@@ -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;

@@ -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  / 

-            (float) (iwidth - leftBand - rightBand));

-    build_filter(&s->v_filters[0][0], (float) s->pad_oheight / 

-            (float) (iheight - topBand - bottomBand));

+    av_build_filter(&s->h_filters[0][0], (float) s->pad_owidth  / 

+            (float) (iwidth - leftBand - rightBand), NB_TAPS, NB_PHASES, 1<<FILTER_BITS, 0);

+    av_build_filter(&s->v_filters[0][0], (float) s->pad_oheight / 

+            (float) (iheight - topBand - bottomBand), NB_TAPS, NB_PHASES, 1<<FILTER_BITS, 0);

     return s;

 fail:

@@ -24,103 +24,17 @@

 #include "avcodec.h"

-typedef struct {

-    /* 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 AVResampleContext;

 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++) {

-        init_mono_resample(&s->channel_ctx[i], s->ratio);

-    }

+    s->resample_context= av_resample_init(output_rate, input_rate);

+    

     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) );

-    bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );

+    for(i=0; i<s->filter_channels; i++){

+        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);

 }

new file mode 100644

@@ -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;

+}