Mercurial > libavcodec.hg
view synth_filter.c @ 11792:a211c41465f3 libavcodec
aacenc: Remove unnecessary variables and scopes in the TLS.
author | alexc |
---|---|
date | Fri, 28 May 2010 20:42:01 +0000 |
parents | 18f17f44de37 |
children |
line wrap: on
line source
/* * copyright (c) 2008 Michael Niedermayer <michaelni@gmx.at> * * 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 "fft.h" #include "synth_filter.h" static void 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; } av_cold void ff_synth_filter_init(SynthFilterContext *c) { c->synth_filter_float = synth_filter_float; if (ARCH_ARM) ff_synth_filter_init_arm(c); }