PeerStreamer

/*

 * The simplest mpeg audio layer 2 encoder

 * Copyright (c) 2000, 2001 Fabrice Bellard

 *

 * This file is part of Libav.

 *

 * Libav 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.

 *

 * Libav 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 Libav; if not, write to the Free Software

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

 */

/**

 * @file

 * The simplest mpeg audio layer 2 encoder.

 */

#include "avcodec.h"

#include "put_bits.h"

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

#define WFRAC_BITS  14   /* fractional bits for window */

#include "mpegaudio.h"

/* currently, cannot change these constants (need to modify

   quantization stage) */

#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)

#define SAMPLES_BUF_SIZE 4096

typedef struct MpegAudioContext {

    PutBitContext pb;

    int nb_channels;

    int lsf;           /* 1 if mpeg2 low bitrate selected */

    int bitrate_index; /* bit rate */

    int freq_index;

    int frame_size; /* frame size, in bits, without padding */

    /* padding computation */

    int frame_frac, frame_frac_incr, do_padding;

    short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */

    int samples_offset[MPA_MAX_CHANNELS];       /* offset in samples_buf */

    int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];

    unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */

    /* code to group 3 scale factors */

    unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];

    int sblimit; /* number of used subbands */

    const unsigned char *alloc_table;

} MpegAudioContext;

/* define it to use floats in quantization (I don't like floats !) */

#define USE_FLOATS

#include "mpegaudiodata.h"

#include "mpegaudiotab.h"

static av_cold int MPA_encode_init(AVCodecContext *avctx)

{

    MpegAudioContext *s = avctx->priv_data;

    int freq = avctx->sample_rate;

    int bitrate = avctx->bit_rate;

    int channels = avctx->channels;

    int i, v, table;

    float a;

    if (channels <= 0 || channels > 2){

        av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);

        return -1;

    }

    bitrate = bitrate / 1000;

    s->nb_channels = channels;

    avctx->frame_size = MPA_FRAME_SIZE;

    /* encoding freq */

    s->lsf = 0;

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

        if (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 = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);

    /* number of used subbands */

    s->sblimit = ff_mpa_sblimit_table[table];

    s->alloc_table = ff_mpa_alloc_tables[table];

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

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

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

        s->samples_offset[i] = 0;

    for(i=0;i<257;i++) {

        int v;

        v = ff_mpa_enwindow[i];

#if WFRAC_BITS != 16

        v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);

#endif

        filter_bank[i] = v;

        if ((i & 63) != 0)

            v = -v;

        if (i != 0)

            filter_bank[512 - i] = v;

    }

    for(i=0;i<64;i++) {

        v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));

        if (v <= 0)

            v = 1;

        scale_factor_table[i] = v;

#ifdef USE_FLOATS

        scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);

#else

#define P 15

        scale_factor_shift[i] = 21 - P - (i / 3);

        scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);

#endif

    }

    for(i=0;i<128;i++) {

        v = i - 64;

        if (v <= -3)

            v = 0;

        else if (v < 0)

            v = 1;

        else if (v == 0)

            v = 2;

        else if (v < 3)

            v = 3;

        else

            v = 4;

        scale_diff_table[i] = v;

    }

    for(i=0;i<17;i++) {

        v = ff_mpa_quant_bits[i];

        if (v < 0)

            v = -v;

        else

            v = v * 3;

        total_quant_bits[i] = 12 * v;

    }

    avctx->coded_frame= avcodec_alloc_frame();

    avctx->coded_frame->key_frame= 1;

    return 0;

}

/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */

static void idct32(int *out, int *tab)

{

    int i, j;

    int *t, *t1, xr;

    const int *xp = costab32;

    for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];

    t = tab + 30;

    t1 = tab + 2;

    do {

        t[0] += t[-4];

        t[1] += t[1 - 4];

        t -= 4;

    } while (t != t1);

    t = tab + 28;

    t1 = tab + 4;

    do {

        t[0] += t[-8];

        t[1] += t[1-8];

        t[2] += t[2-8];

        t[3] += t[3-8];

        t -= 8;

    } while (t != t1);

    t = tab;

    t1 = tab + 32;

    do {

        t[ 3] = -t[ 3];

        t[ 6] = -t[ 6];

        t[11] = -t[11];

        t[12] = -t[12];

        t[13] = -t[13];

        t[15] = -t[15];

        t += 16;

    } while (t != t1);

    t = tab;

    t1 = tab + 8;

    do {

        int x1, x2, x3, x4;

        x3 = MUL(t[16], FIX(SQRT2*0.5));

        x4 = t[0] - x3;

        x3 = t[0] + x3;

        x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));

        x1 = MUL((t[8] - x2), xp[0]);

        x2 = MUL((t[8] + x2), xp[1]);

        t[ 0] = x3 + x1;

        t[ 8] = x4 - x2;

        t[16] = x4 + x2;

        t[24] = x3 - x1;

        t++;

    } while (t != t1);

    xp += 2;

    t = tab;

    t1 = tab + 4;

    do {

        xr = MUL(t[28],xp[0]);

        t[28] = (t[0] - xr);

        t[0] = (t[0] + xr);

        xr = MUL(t[4],xp[1]);

        t[ 4] = (t[24] - xr);

        t[24] = (t[24] + xr);

        xr = MUL(t[20],xp[2]);

        t[20] = (t[8] - xr);

        t[ 8] = (t[8] + xr);

        xr = MUL(t[12],xp[3]);

        t[12] = (t[16] - xr);

        t[16] = (t[16] + xr);

        t++;

    } while (t != t1);

    xp += 4;

    for (i = 0; i < 4; i++) {

        xr = MUL(tab[30-i*4],xp[0]);

        tab[30-i*4] = (tab[i*4] - xr);

        tab[   i*4] = (tab[i*4] + xr);

        xr = MUL(tab[ 2+i*4],xp[1]);

        tab[ 2+i*4] = (tab[28-i*4] - xr);

        tab[28-i*4] = (tab[28-i*4] + xr);

        xr = MUL(tab[31-i*4],xp[0]);

        tab[31-i*4] = (tab[1+i*4] - xr);

        tab[ 1+i*4] = (tab[1+i*4] + xr);

        xr = MUL(tab[ 3+i*4],xp[1]);

        tab[ 3+i*4] = (tab[29-i*4] - xr);

        tab[29-i*4] = (tab[29-i*4] + xr);

        xp += 2;

    }

    t = tab + 30;

    t1 = tab + 1;

    do {

        xr = MUL(t1[0], *xp);

        t1[0] = (t[0] - xr);

        t[0] = (t[0] + xr);

        t -= 2;

        t1 += 2;

        xp++;

    } while (t >= tab);

    for(i=0;i<32;i++) {

        out[i] = tab[bitinv32[i]];

    }

}

#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)

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

{

    short *p, *q;

    int sum, offset, i, j;

    int tmp[64];

    int tmp1[32];

    int *out;

    //    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 > 1) {

                n = av_log2(vmax);

                /* n is the position of the MSB of vmax. now

                   use at most 2 compares to find the index */

                index = (21 - n) * 3 - 3;

                if (index >= 0) {

                    while (vmax <= scale_factor_table[index+1])

                        index++;

                } else {

                    index = 0; /* very unlikely case of overflow */

                }

            } else {

                index = 62; /* value 63 is not allowed */

            }

            av_dlog(NULL, "%2d:%d in=%x %x %d\n",

                    j, i, vmax, scale_factor_table[index], index);

            /* store the scale factor */

            assert(index >=0 && index <= 63);

            sf[i] = index;

        }

        /* compute the transmission factor : look if the scale factors

           are close enough to each other */

        d1 = scale_diff_table[sf[0] - sf[1] + 64];

        d2 = scale_diff_table[sf[1] - sf[2] + 64];

        /* handle the 25 cases */

        switch(d1 * 5 + d2) {

        case 0*5+0:

        case 0*5+4:

        case 3*5+4:

        case 4*5+0:

        case 4*5+4:

            code = 0;

            break;

        case 0*5+1:

        case 0*5+2:

        case 4*5+1:

        case 4*5+2:

            code = 3;

            sf[2] = sf[1];

            break;

        case 0*5+3:

        case 4*5+3:

            code = 3;

            sf[1] = sf[2];

            break;

        case 1*5+0:

        case 1*5+4:

        case 2*5+4:

            code = 1;

            sf[1] = sf[0];

            break;

        case 1*5+1:

        case 1*5+2:

        case 2*5+0:

        case 2*5+1:

        case 2*5+2:

            code = 2;

            sf[1] = sf[2] = sf[0];

            break;

        case 2*5+3:

        case 3*5+3:

            code = 2;

            sf[0] = sf[1] = sf[2];

            break;

        case 3*5+0:

        case 3*5+1:

        case 3*5+2:

            code = 2;

            sf[0] = sf[2] = sf[1];

            break;

        case 1*5+3:

            code = 2;

            if (sf[0] > sf[2])

              sf[0] = sf[2];

            sf[1] = sf[2] = sf[0];

            break;

        default:

            assert(0); //cannot happen

            code = 0;           /* kill warning */

        }

        av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,

                sf[0], sf[1], sf[2], d1, d2, code);

        scale_code[j] = code;

        sf += 3;

    }

}

/* The most important function : psycho acoustic module. In this

   encoder there is basically none, so this is the worst you can do,

   but also this is the simpler. */

static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])

{

    int i;

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

        smr[i] = (int)(fixed_smr[i] * 10);

    }

}

#define SB_NOTALLOCATED  0

#define SB_ALLOCATED     1

#define SB_NOMORE        2

/* Try to maximize the smr while using a number of bits inferior to

   the frame size. I tried to make the code simpler, faster and

   smaller than other encoders :-) */

static void compute_bit_allocation(MpegAudioContext *s,

                                   short smr1[MPA_MAX_CHANNELS][SBLIMIT],

                                   unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],

                                   int *padding)

{

    int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;

    int incr;

    short smr[MPA_MAX_CHANNELS][SBLIMIT];

    unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];

    const unsigned char *alloc;

    memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);

    memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);

    memset(bit_alloc, 0, s->nb_channels * SBLIMIT);

    /* compute frame size and padding */

    max_frame_size = s->frame_size;

    s->frame_frac += s->frame_frac_incr;

    if (s->frame_frac >= 65536) {

        s->frame_frac -= 65536;

        s->do_padding = 1;

        max_frame_size += 8;

    } else {

        s->do_padding = 0;

    }

    /* compute the header + bit alloc size */

    current_frame_size = 32;

    alloc = s->alloc_table;

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

        incr = alloc[0];

        current_frame_size += incr * s->nb_channels;

        alloc += 1 << incr;

    }

    for(;;) {

        /* look for the subband with the largest signal to mask ratio */

        max_sb = -1;

        max_ch = -1;

        max_smr = INT_MIN;

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

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

                if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {

                    max_smr = smr[ch][i];

                    max_sb = i;

                    max_ch = ch;

                }

            }

        }

        av_dlog(NULL, "current=%d max=%d max_sb=%d alloc=%d\n",

                current_frame_size, max_frame_size, max_sb,

                bit_alloc[max_sb]);

        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);

}

/*

 * Output the mpeg audio layer 2 frame. Note how the code is small

 * compared to other encoders :-)

 */

static void encode_frame(MpegAudioContext *s,

                         unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],

                         int padding)

{

    int i, j, k, l, bit_alloc_bits, b, ch;

    unsigned char *sf;

    int q[3];

    PutBitContext *p = &s->pb;

    /* header */

    put_bits(p, 12, 0xfff);

    put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */

    put_bits(p, 2, 4-2);  /* layer 2 */

    put_bits(p, 1, 1); /* no error protection */

    put_bits(p, 4, s->bitrate_index);

    put_bits(p, 2, s->freq_index);

    put_bits(p, 1, s->do_padding); /* use padding */

    put_bits(p, 1, 0);             /* private_bit */

    put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);

    put_bits(p, 2, 0); /* mode_ext */

    put_bits(p, 1, 0); /* no copyright */

    put_bits(p, 1, 1); /* original */

    put_bits(p, 2, 0); /* no emphasis */

    /* bit allocation */

    j = 0;

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

        bit_alloc_bits = s->alloc_table[j];

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

            put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);

        }

        j += 1 << bit_alloc_bits;

    }

    /* scale codes */

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

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

            if (bit_alloc[ch][i])

                put_bits(p, 2, s->scale_code[ch][i]);

        }

    }

    /* scale factors */

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

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

            if (bit_alloc[ch][i]) {

                sf = &s->scale_factors[ch][i][0];

                switch(s->scale_code[ch][i]) {

                case 0:

                    put_bits(p, 6, sf[0]);

                    put_bits(p, 6, sf[1]);

                    put_bits(p, 6, sf[2]);

                    break;

                case 3:

                case 1:

                    put_bits(p, 6, sf[0]);

                    put_bits(p, 6, sf[2]);

                    break;

                case 2:

                    put_bits(p, 6, sf[0]);

                    break;

                }

            }

        }

    }

    /* quantization & write sub band samples */

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

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

            j = 0;

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

                bit_alloc_bits = s->alloc_table[j];

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

                    b = bit_alloc[ch][i];

                    if (b) {

                        int qindex, steps, m, sample, bits;

                        /* we encode 3 sub band samples of the same sub band at a time */

                        qindex = s->alloc_table[j+b];

                        steps = ff_mpa_quant_steps[qindex];

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

                            sample = s->sb_samples[ch][k][l + m][i];

                            /* divide by scale factor */

#ifdef USE_FLOATS

                            {

                                float a;

                                a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];

                                q[m] = (int)((a + 1.0) * steps * 0.5);

                            }

#else

                            {

                                int q1, e, shift, mult;

                                e = s->scale_factors[ch][i][k];

                                shift = scale_factor_shift[e];

                                mult = scale_factor_mult[e];

                                /* normalize to P bits */

                                if (shift < 0)

                                    q1 = sample << (-shift);

                                else

                                    q1 = sample >> shift;

                                q1 = (q1 * mult) >> P;

                                q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);

                            }

#endif

                            if (q[m] >= steps)

                                q[m] = steps - 1;

                            assert(q[m] >= 0 && q[m] < steps);

                        }

                        bits = ff_mpa_quant_bits[qindex];

                        if (bits < 0) {

                            /* group the 3 values to save bits */

                            put_bits(p, -bits,

                                     q[0] + steps * (q[1] + steps * q[2]));

                        } else {

                            put_bits(p, bits, q[0]);

                            put_bits(p, bits, q[1]);

                            put_bits(p, bits, q[2]);

                        }

                    }

                }

                /* next subband in alloc table */

                j += 1 << bit_alloc_bits;

            }

        }

    }

    /* padding */

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

        put_bits(p, 1, 0);

    /* flush */

    flush_put_bits(p);

}

static int MPA_encode_frame(AVCodecContext *avctx,

                            unsigned char *frame, int buf_size, void *data)

{

    MpegAudioContext *s = avctx->priv_data;

    const short *samples = data;

    short smr[MPA_MAX_CHANNELS][SBLIMIT];

    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];

    int padding, i;

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

        filter(s, i, samples + i, s->nb_channels);

    }

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

        compute_scale_factors(s->scale_code[i], s->scale_factors[i],

                              s->sb_samples[i], s->sblimit);

    }

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

        psycho_acoustic_model(s, smr[i]);

    }

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

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

    encode_frame(s, bit_alloc, padding);

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

}

static av_cold int MPA_encode_close(AVCodecContext *avctx)

{

    av_freep(&avctx->coded_frame);

    return 0;

}

AVCodec ff_mp2_encoder = {

    "mp2",

    AVMEDIA_TYPE_AUDIO,

    CODEC_ID_MP2,

    sizeof(MpegAudioContext),

    MPA_encode_init,

    MPA_encode_frame,

    MPA_encode_close,

    NULL,

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

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

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

};

#undef FIX