* 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
- *
*/
/**
- * @file resample2.c
+ * @file libavcodec/resample2.c
* audio resampling
* @author Michael Niedermayer <michaelni@gmx.at>
*/
#include "avcodec.h"
-#include "common.h"
#include "dsputil.h"
#ifndef CONFIG_RESAMPLE_HP
#define FELEM int16_t
#define FELEM2 int32_t
+#define FELEML int64_t
#define FELEM_MAX INT16_MAX
#define FELEM_MIN INT16_MIN
-#else
+#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;
double t=1;
int i;
+ x= x*x/4;
for(i=1; i<50; i++){
- t *= i;
- v += pow(x*x/4, i)/(t*t);
+ t *= x/(i*i);
+ v += t;
}
return v;
}
* builds 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
*/
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;
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 = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
- filter[ph * tap_count + i] = 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
maxsf= FFMAX(maxsf, sf);
minsf= FFMIN(minsf, sf);
if(i%11==0){
- av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%f-%f sf:%f-%f\n", i, ss, maxff, minff, maxsf, minsf);
+ 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
}
-/**
- * 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);
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);
+ av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE);
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];
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++){
- 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;
projects / frescor / ffmpeg.git / blobdiff