/*
  * The simplest mpeg audio layer 2 encoder

  * Copyright (c) 2000, 2001 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

  * The simplest mpeg audio layer 2 encoder.
  */
 
 #include "avcodec.h"

 #include "internal.h"

 #include "put_bits.h"

 #define FRAC_BITS   15   /* fractional bits for sb_samples and dct */
 #define WFRAC_BITS  14   /* fractional bits for window */
 

 #include "mpegaudio.h"
 
 /* currently, cannot change these constants (need to modify
    quantization stage) */
 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
 
 #define SAMPLES_BUF_SIZE 4096
 
 typedef struct MpegAudioContext {
     PutBitContext pb;
     int nb_channels;
     int lsf;           /* 1 if mpeg2 low bitrate selected */
     int bitrate_index; /* bit rate */
     int freq_index;
     int frame_size; /* frame size, in bits, without padding */
     /* padding computation */
     int frame_frac, frame_frac_incr, do_padding;
     short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
     int samples_offset[MPA_MAX_CHANNELS];       /* offset in samples_buf */
     int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
     unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
     /* code to group 3 scale factors */
     unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
     int sblimit; /* number of used subbands */
     const unsigned char *alloc_table;
 } MpegAudioContext;
 
 /* define it to use floats in quantization (I don't like floats !) */

 #define USE_FLOATS

 
 #include "mpegaudiodata.h"
 #include "mpegaudiotab.h"
 

 static av_cold int MPA_encode_init(AVCodecContext *avctx)

 {
     MpegAudioContext *s = avctx->priv_data;
     int freq = avctx->sample_rate;
     int bitrate = avctx->bit_rate;
     int channels = avctx->channels;
     int i, v, table;
     float a;
 
     if (channels <= 0 || channels > 2){
         av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
         return -1;
     }
     bitrate = bitrate / 1000;
     s->nb_channels = channels;
     avctx->frame_size = MPA_FRAME_SIZE;
 
     /* encoding freq */
     s->lsf = 0;
     for(i=0;i<3;i++) {

         if (avpriv_mpa_freq_tab[i] == freq)

             break;

         if ((avpriv_mpa_freq_tab[i] / 2) == freq) {

             s->lsf = 1;
             break;
         }
     }
     if (i == 3){
         av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
         return -1;
     }
     s->freq_index = i;
 
     /* encoding bitrate & frequency */
     for(i=0;i<15;i++) {

         if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate)

             break;
     }
     if (i == 15){
         av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
         return -1;
     }
     s->bitrate_index = i;
 
     /* compute total header size & pad bit */
 
     a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
     s->frame_size = ((int)a) * 8;
 
     /* frame fractional size to compute padding */
     s->frame_frac = 0;
     s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
 
     /* select the right allocation table */
     table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
 
     /* number of used subbands */
     s->sblimit = ff_mpa_sblimit_table[table];
     s->alloc_table = ff_mpa_alloc_tables[table];
 

     av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",

             bitrate, freq, s->frame_size, table, s->frame_frac_incr);

 
     for(i=0;i<s->nb_channels;i++)
         s->samples_offset[i] = 0;
 
     for(i=0;i<257;i++) {
         int v;
         v = ff_mpa_enwindow[i];
 #if WFRAC_BITS != 16
         v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
 #endif
         filter_bank[i] = v;
         if ((i & 63) != 0)
             v = -v;
         if (i != 0)
             filter_bank[512 - i] = v;
     }
 
     for(i=0;i<64;i++) {
         v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
         if (v <= 0)
             v = 1;
         scale_factor_table[i] = v;
 #ifdef USE_FLOATS
         scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
 #else
 #define P 15
         scale_factor_shift[i] = 21 - P - (i / 3);
         scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
 #endif
     }
     for(i=0;i<128;i++) {
         v = i - 64;
         if (v <= -3)
             v = 0;
         else if (v < 0)
             v = 1;
         else if (v == 0)
             v = 2;
         else if (v < 3)
             v = 3;
         else
             v = 4;
         scale_diff_table[i] = v;
     }
 
     for(i=0;i<17;i++) {
         v = ff_mpa_quant_bits[i];
         if (v < 0)
             v = -v;
         else
             v = v * 3;
         total_quant_bits[i] = 12 * v;
     }
 
     avctx->coded_frame= avcodec_alloc_frame();
     avctx->coded_frame->key_frame= 1;
 
     return 0;
 }
 
 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
 static void idct32(int *out, int *tab)
 {
     int i, j;
     int *t, *t1, xr;
     const int *xp = costab32;
 
     for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
 
     t = tab + 30;
     t1 = tab + 2;
     do {
         t[0] += t[-4];
         t[1] += t[1 - 4];
         t -= 4;
     } while (t != t1);
 
     t = tab + 28;
     t1 = tab + 4;
     do {
         t[0] += t[-8];
         t[1] += t[1-8];
         t[2] += t[2-8];
         t[3] += t[3-8];
         t -= 8;
     } while (t != t1);
 
     t = tab;
     t1 = tab + 32;
     do {
         t[ 3] = -t[ 3];
         t[ 6] = -t[ 6];
 
         t[11] = -t[11];
         t[12] = -t[12];
         t[13] = -t[13];
         t[15] = -t[15];
         t += 16;
     } while (t != t1);
 
 
     t = tab;
     t1 = tab + 8;
     do {
         int x1, x2, x3, x4;
 
         x3 = MUL(t[16], FIX(SQRT2*0.5));
         x4 = t[0] - x3;
         x3 = t[0] + x3;
 
         x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
         x1 = MUL((t[8] - x2), xp[0]);
         x2 = MUL((t[8] + x2), xp[1]);
 
         t[ 0] = x3 + x1;
         t[ 8] = x4 - x2;
         t[16] = x4 + x2;
         t[24] = x3 - x1;
         t++;
     } while (t != t1);
 
     xp += 2;
     t = tab;
     t1 = tab + 4;
     do {
         xr = MUL(t[28],xp[0]);
         t[28] = (t[0] - xr);
         t[0] = (t[0] + xr);
 
         xr = MUL(t[4],xp[1]);
         t[ 4] = (t[24] - xr);
         t[24] = (t[24] + xr);
 
         xr = MUL(t[20],xp[2]);
         t[20] = (t[8] - xr);
         t[ 8] = (t[8] + xr);
 
         xr = MUL(t[12],xp[3]);
         t[12] = (t[16] - xr);
         t[16] = (t[16] + xr);
         t++;
     } while (t != t1);
     xp += 4;
 
     for (i = 0; i < 4; i++) {
         xr = MUL(tab[30-i*4],xp[0]);
         tab[30-i*4] = (tab[i*4] - xr);
         tab[   i*4] = (tab[i*4] + xr);
 
         xr = MUL(tab[ 2+i*4],xp[1]);
         tab[ 2+i*4] = (tab[28-i*4] - xr);
         tab[28-i*4] = (tab[28-i*4] + xr);
 
         xr = MUL(tab[31-i*4],xp[0]);
         tab[31-i*4] = (tab[1+i*4] - xr);
         tab[ 1+i*4] = (tab[1+i*4] + xr);
 
         xr = MUL(tab[ 3+i*4],xp[1]);
         tab[ 3+i*4] = (tab[29-i*4] - xr);
         tab[29-i*4] = (tab[29-i*4] + xr);
 
         xp += 2;
     }
 
     t = tab + 30;
     t1 = tab + 1;
     do {
         xr = MUL(t1[0], *xp);
         t1[0] = (t[0] - xr);
         t[0] = (t[0] + xr);
         t -= 2;
         t1 += 2;
         xp++;
     } while (t >= tab);
 
     for(i=0;i<32;i++) {
         out[i] = tab[bitinv32[i]];
     }
 }
 
 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
 

 static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)

 {
     short *p, *q;
     int sum, offset, i, j;
     int tmp[64];
     int tmp1[32];
     int *out;
 
     offset = s->samples_offset[ch];
     out = &s->sb_samples[ch][0][0][0];
     for(j=0;j<36;j++) {
         /* 32 samples at once */
         for(i=0;i<32;i++) {
             s->samples_buf[ch][offset + (31 - i)] = samples[0];
             samples += incr;
         }
 
         /* filter */
         p = s->samples_buf[ch] + offset;
         q = filter_bank;
         /* maxsum = 23169 */
         for(i=0;i<64;i++) {
             sum = p[0*64] * q[0*64];
             sum += p[1*64] * q[1*64];
             sum += p[2*64] * q[2*64];
             sum += p[3*64] * q[3*64];
             sum += p[4*64] * q[4*64];
             sum += p[5*64] * q[5*64];
             sum += p[6*64] * q[6*64];
             sum += p[7*64] * q[7*64];
             tmp[i] = sum;
             p++;
             q++;
         }
         tmp1[0] = tmp[16] >> WSHIFT;
         for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
         for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
 
         idct32(out, tmp1);
 
         /* advance of 32 samples */
         offset -= 32;
         out += 32;
         /* handle the wrap around */
         if (offset < 0) {
             memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
                     s->samples_buf[ch], (512 - 32) * 2);
             offset = SAMPLES_BUF_SIZE - 512;
         }
     }
     s->samples_offset[ch] = offset;
 }
 
 static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
                                   unsigned char scale_factors[SBLIMIT][3],
                                   int sb_samples[3][12][SBLIMIT],
                                   int sblimit)
 {
     int *p, vmax, v, n, i, j, k, code;
     int index, d1, d2;
     unsigned char *sf = &scale_factors[0][0];
 
     for(j=0;j<sblimit;j++) {
         for(i=0;i<3;i++) {
             /* find the max absolute value */
             p = &sb_samples[i][0][j];
             vmax = abs(*p);
             for(k=1;k<12;k++) {
                 p += SBLIMIT;
                 v = abs(*p);
                 if (v > vmax)
                     vmax = v;
             }
             /* compute the scale factor index using log 2 computations */

             if (vmax > 1) {

                 n = av_log2(vmax);
                 /* n is the position of the MSB of vmax. now
                    use at most 2 compares to find the index */
                 index = (21 - n) * 3 - 3;
                 if (index >= 0) {
                     while (vmax <= scale_factor_table[index+1])
                         index++;
                 } else {
                     index = 0; /* very unlikely case of overflow */
                 }
             } else {
                 index = 62; /* value 63 is not allowed */
             }
 

             av_dlog(NULL, "%2d:%d in=%x %x %d\n",
                     j, i, vmax, scale_factor_table[index], index);

             /* store the scale factor */
             assert(index >=0 && index <= 63);
             sf[i] = index;
         }
 
         /* compute the transmission factor : look if the scale factors
            are close enough to each other */
         d1 = scale_diff_table[sf[0] - sf[1] + 64];
         d2 = scale_diff_table[sf[1] - sf[2] + 64];
 
         /* handle the 25 cases */
         switch(d1 * 5 + d2) {
         case 0*5+0:
         case 0*5+4:
         case 3*5+4:
         case 4*5+0:
         case 4*5+4:
             code = 0;
             break;
         case 0*5+1:
         case 0*5+2:
         case 4*5+1:
         case 4*5+2:
             code = 3;
             sf[2] = sf[1];
             break;
         case 0*5+3:
         case 4*5+3:
             code = 3;
             sf[1] = sf[2];
             break;
         case 1*5+0:
         case 1*5+4:
         case 2*5+4:
             code = 1;
             sf[1] = sf[0];
             break;
         case 1*5+1:
         case 1*5+2:
         case 2*5+0:
         case 2*5+1:
         case 2*5+2:
             code = 2;
             sf[1] = sf[2] = sf[0];
             break;
         case 2*5+3:
         case 3*5+3:
             code = 2;
             sf[0] = sf[1] = sf[2];
             break;
         case 3*5+0:
         case 3*5+1:
         case 3*5+2:
             code = 2;
             sf[0] = sf[2] = sf[1];
             break;
         case 1*5+3:
             code = 2;
             if (sf[0] > sf[2])
               sf[0] = sf[2];
             sf[1] = sf[2] = sf[0];
             break;
         default:

             assert(0); //cannot happen

             code = 0;           /* kill warning */
         }
 

         av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,
                 sf[0], sf[1], sf[2], d1, d2, code);

         scale_code[j] = code;
         sf += 3;
     }
 }
 
 /* The most important function : psycho acoustic module. In this
    encoder there is basically none, so this is the worst you can do,
    but also this is the simpler. */
 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
 {
     int i;
 
     for(i=0;i<s->sblimit;i++) {
         smr[i] = (int)(fixed_smr[i] * 10);
     }
 }
 
 
 #define SB_NOTALLOCATED  0
 #define SB_ALLOCATED     1
 #define SB_NOMORE        2
 
 /* Try to maximize the smr while using a number of bits inferior to
    the frame size. I tried to make the code simpler, faster and
    smaller than other encoders :-) */
 static void compute_bit_allocation(MpegAudioContext *s,
                                    short smr1[MPA_MAX_CHANNELS][SBLIMIT],
                                    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
                                    int *padding)
 {
     int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
     int incr;
     short smr[MPA_MAX_CHANNELS][SBLIMIT];
     unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
     const unsigned char *alloc;
 
     memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
     memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
     memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
 
     /* compute frame size and padding */
     max_frame_size = s->frame_size;
     s->frame_frac += s->frame_frac_incr;
     if (s->frame_frac >= 65536) {
         s->frame_frac -= 65536;
         s->do_padding = 1;
         max_frame_size += 8;
     } else {
         s->do_padding = 0;
     }
 
     /* compute the header + bit alloc size */
     current_frame_size = 32;
     alloc = s->alloc_table;
     for(i=0;i<s->sblimit;i++) {
         incr = alloc[0];
         current_frame_size += incr * s->nb_channels;
         alloc += 1 << incr;
     }
     for(;;) {
         /* look for the subband with the largest signal to mask ratio */
         max_sb = -1;
         max_ch = -1;

         max_smr = INT_MIN;

         for(ch=0;ch<s->nb_channels;ch++) {
             for(i=0;i<s->sblimit;i++) {
                 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
                     max_smr = smr[ch][i];
                     max_sb = i;
                     max_ch = ch;
                 }
             }
         }
         if (max_sb < 0)
             break;

         av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n",
                 current_frame_size, max_frame_size, max_sb, max_ch,
                 bit_alloc[max_ch][max_sb]);

 
         /* find alloc table entry (XXX: not optimal, should use
            pointer table) */
         alloc = s->alloc_table;
         for(i=0;i<max_sb;i++) {
             alloc += 1 << alloc[0];
         }
 
         if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
             /* nothing was coded for this band: add the necessary bits */
             incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
             incr += total_quant_bits[alloc[1]];
         } else {
             /* increments bit allocation */
             b = bit_alloc[max_ch][max_sb];
             incr = total_quant_bits[alloc[b + 1]] -
                 total_quant_bits[alloc[b]];
         }
 
         if (current_frame_size + incr <= max_frame_size) {
             /* can increase size */
             b = ++bit_alloc[max_ch][max_sb];
             current_frame_size += incr;
             /* decrease smr by the resolution we added */
             smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
             /* max allocation size reached ? */
             if (b == ((1 << alloc[0]) - 1))
                 subband_status[max_ch][max_sb] = SB_NOMORE;
             else
                 subband_status[max_ch][max_sb] = SB_ALLOCATED;
         } else {
             /* cannot increase the size of this subband */
             subband_status[max_ch][max_sb] = SB_NOMORE;
         }
     }
     *padding = max_frame_size - current_frame_size;
     assert(*padding >= 0);
 }
 
 /*
  * Output the mpeg audio layer 2 frame. Note how the code is small
  * compared to other encoders :-)
  */
 static void encode_frame(MpegAudioContext *s,
                          unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
                          int padding)
 {
     int i, j, k, l, bit_alloc_bits, b, ch;
     unsigned char *sf;
     int q[3];
     PutBitContext *p = &s->pb;
 
     /* header */
 
     put_bits(p, 12, 0xfff);
     put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
     put_bits(p, 2, 4-2);  /* layer 2 */
     put_bits(p, 1, 1); /* no error protection */
     put_bits(p, 4, s->bitrate_index);
     put_bits(p, 2, s->freq_index);
     put_bits(p, 1, s->do_padding); /* use padding */
     put_bits(p, 1, 0);             /* private_bit */
     put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
     put_bits(p, 2, 0); /* mode_ext */
     put_bits(p, 1, 0); /* no copyright */
     put_bits(p, 1, 1); /* original */
     put_bits(p, 2, 0); /* no emphasis */
 
     /* bit allocation */
     j = 0;
     for(i=0;i<s->sblimit;i++) {
         bit_alloc_bits = s->alloc_table[j];
         for(ch=0;ch<s->nb_channels;ch++) {
             put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
         }
         j += 1 << bit_alloc_bits;
     }
 
     /* scale codes */
     for(i=0;i<s->sblimit;i++) {
         for(ch=0;ch<s->nb_channels;ch++) {
             if (bit_alloc[ch][i])
                 put_bits(p, 2, s->scale_code[ch][i]);
         }
     }
 
     /* scale factors */
     for(i=0;i<s->sblimit;i++) {
         for(ch=0;ch<s->nb_channels;ch++) {
             if (bit_alloc[ch][i]) {
                 sf = &s->scale_factors[ch][i][0];
                 switch(s->scale_code[ch][i]) {
                 case 0:
                     put_bits(p, 6, sf[0]);
                     put_bits(p, 6, sf[1]);
                     put_bits(p, 6, sf[2]);
                     break;
                 case 3:
                 case 1:
                     put_bits(p, 6, sf[0]);
                     put_bits(p, 6, sf[2]);
                     break;
                 case 2:
                     put_bits(p, 6, sf[0]);
                     break;
                 }
             }
         }
     }
 
     /* quantization & write sub band samples */
 
     for(k=0;k<3;k++) {
         for(l=0;l<12;l+=3) {
             j = 0;
             for(i=0;i<s->sblimit;i++) {
                 bit_alloc_bits = s->alloc_table[j];
                 for(ch=0;ch<s->nb_channels;ch++) {
                     b = bit_alloc[ch][i];
                     if (b) {
                         int qindex, steps, m, sample, bits;
                         /* we encode 3 sub band samples of the same sub band at a time */
                         qindex = s->alloc_table[j+b];
                         steps = ff_mpa_quant_steps[qindex];
                         for(m=0;m<3;m++) {
                             sample = s->sb_samples[ch][k][l + m][i];
                             /* divide by scale factor */
 #ifdef USE_FLOATS
                             {
                                 float a;
                                 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
                                 q[m] = (int)((a + 1.0) * steps * 0.5);
                             }
 #else
                             {
                                 int q1, e, shift, mult;
                                 e = s->scale_factors[ch][i][k];
                                 shift = scale_factor_shift[e];
                                 mult = scale_factor_mult[e];
 
                                 /* normalize to P bits */
                                 if (shift < 0)
                                     q1 = sample << (-shift);
                                 else
                                     q1 = sample >> shift;
                                 q1 = (q1 * mult) >> P;
                                 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
                             }
 #endif
                             if (q[m] >= steps)
                                 q[m] = steps - 1;
                             assert(q[m] >= 0 && q[m] < steps);
                         }
                         bits = ff_mpa_quant_bits[qindex];
                         if (bits < 0) {
                             /* group the 3 values to save bits */
                             put_bits(p, -bits,
                                      q[0] + steps * (q[1] + steps * q[2]));
                         } else {
                             put_bits(p, bits, q[0]);
                             put_bits(p, bits, q[1]);
                             put_bits(p, bits, q[2]);
                         }
                     }
                 }
                 /* next subband in alloc table */
                 j += 1 << bit_alloc_bits;
             }
         }
     }
 
     /* padding */
     for(i=0;i<padding;i++)
         put_bits(p, 1, 0);
 
     /* flush */
     flush_put_bits(p);
 }
 
 static int MPA_encode_frame(AVCodecContext *avctx,
                             unsigned char *frame, int buf_size, void *data)
 {
     MpegAudioContext *s = avctx->priv_data;

     const short *samples = data;

     short smr[MPA_MAX_CHANNELS][SBLIMIT];
     unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
     int padding, i;
 
     for(i=0;i<s->nb_channels;i++) {
         filter(s, i, samples + i, s->nb_channels);
     }
 
     for(i=0;i<s->nb_channels;i++) {
         compute_scale_factors(s->scale_code[i], s->scale_factors[i],
                               s->sb_samples[i], s->sblimit);
     }
     for(i=0;i<s->nb_channels;i++) {
         psycho_acoustic_model(s, smr[i]);
     }
     compute_bit_allocation(s, smr, bit_alloc, &padding);
 
     init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
 
     encode_frame(s, bit_alloc, padding);
 

     return put_bits_ptr(&s->pb) - s->pb.buf;

 }
 

 static av_cold int MPA_encode_close(AVCodecContext *avctx)

 {
     av_freep(&avctx->coded_frame);
     return 0;
 }
 

 static const AVCodecDefault mp2_defaults[] = {
     { "b",    "128k" },
     { NULL },
 };
 

 AVCodec ff_mp2_encoder = {

     .name           = "mp2",
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = CODEC_ID_MP2,
     .priv_data_size = sizeof(MpegAudioContext),
     .init           = MPA_encode_init,
     .encode         = MPA_encode_frame,
     .close          = MPA_encode_close,

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

     .supported_samplerates= (const int[]){44100, 48000,  32000, 22050, 24000, 16000, 0},

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

     .defaults       = mp2_defaults,

 };