avcodec: split mp2 encoder into float and fixed

This makes the USE_FLOATS == 0 available to the end user
More float optimizations can easily be added as well now
common code should be factored out into a common file once all
fixed point & floating point optimizations are done, this is to
avoid having to move code back and forth between files.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 779 insertions, 0 deletions

diff --git a/libavcodec/mpegaudioenc_template.c b/libavcodec/mpegaudioenc_template.c
new file mode 100644
index 0000000000..a567dcdbb4
--- /dev/null
+++ b/libavcodec/mpegaudioenc_template.c
@@ -0,0 +1,779 @@
+/*
+ * 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 "libavutil/channel_layout.h"
+
+#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"
+#include "mpegaudiodsp.h"
+#include "mpegaudiodata.h"
+#include "mpegaudiotab.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;
+    int16_t filter_bank[512];
+    int scale_factor_table[64];
+    unsigned char scale_diff_table[128];
+#ifdef USE_FLOATS
+    float scale_factor_inv_table[64];
+#else
+    int8_t scale_factor_shift[64];
+    unsigned short scale_factor_mult[64];
+#endif
+    unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
+} MpegAudioContext;
+
+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 AVERROR(EINVAL);
+    }
+    bitrate = bitrate / 1000;
+    s->nb_channels = channels;
+    avctx->frame_size = MPA_FRAME_SIZE;
+    avctx->delay      = 512 - 32 + 1;
+
+    /* 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 AVERROR(EINVAL);
+    }
+    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 AVERROR(EINVAL);
+    }
+    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
+        s->filter_bank[i] = v;
+        if ((i & 63) != 0)
+            v = -v;
+        if (i != 0)
+            s->filter_bank[512 - i] = v;
+    }
+
+    for(i=0;i<64;i++) {
+        v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
+        if (v <= 0)
+            v = 1;
+        s->scale_factor_table[i] = v;
+#ifdef USE_FLOATS
+        s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
+#else
+#define P 15
+        s->scale_factor_shift[i] = 21 - P - (i / 3);
+        s->scale_factor_mult[i] = (1 << P) * exp2((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;
+        s->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;
+        s->total_quant_bits[i] = 12 * v;
+    }
+
+    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 = s->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(MpegAudioContext *s,
+                                  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 <= s->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, s->scale_factor_table[index], index);
+            /* store the scale factor */
+            av_assert2(index >=0 && index <= 63);
+            sf[i] = index;
+        }
+
+        /* compute the transmission factor : look if the scale factors
+           are close enough to each other */
+        d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
+        d2 = s->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:
+            av_assert2(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 += s->total_quant_bits[alloc[1]];
+        } else {
+            /* increments bit allocation */
+            b = bit_alloc[max_ch][max_sb];
+            incr = s->total_quant_bits[alloc[b + 1]] -
+                s->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;
+    av_assert0(*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 * s->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 = s->scale_factor_shift[e];
+                                mult = s->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);
+                                if (q[m] < 0)
+                                    q[m] = 0;
+                            }
+#endif
+                            if (q[m] >= steps)
+                                q[m] = steps - 1;
+                            av_assert2(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, AVPacket *avpkt,
+                            const AVFrame *frame, int *got_packet_ptr)
+{
+    MpegAudioContext *s = avctx->priv_data;
+    const int16_t *samples = (const int16_t *)frame->data[0];
+    short smr[MPA_MAX_CHANNELS][SBLIMIT];
+    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
+    int padding, i, ret;
+
+    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, 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);
+
+    if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0)
+        return ret;
+
+    init_put_bits(&s->pb, avpkt->data, avpkt->size);
+
+    encode_frame(s, bit_alloc, padding);
+
+    if (frame->pts != AV_NOPTS_VALUE)
+        avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
+
+    avpkt->size = put_bits_count(&s->pb) / 8;
+    *got_packet_ptr = 1;
+    return 0;
+}
+
+static const AVCodecDefault mp2_defaults[] = {
+    { "b",    "128k" },
+    { NULL },
+};
+