* 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 "avcodec.h"
-#include "common.h"
#include "dsputil.h"
#ifndef CONFIG_RESAMPLE_HP
#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
* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
*/
void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
- int ph, i, v;
+ int ph, i;
double x, y, w, tab[tap_count];
const int center= (tap_count-1)/2;
}
/**
- * initalizes a audio resampler.
- * note, if either rate is not a integer then simply scale both rates up so they are
+ * Initializes an audio resampler.
+ * Note, if either rate is not an 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));
/**
* Compensates samplerate/timestamp drift. The compensation is done by changing
- * the resampler parameters, so no audible clicks or similar distortions ocur
+ * the resampler parameters, so no audible clicks or similar distortions occur
* @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
*
* @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
+ * @return the number of samples written in dst or -1 if an error occurred
*/
int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
int dst_index, i;
}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;