Remove unneeded avio wrapper functions.

[ffmpeg] / libavcodec / resample2.c

diff --git a/libavcodec/resample2.c b/libavcodec/resample2.c
index ce7daa38a822e7165216448772a600926e6fa89f..da1443d98f900ab505470b91e7021edb026cd008 100644 (file)
--- a/libavcodec/resample2.c
+++ b/libavcodec/resample2.c
@@ -17,7 +17,6 @@
  * 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
- *
  */
 
 /**
@@ -27,23 +26,33 @@
  */
 
 #include "avcodec.h"
-#include "common.h"
 #include "dsputil.h"
 
-#if 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
-#else
-#define FILTER_SHIFT 22
+#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
 
 
@@ -69,9 +78,10 @@ static double bessel(double x){
     double t=1;
     int i;
 
+    x= x*x/4;
     for(i=1; i<50; i++){
-        t *= i;
-        v += pow(x*x/4, i)/(t*t);
+        t *= x/(i*i);
+        v += t;
     }
     return v;
 }
@@ -80,10 +90,10 @@ static double bessel(double x){
  * 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
+ * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
  */
 void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
-    int ph, i, v;
+    int ph, i;
     double x, y, w, tab[tap_count];
     const int center= (tap_count-1)/2;
 
@@ -108,9 +118,9 @@ void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_coun
                 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;
             }
 
@@ -120,8 +130,11 @@ void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_coun
 
         /* normalize so that an uniform color remains the same */
         for(i=0;i<tap_count;i++) {
-            v = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
-            filter[ph * tap_count + i] = 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
@@ -153,7 +166,7 @@ void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_coun
             maxsf= FFMAX(maxsf, sf);
             minsf= FFMIN(minsf, sf);
             if(i%11==0){
-                av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%f-%f sf:%f-%f\n", i, ss, maxff, minff, maxsf, minsf);
+                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;
             }
@@ -163,8 +176,8 @@ void av_build_filter(FELEM *filter, double factor, int tap_count, int phase_coun
 }
 
 /**
- * initalizes a audio resampler.
- * note, if either rate is not a integer then simply scale both rates up so they are
+ * Initializes an audio resampler.
+ * Note, if either rate is not an integer then simply scale both rates up so they are.
  */
 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));
@@ -177,7 +190,7 @@ AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size,
 
     c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
     c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
-    av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, 1);
+    av_build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE);
     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];
 
@@ -253,21 +266,24 @@ int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int
         }else if(sample_index + c->filter_length > src_size){
             break;
         }else if(c->linear){
-            int64_t v=0;
-            int sub_phase= (frac<<8) / c->src_incr;
+            FELEM2 v2=0;
             for(i=0; i<c->filter_length; i++){
-                int64_t coeff= filter[i]*(256 - 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>>8;
+            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;