diff options
author | Fabrice Bellard <fabrice@bellard.org> | 2001-09-18 01:19:09 +0400 |
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committer | Fabrice Bellard <fabrice@bellard.org> | 2001-09-18 01:19:09 +0400 |
commit | afa982fdae1b49a8aee00a27da876bba10ba1073 (patch) | |
tree | 5baf9343cb2a25c335d1ec3e1d5b1f63b314ab32 | |
parent | c34270f5e89fd95b933c483d602f2f298557d04b (diff) |
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
-rw-r--r-- | libavcodec/mpegaudio.c | 54 |
1 files changed, 24 insertions, 30 deletions
diff --git a/libavcodec/mpegaudio.c b/libavcodec/mpegaudio.c index 72661bbc0d..e8ec337466 100644 --- a/libavcodec/mpegaudio.c +++ b/libavcodec/mpegaudio.c @@ -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); |