--- SDL_audiotypecvt_old.c
+++ SDL_audiotypecvt.c
@@ -216,19 +216,31 @@
     }
 }

+/* Works a bit glitchy, need to more work on it for Arbitrary resapling! */
+#define TRY_INTERPOLATE_ARBITRARY
+/* Without it sound is a bit "metallic" after up-sampling and some down-sampling,
+   but stable and no clicks and no clips */
+
 void
 SDL_Upsample_Arbitrary(SDL_AudioCVT *cvt, const int channels)
 {
-    const int srcsize = cvt->len_cvt - (64 * channels);
-    const int dstsize = (int) ((((double)(cvt->len_cvt/(channels*4))) * cvt->rate_incr)) * (channels*4);
-    register int eps = 0;
-    float *dst = ((float *) (cvt->buf + dstsize)) - channels;
-    const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - channels;
-    const float *target = ((const float *) cvt->buf);
-    const size_t cpy = sizeof (float) * channels;
+    const size_t cpy = sizeof (float) * (size_t)channels;
+    const int s_size = cvt->len_cvt / (int)cpy;
+    const int d_size = (int)(((double)s_size) * cvt->rate_incr);
+    float *target = ((float *) cvt->buf);
+          float *dst = target + (d_size * channels) - channels;
+    const float *src = target + (s_size * channels) - channels;
+
+    #ifdef TRY_INTERPOLATE_ARBITRARY
+    float last_sample[8];
+    #endif
     float sample[8];
-    float last_sample[8];
-    int i;
+    #ifdef TRY_INTERPOLATE_ARBITRARY
+    int i;
+    #endif
+
+    register double eps = 0.0;
+    double offset   = 1.0 / cvt->rate_incr;

 #if DEBUG_CONVERT
     fprintf(stderr, "Upsample arbitrary (x%f), %d channels.\n", cvt->rate_incr, channels);
@@ -236,17 +248,27 @@

     SDL_assert(channels <= 8);

+    #ifndef TRY_INTERPOLATE_ARBITRARY
+    SDL_memcpy(sample, src, cpy);
+    #else
     for (i = 0; i < channels; i++) {
-        sample[i] = (float) ((((double) src[i]) + ((double) src[i - channels])) * 0.5);
+        sample[i] = (float) ((((double) src[i]) + ((double)(src - channels)[i])) * 0.5);
     }
     SDL_memcpy(last_sample, src, cpy);
+    #endif

     while (dst > target) {
         SDL_memcpy(dst, sample, cpy);
         dst -= channels;
-        eps += srcsize;
-        if ((eps << 1) >= dstsize) {
+        eps += offset;
+        if(eps >= 1.0) {
+            #ifndef TRY_INTERPOLATE_ARBITRARY
             if (src > target) {
+                src -= channels;
+                SDL_memcpy(sample, src, cpy);
+            }
+            #else
+            if(src > target) {
                 src -= channels;
                 for (i = 0; i < channels; i++) {
                     sample[i] = (float) ((((double) src[i]) + ((double) last_sample[i])) * 0.5);
@@ -255,11 +277,12 @@

             }
             SDL_memcpy(last_sample, src, cpy);
-            eps -= dstsize;
-        }
-    }
-
-    cvt->len_cvt = dstsize;
+            #endif
+            eps -= 1.0;
+        }
+    }
+
+    cvt->len_cvt = d_size * (int)cpy;
     if (cvt->filters[++cvt->filter_index]) {
         cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
     }
@@ -268,16 +291,20 @@
 void
 SDL_Downsample_Arbitrary(SDL_AudioCVT *cvt, const int channels)
 {
-    const int srcsize = cvt->len_cvt - (64 * channels);
-    const int dstsize = (int) (((double)(cvt->len_cvt/(channels*4))) * cvt->rate_incr) * (channels*4);
-    register int eps = 0;
-    float *dst = (float *) cvt->buf;
-    const float *src = (float *) cvt->buf;
-    const float *target = (const float *) (cvt->buf + dstsize);
-    const size_t cpy = sizeof (float) * channels;
+    const size_t cpy = sizeof(float) * (size_t)channels;
+    const int s_size = cvt->len_cvt / (int)cpy;
+    const int d_size = (int)(((double)s_size) * cvt->rate_incr);
+    float *dst = (float *)cvt->buf;
+    const float *src = (float *)cvt->buf;
+    const float *target = src + (s_size * channels);
+    #ifdef TRY_INTERPOLATE_ARBITRARY
     float last_sample[8];
     float sample[8];
     int i;
+    #endif
+
+    register double eps     = 0.0;
+    double offset  = cvt->rate_incr;

 #if DEBUG_CONVERT
     fprintf(stderr, "Downsample arbitrary (x%f), %d channels.\n", cvt->rate_incr, channels);
@@ -285,24 +312,31 @@

     SDL_assert(channels <= 8);

+    #ifdef TRY_INTERPOLATE_ARBITRARY
     SDL_memcpy(sample, src, cpy);
     SDL_memcpy(last_sample, src, cpy);
-
-    while (dst < target) {
+    #endif
+
+    while(dst < target) {
         src += channels;
-        eps += dstsize;
-        if ((eps << 1) >= srcsize) {
+        eps += offset;
+        if(eps >= 1.0) {
+            #ifndef TRY_INTERPOLATE_ARBITRARY
+            dst += channels;
+            SDL_memcpy(dst, src, cpy);
+            #else
             SDL_memcpy(dst, sample, cpy);
             dst += channels;
             for (i = 0; i < channels; i++) {
                 sample[i] = (float) ((((double) src[i]) + ((double) last_sample[i])) * 0.5);
             }
-            SDL_memcpy(last_sample, src, cpy);
-            eps -= srcsize;
-        }
-    }
-
-    cvt->len_cvt = dstsize;
+            SDL_memcpy(last_sample, sample, cpy);
+            #endif
+            eps -= 1.0;
+        }
+    }
+
+    cvt->len_cvt = d_size * (int)cpy;
     if (cvt->filters[++cvt->filter_index]) {
         cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
     }
@@ -312,13 +346,16 @@
 SDL_Upsample_Multiple(SDL_AudioCVT *cvt, const int channels)
 {
     const int multiple = (int) cvt->rate_incr;
-    const int dstsize = cvt->len_cvt * multiple;
-    float *buf = (float *) cvt->buf;
-    float *dst = ((float *) (cvt->buf + dstsize)) - channels;
-    const float *src = ((float *) (cvt->buf + cvt->len_cvt)) - channels;
-    const float *target = buf + channels;
-    const size_t cpy = sizeof (float) * channels;
+    const size_t cpy = sizeof (float) * (size_t)channels;
+    const int s_size = cvt->len_cvt / (int)cpy;
+    const int d_size = s_size * multiple;
+    float *buf = ((float *)cvt->buf);
+    float *target = buf/*+ channels*/;
+          float *dst = buf + (d_size * channels) - channels;
+    const float *src = buf + (s_size * channels) - channels;
+    #ifdef TRY_INTERPOLATE_ARBITRARY
     float last_sample[8];
+    #endif
     int i;

 #if DEBUG_CONVERT
@@ -327,11 +364,14 @@

     SDL_assert(channels <= 8);

+    #ifdef TRY_INTERPOLATE_ARBITRARY
     SDL_memcpy(last_sample, src, cpy);
+    #endif

     while (dst > target) {
         SDL_assert(src >= buf);

+        #ifdef TRY_INTERPOLATE_ARBITRARY
         for (i = 0; i < channels; i++) {
             dst[i] = (float) ((((double)src[i]) + ((double)last_sample[i])) * 0.5);
         }
@@ -346,9 +386,16 @@
         if (src > buf) {
             SDL_memcpy(last_sample, src - channels, cpy);
         }
-    }
-
-    cvt->len_cvt = dstsize;
+        #else
+        for (i = 0; i < multiple; i++) {
+            SDL_memcpy(dst, src, cpy);
+            dst -= channels;
+        }
+        src -= channels;
+        #endif
+    }
+
+    cvt->len_cvt = d_size * (int)cpy;
     if (cvt->filters[++cvt->filter_index]) {
         cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
     }
@@ -358,32 +405,43 @@
 SDL_Downsample_Multiple(SDL_AudioCVT *cvt, const int channels)
 {
     const int multiple = (int) (1.0 / cvt->rate_incr);
-    const int dstsize = cvt->len_cvt / multiple;
-    float *dst = (float *) cvt->buf;
-    const float *src = (float *) cvt->buf;
-    const float *target = (const float *) (cvt->buf + dstsize);
-    const size_t cpy = sizeof (float) * channels;
+    const size_t cpy = sizeof(float) * (size_t)channels;
+    const int s_size = cvt->len_cvt / (int)cpy;
+    const int d_size = s_size / multiple;
+          float *dst = (float *)cvt->buf;
+    const float *src = (float *)cvt->buf;
+    const float *target = src + (s_size * channels);
+
+    #ifdef TRY_INTERPOLATE_ARBITRARY
     float last_sample[8];
     int i;
+    #endif

 #if DEBUG_CONVERT
     fprintf(stderr, "Downsample (x%d), %d channels.\n", multiple, channels);
 #endif

     SDL_assert(channels <= 8);
+    #ifdef TRY_INTERPOLATE_ARBITRARY
     SDL_memcpy(last_sample, src, cpy);
+    #endif

     while (dst < target) {
+        #ifdef TRY_INTERPOLATE_ARBITRARY
         for (i = 0; i < channels; i++) {
             dst[i] = (float) ((((double)src[i]) + ((double)last_sample[i])) * 0.5);
         }
         dst += channels;
-
         SDL_memcpy(last_sample, src, cpy);
+        #else
+        dst += channels;
+        SDL_memcpy(dst, src, cpy);
+        #endif
+
         src += (channels * multiple);
     }

-    cvt->len_cvt = dstsize;
+    cvt->len_cvt = d_size * (int)cpy;
     if (cvt->filters[++cvt->filter_index]) {
         cvt->filters[cvt->filter_index](cvt, AUDIO_F32SYS);
     }