Mercurial > libavcodec.hg
view synth_filter.c @ 11401:e340262ba532 libavcodec
Add an HE-AAC v1 decoder.
A large portion of this code was orignally authored by Robert Swain. The rest
was written by me. Full history is available at:
svn://svn.ffmpeg.org/soc/aac-sbr
http://github.com/aconverse/ffmpeg-heaac/tree/sbr_pub
author | alexc |
---|---|
date | Mon, 08 Mar 2010 04:33:02 +0000 |
parents | 4b3da727d832 |
children | 18f17f44de37 |
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/* * 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" 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; }