mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-27 09:52:17 +00:00
e052f06531
* commit '0f24a3ca999a702f83af9307f9f47b6fdeb546a5': lavc: remove disabled FF_API_OLD_ENCODE_VIDEO cruft lavc: remove disabled FF_API_OLD_ENCODE_AUDIO cruft lavc: remove disabled FF_API_OLD_DECODE_AUDIO cruft Conflicts: libavcodec/flacenc.c libavcodec/libgsm.c libavcodec/utils.c libavcodec/version.h The compatibility wrapers are left as they likely sre still in wide use. They will be removed when they break or otherwise cause work without an volunteer being available. Merged-by: Michael Niedermayer <michaelni@gmx.at>
789 lines
23 KiB
C
789 lines
23 KiB
C
/*
|
|
* 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"
|
|
|
|
/* 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 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
|
|
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)(exp2((3 - i) / 3.0) * (1 << 20));
|
|
if (v <= 0)
|
|
v = 1;
|
|
scale_factor_table[i] = v;
|
|
#ifdef USE_FLOATS
|
|
scale_factor_inv_table[i] = exp2(-(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) * 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;
|
|
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;
|
|
}
|
|
|
|
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 = 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(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 */
|
|
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 = 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_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 += 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;
|
|
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 * 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;
|
|
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->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 },
|
|
};
|
|
|
|
AVCodec ff_mp2_encoder = {
|
|
.name = "mp2",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_MP2,
|
|
.priv_data_size = sizeof(MpegAudioContext),
|
|
.init = MPA_encode_init,
|
|
.encode2 = MPA_encode_frame,
|
|
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
|
|
AV_SAMPLE_FMT_NONE },
|
|
.supported_samplerates = (const int[]){
|
|
44100, 48000, 32000, 22050, 24000, 16000, 0
|
|
},
|
|
.channel_layouts = (const uint64_t[]){ AV_CH_LAYOUT_MONO,
|
|
AV_CH_LAYOUT_STEREO,
|
|
0 },
|
|
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
|
|
.defaults = mp2_defaults,
|
|
};
|