make wmv1 and wmv2 playable with M$ DMO decoder

[ffmpeg] / libavcodec / resample2.c

diff --git a/libavcodec/resample2.c b/libavcodec/resample2.c
index 70d8a1d956c9c4cb3b16a31985009e85e2bf4791..ffd6fc71c637475cd7e73f888901198968091e83 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
- *
  */
 
 /**
@@ -27,7 +26,6 @@
  */
 
 #include "avcodec.h"
-#include "common.h"
 #include "dsputil.h"
 
 #ifndef CONFIG_RESAMPLE_HP
@@ -35,17 +33,26 @@
 
 #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
-#else
+#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
 
 
@@ -86,7 +93,7 @@ static double bessel(double x){
  * @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;
 
@@ -123,8 +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
@@ -156,7 +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;
             }
@@ -166,8 +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));
@@ -256,21 +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(lrintf(val), -32768, 32767);
+#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;