* audio resampling
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
- * This library is free software; you can redistribute it and/or
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library is distributed in the hope that it will be useful,
+ * FFmpeg 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 this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*
*/
-
+
/**
* @file resample2.c
* audio resampling
#include "common.h"
#include "dsputil.h"
-#define PHASE_SHIFT 10
-#define PHASE_COUNT (1<<PHASE_SHIFT)
-#define PHASE_MASK (PHASE_COUNT-1)
+#if 1
#define FILTER_SHIFT 15
+#define FELEM int16_t
+#define FELEM2 int32_t
+#define FELEM_MAX INT16_MAX
+#define FELEM_MIN INT16_MIN
+#else
+#define FILTER_SHIFT 22
+
+#define FELEM int32_t
+#define FELEM2 int64_t
+#define FELEM_MAX INT32_MAX
+#define FELEM_MIN INT32_MIN
+#endif
+
+
typedef struct AVResampleContext{
- short *filter_bank;
+ FELEM *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
int frac;
int src_incr;
int compensation_distance;
+ int phase_shift;
+ int phase_mask;
+ int linear;
}AVResampleContext;
/**
* 0th order modified bessel function of the first kind.
*/
-double bessel(double x){
+static double bessel(double x){
double v=1;
double t=1;
int i;
-
+
for(i=1; i<50; i++){
t *= i;
v += pow(x*x/4, i)/(t*t);
* @param scale wanted sum of coefficients for each filter
* @param type 0->cubic, 1->blackman nuttall windowed sinc, 2->kaiser windowed sinc beta=16
*/
-void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type){
+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];
const int center= (tap_count-1)/2;
/* normalize so that an uniform color remains the same */
for(i=0;i<tap_count;i++) {
- v = clip(lrintf(tab[i] * scale / norm + e), -32768, 32767);
+ v = av_clip(lrintf(tab[i] * scale / norm + e), FELEM_MIN, FELEM_MAX);
filter[ph * tap_count + i] = v;
e += tab[i] * scale / norm - v;
}
* 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){
+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 / (double)in_rate, 1.0);
+ double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
+ int phase_count= 1<<phase_shift;
- memset(c, 0, sizeof(AVResampleContext));
+ c->phase_shift= phase_shift;
+ c->phase_mask= phase_count-1;
+ c->linear= linear;
- c->filter_length= ceil(16.0/factor);
- c->filter_bank= av_mallocz(c->filter_length*(PHASE_COUNT+1)*sizeof(short));
- av_build_filter(c->filter_bank, factor, c->filter_length, PHASE_COUNT, 1<<FILTER_SHIFT, 1);
- c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 1]= (1<<FILTER_SHIFT)-1;
- c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1)/2 + 2]= 1;
+ 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, 1);
+ 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->src_incr= out_rate;
- c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT;
- c->index= -PHASE_COUNT*((c->filter_length-1)/2);
+ c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
+ c->index= -phase_count*((c->filter_length-1)/2);
return 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;
int dst_incr_frac= c->dst_incr % c->src_incr;
int dst_incr= c->dst_incr / c->src_incr;
int compensation_distance= c->compensation_distance;
-
+
+ if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
+ 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;
+ }
+ frac += dst_index * dst_incr_frac;
+ index += dst_index * dst_incr;
+ index += frac / c->src_incr;
+ frac %= c->src_incr;
+ }else{
for(dst_index=0; dst_index < dst_size; dst_index++){
- short *filter= c->filter_bank + c->filter_length*(index & PHASE_MASK);
- int sample_index= index >> PHASE_SHIFT;
- int val=0;
-
+ 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];
+ val += src[FFABS(sample_index + i) % src_size] * filter[i];
}else if(sample_index + c->filter_length > src_size){
break;
- }else{
-#if 0
+ }else if(c->linear){
int64_t v=0;
- int sub_phase= (frac<<12) / c->src_incr;
+ int sub_phase= (frac<<8) / c->src_incr;
for(i=0; i<c->filter_length; i++){
- int64_t coeff= filter[i]*(4096 - sub_phase) + filter[i + c->filter_length]*sub_phase;
+ int64_t coeff= filter[i]*(256 - sub_phase) + filter[i + c->filter_length]*sub_phase;
v += src[sample_index + i] * coeff;
}
- val= v>>12;
-#else
+ val= v>>8;
+ }else{
for(i=0; i<c->filter_length; i++){
- val += src[sample_index + i] * filter[i];
+ val += src[sample_index + i] * (FELEM2)filter[i];
}
-#endif
}
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst_incr= c->ideal_dst_incr / c->src_incr;
}
}
- *consumed= FFMAX(index, 0) >> PHASE_SHIFT;
- index= FFMIN(index, 0);
+ }
+ *consumed= FFMAX(index, 0) >> c->phase_shift;
+ if(index>=0) index &= c->phase_mask;
if(compensation_distance){
compensation_distance -= dst_index;
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;
}
projects / ffmpeg / blobdiff