/*
* The simplest mpeg audio layer 2 encoder
- * Copyright (c) 2000 Gerard Lantau.
+ * Copyright (c) 2000, 2001 Fabrice Bellard.
*
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU 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 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 program is distributed in the hope that it will be useful,
+ * 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 General Public License for more details.
+ * 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 General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ * 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., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
+
+/**
+ * @file mpegaudio.c
+ * The simplest mpeg audio layer 2 encoder.
+ */
+
#include "avcodec.h"
-#include <math.h>
+#include "bitstream.h"
#include "mpegaudio.h"
-#define DCT_BITS 14 /* number of bits for the DCT */
-#define MUL(a,b) (((a) * (b)) >> DCT_BITS)
-#define FIX(a) ((int)((a) * (1 << DCT_BITS)))
+/* currently, cannot change these constants (need to modify
+ quantization stage) */
+#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
+#define FIX(a) ((int)((a) * (1 << FRAC_BITS)))
#define SAMPLES_BUF_SIZE 4096
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 */
+ int64_t 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 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];
+ unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
int sblimit; /* number of used subbands */
const unsigned char *alloc_table;
} MpegAudioContext;
#include "mpegaudiotab.h"
-int MPA_encode_init(AVCodecContext *avctx)
+static int MPA_encode_init(AVCodecContext *avctx)
{
MpegAudioContext *s = avctx->priv_data;
int freq = avctx->sample_rate;
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)
+ if (mpa_freq_tab[i] == freq)
break;
if ((mpa_freq_tab[i] / 2) == freq) {
s->lsf = 1;
break;
}
}
- if (i == 3)
+ 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 (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
+ if (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
break;
}
- if (i == 15)
+ 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 = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
s->alloc_table = alloc_tables[table];
#ifdef DEBUG
- printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
+ av_log(avctx, AV_LOG_DEBUG, "%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<257;i++) {
int v;
- v = (mpa_enwindow[i] + 2) >> 2;
+ 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;
v = 2;
else if (v < 3)
v = 3;
- else
+ else
v = 4;
scale_diff_table[i] = v;
}
for(i=0;i<17;i++) {
v = quant_bits[i];
- if (v < 0)
+ 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 sblimit, int left_shift)
+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[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[ 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]);
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);
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;
}
} while (t >= tab);
for(i=0;i<32;i++) {
- out[i] = tab[bitinv32[i]] << left_shift;
+ 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, norm, n;
- short tmp[64];
+ int sum, offset, i, j;
+ int tmp[64];
int tmp1[32];
int *out;
sum += p[5*64] * q[5*64];
sum += p[6*64] * q[6*64];
sum += p[7*64] * q[7*64];
- tmp[i] = sum >> 14;
+ tmp[i] = sum;
p++;
q++;
}
- tmp1[0] = tmp[16];
- for( i=1; i<=16; i++ ) tmp1[i] = tmp[i+16]+tmp[16-i];
- for( i=17; i<=31; i++ ) tmp1[i] = tmp[i+16]-tmp[80-i];
-
- /* integer IDCT 32 with normalization. XXX: There may be some
- overflow left */
- norm = 0;
- for(i=0;i<32;i++) {
- norm |= abs(tmp1[i]);
- }
- n = av_log2(norm) - 12;
- if (n > 0) {
- for(i=0;i<32;i++)
- tmp1[i] >>= n;
- } else {
- n = 0;
- }
+ 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, s->sblimit, n);
+ 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),
+ memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
s->samples_buf[ch], (512 - 32) * 2);
offset = SAMPLES_BUF_SIZE - 512;
}
}
static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
- unsigned char scale_factors[SBLIMIT][3],
+ 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 */
/* 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
+ /* 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) {
index = 0; /* very unlikely case of overflow */
}
} else {
- index = 63;
+ index = 62; /* value 63 is not allowed */
}
-
+
#if 0
- printf("%2d:%d in=%x %x %d\n",
+ printf("%2d:%d in=%x %x %d\n",
j, i, vmax, scale_factor_table[index], index);
#endif
/* store the scale factor */
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:
sf[1] = sf[2] = sf[0];
break;
default:
- abort();
+ assert(0); //cant happen
+ code = 0; /* kill warning */
}
-
+
#if 0
- printf("%d: %2d %2d %2d %d %d -> %d\n", j,
+ printf("%d: %2d %2d %2d %d %d -> %d\n", j,
sf[0], sf[1], sf[2], d1, d2, code);
#endif
scale_code[j] = code;
/* 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,
+static void compute_bit_allocation(MpegAudioContext *s,
short smr1[MPA_MAX_CHANNELS][SBLIMIT],
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
int *padding)
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 0
- printf("current=%d max=%d max_sb=%d alloc=%d\n",
+ printf("current=%d max=%d max_sb=%d alloc=%d\n",
current_frame_size, max_frame_size, max_sb,
bit_alloc[max_sb]);
-#endif
+#endif
if (max_sb < 0)
break;
-
+
/* find alloc table entry (XXX: not optimal, should use
pointer table) */
alloc = s->alloc_table;
} else {
/* increments bit allocation */
b = bit_alloc[max_ch][max_sb];
- incr = total_quant_bits[alloc[b + 1]] -
+ incr = total_quant_bits[alloc[b + 1]] -
total_quant_bits[alloc[b]];
}
}
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])
+ if (bit_alloc[ch][i])
put_bits(p, 2, s->scale_code[ch][i]);
}
}
}
}
}
-
+
/* quantization & write sub band samples */
for(k=0;k<3;k++) {
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);
bits = quant_bits[qindex];
if (bits < 0) {
/* group the 3 values to save bits */
- put_bits(p, -bits,
+ put_bits(p, -bits,
q[0] + steps * (q[1] + steps * q[2]));
#if 0
- printf("%d: gr1 %d\n",
+ 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",
+ printf("%d: gr3 %d %d %d\n",
i, q[0], q[1], q[2]);
-#endif
+#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;
+ j += 1 << bit_alloc_bits;
}
}
}
flush_put_bits(p);
}
-int MPA_encode_frame(AVCodecContext *avctx,
- unsigned char *frame, int buf_size, void *data)
+static int MPA_encode_frame(AVCodecContext *avctx,
+ unsigned char *frame, int buf_size, void *data)
{
MpegAudioContext *s = avctx->priv_data;
short *samples = data;
}
for(i=0;i<s->nb_channels;i++) {
- compute_scale_factors(s->scale_code[i], s->scale_factors[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++) {
}
compute_bit_allocation(s, smr, bit_alloc, &padding);
- init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
+ init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
encode_frame(s, bit_alloc, padding);
-
+
s->nb_samples += MPA_FRAME_SIZE;
- return s->pb.buf_ptr - s->pb.buf;
+ return pbBufPtr(&s->pb) - s->pb.buf;
}
+static int MPA_encode_close(AVCodecContext *avctx)
+{
+ av_freep(&avctx->coded_frame);
+ return 0;
+}
+#ifdef CONFIG_MP2_ENCODER
AVCodec mp2_encoder = {
"mp2",
CODEC_TYPE_AUDIO,
sizeof(MpegAudioContext),
MPA_encode_init,
MPA_encode_frame,
+ MPA_encode_close,
NULL,
};
+#endif // CONFIG_MP2_ENCODER
+
+#undef FIX
projects / ffmpeg / blobdiff