* 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 resample2.c
+ * @file
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "avcodec.h"
-#include "common.h"
#include "dsputil.h"
+#include "libavutil/common.h"
+
+#if FF_API_AVCODEC_RESAMPLE
#ifndef CONFIG_RESAMPLE_HP
#define FILTER_SHIFT 15
#else
#define FILTER_SHIFT 0
-#define FELEM long double
-#define FELEM2 long double
-#define FELEML long double
+#define FELEM double
+#define FELEM2 double
+#define FELEML double
#define WINDOW_TYPE 24
#endif
typedef struct AVResampleContext{
+ const AVClass *av_class;
FELEM *filter_bank;
int filter_length;
int ideal_dst_incr;
*/
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){
- int ph, i, v;
- double x, y, w, tab[tap_count];
+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;
+ 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;
}
-/**
- * initalizes a audio resampler.
- * note, if either rate is not a integer then simply scale both rates up so they are
- */
AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
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){
av_freep(&c);
}
-/**
- * Compensates samplerate/timestamp drift. The compensation is done by changing
- * the resampler parameters, so no audible clicks or similar distortions ocur
- * @param compensation_distance distance in output samples over which the compensation should be performed
- * @param sample_delta number of output samples which should be output less
- *
- * example: av_resample_compensate(c, 10, 500)
- * here instead of 510 samples only 500 samples would be output
- *
- * note, due to rounding the actual compensation might be slightly different,
- * especially if the compensation_distance is large and the in_rate used during init is small
- */
void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
// sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
c->compensation_distance= compensation_distance;
c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
}
-/**
- * resamples.
- * @param src an array of unconsumed samples
- * @param consumed the number of samples of src which have been consumed are returned here
- * @param src_size the number of unconsumed samples available
- * @param dst_size the amount of space in samples available in dst
- * @param update_ctx if this is 0 then the context wont be modified, that way several channels can be resampled with the same context
- * @return the number of samples written in dst or -1 if an error occured
- */
int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
int dst_index, i;
int index= c->index;
}else if(sample_index + c->filter_length > src_size){
break;
}else if(c->linear){
- int64_t v=0;
- int sub_phase= (frac<<8) / c->src_incr;
+ FELEM2 v2=0;
for(i=0; i<c->filter_length; i++){
- FELEML coeff= filter[i]*(256 - sub_phase) + filter[i + c->filter_length]*sub_phase;
- v += src[sample_index + i] * coeff;
+ val += src[sample_index + i] * (FELEM2)filter[i];
+ v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
}
- val= v>>8;
+ val+=(v2-val)*(FELEML)frac / c->src_incr;
}else{
for(i=0; i<c->filter_length; i++){
val += src[sample_index + i] * (FELEM2)filter[i];
}
#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
- dst[dst_index] = av_clip(lrintf(val), -32768, 32767);
+ dst[dst_index] = av_clip_int16(lrintf(val));
#else
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
return dst_index;
}
+
+#endif
projects / ffmpeg / blobdiff