*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
-
+
/**
* @file resample2.c
* audio resampling
double v=1;
double t=1;
int i;
-
+
for(i=1; i<50; i++){
t *= i;
v += pow(x*x/4, i)/(t*t);
AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
int phase_count= 1<<phase_shift;
-
+
c->phase_shift= phase_shift;
c->phase_mask= phase_count-1;
c->linear= linear;
* 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,
+ * 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){
int64_t index2= ((int64_t)index)<<32;
int64_t incr= (1LL<<32) * c->dst_incr / c->src_incr;
dst_size= FFMIN(dst_size, (src_size-1-index) * (int64_t)c->src_incr / c->dst_incr);
-
+
for(dst_index=0; dst_index < dst_size; dst_index++){
dst[dst_index] = src[index2>>32];
index2 += incr;
FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
int sample_index= index >> c->phase_shift;
FELEM2 val=0;
-
+
if(sample_index < 0){
for(i=0; i<c->filter_length; i++)
val += src[ABS(sample_index + i) % src_size] * filter[i];
c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
c->compensation_distance= compensation_distance;
}
-#if 0
+#if 0
if(update_ctx && !c->compensation_distance){
#undef rand
av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
}
#endif
-
+
return dst_index;
}