Mercurial > mplayer.hg
view libfaad2/ssr_ipqf.h @ 11511:6e580b901205
original config:
> ao_data.bps=channels*rate;
> if(format != AFMT_U8 && format != AFMT_S8)
> ao_data.bps*=2;
fallback config, before patch:
> ao_data.bps=ao_data.channels * ao_data.samplerate;
since we forced the format to S16_LE in fallback, we should double bps
to be consistent with an original config of the same settings.
author | joey |
---|---|
date | Sun, 23 Nov 2003 17:04:19 +0000 |
parents | 3185f64f6350 |
children | 4a370c80fe5c |
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/* ** FAAD - Freeware Advanced Audio Decoder ** Copyright (C) 2002 M. Bakker ** ** 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. ** ** This program 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 General Public License for more details. ** ** You should have received a copy of the GNU General Public License ** along with this program; if not, write to the Free Software ** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. ** ** $Id: ssr_ipqf.h,v 1.4 2003/09/09 18:09:52 menno Exp $ **/ #ifndef __SSR_IPQF_H__ #define __SSR_IPQF_H__ #ifdef __cplusplus extern "C" { #endif void ssr_ipqf(ssr_info *ssr, real_t *in_data, real_t *out_data, real_t buffer[SSR_BANDS][96/4], uint16_t frame_len, uint8_t bands); #ifdef __cplusplus } #endif #endif