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[patch] More accurate resampling (Up to 2a0d717d55be revision) with kept interpolator

SDL_audiotypecvt.c.patch (text/plain), 8.42 KB, created by Vitaly Novichkov on 2017-01-08 00:08:47 UTC

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Creator: Vitaly Novichkov

Created: 2017-01-08 00:08:47 UTC

Size: 8.42 KB

patch

obsolete

>--- 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);
>     }
>

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Attachments on bug 3527: 2653 | 2655 | 2656 | 2663 | 2664 | 2665 | 2666 | 2667 | 2668

We are currently migrating Bugzilla to GitHub issues.Any changes made to the bug tracker now will be lost, so please do not post new bugs or make changes to them.When we're done, all bug URLs will redirect to their equivalent location on the new bug tracker.

We are currently migrating Bugzilla to GitHub issues.
Any changes made to the bug tracker now will be lost, so please do not post new bugs or make changes to them.
When we're done, all bug URLs will redirect to their equivalent location on the new bug tracker.