remove dependency of mpeg audio encoder over mpeg audio decoder

Originally committed as revision 9082 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 18 insertions, 770 deletions

diff --git a/libavcodec/mpegaudio.c b/libavcodec/mpegaudio.c
index 2292a77a99..663427a43b 100644
--- a/libavcodec/mpegaudio.c
+++ b/libavcodec/mpegaudio.c
@@ -1,6 +1,6 @@
 /*
- * The simplest mpeg audio layer 2 encoder
- * Copyright (c) 2000, 2001 Fabrice Bellard.
+ * MPEG Audio common code
+ * Copyright (c) 2001, 2002 Fabrice Bellard.
  *
  * This file is part of FFmpeg.
  *
@@ -21,782 +21,30 @@
 
 /**
  * @file mpegaudio.c
- * The simplest mpeg audio layer 2 encoder.
+ * MPEG Audio common code.
  */
 
-#include "avcodec.h"
-#include "bitstream.h"
 #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 FIX(a)   ((int)((a) * (1 << FRAC_BITS)))
 
-#define SAMPLES_BUF_SIZE 4096
-
-typedef struct MpegAudioContext {
-    PutBitContext pb;
-    int nb_channels;
-    int freq, bit_rate;
-    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 */
-    int64_t nb_samples; /* total number of samples encoded */
-    /* 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 int MPA_encode_init(AVCodecContext *avctx)
+/* bitrate is in kb/s */
+int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf)
 {
-    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;
-    s->freq = freq;
-    s->bit_rate = bitrate * 1000;
-    avctx->frame_size = MPA_FRAME_SIZE;
-
-    /* encoding freq */
-    s->lsf = 0;
-    for(i=0;i<3;i++) {
-        if (ff_mpa_freq_tab[i] == freq)
-            break;
-        if ((ff_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 (ff_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 = 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];
+    int ch_bitrate, table;
 
-#ifdef DEBUG
-    av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
-           bitrate, freq, s->frame_size, table, s->frame_frac_incr);
-#endif
-
-    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;
+    ch_bitrate = bitrate / nb_channels;
+    if (!lsf) {
+        if ((freq == 48000 && ch_bitrate >= 56) ||
+            (ch_bitrate >= 56 && ch_bitrate <= 80))
+            table = 0;
+        else if (freq != 48000 && ch_bitrate >= 96)
+            table = 1;
+        else if (freq != 32000 && ch_bitrate <= 48)
+            table = 2;
         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, short *samples, int incr)
-{
-    short *p, *q;
-    int sum, offset, i, j;
-    int tmp[64];
-    int tmp1[32];
-    int *out;
-
-    //    print_pow1(samples, 1152);
-
-    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;
-
-    //    print_pow(s->sb_samples, 1152);
-}
-
-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 > 0) {
-                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 */
-            }
-
-#if 0
-            printf("%2d:%d in=%x %x %d\n",
-                   j, i, vmax, scale_factor_table[index], index);
-#endif
-            /* 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); //cant happen
-            code = 0;           /* kill warning */
-        }
-
-#if 0
-        printf("%d: %2d %2d %2d %d %d -> %d\n", j,
-               sf[0], sf[1], sf[2], d1, d2, code);
-#endif
-        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;
+            table = 3;
     } 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 = 0x80000000;
-        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 0
-        printf("current=%d max=%d max_sb=%d alloc=%d\n",
-               current_frame_size, max_frame_size, max_sb,
-               bit_alloc[max_sb]);
-#endif
-        if (max_sb < 0)
-            break;
-
-        /* 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);
-
-#if 0
-    for(i=0;i<s->sblimit;i++) {
-        printf("%d ", bit_alloc[i]);
-    }
-    printf("\n");
-#endif
-}
-
-/*
- * 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]));
-#if 0
-                            printf("%d: gr1 %d\n",
-                                   i, q[0] + steps * (q[1] + steps * q[2]));
-#endif
-                        } else {
-#if 0
-                            printf("%d: gr3 %d %d %d\n",
-                                   i, q[0], q[1], q[2]);
-#endif
-                            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;
-    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]);
+        table = 4;
     }
-    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);
-
-    s->nb_samples += MPA_FRAME_SIZE;
-    return pbBufPtr(&s->pb) - s->pb.buf;
-}
-
-static int MPA_encode_close(AVCodecContext *avctx)
-{
-    av_freep(&avctx->coded_frame);
-    return 0;
+    return table;
 }
-
-AVCodec mp2_encoder = {
-    "mp2",
-    CODEC_TYPE_AUDIO,
-    CODEC_ID_MP2,
-    sizeof(MpegAudioContext),
-    MPA_encode_init,
-    MPA_encode_frame,
-    MPA_encode_close,
-    NULL,
-};
-
-#undef FIX