view synth_filter.c @ 10469:23ca6d60184d libavcodec

Add support for hardcoding the mpegaudiodec tables. Reduces .bss size by about 194 kB.

/*
 * copyright (c) 2008 Michael Niedermayer <[e-mail]>
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg 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 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg 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
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "synth_filter.h"

void ff_synth_filter_float(FFTContext *imdct,
                         float *synth_buf_ptr, int *synth_buf_offset,
                         float synth_buf2[32], const float window[512],
                         float out[32], const float in[32], float scale, float bias)
{
    float *synth_buf= synth_buf_ptr + *synth_buf_offset;
    int i, j;

    ff_imdct_half(imdct, synth_buf, in);

    for (i = 0; i < 16; i++){
        float a= synth_buf2[i   ];
        float b= synth_buf2[i+16];
        float c= 0;
        float d= 0;
        for (j = 0; j < 512 - *synth_buf_offset; j += 64){
            a += window[i+j   ]*(-synth_buf[15-i+j]);
            b += window[i+j+16]*( synth_buf[   i+j]);
            c += window[i+j+32]*( synth_buf[16+i+j]);
            d += window[i+j+48]*( synth_buf[31-i+j]);
        }
        for (     ; j < 512; j += 64){
            a += window[i+j   ]*(-synth_buf[15-i+j-512]);
            b += window[i+j+16]*( synth_buf[   i+j-512]);
            c += window[i+j+32]*( synth_buf[16+i+j-512]);
            d += window[i+j+48]*( synth_buf[31-i+j-512]);
        }
        out[i   ] = a * scale + bias;
        out[i+16] = b * scale + bias;
        synth_buf2[i   ] = c;
        synth_buf2[i+16] = d;
    }
    *synth_buf_offset= (*synth_buf_offset-32)&511;
}