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corrected mpeg audio encoding overflows - now it should give correct quality even for very high volumes 

Originally committed as revision 132 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Fabrice Bellard 2001-09-17 21:19:09 +00:00
parent c34270f5e8
commit afa982fdae
1 changed files with 24 additions and 30 deletions
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54
libavcodec/mpegaudio.c
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@ -1,6 +1,6 @@
/*
* The simplest mpeg audio layer 2 encoder
* Copyright (c) 2000 Gerard Lantau.
* Copyright (c) 2000, 2001 Gerard Lantau.
*
* 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
@ -20,9 +20,12 @@
#include <math.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 FRAC_BITS 15
#define WFRAC_BITS 14
#define MUL(a,b) (((INT64)(a) * (INT64)(b)) >> FRAC_BITS)
#define FIX(a) ((int)((a) * (1 << FRAC_BITS)))
#define SAMPLES_BUF_SIZE 4096
@ -119,7 +122,10 @@ int MPA_encode_init(AVCodecContext *avctx)
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;
@ -168,7 +174,7 @@ int MPA_encode_init(AVCodecContext *avctx)
}
/* 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;
@ -283,15 +289,17 @@ static void idct32(int *out, int *tab, int sblimit, int left_shift)
} 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;
@ -319,29 +327,15 @@ static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
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];
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;
/* 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;
}
idct32(out, tmp1, s->sblimit, n);
idct32(out, tmp1);
/* advance of 32 samples */
offset -= 32;
@ -391,9 +385,9 @@ static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
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);