polyphase kaiser windowed sinc and blackman nuttall windowed sinc audio resample filters

Originally committed as revision 3228 to svn://svn.ffmpeg.org/ffmpeg/trunk

1 files changed, 214 insertions, 0 deletions

diff --git a/libavcodec/resample2.c b/libavcodec/resample2.c
new file mode 100644
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--- /dev/null
+++ b/libavcodec/resample2.c
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+/*
+ * audio resampling
+ * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * This library 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.
+ *
+ * This library 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
+ *
+ */
+ 
+/**
+ * @file resample2.c
+ * audio resampling
+ * @author Michael Niedermayer <michaelni@gmx.at>
+ */
+
+#include "avcodec.h"
+#include "common.h"
+
+#define PHASE_SHIFT 10
+#define PHASE_COUNT (1<<PHASE_SHIFT)
+#define PHASE_MASK (PHASE_COUNT-1)
+#define FILTER_SHIFT 15
+
+typedef struct AVResampleContext{
+    short *filter_bank;
+    int filter_length;
+    int ideal_dst_incr;
+    int dst_incr;
+    int index;
+    int frac;
+    int src_incr;
+    int compensation_distance;
+}AVResampleContext;
+
+/**
+ * 0th order modified bessel function of the first kind.
+ */
+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);
+    }
+    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
+ */
+void av_build_filter(int16_t *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;
+
+    /* 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;
+            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;
+            case 2:
+                w = 2.0*x / (factor*tap_count*M_PI);
+                y *= bessel(16*sqrt(FFMAX(1-w*w, 0))) / bessel(16);
+                break;
+            }
+
+            tab[i] = y;
+            norm += y;
+        }
+
+        /* 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);
+            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 *c= av_mallocz(sizeof(AVResampleContext));
+    double factor= FFMIN(out_rate / (double)in_rate, 1.0);
+
+    memset(c, 0, sizeof(AVResampleContext));
+
+    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) + 1]= (1<<FILTER_SHIFT)-1;
+    c->filter_bank[c->filter_length*PHASE_COUNT + (c->filter_length-1) + 2]= 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);
+
+    return c;
+}
+
+void av_resample_close(AVResampleContext *c){
+    av_freep(&c->filter_bank);
+    av_freep(&c);
+}
+
+void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
+    assert(!c->compensation_distance); //FIXME
+
+    c->compensation_distance= compensation_distance;
+    c->dst_incr-= c->ideal_dst_incr * 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 frac= c->frac;
+    int dst_incr_frac= c->dst_incr % c->src_incr;
+    int dst_incr=      c->dst_incr / c->src_incr;
+    
+    if(c->compensation_distance && c->compensation_distance < dst_size)
+        dst_size= c->compensation_distance;
+    
+    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;
+        
+        if(sample_index < 0){
+            for(i=0; i<c->filter_length; i++)
+                val += src[ABS(sample_index + i)] * filter[i];
+        }else if(sample_index + c->filter_length > src_size){
+            break;
+        }else{
+#if 0
+            int64_t v=0;
+            int sub_phase= (frac<<12) / 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;
+                v += src[sample_index + i] * coeff;
+            }
+            val= v>>12;
+#else
+            for(i=0; i<c->filter_length; i++){
+                val += src[sample_index + i] * filter[i];
+            }
+#endif
+        }
+
+        val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
+        dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
+
+        frac += dst_incr_frac;
+        index += dst_incr;
+        if(frac >= c->src_incr){
+            frac -= c->src_incr;
+            index++;
+        }
+    }
+    if(update_ctx){
+        if(c->compensation_distance){
+            c->compensation_distance -= index;
+            if(!c->compensation_distance)
+                c->dst_incr= c->ideal_dst_incr;
+        }
+        c->frac= frac;
+        c->index=0;
+    }
+    *consumed= index >> PHASE_SHIFT;
+    return dst_index;
+}