + 18;

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

         AA(0);
         AA(1);
         AA(2);
         AA(3);
         AA(4);
         AA(5);
         AA(6);
         AA(7);

 
         ptr += 18;

     }
 }

 #endif /* compute_antialias */

 static void compute_imdct(MPADecodeContext *s, GranuleDef *g,
                           INTFLOAT *sb_samples, INTFLOAT *mdct_buf)

 {

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

     INTFLOAT out2[12];
     int i, j, mdct_long_end, sblimit;

 
     /* find last non zero block */

     ptr  = g->sb_hybrid + 576;

     ptr1 = g->sb_hybrid + 2 * 18;
     while (ptr >= ptr1) {

         int32_t *p;

         ptr -= 6;

         p    = (int32_t*)ptr;
         if (p[0] | p[1] | p[2] | p[3] | p[4] | p[5])

             break;
     }
     sblimit = ((ptr - g->sb_hybrid) / 18) + 1;
 
     if (g->block_type == 2) {
         /* XXX: check for 8000 Hz */
         if (g->switch_point)
             mdct_long_end = 2;
         else
             mdct_long_end = 0;
     } else {
         mdct_long_end = sblimit;
     }
 

     s->mpadsp.RENAME(imdct36_blocks)(sb_samples, mdct_buf, g->sb_hybrid,
                                      mdct_long_end, g->switch_point,
                                      g->block_type);
 
     buf = mdct_buf + 4*18*(mdct_long_end >> 2) + (mdct_long_end & 3);
     ptr = g->sb_hybrid + 18 * mdct_long_end;
 

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

         /* select frequency inversion */

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

         out_ptr = sb_samples + j;

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

             *out_ptr = buf[4*i];

             out_ptr += SBLIMIT;
         }
         imdct12(out2, ptr + 0);

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

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

             out_ptr += SBLIMIT;
         }

         imdct12(out2, ptr + 1);

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

             *out_ptr     = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*2)];
             buf[4*(i + 6*0)] = MULH3(out2[i + 6], win[i + 6], 1);

             out_ptr += SBLIMIT;
         }
         imdct12(out2, ptr + 2);

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

             buf[4*(i + 6*0)] = MULH3(out2[i    ], win[i    ], 1) + buf[4*(i + 6*0)];
             buf[4*(i + 6*1)] = MULH3(out2[i + 6], win[i + 6], 1);
             buf[4*(i + 6*2)] = 0;

         }

         ptr += 18;

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

     }
     /* zero bands */

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

         /* overlap */
         out_ptr = sb_samples + j;

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

             *out_ptr = buf[4*i];
             buf[4*i]   = 0;

             out_ptr += SBLIMIT;
         }

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

     }
 }
 
 /* main layer3 decoding function */
 static int mp_decode_layer3(MPADecodeContext *s)
 {

     int nb_granules, main_data_begin;

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

     GranuleDef *g;

     int16_t exponents[576]; //FIXME try INTFLOAT

 
     /* read side info */
     if (s->lsf) {
         main_data_begin = get_bits(&s->gb, 8);

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

         nb_granules = 1;
     } else {
         main_data_begin = get_bits(&s->gb, 9);
         if (s->nb_channels == 2)

             skip_bits(&s->gb, 3);

         else

             skip_bits(&s->gb, 5);

         nb_granules = 2;

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

             s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */
             s->granules[ch][1].scfsi = get_bits(&s->gb, 4);

         }
     }

     for (gr = 0; gr < nb_granules; gr++) {
         for (ch = 0; ch < s->nb_channels; ch++) {

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

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

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

             g->big_values     = get_bits(&s->gb,  9);
             if (g->big_values > 288) {

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

                 return AVERROR_INVALIDDATA;

             }
 

             g->global_gain = get_bits(&s->gb, 8);
             /* if MS stereo only is selected, we precompute the
                1/sqrt(2) renormalization factor */

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

                 MODE_EXT_MS_STEREO)
                 g->global_gain -= 2;
             if (s->lsf)
                 g->scalefac_compress = get_bits(&s->gb, 9);
             else
                 g->scalefac_compress = get_bits(&s->gb, 4);

             blocksplit_flag = get_bits1(&s->gb);

             if (blocksplit_flag) {
                 g->block_type = get_bits(&s->gb, 2);

                 if (g->block_type == 0) {

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

                     return AVERROR_INVALIDDATA;

                 }

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

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

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

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

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

                 ff_init_short_region(s, g);

             } else {

                 int region_address1, region_address2;

                 g->block_type = 0;
                 g->switch_point = 0;

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

                     g->table_select[i] = get_bits(&s->gb, 5);
                 /* compute huffman coded region sizes */
                 region_address1 = get_bits(&s->gb, 4);
                 region_address2 = get_bits(&s->gb, 3);

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

                         region_address1, region_address2);

                 ff_init_long_region(s, g, region_address1, region_address2);

             }

             ff_region_offset2size(g);
             ff_compute_band_indexes(s, g);

             g->preflag = 0;
             if (!s->lsf)

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

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

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

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

                     g->block_type, g->switch_point);
         }
     }
 

     if (!s->adu_mode) {

         int skip;

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

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

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

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

         av_dlog(s->avctx, "seekback:%d, lastbuf:%d\n",
                 main_data_begin, s->last_buf_size);

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

         s->in_gb = s->gb;

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

 #if !UNCHECKED_BITSTREAM_READER

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

 #endif

         s->last_buf_size <<= 3;
         for (gr = 0; gr < nb_granules && (s->last_buf_size >> 3) < main_data_begin; gr++) {
             for (ch = 0; ch < s->nb_channels; ch++) {
                 g = &s->granules[ch][gr];
                 s->last_buf_size += g->part2_3_length;
                 memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));

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

             }
         }
         skip = s->last_buf_size - 8 * main_data_begin;
         if (skip >= s->gb.size_in_bits && s->in_gb.buffer) {
             skip_bits_long(&s->in_gb, skip - s->gb.size_in_bits);
             s->gb           = s->in_gb;
             s->in_gb.buffer = NULL;
         } else {
             skip_bits_long(&s->gb, skip);
         }
     } else {
         gr = 0;

     }

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

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

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

             bits_pos = get_bits_count(&s->gb);

             if (!s->lsf) {

                 uint8_t *sc;

                 int slen, slen1, slen2;
 
                 /* MPEG1 scale factors */
                 slen1 = slen_table[0][g->scalefac_compress];
                 slen2 = slen_table[1][g->scalefac_compress];

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

                 if (g->block_type == 2) {
                     n = g->switch_point ? 17 : 18;
                     j = 0;

                     if (slen1) {
                         for (i = 0; i < n; i++)

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

                     } else {
                         for (i = 0; i < n; i++)

                             g->scale_factors[j++] = 0;
                     }

                     if (slen2) {
                         for (i = 0; i < 18; i++)

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

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

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

                     } else {
                         for (i = 0; i < 21; i++)

                             g->scale_factors[j++] = 0;
                     }

                 } else {

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

                     j = 0;

                     for (k = 0; k < 4; k++) {
                         n = k == 0 ? 6 : 5;

                         if ((g->scfsi & (0x8 >> k)) == 0) {
                             slen = (k < 2) ? slen1 : slen2;

                             if (slen) {
                                 for (i = 0; i < n; i++)

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

                             } else {
                                 for (i = 0; i < n; i++)

                                     g->scale_factors[j++] = 0;
                             }

                         } else {
                             /* simply copy from last granule */

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

                                 g->scale_factors[j] = sc[j];
                                 j++;
                             }
                         }
                     }
                     g->scale_factors[j++] = 0;
                 }
             } else {
                 int tindex, tindex2, slen[4], sl, sf;
 
                 /* LSF scale factors */

                 if (g->block_type == 2)

                     tindex = g->switch_point ? 2 : 1;

                 else

                     tindex = 0;

                 sf = g->scalefac_compress;
                 if ((s->mode_ext & MODE_EXT_I_STEREO) && ch == 1) {
                     /* intensity stereo case */
                     sf >>= 1;
                     if (sf < 180) {
                         lsf_sf_expand(slen, sf, 6, 6, 0);
                         tindex2 = 3;
                     } else if (sf < 244) {
                         lsf_sf_expand(slen, sf - 180, 4, 4, 0);
                         tindex2 = 4;
                     } else {
                         lsf_sf_expand(slen, sf - 244, 3, 0, 0);
                         tindex2 = 5;
                     }
                 } else {
                     /* normal case */
                     if (sf < 400) {
                         lsf_sf_expand(slen, sf, 5, 4, 4);
                         tindex2 = 0;
                     } else if (sf < 500) {
                         lsf_sf_expand(slen, sf - 400, 5, 4, 0);
                         tindex2 = 1;
                     } else {
                         lsf_sf_expand(slen, sf - 500, 3, 0, 0);
                         tindex2 = 2;
                         g->preflag = 1;
                     }
                 }
 
                 j = 0;

                 for (k = 0; k < 4; k++) {
                     n  = lsf_nsf_table[tindex2][tindex][k];

                     sl = slen[k];

                     if (sl) {
                         for (i = 0; i < n; i++)

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

                     } else {
                         for (i = 0; i < n; i++)

                             g->scale_factors[j++] = 0;
                     }

                 }
                 /* XXX: should compute exact size */

                 for (; j < 40; j++)

                     g->scale_factors[j] = 0;
             }
 
             exponents_from_scale_factors(s, g, exponents);
 
             /* read Huffman coded residue */

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

         } /* ch */
 

         if (s->mode == MPA_JSTEREO)

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

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

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

 
             reorder_block(s, g);

             compute_antialias(s, g);

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

         }
     } /* gr */

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

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

     return nb_granules * 18;
 }
 

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

                            const uint8_t *buf, int buf_size)

 {

     int i, nb_frames, ch, ret;

     OUT_INT *samples_ptr;

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

     /* skip error protection field */
     if (s->error_protection)

         skip_bits(&s->gb, 16);

 
     switch(s->layer) {
     case 1:

         s->avctx->frame_size = 384;

         nb_frames = mp_decode_layer1(s);
         break;
     case 2:

         s->avctx->frame_size = 1152;

         nb_frames = mp_decode_layer2(s);
         break;
     case 3:

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

     default:
         nb_frames = mp_decode_layer3(s);

         s->last_buf_size=0;

         if (s->in_gb.buffer) {

             align_get_bits(&s->gb);

             i = get_bits_left(&s->gb)>>3;
             if (i >= 0 && i <= BACKSTEP_SIZE) {

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

             } else

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

             s->gb           = s->in_gb;
             s->in_gb.buffer = NULL;

         }
 

         align_get_bits(&s->gb);

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

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

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

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

             i = FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);

         }

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

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

         s->last_buf_size += i;

     }

     if(nb_frames < 0)
         return nb_frames;
 

     /* get output buffer */
     if (!samples) {
         s->frame.nb_samples = s->avctx->frame_size;

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

             av_log(s->avctx, AV_LOG_ERROR, "get_buffer() failed\n");
             return ret;
         }

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

     }

     /* apply the synthesis filter */

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

         int sample_stride;
         if (s->avctx->sample_fmt == OUT_FMT_P) {
             samples_ptr   = samples[ch];
             sample_stride = 1;
         } else {
             samples_ptr   = samples[0] + ch;
             sample_stride = s->nb_channels;
         }

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

             RENAME(ff_mpa_synth_filter)(&s->mpadsp, s->synth_buf[ch],
                                         &(s->synth_buf_offset[ch]),
                                         RENAME(ff_mpa_synth_window),
                                         &s->dither_state, samples_ptr,
                                         sample_stride, s->sb_samples[ch][i]);
             samples_ptr += 32 * sample_stride;

         }
     }

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

 }
 

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

                         AVPacket *avpkt)

 {

     const uint8_t *buf  = avpkt->data;
     int buf_size        = avpkt->size;

     MPADecodeContext *s = avctx->priv_data;

     uint32_t header;

     int ret;

     while(buf_size && !*buf){

         buf++;

         buf_size--;
     }

     if (buf_size < HEADER_SIZE)

         return AVERROR_INVALIDDATA;

     header = AV_RB32(buf);

     if (header>>8 == AV_RB32("TAG")>>8) {
         av_log(avctx, AV_LOG_DEBUG, "discarding ID3 tag\n");
         return buf_size;
     }

     if (ff_mpa_check_header(header) < 0) {

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

         return AVERROR_INVALIDDATA;

     }
 

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

         /* free format: prepare to compute frame size */
         s->frame_size = -1;

         return AVERROR_INVALIDDATA;

     }
     /* update codec info */

     avctx->channels       = s->nb_channels;

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

     if (!avctx->bit_rate)
         avctx->bit_rate = s->bit_rate;

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

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

         return AVERROR_INVALIDDATA;

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

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

         buf_size= s->frame_size;

     }

     ret = mp_decode_frame(s, NULL, buf, buf_size);
     if (ret >= 0) {

         *got_frame_ptr   = 1;
         *(AVFrame *)data = s->frame;

         avctx->sample_rate = s->sample_rate;
         //FIXME maybe move the other codec info stuff from above here too

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

         /* Only return an error if the bad frame makes up the whole packet or
          * the error is related to buffer management.
          * If there is more data in the packet, just consume the bad frame
          * instead of returning an error, which would discard the whole
          * packet. */

         *got_frame_ptr = 0;

         if (buf_size == avpkt->size || ret != AVERROR_INVALIDDATA)
             return ret;

     }

     s->frame_size = 0;

     return buf_size;

 }
 

 static void mp_flush(MPADecodeContext *ctx)
 {
     memset(ctx->synth_buf, 0, sizeof(ctx->synth_buf));
     ctx->last_buf_size = 0;
 }
 

 static void flush(AVCodecContext *avctx)
 {

     mp_flush(avctx->priv_data);

 }
 

 #if CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER

 static int decode_frame_adu(AVCodecContext *avctx, void *data,
                             int *got_frame_ptr, AVPacket *avpkt)

 {

     const uint8_t *buf  = avpkt->data;
     int buf_size        = avpkt->size;

     MPADecodeContext *s = avctx->priv_data;
     uint32_t header;

     int len, ret;

     int av_unused out_size;

 
     len = buf_size;
 
     // Discard too short frames
     if (buf_size < HEADER_SIZE) {

         av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
         return AVERROR_INVALIDDATA;

     }
 
 
     if (len > MPA_MAX_CODED_FRAME_SIZE)
         len = MPA_MAX_CODED_FRAME_SIZE;
 
     // Get header and restore sync word

     header = AV_RB32(buf) | 0xffe00000;

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

         av_log(avctx, AV_LOG_ERROR, "Invalid frame header\n");
         return AVERROR_INVALIDDATA;

     }
 

     avpriv_mpegaudio_decode_header((MPADecodeHeader *)s, header);

     /* update codec info */
     avctx->sample_rate = s->sample_rate;

     avctx->channels    = s->nb_channels;

     if (!avctx->bit_rate)
         avctx->bit_rate = s->bit_rate;

     s->frame_size = len;

     ret = mp_decode_frame(s, NULL, buf, buf_size);
     if (ret < 0) {

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

         return ret;

     }

 
     *got_frame_ptr   = 1;
     *(AVFrame *)data = s->frame;

 
     return buf_size;
 }

 #endif /* CONFIG_MP3ADU_DECODER || CONFIG_MP3ADUFLOAT_DECODER */

 #if CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER

 /**
  * Context for MP3On4 decoder
  */
 typedef struct MP3On4DecodeContext {

     AVFrame *frame;

     int frames;                     ///< number of mp3 frames per block (number of mp3 decoder instances)
     int syncword;                   ///< syncword patch
     const uint8_t *coff;            ///< channel offsets in output buffer

     MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
 } MP3On4DecodeContext;
 

 #include "mpeg4audio.h"
 

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

 
 /* number of mp3 decoder instances */
 static const uint8_t mp3Frames[8] = { 0, 1, 1, 2, 3, 3, 4, 5 };
 

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

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

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

 };
 

 /* mp3on4 channel layouts */
 static const int16_t chan_layout[8] = {
     0,
     AV_CH_LAYOUT_MONO,
     AV_CH_LAYOUT_STEREO,
     AV_CH_LAYOUT_SURROUND,
     AV_CH_LAYOUT_4POINT0,
     AV_CH_LAYOUT_5POINT0,
     AV_CH_LAYOUT_5POINT1,
     AV_CH_LAYOUT_7POINT1
 };

 static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
 {
     MP3On4DecodeContext *s = avctx->priv_data;
     int i;
 
     for (i = 0; i < s->frames; i++)
         av_free(s->mp3decctx[i]);
 
     return 0;
 }
 

 
 static int decode_init_mp3on4(AVCodecContext * avctx)
 {
     MP3On4DecodeContext *s = avctx->priv_data;

     MPEG4AudioConfig cfg;

     int i;
 
     if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
         av_log(avctx, AV_LOG_ERROR, "Codec extradata missing or too short.\n");

         return AVERROR_INVALIDDATA;

     }
 

     avpriv_mpeg4audio_get_config(&cfg, avctx->extradata,
                                  avctx->extradata_size * 8, 1);

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

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

         return AVERROR_INVALIDDATA;

     }

     s->frames             = mp3Frames[cfg.chan_config];
     s->coff               = chan_offset[cfg.chan_config];
     avctx->channels       = ff_mpeg4audio_channels[cfg.chan_config];

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

     if (cfg.sample_rate < 16000)
         s->syncword = 0xffe00000;
     else
         s->syncword = 0xfff00000;
 

     /* Init the first mp3 decoder in standard way, so that all tables get builded
      * We replace avctx->priv_data with the context of the first decoder so that
      * decode_init() does not have to be changed.

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

      */
     // Allocate zeroed memory for the first decoder context
     s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));

     if (!s->mp3decctx[0])
         goto alloc_fail;

     // Put decoder context in place to make init_decode() happy
     avctx->priv_data = s->mp3decctx[0];
     decode_init(avctx);

     s->frame = avctx->coded_frame;

     // Restore mp3on4 context pointer
     avctx->priv_data = s;
     s->mp3decctx[0]->adu_mode = 1; // Set adu mode
 
     /* Create a separate codec/context for each frame (first is already ok).
      * Each frame is 1 or 2 channels - up to 5 frames allowed
      */
     for (i = 1; i < s->frames; i++) {
         s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));

         if (!s->mp3decctx[i])
             goto alloc_fail;

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

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

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

     }
 
     return 0;

 alloc_fail:
     decode_close_mp3on4(avctx);
     return AVERROR(ENOMEM);

 }
 
 

 static void flush_mp3on4(AVCodecContext *avctx)

 {
     int i;

     MP3On4DecodeContext *s = avctx->priv_data;

     for (i = 0; i < s->frames; i++)
         mp_flush(s->mp3decctx[i]);

 }
 
 

 static int decode_frame_mp3on4(AVCodecContext *avctx, void *data,
                                int *got_frame_ptr, AVPacket *avpkt)

 {

     const uint8_t *buf     = avpkt->data;
     int buf_size           = avpkt->size;

     MP3On4DecodeContext *s = avctx->priv_data;
     MPADecodeContext *m;

     int fsize, len = buf_size, out_size = 0;

     uint32_t header;

     OUT_INT **out_samples;
     OUT_INT *outptr[2];
     int fr, ch, ret;

     /* get output buffer */

     s->frame->nb_samples = MPA_FRAME_SIZE;

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

         av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
         return ret;

     }

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

     // Discard too short frames

     if (buf_size < HEADER_SIZE)

         return AVERROR_INVALIDDATA;

     avctx->bit_rate = 0;
 

     ch = 0;

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

         fsize = AV_RB16(buf) >> 4;

         fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);

         m     = s->mp3decctx[fr];

         av_assert1(m);

         if (fsize < HEADER_SIZE) {
             av_log(avctx, AV_LOG_ERROR, "Frame size smaller than header size\n");
             return AVERROR_INVALIDDATA;
         }

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

         if (ff_mpa_check_header(header) < 0) // Bad header, discard block
             break;

         avpriv_mpegaudio_decode_header((MPADecodeHeader *)m, header);

         if (ch + m->nb_channels > avctx->channels ||
             s->coff[fr] + m->nb_channels > avctx->channels) {

             av_log(avctx, AV_LOG_ERROR, "frame channel count exceeds codec "
                                         "channel count\n");
             return AVERROR_INVALIDDATA;
         }
         ch += m->nb_channels;
 

         outptr[0] = out_samples[s->coff[fr]];
         if (m->nb_channels > 1)
             outptr[1] = out_samples[s->coff[fr] + 1];
 

         if ((ret = mp_decode_frame(m, outptr, buf, fsize)) < 0)
             return ret;
 
         out_size += ret;

         buf      += fsize;
         len      -= fsize;

         avctx->bit_rate += m->bit_rate;

     }
 
     /* update codec info */
     avctx->sample_rate = s->mp3decctx[0]->sample_rate;
 

     s->frame->nb_samples = out_size / (avctx->channels * sizeof(OUT_INT));
     *got_frame_ptr   = 1;
     *(AVFrame *)data = *s->frame;
 

     return buf_size;
 }

 #endif /* CONFIG_MP3ON4_DECODER || CONFIG_MP3ON4FLOAT_DECODER */

 #if !CONFIG_FLOAT

 #if CONFIG_MP1_DECODER

 AVCodec ff_mp1_decoder = {
     .name           = "mp1",
     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_MP1,

     .priv_data_size = sizeof(MPADecodeContext),
     .init           = decode_init,
     .decode         = decode_frame,

     .capabilities   = CODEC_CAP_DR1,

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

     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
                                                       AV_SAMPLE_FMT_S16,
                                                       AV_SAMPLE_FMT_NONE },

 };
 #endif

 #if CONFIG_MP2_DECODER

 AVCodec ff_mp2_decoder = {
     .name           = "mp2",
     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_MP2,

     .priv_data_size = sizeof(MPADecodeContext),
     .init           = decode_init,
     .decode         = decode_frame,

     .capabilities   = CODEC_CAP_DR1,

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

     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
                                                       AV_SAMPLE_FMT_S16,
                                                       AV_SAMPLE_FMT_NONE },

 };

 #endif

 #if CONFIG_MP3_DECODER

 AVCodec ff_mp3_decoder = {
     .name           = "mp3",
     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_MP3,

     .priv_data_size = sizeof(MPADecodeContext),
     .init           = decode_init,
     .decode         = decode_frame,

     .capabilities   = CODEC_CAP_DR1,

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

     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
                                                       AV_SAMPLE_FMT_S16,
                                                       AV_SAMPLE_FMT_NONE },

 };

 #endif

 #if CONFIG_MP3ADU_DECODER

 AVCodec ff_mp3adu_decoder = {
     .name           = "mp3adu",
     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_MP3ADU,

     .priv_data_size = sizeof(MPADecodeContext),
     .init           = decode_init,
     .decode         = decode_frame_adu,

     .capabilities   = CODEC_CAP_DR1,

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

     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
                                                       AV_SAMPLE_FMT_S16,
                                                       AV_SAMPLE_FMT_NONE },

 };

 #endif

 #if CONFIG_MP3ON4_DECODER

 AVCodec ff_mp3on4_decoder = {
     .name           = "mp3on4",
     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_MP3ON4,

     .priv_data_size = sizeof(MP3On4DecodeContext),
     .init           = decode_init_mp3on4,
     .close          = decode_close_mp3on4,
     .decode         = decode_frame_mp3on4,

     .capabilities   = CODEC_CAP_DR1,

     .flush          = flush_mp3on4,

     .long_name      = NULL_IF_CONFIG_SMALL("MP3onMP4"),

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

 };

 #endif

 #endif