comparison mpegaudio.c @ 89:2e88e3afecd0 libavcodec

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

88:06f63b58d2a8

/*
 * The simplest mpeg audio layer 2 encoder
 * Copyright (c) 2000 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
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.

 */
#include "avcodec.h"
#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)))

#define SAMPLES_BUF_SIZE 4096

typedef struct MpegAudioContext {
    PutBitContext pb;

    for(i=0;i<s->nb_channels;i++)
        s->samples_offset[i] = 0;

    for(i=0;i<257;i++) {
        int v;
        v = (mpa_enwindow[i] + 2) >> 2;
        filter_bank[i] = v;
        if ((i & 63) != 0)
            v = -v;
        if (i != 0)
            filter_bank[512 - i] = v;

    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)
{
    int i, j;
    int *t, *t1, xr;
    const int *xp = costab32;

        t1 += 2;
        xp++;
    } while (t >= tab);

    for(i=0;i<32;i++) {
        out[i] = tab[bitinv32[i]] << left_shift;
    }
}

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 tmp1[32];
    int *out;

    //    print_pow1(samples, 1152);

            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 >> 14;
            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;
        }

        idct32(out, tmp1, s->sblimit, n);

        /* advance of 32 samples */
        offset -= 32;
        out += 32;
        /* handle the wrap around */

                        index++;
                } else {
                    index = 0; /* very unlikely case of overflow */
                }
            } else {
                index = 63;
            }
            
#if 0
            printf("%2d:%d in=%x %x %d\n", 
                   j, i, vmax, scale_factor_table[index], index);
#endif
            /* store the scale factor */

89:2e88e3afecd0

/*
 * The simplest mpeg audio layer 2 encoder
 * 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
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.

 */
#include "avcodec.h"
#include <math.h>
#include "mpegaudio.h"

/* 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

typedef struct MpegAudioContext {
    PutBitContext pb;

    for(i=0;i<s->nb_channels;i++)
        s->samples_offset[i] = 0;

    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;

    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;

        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, short *samples, int incr)
{
    short *p, *q;
    int sum, offset, i, j;
    int tmp[64];
    int tmp1[32];
    int *out;

    //    print_pow1(samples, 1152);

            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 */

                        index++;
                } else {
                    index = 0; /* very unlikely case of overflow */
                }
            } else {
                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);
#endif
            /* store the scale factor */