X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fresample2.c;h=48c20c2cbb18b0ac132089e4539379b3eaa5e8f3;hb=3ab770001817e0f52114a9876819f07fcd8ed93a;hp=44761825cf3d034c1028fdc7a74f6f4ca0fff298;hpb=2ac615da82459989bc02f33fd45a464ab42a77b8;p=ffmpeg

diff --git a/libavcodec/resample2.c b/libavcodec/resample2.c
index 44761825cf3..48c20c2cbb1 100644
--- a/libavcodec/resample2.c
+++ b/libavcodec/resample2.c
@@ -2,39 +2,63 @@
  * audio resampling
  * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
  *
- * This library is free software; you can redistribute it and/or
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ * version 2.1 of the License, or (at your option) any later version.
  *
- * This library is distributed in the hope that it will be useful,
+ * Libav 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
- *
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
- 
+
 /**
- * @file resample2.c
+ * @file
  * audio resampling
  * @author Michael Niedermayer <michaelni@gmx.at>
  */
 
 #include "avcodec.h"
-#include "common.h"
 #include "dsputil.h"
 
-#define PHASE_SHIFT 10
-#define PHASE_COUNT (1<<PHASE_SHIFT)
-#define PHASE_MASK (PHASE_COUNT-1)
+#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{
-    short *filter_bank;
+    const AVClass *av_class;
+    FELEM *filter_bank;
     int filter_length;
     int ideal_dst_incr;
     int dst_incr;
@@ -42,41 +66,51 @@ typedef struct AVResampleContext{
     int frac;
     int src_incr;
     int compensation_distance;
+    int phase_shift;
+    int phase_mask;
+    int linear;
 }AVResampleContext;
 
 /**
  * 0th order modified bessel function of the first kind.
  */
-double bessel(double x){
+static double bessel(double x){
     double v=1;
+    double lastv=0;
     double t=1;
     int i;
-    
-    for(i=1; i<50; i++){
-        t *= i;
-        v += pow(x*x/4, i)/(t*t);
+
+    x= x*x/4;
+    for(i=1; v != lastv; i++){
+        lastv=v;
+        t *= x/(i*i);
+        v += t;
     }
     return v;
 }
 
 /**
- * builds a polyphase filterbank.
+ * Build 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
+ * @return 0 on success, negative on error
  */
-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];
+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;
-        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;
@@ -92,9 +126,9 @@ void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_co
                 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;
             }
 
@@ -104,34 +138,84 @@ void av_build_filter(int16_t *filter, double factor, int tap_count, int phase_co
 
         /* 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;
+#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;
 }
 
-/**
- * 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 *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 / (double)in_rate, 1.0);
+    double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
+    int phase_count= 1<<phase_shift;
 
-    memset(c, 0, sizeof(AVResampleContext));
+    if (!c)
+        return NULL;
 
-    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);
-    memcpy(&c->filter_bank[c->filter_length*PHASE_COUNT+1], c->filter_bank, (c->filter_length-1)*sizeof(short));
-    c->filter_bank[c->filter_length*PHASE_COUNT]= c->filter_bank[c->filter_length - 1];
+    c->phase_shift= phase_shift;
+    c->phase_mask= phase_count-1;
+    c->linear= linear;
+
+    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->src_incr= out_rate;
-    c->ideal_dst_incr= c->dst_incr= in_rate * PHASE_COUNT;
-    c->index= -PHASE_COUNT*((c->filter_length-1)/2);
+    c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
+    c->index= -phase_count*((c->filter_length-1)/2);
 
     return c;
+error:
+    av_free(c->filter_bank);
+    av_free(c);
+    return NULL;
 }
 
 void av_resample_close(AVResampleContext *c){
@@ -145,15 +229,6 @@ void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensa
     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;
@@ -161,35 +236,50 @@ int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int
     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;
+        }
+        frac += dst_index * dst_incr_frac;
+        index += dst_index * dst_incr;
+        index += frac / c->src_incr;
+        frac %= c->src_incr;
+  }else{
     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;
-                
+        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[ABS(sample_index + i) % src_size] * filter[i];
+                val += src[FFABS(sample_index + i) % src_size] * 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;
+        }else if(c->linear){
+            FELEM2 v2=0;
             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 += src[sample_index + i] * (FELEM2)filter[i];
+                v2  += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
             }
-            val= v>>12;
-#else
+            val+=(v2-val)*(FELEML)frac / c->src_incr;
+        }else{
             for(i=0; i<c->filter_length; i++){
-                val += src[sample_index + i] * filter[i];
+                val += src[sample_index + i] * (FELEM2)filter[i];
             }
-#endif
         }
 
+#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;
@@ -204,8 +294,9 @@ int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int
             dst_incr=      c->ideal_dst_incr / c->src_incr;
         }
     }
-    *consumed= FFMAX(index, 0) >> PHASE_SHIFT;
-    index= FFMIN(index, 0);
+  }
+    *consumed= FFMAX(index, 0) >> c->phase_shift;
+    if(index>=0) index &= c->phase_mask;
 
     if(compensation_distance){
         compensation_distance -= dst_index;
@@ -217,13 +308,13 @@ int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int
         c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
         c->compensation_distance= compensation_distance;
     }
-#if 0    
+#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;
 }