sound, prepare FM filtering

pico/sound/resampler.h

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+/* Configurable fixed point resampling SINC filter for mono and stereo audio.
+ *
+ * (C) 2022 kub
+ *
+ * This work is licensed under the terms of any of these licenses
+ * (at your option):
+ *  - GNU GPL, version 2 or later.
+ *  - MAME license.
+ * See COPYING file in the top-level directory.
+ */
+
+struct resampler {
+  int	stereo;         // mono or stereo?
+  int   taps;           // taps to compute per output sample
+  int   interpolation;  // upsampling factor (numerator)
+  int   decimation;     // downsampling factor (denominator)
+  int   ratio_int;      // floor(decimation/interpolation)
+  u32   interp_inv;     // Q16, 1.0/interpolation
+  s16   *filter;        // filter taps
+  s32   *buffer;        // filter history and input buffer (w/o zero stuffing)
+  int   buffer_sz;      // buffer size in frames
+  int   buffer_idx;     // buffer offset
+  int	phase;          // filter phase for last output sample
+};
+typedef struct resampler resampler_t;
+
+
+/* Release a resampler */
+void resampler_free(resampler_t *r);
+/* Create a resampler with upsampling factor :interpolation and downsampling
+ * factor :decimation, Kaiser windowed SINC polyphase FIR with bank size :taps.
+ * The created filter has a size of :taps*:interpolation for upsampling and
+ * :taps*:decimation for downsampling. :taps is limiting the cost per sample and
+ * should be big enough to avoid inaccuracy (>= 8, higher is more accurate).
+ * :cutoff is in [0..1] with 1 representing the Nyquist rate after decimation.
+ * :beta is the Kaiser window beta.
+ * :max_input is the maximum length in a resampler_update call */
+resampler_t *resampler_new(unsigned taps, unsigned interpolation, unsigned decimation,
+       double cutoff, double beta, unsigned max_input, int stereo);
+/* Obtain :length resampled audio frames in :buffer. Use :get_samples to obtain
+ * the needed amount of input samples */
+void resampler_update(resampler_t *r, s32 *buffer, int length,
+       void (*generate_samples)(s32 *buffer, int length, int stereo));
+