audio: Replaced the resampler. Again.

This time it's using real math from a real whitepaper instead of my previous
amateur, fast-but-low-quality attempt. The new resampler does "bandlimited
interpolation," as described here: https://ccrma.stanford.edu/~jos/resample/

The output appears to sound cleaner, especially at high frequencies, and of
course works with non-power-of-two rate conversions.

There are some obvious optimizations to be done to this still, and there is
other fallout: this doesn't resample a buffer in-place, the 2-channels-Sint16
fast path is gone because this resampler does a _lot_ of floating point math.
There is a nasty hack to make it work with SDL_AudioCVT.

It's possible these issues are solvable, but they aren't solved as of yet.
Still, I hope this effort is slouching in the right direction.

src/audio/SDL_audio.c

@@ -1543,6 +1543,8 @@ SDL_AudioQuit(void)
 #ifdef HAVE_LIBSAMPLERATE_H
     UnloadLibSampleRate();
 #endif
+
+    SDL_FreeResampleFilter();
 }
 
 #define NUM_FORMATS 10

src/audio/SDL_audio_c.h

@@ -69,6 +69,11 @@ extern SDL_AudioFilter SDL_Convert_F32_to_S16;
 extern SDL_AudioFilter SDL_Convert_F32_to_U16;
 extern SDL_AudioFilter SDL_Convert_F32_to_S32;
 
+/* You need to call SDL_PrepareResampleFilter() before using the internal resampler.
+   SDL_AudioQuit() calls SDL_FreeResamplerFilter(), you should never call it yourself. */
+int SDL_PrepareResampleFilter(void);
+void SDL_FreeResampleFilter(void);
+
 
 /* SDL_AudioStream is a new audio conversion interface. It
     might eventually become a public API.