* 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 Libav.
+ *
+ * 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 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,
+ * 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 this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
- *
+ * 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"
-#if 1
+#ifndef CONFIG_RESAMPLE_HP
#define FILTER_SHIFT 15
#define FELEM int16_t
#define FELEM2 int32_t
+#define FELEML int64_t
#define FELEM_MAX INT16_MAX
#define FELEM_MIN INT16_MIN
-#else
-#define FILTER_SHIFT 22
+#define WINDOW_TYPE 9
+#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
+#define FILTER_SHIFT 30
#define FELEM int32_t
#define FELEM2 int64_t
+#define FELEML int64_t
#define FELEM_MAX INT32_MAX
#define FELEM_MIN INT32_MIN
+#define WINDOW_TYPE 12
+#else
+#define FILTER_SHIFT 0
+
+#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;
/**
* 0th order modified bessel function of the first kind.
*/
-double bessel(double x){
+static double bessel(double x){
double v=1;
+ double lastv=0;
double t=1;
int i;
-
- for(i=1; i<50; i++){
- t *= i;
- v += pow(x*x/4, i)/(t*t);
+
+ x= x*x/4;
+ for(i=1; v != lastv; i++){
+ lastv=v;
+ t *= x/(i*i);
+ v += t;
}
return v;
}
/**
- * 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->kaiser windowed sinc beta=16
+ * @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;
for(ph=0;ph<phase_count;ph++) {
double norm = 0;
- double e= 0;
for(i=0;i<tap_count;i++) {
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
w = 2.0*x / (factor*tap_count) + M_PI;
y *= 0.3635819 - 0.4891775 * cos(w) + 0.1365995 * cos(2*w) - 0.0106411 * cos(3*w);
break;
- case 2:
+ default:
w = 2.0*x / (factor*tap_count*M_PI);
- y *= bessel(16*sqrt(FFMAX(1-w*w, 0)));
+ y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
break;
}
/* normalize so that an uniform color remains the same */
for(i=0;i<tap_count;i++) {
- v = clip(lrintf(tab[i] * scale / norm + e), FELEM_MIN, FELEM_MAX);
- filter[ph * tap_count + i] = v;
- e += tab[i] * scale / norm - v;
+#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
+ filter[ph * tap_count + i] = tab[i] / norm;
+#else
+ filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
+#endif
+ }
+ }
+#if 0
+ {
+#define LEN 1024
+ int j,k;
+ double sine[LEN + tap_count];
+ double filtered[LEN];
+ double maxff=-2, minff=2, maxsf=-2, minsf=2;
+ for(i=0; i<LEN; i++){
+ double ss=0, sf=0, ff=0;
+ for(j=0; j<LEN+tap_count; j++)
+ sine[j]= cos(i*j*M_PI/LEN);
+ for(j=0; j<LEN; j++){
+ double sum=0;
+ ph=0;
+ for(k=0; k<tap_count; k++)
+ sum += filter[ph * tap_count + k] * sine[k+j];
+ filtered[j]= sum / (1<<FILTER_SHIFT);
+ ss+= sine[j + center] * sine[j + center];
+ ff+= filtered[j] * filtered[j];
+ sf+= sine[j + center] * filtered[j];
+ }
+ ss= sqrt(2*ss/LEN);
+ ff= sqrt(2*ff/LEN);
+ sf= 2*sf/LEN;
+ maxff= FFMAX(maxff, ff);
+ minff= FFMIN(minff, ff);
+ maxsf= FFMAX(maxsf, sf);
+ minsf= FFMIN(minsf, sf);
+ if(i%11==0){
+ av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);
+ minff=minsf= 2;
+ maxff=maxsf= -2;
+ }
}
}
+#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;
-
- memset(c, 0, sizeof(AVResampleContext));
-
+
+ if (!c)
+ return NULL;
+
c->phase_shift= phase_shift;
c->phase_mask= phase_count-1;
c->linear= linear;
- c->filter_length= FFMAX(ceil(filter_size/factor), 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);
+ 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;
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){
- assert(index >= 0);
- for(dst_index=0; dst_index < dst_size; dst_index++){
- if(index < src_size)
- dst[dst_index] = src[index];
- else
- break;
+ 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);
- frac += dst_incr_frac;
- index += dst_incr;
- if(frac >= c->src_incr){
- frac -= c->src_incr;
- index++;
- }
+ 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++){
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(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++){
- int64_t 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_int16(lrintf(val));
+#else
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
+#endif
frac += dst_incr_frac;
index += dst_incr;
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