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FFmpeg/libavcodec/mpegaudioenc.c
Michael Niedermayer 80d156d7fd Merge remote-tracking branch 'qatar/master' 
* qatar/master:
  qdm2: Use floating point synthesis filter.
  h264: correct border check.
  h264: fix loopfilter with threading at slice boundaries.
  Fix ff_mpa_synth_filter_fixed() prototype
  Rename costablegen.c ---> cos_tablegen.c.
  Collapse tableprint.c into tableprint.h.
  Simplify trig table rules
  Remove potentially unstable filenames from comments in generated files.
  Ignore generated tables and generated table generator programs.
  Simplify CLEANFILES make variable by using wildcards.
  Remove silly insults from avformat_version() Doxygen documentation.
  mpegaudiodsp: fix x86 and ppc makefiles
  configure: Adjust AVX assembler check.
  mpegaudio: remove unused version of SAME_HEADER_MASK
  mpegaudio: remove useless #undef at end of file
  asfdec: add missing #include for av_bswap32()
  mpegaudio: merge two #if CONFIG_FLOAT blocks
  mpegaudio: move some struct definitions from mpegaudio.h
  Move some mpegaudio functions to new mpegaudiodsp subsystem

Conflicts:
	libavcodec/h264.c
	libavcodec/x86/Makefile

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-05-20 05:48:22 +02:00

785 lines
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/*
* The simplest mpeg audio layer 2 encoder
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* The simplest mpeg audio layer 2 encoder.
*/
#include "avcodec.h"
#include "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 */
}
#if 0
printf("%2d:%d in=%x %x %d\n",
j, i, vmax, scale_factor_table[index], index);
#endif
/* store the scale factor */
assert(index >=0 && index <= 63);
sf[i] = index;
}
/* compute the transmission factor : look if the scale factors
are close enough to each other */
d1 = scale_diff_table[sf[0] - sf[1] + 64];
d2 = scale_diff_table[sf[1] - sf[2] + 64];
/* handle the 25 cases */
switch(d1 * 5 + d2) {
case 0*5+0:
case 0*5+4:
case 3*5+4:
case 4*5+0:
case 4*5+4:
code = 0;
break;
case 0*5+1:
case 0*5+2:
case 4*5+1:
case 4*5+2:
code = 3;
sf[2] = sf[1];
break;
case 0*5+3:
case 4*5+3:
code = 3;
sf[1] = sf[2];
break;
case 1*5+0:
case 1*5+4:
case 2*5+4:
code = 1;
sf[1] = sf[0];
break;
case 1*5+1:
case 1*5+2:
case 2*5+0:
case 2*5+1:
case 2*5+2:
code = 2;
sf[1] = sf[2] = sf[0];
break;
case 2*5+3:
case 3*5+3:
code = 2;
sf[0] = sf[1] = sf[2];
break;
case 3*5+0:
case 3*5+1:
case 3*5+2:
code = 2;
sf[0] = sf[2] = sf[1];
break;
case 1*5+3:
code = 2;
if (sf[0] > sf[2])
sf[0] = sf[2];
sf[1] = sf[2] = sf[0];
break;
default:
assert(0); //cannot happen
code = 0; /* kill warning */
}
#if 0
printf("%d: %2d %2d %2d %d %d -> %d\n", j,
sf[0], sf[1], sf[2], d1, d2, code);
#endif
scale_code[j] = code;
sf += 3;
}
}
/* The most important function : psycho acoustic module. In this
encoder there is basically none, so this is the worst you can do,
but also this is the simpler. */
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
{
int i;
for(i=0;i<s->sblimit;i++) {
smr[i] = (int)(fixed_smr[i] * 10);
}
}
#define SB_NOTALLOCATED 0
#define SB_ALLOCATED 1
#define SB_NOMORE 2
/* Try to maximize the smr while using a number of bits inferior to
the frame size. I tried to make the code simpler, faster and
smaller than other encoders :-) */
static void compute_bit_allocation(MpegAudioContext *s,
short smr1[MPA_MAX_CHANNELS][SBLIMIT],
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
int *padding)
{
int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
int incr;
short smr[MPA_MAX_CHANNELS][SBLIMIT];
unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
const unsigned char *alloc;
memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
/* compute frame size and padding */
max_frame_size = s->frame_size;
s->frame_frac += s->frame_frac_incr;
if (s->frame_frac >= 65536) {
s->frame_frac -= 65536;
s->do_padding = 1;
max_frame_size += 8;
} 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 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);
}
/*
* 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)"),
};
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