[libav.git] / libavcodec / mpegaudio.c

/*
 * The simplest mpeg audio layer 2 encoder
 * Copyright (c) 2000, 2001 Fabrice Bellard.
 *
 * This library 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 of the License, or (at your option) any later version.
 *
 * This library 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 this library; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
 */
#include "avcodec.h"
#include "mpegaudio.h"

/* currently, cannot change these constants (need to modify
   quantization stage) */
#define FRAC_BITS 15
#define WFRAC_BITS  14
#define MUL(a,b) (((INT64)(a) * (INT64)(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 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 "mpegaudiotab.h"

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 > 2)
        return -1;
    bitrate = bitrate / 1000;
    s->nb_channels = channels;
    s->freq = freq;
    s->bit_rate = bitrate * 1000;
    avctx->frame_size = MPA_FRAME_SIZE;
    avctx->key_frame = 1; /* always key frame */

    /* encoding freq */
    s->lsf = 0;
    for(i=0;i<3;i++) {
        if (mpa_freq_tab[i] == freq) 
            break;
        if ((mpa_freq_tab[i] / 2) == freq) {
            s->lsf = 1;
            break;
        }
    }
    if (i == 3)
        return -1;
    s->freq_index = i;

    /* encoding bitrate & frequency */
    for(i=0;i<15;i++) {
        if (mpa_bitrate_tab[s->lsf][1][i] == bitrate) 
            break;
    }
    if (i == 15)
        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 = sblimit_table[table];
    s->alloc_table = alloc_tables[table];

#ifdef DEBUG
    printf("%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 = 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 = quant_bits[i];
        if (v < 0) 
            v = -v;
        else
            v = v * 3;
        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, 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:
            av_abort();
        }
        
#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;
    } 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 = 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 = 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);
}

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]);
    }
    compute_bit_allocation(s, smr, bit_alloc, &padding);

    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);

    encode_frame(s, bit_alloc, padding);
    
    s->nb_samples += MPA_FRAME_SIZE;
    return pbBufPtr(&s->pb) - s->pb.buf;
}


AVCodec mp2_encoder = {
    "mp2",
    CODEC_TYPE_AUDIO,
    CODEC_ID_MP2,
    sizeof(MpegAudioContext),
    MPA_encode_init,
    MPA_encode_frame,
    NULL,
};

#undef FIX
Commit	Line	Data
de6d9b64 FB	1	/*
de6d9b64 FB	2	* The simplest mpeg audio layer 2 encoder
ff4ec49e	3	* Copyright (c) 2000, 2001 Fabrice Bellard.
de6d9b64	4	*
ff4ec49e FB	5	* This library is free software; you can redistribute it and/or
	6	* modify it under the terms of the GNU Lesser General Public
	7	* License as published by the Free Software Foundation; either
	8	* version 2 of the License, or (at your option) any later version.
de6d9b64	9	*
ff4ec49e	10	* This library is distributed in the hope that it will be useful,
de6d9b64	11	* but WITHOUT ANY WARRANTY; without even the implied warranty of
ff4ec49e FB	12	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
ff4ec49e FB	13	* Lesser General Public License for more details.
de6d9b64	14	*
ff4ec49e FB	15	* You should have received a copy of the GNU Lesser General Public
	16	* License along with this library; if not, write to the Free Software
	17	* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
de6d9b64	18	*/
de6d9b64 FB	19	#include "avcodec.h"
	20	#include "mpegaudio.h"
	21
afa982fd FB	22	/* currently, cannot change these constants (need to modify
	23	quantization stage) */
	24	#define FRAC_BITS 15
	25	#define WFRAC_BITS 14
	26	#define MUL(a,b) (((INT64)(a) * (INT64)(b)) >> FRAC_BITS)
	27	#define FIX(a) ((int)((a) * (1 << FRAC_BITS)))
2456e28d FB	28
	29	#define SAMPLES_BUF_SIZE 4096
	30
	31	typedef struct MpegAudioContext {
	32	PutBitContext pb;
	33	int nb_channels;
	34	int freq, bit_rate;
	35	int lsf; /* 1 if mpeg2 low bitrate selected */
	36	int bitrate_index; /* bit rate */
	37	int freq_index;
	38	int frame_size; /* frame size, in bits, without padding */
	39	INT64 nb_samples; /* total number of samples encoded */
	40	/* padding computation */
	41	int frame_frac, frame_frac_incr, do_padding;
	42	short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
	43	int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
	44	int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
	45	unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
	46	/* code to group 3 scale factors */
	47	unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
	48	int sblimit; /* number of used subbands */
	49	const unsigned char *alloc_table;
	50	} MpegAudioContext;
	51
de6d9b64 FB	52	/* define it to use floats in quantization (I don't like floats !) */
	53	//#define USE_FLOATS
	54
de6d9b64 FB	55	#include "mpegaudiotab.h"
	56
	57	int MPA_encode_init(AVCodecContext *avctx)
	58	{
	59	MpegAudioContext *s = avctx->priv_data;
	60	int freq = avctx->sample_rate;
	61	int bitrate = avctx->bit_rate;
	62	int channels = avctx->channels;
2456e28d	63	int i, v, table;
de6d9b64 FB	64	float a;
	65
	66	if (channels > 2)
	67	return -1;
	68	bitrate = bitrate / 1000;
	69	s->nb_channels = channels;
	70	s->freq = freq;
	71	s->bit_rate = bitrate * 1000;
	72	avctx->frame_size = MPA_FRAME_SIZE;
	73	avctx->key_frame = 1; /* always key frame */
	74
	75	/* encoding freq */
	76	s->lsf = 0;
	77	for(i=0;i<3;i++) {
2456e28d	78	if (mpa_freq_tab[i] == freq)
de6d9b64	79	break;
2456e28d	80	if ((mpa_freq_tab[i] / 2) == freq) {
de6d9b64 FB	81	s->lsf = 1;
	82	break;
	83	}
	84	}
	85	if (i == 3)
	86	return -1;
	87	s->freq_index = i;
	88
	89	/* encoding bitrate & frequency */
	90	for(i=0;i<15;i++) {
2456e28d	91	if (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
de6d9b64 FB	92	break;
	93	}
	94	if (i == 15)
	95	return -1;
	96	s->bitrate_index = i;
	97
	98	/* compute total header size & pad bit */
	99
	100	a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
	101	s->frame_size = ((int)a) * 8;
	102
	103	/* frame fractional size to compute padding */
	104	s->frame_frac = 0;
	105	s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
	106
	107	/* select the right allocation table */
2456e28d FB	108	table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
2456e28d FB	109
de6d9b64 FB	110	/* number of used subbands */
	111	s->sblimit = sblimit_table[table];
	112	s->alloc_table = alloc_tables[table];
	113
	114	#ifdef DEBUG
	115	printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
	116	bitrate, freq, s->frame_size, table, s->frame_frac_incr);
	117	#endif
	118
	119	for(i=0;i<s->nb_channels;i++)
	120	s->samples_offset[i] = 0;
	121
2456e28d FB	122	for(i=0;i<257;i++) {
2456e28d FB	123	int v;
afa982fd FB	124	v = mpa_enwindow[i];
	125	#if WFRAC_BITS != 16
	126	v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
	127	#endif
2456e28d FB	128	filter_bank[i] = v;
	129	if ((i & 63) != 0)
	130	v = -v;
	131	if (i != 0)
	132	filter_bank[512 - i] = v;
de6d9b64	133	}
2456e28d	134
de6d9b64 FB	135	for(i=0;i<64;i++) {
	136	v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
	137	if (v <= 0)
	138	v = 1;
	139	scale_factor_table[i] = v;
	140	#ifdef USE_FLOATS
	141	scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
	142	#else
	143	#define P 15
	144	scale_factor_shift[i] = 21 - P - (i / 3);
	145	scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
	146	#endif
	147	}
	148	for(i=0;i<128;i++) {
	149	v = i - 64;
	150	if (v <= -3)
	151	v = 0;
	152	else if (v < 0)
	153	v = 1;
	154	else if (v == 0)
	155	v = 2;
	156	else if (v < 3)
	157	v = 3;
	158	else
	159	v = 4;
	160	scale_diff_table[i] = v;
	161	}
	162
	163	for(i=0;i<17;i++) {
	164	v = quant_bits[i];
	165	if (v < 0)
	166	v = -v;
	167	else
	168	v = v * 3;
	169	total_quant_bits[i] = 12 * v;
	170	}
	171
	172	return 0;
	173	}
	174
2456e28d	175	/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
afa982fd	176	static void idct32(int out, int tab)
de6d9b64 FB	177	{
	178	int i, j;
	179	int t, t1, xr;
	180	const int *xp = costab32;
	181
	182	for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
	183
	184	t = tab + 30;
	185	t1 = tab + 2;
	186	do {
	187	t[0] += t[-4];
	188	t[1] += t[1 - 4];
	189	t -= 4;
	190	} while (t != t1);
	191
	192	t = tab + 28;
	193	t1 = tab + 4;
	194	do {
	195	t[0] += t[-8];
	196	t[1] += t[1-8];
	197	t[2] += t[2-8];
	198	t[3] += t[3-8];
	199	t -= 8;
	200	} while (t != t1);
	201
	202	t = tab;
	203	t1 = tab + 32;
	204	do {
	205	t[ 3] = -t[ 3];
	206	t[ 6] = -t[ 6];
	207
	208	t[11] = -t[11];
	209	t[12] = -t[12];
	210	t[13] = -t[13];
	211	t[15] = -t[15];
	212	t += 16;
	213	} while (t != t1);
	214
	215
	216	t = tab;
	217	t1 = tab + 8;
	218	do {
	219	int x1, x2, x3, x4;
	220
	221	x3 = MUL(t[16], FIX(SQRT2*0.5));
	222	x4 = t[0] - x3;
	223	x3 = t[0] + x3;
	224
	225	x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
	226	x1 = MUL((t[8] - x2), xp[0]);
	227	x2 = MUL((t[8] + x2), xp[1]);
	228
	229	t[ 0] = x3 + x1;
	230	t[ 8] = x4 - x2;
	231	t[16] = x4 + x2;
	232	t[24] = x3 - x1;
	233	t++;
	234	} while (t != t1);
	235
	236	xp += 2;
	237	t = tab;
	238	t1 = tab + 4;
	239	do {
	240	xr = MUL(t[28],xp[0]);
241	t[28] = (t[0] - xr);
242	t[0] = (t[0] + xr);
243
244	xr = MUL(t[4],xp[1]);
245	t[ 4] = (t[24] - xr);
246	t[24] = (t[24] + xr);
247
248	xr = MUL(t[20],xp[2]);
249	t[20] = (t[8] - xr);
250	t[ 8] = (t[8] + xr);
251
252	xr = MUL(t[12],xp[3]);
253	t[12] = (t[16] - xr);
254	t[16] = (t[16] + xr);
255	t++;
256	} while (t != t1);
257	xp += 4;
258
259	for (i = 0; i < 4; i++) {
260	xr = MUL(tab[30-i*4],xp[0]);
261	tab[30-i4] = (tab[i4] - xr);
262	tab[ i4] = (tab[i4] + xr);
263
264	xr = MUL(tab[ 2+i*4],xp[1]);
265	tab[ 2+i4] = (tab[28-i4] - xr);
266	tab[28-i4] = (tab[28-i4] + xr);
267
268	xr = MUL(tab[31-i*4],xp[0]);
269	tab[31-i4] = (tab[1+i4] - xr);
270	tab[ 1+i4] = (tab[1+i4] + xr);
271
272	xr = MUL(tab[ 3+i*4],xp[1]);
273	tab[ 3+i4] = (tab[29-i4] - xr);
274	tab[29-i4] = (tab[29-i4] + xr);
275
276	xp += 2;
277	}
278
279	t = tab + 30;
280	t1 = tab + 1;
281	do {
282	xr = MUL(t1[0], *xp);
283	t1[0] = (t[0] - xr);
284	t[0] = (t[0] + xr);
285	t -= 2;
286	t1 += 2;
287	xp++;
288	} while (t >= tab);
289
290	for(i=0;i<32;i++) {
afa982fd	291	out[i] = tab[bitinv32[i]];
de6d9b64 FB	292	}
	293	}
	294
afa982fd FB	295	#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
afa982fd FB	296
de6d9b64 FB	297	static void filter(MpegAudioContext s, int ch, short samples, int incr)
	298	{
	299	short p, q;
afa982fd FB	300	int sum, offset, i, j;
afa982fd FB	301	int tmp[64];
de6d9b64 FB	302	int tmp1[32];
	303	int *out;
	304
	305	// print_pow1(samples, 1152);
	306
	307	offset = s->samples_offset[ch];
	308	out = &s->sb_samples[ch][0][0][0];
	309	for(j=0;j<36;j++) {
	310	/* 32 samples at once */
	311	for(i=0;i<32;i++) {
	312	s->samples_buf[ch][offset + (31 - i)] = samples[0];
	313	samples += incr;
	314	}
	315
	316	/* filter */
	317	p = s->samples_buf[ch] + offset;
	318	q = filter_bank;
	319	/* maxsum = 23169 */
	320	for(i=0;i<64;i++) {
	321	sum = p[064] q[0*64];
	322	sum += p[164] q[1*64];
	323	sum += p[264] q[2*64];
	324	sum += p[364] q[3*64];
	325	sum += p[464] q[4*64];
	326	sum += p[564] q[5*64];
	327	sum += p[664] q[6*64];
	328	sum += p[764] q[7*64];
afa982fd	329	tmp[i] = sum;
de6d9b64 FB	330	p++;
	331	q++;
	332	}
afa982fd FB	333	tmp1[0] = tmp[16] >> WSHIFT;
	334	for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
	335	for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
de6d9b64	336
afa982fd	337	idct32(out, tmp1);
de6d9b64 FB	338
	339	/* advance of 32 samples */
	340	offset -= 32;
	341	out += 32;
	342	/* handle the wrap around */
	343	if (offset < 0) {
	344	memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
	345	s->samples_buf[ch], (512 - 32) * 2);
	346	offset = SAMPLES_BUF_SIZE - 512;
	347	}
	348	}
	349	s->samples_offset[ch] = offset;
	350
	351	// print_pow(s->sb_samples, 1152);
	352	}
	353
	354	static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
	355	unsigned char scale_factors[SBLIMIT][3],
	356	int sb_samples[3][12][SBLIMIT],
	357	int sblimit)
	358	{
	359	int *p, vmax, v, n, i, j, k, code;
	360	int index, d1, d2;
	361	unsigned char *sf = &scale_factors[0][0];
	362
	363	for(j=0;j<sblimit;j++) {
	364	for(i=0;i<3;i++) {
	365	/* find the max absolute value */
	366	p = &sb_samples[i][0][j];
	367	vmax = abs(*p);
	368	for(k=1;k<12;k++) {
	369	p += SBLIMIT;
	370	v = abs(*p);
	371	if (v > vmax)
	372	vmax = v;
	373	}
	374	/* compute the scale factor index using log 2 computations */
	375	if (vmax > 0) {
935442b5	376	n = av_log2(vmax);
de6d9b64 FB	377	/* n is the position of the MSB of vmax. now
	378	use at most 2 compares to find the index */
	379	index = (21 - n) * 3 - 3;
	380	if (index >= 0) {
	381	while (vmax <= scale_factor_table[index+1])
	382	index++;
	383	} else {
	384	index = 0; /* very unlikely case of overflow */
	385	}
	386	} else {
afa982fd	387	index = 62; /* value 63 is not allowed */
de6d9b64	388	}
afa982fd	389
de6d9b64 FB	390	#if 0
	391	printf("%2d:%d in=%x %x %d\n",
	392	j, i, vmax, scale_factor_table[index], index);
	393	#endif
	394	/* store the scale factor */
	395	assert(index >=0 && index <= 63);
	396	sf[i] = index;
	397	}
	398
	399	/* compute the transmission factor : look if the scale factors
	400	are close enough to each other */
	401	d1 = scale_diff_table[sf[0] - sf[1] + 64];
	402	d2 = scale_diff_table[sf[1] - sf[2] + 64];
	403
	404	/* handle the 25 cases */
	405	switch(d1 * 5 + d2) {
	406	case 0*5+0:
	407	case 0*5+4:
	408	case 3*5+4:
	409	case 4*5+0:
	410	case 4*5+4:
	411	code = 0;
	412	break;
	413	case 0*5+1:
	414	case 0*5+2:
	415	case 4*5+1:
	416	case 4*5+2:
	417	code = 3;
	418	sf[2] = sf[1];
	419	break;
	420	case 0*5+3:
	421	case 4*5+3:
	422	code = 3;
	423	sf[1] = sf[2];
	424	break;
	425	case 1*5+0:
	426	case 1*5+4:
	427	case 2*5+4:
	428	code = 1;
	429	sf[1] = sf[0];
	430	break;
	431	case 1*5+1:
	432	case 1*5+2:
	433	case 2*5+0:
	434	case 2*5+1:
	435	case 2*5+2:
	436	code = 2;
	437	sf[1] = sf[2] = sf[0];
	438	break;
	439	case 2*5+3:
	440	case 3*5+3:
	441	code = 2;
	442	sf[0] = sf[1] = sf[2];
	443	break;
	444	case 3*5+0:
	445	case 3*5+1:
	446	case 3*5+2:
	447	code = 2;
	448	sf[0] = sf[2] = sf[1];
	449	break;
	450	case 1*5+3:
	451	code = 2;
	452	if (sf[0] > sf[2])
	453	sf[0] = sf[2];
454	sf[1] = sf[2] = sf[0];
455	break;
456	default:
02ac3136	457	av_abort();
de6d9b64 FB	458	}
	459
	460	#if 0
	461	printf("%d: %2d %2d %2d %d %d -> %d\n", j,
	462	sf[0], sf[1], sf[2], d1, d2, code);
	463	#endif
	464	scale_code[j] = code;
	465	sf += 3;
	466	}
	467	}
	468
	469	/* The most important function : psycho acoustic module. In this
	470	encoder there is basically none, so this is the worst you can do,
	471	but also this is the simpler. */
	472	static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
	473	{
	474	int i;
	475
	476	for(i=0;i<s->sblimit;i++) {
	477	smr[i] = (int)(fixed_smr[i] * 10);
	478	}
	479	}
	480
	481
	482	#define SB_NOTALLOCATED 0
	483	#define SB_ALLOCATED 1
	484	#define SB_NOMORE 2
	485
	486	/* Try to maximize the smr while using a number of bits inferior to
	487	the frame size. I tried to make the code simpler, faster and
	488	smaller than other encoders :-) */
	489	static void compute_bit_allocation(MpegAudioContext *s,
	490	short smr1[MPA_MAX_CHANNELS][SBLIMIT],
	491	unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
	492	int *padding)
	493	{
	494	int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
	495	int incr;
	496	short smr[MPA_MAX_CHANNELS][SBLIMIT];
	497	unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
	498	const unsigned char *alloc;
	499
	500	memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
	501	memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
	502	memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
	503
	504	/* compute frame size and padding */
	505	max_frame_size = s->frame_size;
	506	s->frame_frac += s->frame_frac_incr;
	507	if (s->frame_frac >= 65536) {
	508	s->frame_frac -= 65536;
	509	s->do_padding = 1;
	510	max_frame_size += 8;
	511	} else {
	512	s->do_padding = 0;
	513	}
	514
	515	/* compute the header + bit alloc size */
	516	current_frame_size = 32;
	517	alloc = s->alloc_table;
	518	for(i=0;i<s->sblimit;i++) {
	519	incr = alloc[0];
	520	current_frame_size += incr * s->nb_channels;
	521	alloc += 1 << incr;
522	}
523	for(;;) {
524	/* look for the subband with the largest signal to mask ratio */
525	max_sb = -1;
526	max_ch = -1;
527	max_smr = 0x80000000;
528	for(ch=0;ch<s->nb_channels;ch++) {
529	for(i=0;i<s->sblimit;i++) {
530	if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
531	max_smr = smr[ch][i];
532	max_sb = i;
533	max_ch = ch;
534	}
535	}
536	}
537	#if 0
538	printf("current=%d max=%d max_sb=%d alloc=%d\n",
539	current_frame_size, max_frame_size, max_sb,
540	bit_alloc[max_sb]);
541	#endif
542	if (max_sb < 0)
543	break;
544
545	/* find alloc table entry (XXX: not optimal, should use
546	pointer table) */
547	alloc = s->alloc_table;
548	for(i=0;i<max_sb;i++) {
549	alloc += 1 << alloc[0];
550	}
551
552	if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
553	/* nothing was coded for this band: add the necessary bits */
554	incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
555	incr += total_quant_bits[alloc[1]];
556	} else {
557	/* increments bit allocation */
558	b = bit_alloc[max_ch][max_sb];
559	incr = total_quant_bits[alloc[b + 1]] -
560	total_quant_bits[alloc[b]];
561	}
562
563	if (current_frame_size + incr <= max_frame_size) {
564	/* can increase size */
565	b = ++bit_alloc[max_ch][max_sb];
566	current_frame_size += incr;
567	/* decrease smr by the resolution we added */
568	smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
569	/* max allocation size reached ? */
570	if (b == ((1 << alloc[0]) - 1))
571	subband_status[max_ch][max_sb] = SB_NOMORE;
572	else
573	subband_status[max_ch][max_sb] = SB_ALLOCATED;
574	} else {
575	/* cannot increase the size of this subband */
576	subband_status[max_ch][max_sb] = SB_NOMORE;
577	}
578	}
579	*padding = max_frame_size - current_frame_size;
580	assert(*padding >= 0);
581
582	#if 0
583	for(i=0;i<s->sblimit;i++) {
584	printf("%d ", bit_alloc[i]);
585	}
586	printf("\n");
587	#endif
588	}
589
590	/*
591	* Output the mpeg audio layer 2 frame. Note how the code is small
592	* compared to other encoders :-)
593	*/
594	static void encode_frame(MpegAudioContext *s,
595	unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
596	int padding)
597	{
598	int i, j, k, l, bit_alloc_bits, b, ch;
599	unsigned char *sf;
600	int q[3];
601	PutBitContext *p = &s->pb;
602
603	/* header */
604
605	put_bits(p, 12, 0xfff);
606	put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
607	put_bits(p, 2, 4-2); /* layer 2 */
608	put_bits(p, 1, 1); /* no error protection */
609	put_bits(p, 4, s->bitrate_index);
610	put_bits(p, 2, s->freq_index);
611	put_bits(p, 1, s->do_padding); /* use padding */
612	put_bits(p, 1, 0); /* private_bit */
613	put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
614	put_bits(p, 2, 0); /* mode_ext */
615	put_bits(p, 1, 0); /* no copyright */
616	put_bits(p, 1, 1); /* original */
617	put_bits(p, 2, 0); /* no emphasis */
618
619	/* bit allocation */
620	j = 0;
621	for(i=0;i<s->sblimit;i++) {
622	bit_alloc_bits = s->alloc_table[j];
623	for(ch=0;ch<s->nb_channels;ch++) {
624	put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
625	}
626	j += 1 << bit_alloc_bits;
627	}
628
629	/* scale codes */
630	for(i=0;i<s->sblimit;i++) {
631	for(ch=0;ch<s->nb_channels;ch++) {
632	if (bit_alloc[ch][i])
633	put_bits(p, 2, s->scale_code[ch][i]);
634	}
635	}
636
637	/* scale factors */
638	for(i=0;i<s->sblimit;i++) {
639	for(ch=0;ch<s->nb_channels;ch++) {
640	if (bit_alloc[ch][i]) {
641	sf = &s->scale_factors[ch][i][0];
642	switch(s->scale_code[ch][i]) {
643	case 0:
644	put_bits(p, 6, sf[0]);
645	put_bits(p, 6, sf[1]);
646	put_bits(p, 6, sf[2]);
647	break;
648	case 3:
649	case 1:
650	put_bits(p, 6, sf[0]);
651	put_bits(p, 6, sf[2]);
652	break;
653	case 2:
654	put_bits(p, 6, sf[0]);
655	break;
656	}
657	}
658	}
659	}
660
661	/* quantization & write sub band samples */
662
663	for(k=0;k<3;k++) {
664	for(l=0;l<12;l+=3) {
665	j = 0;
666	for(i=0;i<s->sblimit;i++) {
667	bit_alloc_bits = s->alloc_table[j];
668	for(ch=0;ch<s->nb_channels;ch++) {
669	b = bit_alloc[ch][i];
670	if (b) {
671	int qindex, steps, m, sample, bits;
672	/* we encode 3 sub band samples of the same sub band at a time */
673	qindex = s->alloc_table[j+b];
674	steps = quant_steps[qindex];
675	for(m=0;m<3;m++) {
676	sample = s->sb_samples[ch][k][l + m][i];
677	/* divide by scale factor */
678	#ifdef USE_FLOATS
679	{
680	float a;
681	a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
682	q[m] = (int)((a + 1.0) * steps * 0.5);
683	}
684	#else
685	{
686	int q1, e, shift, mult;
687	e = s->scale_factors[ch][i][k];
688	shift = scale_factor_shift[e];
689	mult = scale_factor_mult[e];
690
691	/* normalize to P bits */
692	if (shift < 0)
693	q1 = sample << (-shift);
694	else
695	q1 = sample >> shift;
696	q1 = (q1 * mult) >> P;
697	q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
698	}
699	#endif
700	if (q[m] >= steps)
701	q[m] = steps - 1;
702	assert(q[m] >= 0 && q[m] < steps);
703	}
704	bits = quant_bits[qindex];
705	if (bits < 0) {
706	/* group the 3 values to save bits */
707	put_bits(p, -bits,
708	q[0] + steps * (q[1] + steps * q[2]));
709	#if 0
710	printf("%d: gr1 %d\n",
711	i, q[0] + steps * (q[1] + steps * q[2]));
712	#endif
713	} else {
714	#if 0
715	printf("%d: gr3 %d %d %d\n",
716	i, q[0], q[1], q[2]);
717	#endif
718	put_bits(p, bits, q[0]);
719	put_bits(p, bits, q[1]);
720	put_bits(p, bits, q[2]);
721	}
722	}
723	}
724	/* next subband in alloc table */
725	j += 1 << bit_alloc_bits;
726	}
727	}
728	}
729
730	/* padding */
731	for(i=0;i<padding;i++)
732	put_bits(p, 1, 0);
733
734	/* flush */
735	flush_put_bits(p);
736	}
737
738	int MPA_encode_frame(AVCodecContext *avctx,
739	unsigned char frame, int buf_size, void data)
740	{
741	MpegAudioContext *s = avctx->priv_data;
742	short *samples = data;
743	short smr[MPA_MAX_CHANNELS][SBLIMIT];
744	unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
745	int padding, i;
746
747	for(i=0;i<s->nb_channels;i++) {
748	filter(s, i, samples + i, s->nb_channels);
749	}
750
751	for(i=0;i<s->nb_channels;i++) {
752	compute_scale_factors(s->scale_code[i], s->scale_factors[i],
753	s->sb_samples[i], s->sblimit);
754	}
755	for(i=0;i<s->nb_channels;i++) {
756	psycho_acoustic_model(s, smr[i]);
757	}
758	compute_bit_allocation(s, smr, bit_alloc, &padding);
759
760	init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
761
762	encode_frame(s, bit_alloc, padding);
763
764	s->nb_samples += MPA_FRAME_SIZE;
17592475	765	return pbBufPtr(&s->pb) - s->pb.buf;
de6d9b64 FB	766	}
	767
	768
	769	AVCodec mp2_encoder = {
	770	"mp2",
	771	CODEC_TYPE_AUDIO,
	772	CODEC_ID_MP2,
	773	sizeof(MpegAudioContext),
	774	MPA_encode_init,
	775	MPA_encode_frame,
	776	NULL,
	777	};
cd4af68a ZK	778
cd4af68a ZK	779	#undef FIX