swr: rename resample2 to resample.
authorClément Bœsch <ubitux@gmail.com>
Tue, 15 Nov 2011 21:17:58 +0000 (22:17 +0100)
committerClément Bœsch <ubitux@gmail.com>
Wed, 16 Nov 2011 06:27:20 +0000 (07:27 +0100)
libswresample/Makefile
libswresample/resample.c [new file with mode: 0644]
libswresample/resample2.c [deleted file]

index 7435162..cc3038f 100644 (file)
@@ -5,7 +5,7 @@ FFLIBS = avutil
 
 HEADERS = swresample.h
 
-OBJS = swresample.o audioconvert.o resample2.o rematrix.o
+OBJS = swresample.o audioconvert.o resample.o rematrix.o
 
 TESTPROGS = swresample_test
 
diff --git a/libswresample/resample.c b/libswresample/resample.c
new file mode 100644 (file)
index 0000000..02f29ba
--- /dev/null
@@ -0,0 +1,352 @@
+/*
+ * audio resampling
+ * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * 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 2.1 of the License, or (at your option) any later version.
+ *
+ * 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 FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * audio resampling
+ * @author Michael Niedermayer <michaelni@gmx.at>
+ */
+
+#include "libavutil/log.h"
+#include "swresample_internal.h"
+
+#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
+#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;
+    int dst_incr;
+    int index;
+    int frac;
+    int src_incr;
+    int compensation_distance;
+    int phase_shift;
+    int phase_mask;
+    int linear;
+    double factor;
+}AVResampleContext;
+
+/**
+ * 0th order modified bessel function of the first kind.
+ */
+static double bessel(double x){
+    double v=1;
+    double lastv=0;
+    double t=1;
+    int i;
+    static const double inv[100]={
+ 1.0/( 1* 1), 1.0/( 2* 2), 1.0/( 3* 3), 1.0/( 4* 4), 1.0/( 5* 5), 1.0/( 6* 6), 1.0/( 7* 7), 1.0/( 8* 8), 1.0/( 9* 9), 1.0/(10*10),
+ 1.0/(11*11), 1.0/(12*12), 1.0/(13*13), 1.0/(14*14), 1.0/(15*15), 1.0/(16*16), 1.0/(17*17), 1.0/(18*18), 1.0/(19*19), 1.0/(20*20),
+ 1.0/(21*21), 1.0/(22*22), 1.0/(23*23), 1.0/(24*24), 1.0/(25*25), 1.0/(26*26), 1.0/(27*27), 1.0/(28*28), 1.0/(29*29), 1.0/(30*30),
+ 1.0/(31*31), 1.0/(32*32), 1.0/(33*33), 1.0/(34*34), 1.0/(35*35), 1.0/(36*36), 1.0/(37*37), 1.0/(38*38), 1.0/(39*39), 1.0/(40*40),
+ 1.0/(41*41), 1.0/(42*42), 1.0/(43*43), 1.0/(44*44), 1.0/(45*45), 1.0/(46*46), 1.0/(47*47), 1.0/(48*48), 1.0/(49*49), 1.0/(50*50),
+ 1.0/(51*51), 1.0/(52*52), 1.0/(53*53), 1.0/(54*54), 1.0/(55*55), 1.0/(56*56), 1.0/(57*57), 1.0/(58*58), 1.0/(59*59), 1.0/(60*60),
+ 1.0/(61*61), 1.0/(62*62), 1.0/(63*63), 1.0/(64*64), 1.0/(65*65), 1.0/(66*66), 1.0/(67*67), 1.0/(68*68), 1.0/(69*69), 1.0/(70*70),
+ 1.0/(71*71), 1.0/(72*72), 1.0/(73*73), 1.0/(74*74), 1.0/(75*75), 1.0/(76*76), 1.0/(77*77), 1.0/(78*78), 1.0/(79*79), 1.0/(80*80),
+ 1.0/(81*81), 1.0/(82*82), 1.0/(83*83), 1.0/(84*84), 1.0/(85*85), 1.0/(86*86), 1.0/(87*87), 1.0/(88*88), 1.0/(89*89), 1.0/(90*90),
+ 1.0/(91*91), 1.0/(92*92), 1.0/(93*93), 1.0/(94*94), 1.0/(95*95), 1.0/(96*96), 1.0/(97*97), 1.0/(98*98), 1.0/(99*99), 1.0/(10000)
+    };
+
+    x= x*x/4;
+    for(i=0; v != lastv; i++){
+        lastv=v;
+        t *= x*inv[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..16->kaiser windowed sinc beta=2..16
+ * @return 0 on success, negative on error
+ */
+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;
+        for(i=0;i<tap_count;i++) {
+            x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
+            if (x == 0) y = 1.0;
+            else        y = sin(x) / x;
+            switch(type){
+            case 0:{
+                const float d= -0.5; //first order derivative = -0.5
+                x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
+                if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*(            -x*x + x*x*x);
+                else      y=                       d*(-4 + 8*x - 5*x*x + x*x*x);
+                break;}
+            case 1:
+                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;
+            default:
+                w = 2.0*x / (factor*tap_count*M_PI);
+                y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
+                break;
+            }
+
+            tab[i] = y;
+            norm += y;
+        }
+
+        /* normalize so that an uniform color remains the same */
+        for(i=0;i<tap_count;i++) {
+#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;
+}
+
+AVResampleContext *swr_resample_init(AVResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
+    double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
+    int phase_count= 1<<phase_shift;
+
+    if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor
+           || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1)) {
+        c = av_mallocz(sizeof(AVResampleContext));
+        if (!c)
+            return NULL;
+
+        c->phase_shift   = phase_shift;
+        c->phase_mask    = phase_count - 1;
+        c->linear        = linear;
+        c->factor        = factor;
+        c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);
+        c->filter_bank   = av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
+        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->compensation_distance= 0;
+    if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
+        goto error;
+    c->ideal_dst_incr= c->dst_incr;
+
+    c->index= -phase_count*((c->filter_length-1)/2);
+    c->frac= 0;
+
+    return c;
+error:
+    av_free(c->filter_bank);
+    av_free(c);
+    return NULL;
+}
+
+void swr_resample_free(AVResampleContext **c){
+    if(!*c)
+        return;
+    av_freep(&(*c)->filter_bank);
+    av_freep(c);
+}
+
+void swr_compensate(struct SwrContext *s, int sample_delta, int compensation_distance){
+    AVResampleContext *c= s->resample;
+//    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;
+}
+
+int swr_resample(AVResampleContext *c, short *dst, const short *src, int *consumed, int src_size, int dst_size, int update_ctx){
+    int dst_index, i;
+    int index= c->index;
+    int frac= c->frac;
+    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;
+        }
+        index += dst_index * dst_incr;
+        index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
+        frac   = (frac + dst_index * (int64_t)dst_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[FFABS(sample_index + i) % src_size] * filter[i];
+            }else if(sample_index + c->filter_length > src_size){
+                break;
+            }else if(c->linear){
+                FELEM2 v2=0;
+                for(i=0; i<c->filter_length; i++){
+                    val += src[sample_index + i] * (FELEM2)filter[i];
+                    v2  += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
+                }
+                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;
+            if(frac >= c->src_incr){
+                frac -= c->src_incr;
+                index++;
+            }
+
+            if(dst_index + 1 == compensation_distance){
+                compensation_distance= 0;
+                dst_incr_frac= c->ideal_dst_incr % c->src_incr;
+                dst_incr=      c->ideal_dst_incr / c->src_incr;
+            }
+        }
+    }
+    *consumed= FFMAX(index, 0) >> c->phase_shift;
+    if(index>=0) index &= c->phase_mask;
+
+    if(compensation_distance){
+        compensation_distance -= dst_index;
+        assert(compensation_distance > 0);
+    }
+    if(update_ctx){
+        c->frac= frac;
+        c->index= index;
+        c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
+        c->compensation_distance= compensation_distance;
+    }
+#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;
+}
+
+int swr_multiple_resample(AVResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
+    int i, ret= -1;
+
+    for(i=0; i<dst->ch_count; i++){
+        ret= swr_resample(c, (short*)dst->ch[i], (const short*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
+    }
+
+    return ret;
+}
diff --git a/libswresample/resample2.c b/libswresample/resample2.c
deleted file mode 100644 (file)
index 02f29ba..0000000
+++ /dev/null
@@ -1,352 +0,0 @@
-/*
- * audio resampling
- * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
- *
- * 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 2.1 of the License, or (at your option) any later version.
- *
- * 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 FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * audio resampling
- * @author Michael Niedermayer <michaelni@gmx.at>
- */
-
-#include "libavutil/log.h"
-#include "swresample_internal.h"
-
-#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
-#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;
-    int dst_incr;
-    int index;
-    int frac;
-    int src_incr;
-    int compensation_distance;
-    int phase_shift;
-    int phase_mask;
-    int linear;
-    double factor;
-}AVResampleContext;
-
-/**
- * 0th order modified bessel function of the first kind.
- */
-static double bessel(double x){
-    double v=1;
-    double lastv=0;
-    double t=1;
-    int i;
-    static const double inv[100]={
- 1.0/( 1* 1), 1.0/( 2* 2), 1.0/( 3* 3), 1.0/( 4* 4), 1.0/( 5* 5), 1.0/( 6* 6), 1.0/( 7* 7), 1.0/( 8* 8), 1.0/( 9* 9), 1.0/(10*10),
- 1.0/(11*11), 1.0/(12*12), 1.0/(13*13), 1.0/(14*14), 1.0/(15*15), 1.0/(16*16), 1.0/(17*17), 1.0/(18*18), 1.0/(19*19), 1.0/(20*20),
- 1.0/(21*21), 1.0/(22*22), 1.0/(23*23), 1.0/(24*24), 1.0/(25*25), 1.0/(26*26), 1.0/(27*27), 1.0/(28*28), 1.0/(29*29), 1.0/(30*30),
- 1.0/(31*31), 1.0/(32*32), 1.0/(33*33), 1.0/(34*34), 1.0/(35*35), 1.0/(36*36), 1.0/(37*37), 1.0/(38*38), 1.0/(39*39), 1.0/(40*40),
- 1.0/(41*41), 1.0/(42*42), 1.0/(43*43), 1.0/(44*44), 1.0/(45*45), 1.0/(46*46), 1.0/(47*47), 1.0/(48*48), 1.0/(49*49), 1.0/(50*50),
- 1.0/(51*51), 1.0/(52*52), 1.0/(53*53), 1.0/(54*54), 1.0/(55*55), 1.0/(56*56), 1.0/(57*57), 1.0/(58*58), 1.0/(59*59), 1.0/(60*60),
- 1.0/(61*61), 1.0/(62*62), 1.0/(63*63), 1.0/(64*64), 1.0/(65*65), 1.0/(66*66), 1.0/(67*67), 1.0/(68*68), 1.0/(69*69), 1.0/(70*70),
- 1.0/(71*71), 1.0/(72*72), 1.0/(73*73), 1.0/(74*74), 1.0/(75*75), 1.0/(76*76), 1.0/(77*77), 1.0/(78*78), 1.0/(79*79), 1.0/(80*80),
- 1.0/(81*81), 1.0/(82*82), 1.0/(83*83), 1.0/(84*84), 1.0/(85*85), 1.0/(86*86), 1.0/(87*87), 1.0/(88*88), 1.0/(89*89), 1.0/(90*90),
- 1.0/(91*91), 1.0/(92*92), 1.0/(93*93), 1.0/(94*94), 1.0/(95*95), 1.0/(96*96), 1.0/(97*97), 1.0/(98*98), 1.0/(99*99), 1.0/(10000)
-    };
-
-    x= x*x/4;
-    for(i=0; v != lastv; i++){
-        lastv=v;
-        t *= x*inv[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..16->kaiser windowed sinc beta=2..16
- * @return 0 on success, negative on error
- */
-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;
-        for(i=0;i<tap_count;i++) {
-            x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
-            if (x == 0) y = 1.0;
-            else        y = sin(x) / x;
-            switch(type){
-            case 0:{
-                const float d= -0.5; //first order derivative = -0.5
-                x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
-                if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*(            -x*x + x*x*x);
-                else      y=                       d*(-4 + 8*x - 5*x*x + x*x*x);
-                break;}
-            case 1:
-                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;
-            default:
-                w = 2.0*x / (factor*tap_count*M_PI);
-                y *= bessel(type*sqrt(FFMAX(1-w*w, 0)));
-                break;
-            }
-
-            tab[i] = y;
-            norm += y;
-        }
-
-        /* normalize so that an uniform color remains the same */
-        for(i=0;i<tap_count;i++) {
-#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;
-}
-
-AVResampleContext *swr_resample_init(AVResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
-    double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
-    int phase_count= 1<<phase_shift;
-
-    if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor
-           || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1)) {
-        c = av_mallocz(sizeof(AVResampleContext));
-        if (!c)
-            return NULL;
-
-        c->phase_shift   = phase_shift;
-        c->phase_mask    = phase_count - 1;
-        c->linear        = linear;
-        c->factor        = factor;
-        c->filter_length = FFMAX((int)ceil(filter_size/factor), 1);
-        c->filter_bank   = av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
-        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->compensation_distance= 0;
-    if(!av_reduce(&c->src_incr, &c->dst_incr, out_rate, in_rate * (int64_t)phase_count, INT32_MAX/2))
-        goto error;
-    c->ideal_dst_incr= c->dst_incr;
-
-    c->index= -phase_count*((c->filter_length-1)/2);
-    c->frac= 0;
-
-    return c;
-error:
-    av_free(c->filter_bank);
-    av_free(c);
-    return NULL;
-}
-
-void swr_resample_free(AVResampleContext **c){
-    if(!*c)
-        return;
-    av_freep(&(*c)->filter_bank);
-    av_freep(c);
-}
-
-void swr_compensate(struct SwrContext *s, int sample_delta, int compensation_distance){
-    AVResampleContext *c= s->resample;
-//    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;
-}
-
-int swr_resample(AVResampleContext *c, short *dst, const short *src, int *consumed, int src_size, int dst_size, int update_ctx){
-    int dst_index, i;
-    int index= c->index;
-    int frac= c->frac;
-    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;
-        }
-        index += dst_index * dst_incr;
-        index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
-        frac   = (frac + dst_index * (int64_t)dst_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[FFABS(sample_index + i) % src_size] * filter[i];
-            }else if(sample_index + c->filter_length > src_size){
-                break;
-            }else if(c->linear){
-                FELEM2 v2=0;
-                for(i=0; i<c->filter_length; i++){
-                    val += src[sample_index + i] * (FELEM2)filter[i];
-                    v2  += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
-                }
-                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;
-            if(frac >= c->src_incr){
-                frac -= c->src_incr;
-                index++;
-            }
-
-            if(dst_index + 1 == compensation_distance){
-                compensation_distance= 0;
-                dst_incr_frac= c->ideal_dst_incr % c->src_incr;
-                dst_incr=      c->ideal_dst_incr / c->src_incr;
-            }
-        }
-    }
-    *consumed= FFMAX(index, 0) >> c->phase_shift;
-    if(index>=0) index &= c->phase_mask;
-
-    if(compensation_distance){
-        compensation_distance -= dst_index;
-        assert(compensation_distance > 0);
-    }
-    if(update_ctx){
-        c->frac= frac;
-        c->index= index;
-        c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
-        c->compensation_distance= compensation_distance;
-    }
-#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;
-}
-
-int swr_multiple_resample(AVResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){
-    int i, ret= -1;
-
-    for(i=0; i<dst->ch_count; i++){
-        ret= swr_resample(c, (short*)dst->ch[i], (const short*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count);
-    }
-
-    return ret;
-}