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

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

libavcodec/avcodec.h

@@ -1846,6 +1846,7 @@ extern AVCodec ac3_decoder;
 /* resample.c */
 
 struct ReSampleContext;
+struct AVResampleContext;
 
 typedef struct ReSampleContext ReSampleContext;
 
@@ -1854,6 +1855,9 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels,
 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples);
 void audio_resample_close(ReSampleContext *s);
 
+struct AVResampleContext *av_resample_init(int out_rate, int in_rate);
+int av_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx);
+
 /* YUV420 format is assumed ! */
 
 struct ImgReSampleContext;

libavcodec/imgresample.c

@@ -55,6 +55,8 @@ struct ImgReSampleContext {
     uint8_t *line_buf;
 };
 
+void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_count, int scale, int type);
+
 static inline int get_phase(int pos)
 {
     return ((pos) >> (POS_FRAC_BITS - PHASE_BITS)) & ((1 << PHASE_BITS) - 1);
@@ -540,48 +542,6 @@ static void component_resample(ImgReSampleContext *s,
     }
 }
 
-/* XXX: the following filter is quite naive, but it seems to suffice
-   for 4 taps */
-static void build_filter(int16_t *filter, float factor)
-{
-    int ph, i, v;
-    float x, y, tab[NB_TAPS], norm, mult, target;
-
-    /* if upsampling, only need to interpolate, no filter */
-    if (factor > 1.0)
-        factor = 1.0;
-
-    for(ph=0;ph<NB_PHASES;ph++) {
-        norm = 0;
-        for(i=0;i<NB_TAPS;i++) {
-#if 1
-            const float d= -0.5; //first order derivative = -0.5
-            x = fabs(((float)(i - FCENTER) - (float)ph / NB_PHASES) * 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);
-#else
-            x = M_PI * ((float)(i - FCENTER) - (float)ph / NB_PHASES) * factor;
-            if (x == 0)
-                y = 1.0;
-            else
-                y = sin(x) / x;
-#endif
-            tab[i] = y;
-            norm += y;
-        }
-
-        /* normalize so that an uniform color remains the same */
-        target= 1 << FILTER_BITS;
-        for(i=0;i<NB_TAPS;i++) {
-            mult = target / norm;
-            v = lrintf(tab[i] * mult);
-            filter[ph * NB_TAPS + i] = v;
-            norm -= tab[i];
-            target -= v;
-        }
-    }
-}
-
 ImgReSampleContext *img_resample_init(int owidth, int oheight,
                                       int iwidth, int iheight)
 {
@@ -626,10 +586,10 @@ ImgReSampleContext *img_resample_full_init(int owidth, int oheight,
     s->h_incr = ((iwidth - leftBand - rightBand) * POS_FRAC) / s->pad_owidth;
     s->v_incr = ((iheight - topBand - bottomBand) * POS_FRAC) / s->pad_oheight; 
 
-    build_filter(&s->h_filters[0][0], (float) s->pad_owidth  / 
+    av_build_filter(&s->h_filters[0][0], (float) s->pad_owidth  / 
-            (float) (iwidth - leftBand - rightBand));
+            (float) (iwidth - leftBand - rightBand), NB_TAPS, NB_PHASES, 1<<FILTER_BITS, 0);
-    build_filter(&s->v_filters[0][0], (float) s->pad_oheight / 
+    av_build_filter(&s->v_filters[0][0], (float) s->pad_oheight / 
-            (float) (iheight - topBand - bottomBand));
+            (float) (iheight - topBand - bottomBand), NB_TAPS, NB_PHASES, 1<<FILTER_BITS, 0);
 
     return s;
 fail:

libavcodec/resample.c

@@ -24,103 +24,17 @@
 
 #include "avcodec.h"
 
-typedef struct {
+struct AVResampleContext;
-    /* fractional resampling */
-    uint32_t incr; /* fractional increment */
-    uint32_t frac;
-    int last_sample;
-    /* integer down sample */
-    int iratio;  /* integer divison ratio */
-    int icount, isum;
-    int inv;
-} ReSampleChannelContext;
 
 struct ReSampleContext {
-    ReSampleChannelContext channel_ctx[2];
+    struct AVResampleContext *resample_context;
+    short *temp[2];
+    int temp_len;
     float ratio;
     /* channel convert */
     int input_channels, output_channels, filter_channels;
 };
 
-
-#define FRAC_BITS 16
-#define FRAC (1 << FRAC_BITS)
-
-static void init_mono_resample(ReSampleChannelContext *s, float ratio)
-{
-    ratio = 1.0 / ratio;
-    s->iratio = (int)floorf(ratio);
-    if (s->iratio == 0)
-        s->iratio = 1;
-    s->incr = (int)((ratio / s->iratio) * FRAC);
-    s->frac = FRAC;
-    s->last_sample = 0;
-    s->icount = s->iratio;
-    s->isum = 0;
-    s->inv = (FRAC / s->iratio);
-}
-
-/* fractional audio resampling */
-static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
-{
-    unsigned int frac, incr;
-    int l0, l1;
-    short *q, *p, *pend;
-
-    l0 = s->last_sample;
-    incr = s->incr;
-    frac = s->frac;
-
-    p = input;
-    pend = input + nb_samples;
-    q = output;
-
-    l1 = *p++;
-    for(;;) {
-        /* interpolate */
-        *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
-        frac = frac + s->incr;
-        while (frac >= FRAC) {
-            frac -= FRAC;
-            if (p >= pend)
-                goto the_end;
-            l0 = l1;
-            l1 = *p++;
-        }
-    }
- the_end:
-    s->last_sample = l1;
-    s->frac = frac;
-    return q - output;
-}
-
-static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
-{
-    short *q, *p, *pend;
-    int c, sum;
-
-    p = input;
-    pend = input + nb_samples;
-    q = output;
-
-    c = s->icount;
-    sum = s->isum;
-
-    for(;;) {
-        sum += *p++;
-        if (--c == 0) {
-            *q++ = (sum * s->inv) >> FRAC_BITS;
-            c = s->iratio;
-            sum = 0;
-        }
-        if (p >= pend)
-            break;
-    }
-    s->isum = sum;
-    s->icount = c;
-    return q - output;
-}
-
 /* n1: number of samples */
 static void stereo_to_mono(short *output, short *input, int n1)
 {
@@ -210,31 +124,6 @@ static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
     }
 }
 
-static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
-{
-    short *buf1;
-    short *buftmp;
-
-    buf1= (short*)av_malloc( nb_samples * sizeof(short) );
-
-    /* first downsample by an integer factor with averaging filter */
-    if (s->iratio > 1) {
-        buftmp = buf1;
-        nb_samples = integer_downsample(s, buftmp, input, nb_samples);
-    } else {
-        buftmp = input;
-    }
-
-    /* then do a fractional resampling with linear interpolation */
-    if (s->incr != FRAC) {
-        nb_samples = fractional_resample(s, output, buftmp, nb_samples);
-    } else {
-        memcpy(output, buftmp, nb_samples * sizeof(short));
-    }
-    av_free(buf1);
-    return nb_samples;
-}
-
 ReSampleContext *audio_resample_init(int output_channels, int input_channels, 
                                       int output_rate, int input_rate)
 {
@@ -271,16 +160,13 @@ ReSampleContext *audio_resample_init(int output_channels, int input_channels,
     if(s->filter_channels>2)
       s->filter_channels = 2;
 
-    for(i=0;i<s->filter_channels;i++) {
+    s->resample_context= av_resample_init(output_rate, input_rate);
-        init_mono_resample(&s->channel_ctx[i], s->ratio);
+    
-    }
     return s;
 }
 
 /* resample audio. 'nb_samples' is the number of input samples */
 /* XXX: optimize it ! */
-/* XXX: do it with polyphase filters, since the quality here is
-   HORRIBLE. Return the number of samples available in output */
 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
 {
     int i, nb_samples1;
@@ -296,8 +182,11 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
     }
 
     /* XXX: move those malloc to resample init code */
-    bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
+    for(i=0; i<s->filter_channels; i++){
-    bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
+        bufin[i]= (short*) av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
+        memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
+        buftmp2[i] = bufin[i] + s->temp_len;
+    }
     
     /* make some zoom to avoid round pb */
     lenout= (int)(nb_samples * s->ratio) + 16;
@@ -306,27 +195,32 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
 
     if (s->input_channels == 2 &&
         s->output_channels == 1) {
-        buftmp2[0] = bufin[0];
         buftmp3[0] = output;
         stereo_to_mono(buftmp2[0], input, nb_samples);
     } else if (s->output_channels >= 2 && s->input_channels == 1) {
-        buftmp2[0] = input;
         buftmp3[0] = bufout[0];
+        memcpy(buftmp2[0], input, nb_samples*sizeof(short));
     } else if (s->output_channels >= 2) {
-        buftmp2[0] = bufin[0];
-        buftmp2[1] = bufin[1];
         buftmp3[0] = bufout[0];
         buftmp3[1] = bufout[1];
         stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
     } else {
-        buftmp2[0] = input;
         buftmp3[0] = output;
+        memcpy(buftmp2[0], input, nb_samples*sizeof(short));
     }
 
+    nb_samples += s->temp_len;
+
     /* resample each channel */
     nb_samples1 = 0; /* avoid warning */
     for(i=0;i<s->filter_channels;i++) {
-        nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
+        int consumed;
+        int is_last= i+1 == s->filter_channels;
+
+        nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
+        s->temp_len= nb_samples - consumed;
+        s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
+        memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
     }
 
     if (s->output_channels == 2 && s->input_channels == 1) {
@@ -347,5 +241,8 @@ int audio_resample(ReSampleContext *s, short *output, short *input, int nb_sampl
 
 void audio_resample_close(ReSampleContext *s)
 {
+    av_resample_close(s->resample_context);
+    av_freep(&s->temp[0]);
+    av_freep(&s->temp[1]);
     av_free(s);
 }

libavcodec/resample2.c

@@ -0,0 +1,214 @@
+/*
+ * 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;
+}