/*
* 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
-#include <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-#include <math.h>
+
+/**
+ * @file mpegaudio.c
+ * The simplest mpeg audio layer 2 encoder.
+ */
+
#include "avcodec.h"
+#include "bitstream.h"
#include "mpegaudio.h"
-#define NDEBUG
-#include <assert.h>
+/* currently, cannot change these constants (need to modify
+ quantization stage) */
+#define FRAC_BITS 15
+#define WFRAC_BITS 14
+#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
+
+typedef struct MpegAudioContext {
+ PutBitContext pb;
+ int nb_channels;
+ int freq, bit_rate;
+ 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 */
+ 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 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
-#define MPA_STEREO 0
-#define MPA_JSTEREO 1
-#define MPA_DUAL 2
-#define MPA_MONO 3
-
#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;
int bitrate = avctx->bit_rate;
int channels = avctx->channels;
- int i, v, table, ch_bitrate;
+ int i, v, table;
float a;
if (channels > 2)
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 (freq_tab[i] == freq)
+ if (mpa_freq_tab[i] == freq)
break;
- if ((freq_tab[i] / 2) == freq) {
+ 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 (bitrate_tab[1-s->lsf][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 */
s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
/* select the right allocation table */
- ch_bitrate = bitrate / s->nb_channels;
- if (!s->lsf) {
- if ((freq == 48000 && ch_bitrate >= 56) ||
- (ch_bitrate >= 56 && ch_bitrate <= 80))
- table = 0;
- else if (freq != 48000 && ch_bitrate >= 96)
- table = 1;
- else if (freq != 32000 && ch_bitrate <= 48)
- table = 2;
- else
- table = 3;
- } else {
- table = 4;
- }
+ table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
+
/* number of used subbands */
s->sblimit = sblimit_table[table];
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<s->nb_channels;i++)
s->samples_offset[i] = 0;
- for(i=0;i<512;i++) {
- float a = enwindow[i] * 32768.0 * 16.0;
- filter_bank[i] = (int)(a);
+ for(i=0;i<257;i++) {
+ int v;
+ 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;
+ 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)
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 */
-static void idct32(int *out, int *tab, int sblimit, int left_shift)
+/* 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;
} 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 = 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;
}
/* compute the scale factor index using log 2 computations */
if (vmax > 0) {
- n = log2(vmax);
+ 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;
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",
j, i, vmax, scale_factor_table[index], index);
sf[1] = sf[2] = sf[0];
break;
default:
- abort();
+ assert(0); //cant happen
+ code = 0; /* kill warning */
}
#if 0
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;
}
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;
+}
AVCodec mp2_encoder = {
"mp2",
sizeof(MpegAudioContext),
MPA_encode_init,
MPA_encode_frame,
+ MPA_encode_close,
NULL,
};
+
+#undef FIX
projects / ffmpeg / blobdiff