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FFmpeg/libavcodec/mpegaudioenc.c
Anton Khirnov 2df0c32ea1 lavc: use a separate field for exporting audio encoder padding 
Currently, the amount of padding inserted at the beginning by some audio
encoders, is exported through AVCodecContext.delay. However
- the term 'delay' is heavily overloaded and can have multiple different
  meanings even in the case of audio encoding.
- this field has entirely different meanings, depending on whether the
  codec context is used for encoding or decoding (and has yet another
  different meaning for video), preventing generic handling of the codec
  context.

Therefore, add a new field -- AVCodecContext.initial_padding. It could
conceivably be used for decoding as well at a later point.
2014-10-13 19:09:01 +00:00

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/*
* 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 "libavutil/channel_layout.h"
#include "avcodec.h"
#include "internal.h"
#include "put_bits.h"
#define FRAC_BITS 15 /* fractional bits for sb_samples and dct */
#define WFRAC_BITS 14 /* fractional bits for window */
#include "mpegaudio.h"
#include "mpegaudiodsp.h"
#include "mpegaudiodata.h"
#include "mpegaudiotab.h"
/* currently, cannot change these constants (need to modify
quantization stage) */
#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
#define SAMPLES_BUF_SIZE 4096
typedef struct MpegAudioContext {
PutBitContext pb;
int nb_channels;
int lsf; /* 1 if mpeg2 low bitrate selected */
int bitrate_index; /* bit rate */
int freq_index;
int frame_size; /* frame size, in bits, without padding */
/* padding computation */
int frame_frac, frame_frac_incr, do_padding;
short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
/* code to group 3 scale factors */
unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
int sblimit; /* number of used subbands */
const unsigned char *alloc_table;
int16_t filter_bank[512];
int scale_factor_table[64];
unsigned char scale_diff_table[128];
float scale_factor_inv_table[64];
unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
} MpegAudioContext;
static av_cold int MPA_encode_init(AVCodecContext *avctx)
{
MpegAudioContext *s = avctx->priv_data;
int freq = avctx->sample_rate;
int bitrate = avctx->bit_rate;
int channels = avctx->channels;
int i, v, table;
float a;
if (channels <= 0 || channels > 2){
av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
return AVERROR(EINVAL);
}
bitrate = bitrate / 1000;
s->nb_channels = channels;
avctx->frame_size = MPA_FRAME_SIZE;
avctx->initial_padding = 512 - 32 + 1;
/* encoding freq */
s->lsf = 0;
for(i=0;i<3;i++) {
if (avpriv_mpa_freq_tab[i] == freq)
break;
if ((avpriv_mpa_freq_tab[i] / 2) == freq) {
s->lsf = 1;
break;
}
}
if (i == 3){
av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
return AVERROR(EINVAL);
}
s->freq_index = i;
/* encoding bitrate & frequency */
for(i=0;i<15;i++) {
if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
break;
}
if (i == 15){
av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
return AVERROR(EINVAL);
}
s->bitrate_index = i;
/* compute total header size & pad bit */
a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
s->frame_size = ((int)a) * 8;
/* frame fractional size to compute padding */
s->frame_frac = 0;
s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
/* select the right allocation table */
table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
/* number of used subbands */
s->sblimit = ff_mpa_sblimit_table[table];
s->alloc_table = ff_mpa_alloc_tables[table];
av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
bitrate, freq, s->frame_size, table, s->frame_frac_incr);
for(i=0;i<s->nb_channels;i++)
s->samples_offset[i] = 0;
for(i=0;i<257;i++) {
int v;
v = ff_mpa_enwindow[i];
#if WFRAC_BITS != 16
v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
#endif
s->filter_bank[i] = v;
if ((i & 63) != 0)
v = -v;
if (i != 0)
s->filter_bank[512 - i] = v;
}
for(i=0;i<64;i++) {
v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
if (v <= 0)
v = 1;
s->scale_factor_table[i] = v;
s->scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
}
for(i=0;i<128;i++) {
v = i - 64;
if (v <= -3)
v = 0;
else if (v < 0)
v = 1;
else if (v == 0)
v = 2;
else if (v < 3)
v = 3;
else
v = 4;
s->scale_diff_table[i] = v;
}
for(i=0;i<17;i++) {
v = ff_mpa_quant_bits[i];
if (v < 0)
v = -v;
else
v = v * 3;
s->total_quant_bits[i] = 12 * v;
}
return 0;
}
/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
static void idct32(int *out, int *tab)
{
int i, j;
int *t, *t1, xr;
const int *xp = costab32;
for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
t = tab + 30;
t1 = tab + 2;
do {
t[0] += t[-4];
t[1] += t[1 - 4];
t -= 4;
} while (t != t1);
t = tab + 28;
t1 = tab + 4;
do {
t[0] += t[-8];
t[1] += t[1-8];
t[2] += t[2-8];
t[3] += t[3-8];
t -= 8;
} while (t != t1);
t = tab;
t1 = tab + 32;
do {
t[ 3] = -t[ 3];
t[ 6] = -t[ 6];
t[11] = -t[11];
t[12] = -t[12];
t[13] = -t[13];
t[15] = -t[15];
t += 16;
} while (t != t1);
t = tab;
t1 = tab + 8;
do {
int x1, x2, x3, x4;
x3 = MUL(t[16], FIX(SQRT2*0.5));
x4 = t[0] - x3;
x3 = t[0] + x3;
x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
x1 = MUL((t[8] - x2), xp[0]);
x2 = MUL((t[8] + x2), xp[1]);
t[ 0] = x3 + x1;
t[ 8] = x4 - x2;
t[16] = x4 + x2;
t[24] = x3 - x1;
t++;
} while (t != t1);
xp += 2;
t = tab;
t1 = tab + 4;
do {
xr = MUL(t[28],xp[0]);
t[28] = (t[0] - xr);
t[0] = (t[0] + xr);
xr = MUL(t[4],xp[1]);
t[ 4] = (t[24] - xr);
t[24] = (t[24] + xr);
xr = MUL(t[20],xp[2]);
t[20] = (t[8] - xr);
t[ 8] = (t[8] + xr);
xr = MUL(t[12],xp[3]);
t[12] = (t[16] - xr);
t[16] = (t[16] + xr);
t++;
} while (t != t1);
xp += 4;
for (i = 0; i < 4; i++) {
xr = MUL(tab[30-i*4],xp[0]);
tab[30-i*4] = (tab[i*4] - xr);
tab[ i*4] = (tab[i*4] + xr);
xr = MUL(tab[ 2+i*4],xp[1]);
tab[ 2+i*4] = (tab[28-i*4] - xr);
tab[28-i*4] = (tab[28-i*4] + xr);
xr = MUL(tab[31-i*4],xp[0]);
tab[31-i*4] = (tab[1+i*4] - xr);
tab[ 1+i*4] = (tab[1+i*4] + xr);
xr = MUL(tab[ 3+i*4],xp[1]);
tab[ 3+i*4] = (tab[29-i*4] - xr);
tab[29-i*4] = (tab[29-i*4] + xr);
xp += 2;
}
t = tab + 30;
t1 = tab + 1;
do {
xr = MUL(t1[0], *xp);
t1[0] = (t[0] - xr);
t[0] = (t[0] + xr);
t -= 2;
t1 += 2;
xp++;
} while (t >= tab);
for(i=0;i<32;i++) {
out[i] = tab[bitinv32[i]];
}
}
#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
{
short *p, *q;
int sum, offset, i, j;
int tmp[64];
int tmp1[32];
int *out;
offset = s->samples_offset[ch];
out = &s->sb_samples[ch][0][0][0];
for(j=0;j<36;j++) {
/* 32 samples at once */
for(i=0;i<32;i++) {
s->samples_buf[ch][offset + (31 - i)] = samples[0];
samples += incr;
}
/* filter */
p = s->samples_buf[ch] + offset;
q = s->filter_bank;
/* maxsum = 23169 */
for(i=0;i<64;i++) {
sum = p[0*64] * q[0*64];
sum += p[1*64] * q[1*64];
sum += p[2*64] * q[2*64];
sum += p[3*64] * q[3*64];
sum += p[4*64] * q[4*64];
sum += p[5*64] * q[5*64];
sum += p[6*64] * q[6*64];
sum += p[7*64] * q[7*64];
tmp[i] = sum;
p++;
q++;
}
tmp1[0] = tmp[16] >> WSHIFT;
for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
idct32(out, tmp1);
/* advance of 32 samples */
offset -= 32;
out += 32;
/* handle the wrap around */
if (offset < 0) {
memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
s->samples_buf[ch], (512 - 32) * 2);
offset = SAMPLES_BUF_SIZE - 512;
}
}
s->samples_offset[ch] = offset;
}
static void compute_scale_factors(MpegAudioContext *s,
unsigned char scale_code[SBLIMIT],
unsigned char scale_factors[SBLIMIT][3],
int sb_samples[3][12][SBLIMIT],
int sblimit)
{
int *p, vmax, v, n, i, j, k, code;
int index, d1, d2;
unsigned char *sf = &scale_factors[0][0];
for(j=0;j<sblimit;j++) {
for(i=0;i<3;i++) {
/* find the max absolute value */
p = &sb_samples[i][0][j];
vmax = abs(*p);
for(k=1;k<12;k++) {
p += SBLIMIT;
v = abs(*p);
if (v > vmax)
vmax = v;
}
/* compute the scale factor index using log 2 computations */
if (vmax > 1) {
n = av_log2(vmax);
/* n is the position of the MSB of vmax. now
use at most 2 compares to find the index */
index = (21 - n) * 3 - 3;
if (index >= 0) {
while (vmax <= s->scale_factor_table[index+1])
index++;
} else {
index = 0; /* very unlikely case of overflow */
}
} else {
index = 62; /* value 63 is not allowed */
}
av_dlog(NULL, "%2d:%d in=%x %x %d\n",
j, i, vmax, s->scale_factor_table[index], index);
/* store the scale factor */
assert(index >=0 && index <= 63);
sf[i] = index;
}
/* compute the transmission factor : look if the scale factors
are close enough to each other */
d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
d2 = s->scale_diff_table[sf[1] - sf[2] + 64];
/* handle the 25 cases */
switch(d1 * 5 + d2) {
case 0*5+0:
case 0*5+4:
case 3*5+4:
case 4*5+0:
case 4*5+4:
code = 0;
break;
case 0*5+1:
case 0*5+2:
case 4*5+1:
case 4*5+2:
code = 3;
sf[2] = sf[1];
break;
case 0*5+3:
case 4*5+3:
code = 3;
sf[1] = sf[2];
break;
case 1*5+0:
case 1*5+4:
case 2*5+4:
code = 1;
sf[1] = sf[0];
break;
case 1*5+1:
case 1*5+2:
case 2*5+0:
case 2*5+1:
case 2*5+2:
code = 2;
sf[1] = sf[2] = sf[0];
break;
case 2*5+3:
case 3*5+3:
code = 2;
sf[0] = sf[1] = sf[2];
break;
case 3*5+0:
case 3*5+1:
case 3*5+2:
code = 2;
sf[0] = sf[2] = sf[1];
break;
case 1*5+3:
code = 2;
if (sf[0] > sf[2])
sf[0] = sf[2];
sf[1] = sf[2] = sf[0];
break;
default:
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;
}
}
}
if (max_sb < 0)
break;
av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n",
current_frame_size, max_frame_size, max_sb, max_ch,
bit_alloc[max_ch][max_sb]);
/* find alloc table entry (XXX: not optimal, should use
pointer table) */
alloc = s->alloc_table;
for(i=0;i<max_sb;i++) {
alloc += 1 << alloc[0];
}
if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
/* nothing was coded for this band: add the necessary bits */
incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
incr += s->total_quant_bits[alloc[1]];
} else {
/* increments bit allocation */
b = bit_alloc[max_ch][max_sb];
incr = s->total_quant_bits[alloc[b + 1]] -
s->total_quant_bits[alloc[b]];
}
if (current_frame_size + incr <= max_frame_size) {
/* can increase size */
b = ++bit_alloc[max_ch][max_sb];
current_frame_size += incr;
/* decrease smr by the resolution we added */
smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
/* max allocation size reached ? */
if (b == ((1 << alloc[0]) - 1))
subband_status[max_ch][max_sb] = SB_NOMORE;
else
subband_status[max_ch][max_sb] = SB_ALLOCATED;
} else {
/* cannot increase the size of this subband */
subband_status[max_ch][max_sb] = SB_NOMORE;
}
}
*padding = max_frame_size - current_frame_size;
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++) {
float a;
sample = s->sb_samples[ch][k][l + m][i];
/* divide by scale factor */
a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]];
q[m] = (int)((a + 1.0) * steps * 0.5);
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, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
MpegAudioContext *s = avctx->priv_data;
const int16_t *samples = (const int16_t *)frame->data[0];
short smr[MPA_MAX_CHANNELS][SBLIMIT];
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
int padding, i, ret;
for(i=0;i<s->nb_channels;i++) {
filter(s, i, samples + i, s->nb_channels);
}
for(i=0;i<s->nb_channels;i++) {
compute_scale_factors(s, s->scale_code[i], s->scale_factors[i],
s->sb_samples[i], s->sblimit);
}
for(i=0;i<s->nb_channels;i++) {
psycho_acoustic_model(s, smr[i]);
}
compute_bit_allocation(s, smr, bit_alloc, &padding);
if ((ret = ff_alloc_packet(avpkt, MPA_MAX_CODED_FRAME_SIZE))) {
av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
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->initial_padding);
avpkt->size = put_bits_count(&s->pb) / 8;
*got_packet_ptr = 1;
return 0;
}
static const AVCodecDefault mp2_defaults[] = {
{ "b", "384000" },
{ NULL },
};
AVCodec ff_mp2_encoder = {
.name = "mp2",
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
.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 },
.defaults = mp2_defaults,
};
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