* audio resampling
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file libavcodec/resample2.c
+ * @file
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
*/
static double bessel(double x){
double v=1;
+ double lastv=0;
double t=1;
int i;
x= x*x/4;
- for(i=1; i<50; i++){
+ for(i=1; v != lastv; i++){
+ lastv=v;
t *= x/(i*i);
v += t;
}
}
/**
- * builds a polyphase filterbank.
+ * Build a polyphase filterbank.
* @param factor resampling factor
* @param scale wanted sum of coefficients for each filter
* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
+ * @return 0 on success, negative on error
*/
-void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
+static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
int ph, i;
- double x, y, w, tab[tap_count];
+ double x, y, w;
+ double *tab = av_malloc(tap_count * sizeof(*tab));
const int center= (tap_count-1)/2;
+ if (!tab)
+ return AVERROR(ENOMEM);
+
/* if upsampling, only need to interpolate, no filter */
if (factor > 1.0)
factor = 1.0;
}
}
#endif
+
+ av_free(tab);
+ return 0;
}
AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
int phase_count= 1<<phase_shift;
+ if (!c)
+ return NULL;
+
c->phase_shift= phase_shift;
c->phase_mask= phase_count-1;
c->linear= linear;
c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
- av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE);
+ if (!c->filter_bank)
+ goto error;
+ if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
+ goto error;
memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
c->index= -phase_count*((c->filter_length-1)/2);
return c;
+error:
+ av_free(c->filter_bank);
+ av_free(c);
+ return NULL;
}
void av_resample_close(AVResampleContext *c){