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+/* Copyright 2012 Dietrich Epp <depp@zdome.net> */
+#pragma once
+#include "types.h"
+#include "param.h"
+
+#include <vector>
+
+/* Bits used for interpolating between two filters.  The scalar
+   implementation supports up to 16 bits, but the SIMD implementations
+   generally only support 14 so the values can fit in a signed 16-bit
+   word.  We bring all implementations to the same accuracy in order
+   to ensure that all implementations produce the same results.  */
+#define INTERP_BITS 14
+
+/*
+  Filter types
+*/
+enum lfr_ftype_t {
+	LFR_FTYPE_S16,
+	LFR_FTYPE_F32
+};
+
+/*
+  Array of FIR filters.
+
+  Let N=2^log2nfilt, M=nsamp.
+
+  This is an array of N+1 filters, 0..N, with each filter M samples
+  long.  Filter number I will be centered on the sample with offset:
+
+  floor(M/2) + I/N
+
+  This means that filter N will be a copy of filter 0, except shifted
+  to the right by one sample.  This extra filter makes the
+  interpolation code simpler.
+*/
+
+
+/*
+  Information queries that a filter responds to.  Each query gives an
+  integer or floating point result, automatically cast to the type
+  requested.
+*/
+enum LFR_INFO_TYPE {
+    /*
+		The size of the filter, also known as the filter's order.  This
+		does not include oversampling.  The resampler will read this
+		many samples, starting with the current position (rounded down),
+		whenever it calculates a sample.
+
+		In other words, this is the amount of overlap needed when
+		resampling consecutive buffers.
+	*/
+	LFR_INFO_SIZE,
+
+	/*
+		Filter delay.  Note that lfr_filter_delay returns a fixed point
+		number and is usually preferable.
+	*/
+	LFR_INFO_DELAY,
+
+	/*
+		The is the number of bytes used by filter coefficients.
+	*/
+	LFR_INFO_MEMSIZE,
+
+	/*
+		The normalized pass frequency the filter was designed with.
+	*/
+	LFR_INFO_FPASS,
+
+	/*
+		The normalized stop frequency the filter was designed with.
+	*/
+	LFR_INFO_FSTOP,
+
+	/*
+		The stopband attenuation, in dB, that the filter was designed
+		with.
+	*/
+	LFR_INFO_ATTEN,
+
+	LFR_INFO_COUNT
+};
+
+/*
+ * A filter for resampling audio.
+ */
+struct lfr_filter {
+	/*
+		FIR filter coefficient data type.
+	*/
+	lfr_ftype_t type;
+
+	/*
+		FIR filter data.
+	*/
+	std::vector<char> data;
+
+	/*
+		Length of each filter, in samples.  This is chosen to make each
+		filter size a multiple of the SIMD register size.  Most SIMD
+		implementations today use 16 byte registers, so this will
+		usually be a multiple of 8 or 4, depending on the coefficient
+		type.
+	*/
+	int nsamp;
+
+	/*
+		Base 2 logarithm of the number of different filters.  There is
+		always an additional filter at the end of the filter array to
+		simplify interpolation, this extra filter is a copy of the first
+		filter shifted right by one sample.
+	*/
+	int log2nfilt;
+
+	/*
+		Filter delay.  Filters are causal, so this should be
+		non-negative.
+	*/
+	lfr_fixed_t delay;
+
+	/*
+		Design parameters of the filter.
+	*/
+	double f_pass, f_stop, atten;
+
+	lfr_filter(lfr_param& param);
+	int geti(LFR_INFO_TYPE iname, bool convert = true) const;
+	double getf(LFR_INFO_TYPE iname, bool convert = true) const;
+
+
+/*
+	Resample an audio buffer.  Note that this function may need to
+	create intermediate buffers if there is no function which can
+	directly operate on the input and output formats.  No intermediate
+	buffers will be necessary if the following conditions are met:
+
+	- Input and output formats are identical.
+
+	- Sample format is either S16_NATIVE or F32_NATIVE.
+
+	- The number of channels is either 1 (mono) or 2 (stereo).
+
+	pos: Current position relative to the start of the input buffer,
+	expressed as a 32.32 fixed point number.  On return, this will
+	contain the updated position.  Positions outside the input buffer
+	are acceptable, it will be treated as if the input buffer were
+	padded with an unlimited number of zeroes on either side.
+
+	inv_ratio: Inverse of the resampling ratio, expressed as a 32.32
+	fixed point number.  This number is equal to the input sample rate
+	divided by the output sample rate.
+
+	dither: State of the PRNG used for dithering.
+
+	nchan: Number of interleaved channels.
+
+	out, in: Input and output buffers.  The buffers are not permitted to
+	alias each other.
+
+	outlen, inlen: Length of buffers, in frames.  Note that the length
+	type is 'int' instead of 'size_t'; this matches the precision of
+	buffer positions.
+
+	outfmt, infmt: Format of input and output buffers.
+
+	filter: A suitable low-pass filter for resampling at the given
+	ratio.
+*/
+	void resample(
+		lfr_fixed_t *pos, lfr_fixed_t inv_ratio,
+		unsigned *dither, int nchan,
+		void *out, lfr_fmt_t outfmt, int outlen,
+		const void *in, lfr_fmt_t infmt, int inlen);
+private:
+	/*
+		Create a Kaiser-windowed sinc filter.
+	*/
+	void setup_window(double cutoff, double beta);
+};
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