/**************************************************************************/
/*  audio_filter_sw.h                                                     */
/**************************************************************************/
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#ifndef AUDIO_FILTER_SW_H
#define AUDIO_FILTER_SW_H

#include "core/math/math_funcs.h"

class AudioFilterSW {
public:
	struct Coeffs {
		float a1, a2;
		float b0, b1, b2;

		//bool operator==(const Coeffs &p_rv) { return (FLOATS_EQ(a1,p_rv.a1) && FLOATS_EQ(a2,p_rv.a2) && FLOATS_EQ(b1,p_rv.b1) && FLOATS_EQ(b2,p_rv.b2) && FLOATS_EQ(b0,p_rv.b0) ); }
		Coeffs() { a1 = a2 = b0 = b1 = b2 = 0.0; }
	};

	enum Mode {
		BANDPASS,
		HIGHPASS,
		LOWPASS,
		NOTCH,
		PEAK,
		BANDLIMIT,
		LOWSHELF,
		HIGHSHELF

	};

	class Processor { // simple filter processor

		AudioFilterSW *filter;
		Coeffs coeffs;
		float ha1, ha2, hb1, hb2; //history
		Coeffs incr_coeffs;

	public:
		void set_filter(AudioFilterSW *p_filter, bool p_clear_history = true);
		void process(float *p_samples, int p_amount, int p_stride = 1, bool p_interpolate = false);
		void update_coeffs(int p_interp_buffer_len = 0);
		_ALWAYS_INLINE_ void process_one(float &p_sample);
		_ALWAYS_INLINE_ void process_one_interp(float &p_sample);

		Processor();
	};

private:
	float cutoff;
	float resonance;
	float gain;
	float sampling_rate;
	int stages;
	Mode mode;

public:
	float get_response(float p_freq, Coeffs *p_coeffs);

	void set_mode(Mode p_mode);
	void set_cutoff(float p_cutoff);
	void set_resonance(float p_resonance);
	void set_gain(float p_gain);
	void set_sampling_rate(float p_srate);
	void set_stages(int p_stages); //adjust for multiple stages

	void prepare_coefficients(Coeffs *p_coeffs);

	AudioFilterSW();
};

/* inline methods */

void AudioFilterSW::Processor::process_one(float &p_sample) {
	float pre = p_sample;
	p_sample = (p_sample * coeffs.b0 + hb1 * coeffs.b1 + hb2 * coeffs.b2 + ha1 * coeffs.a1 + ha2 * coeffs.a2);
	ha2 = ha1;
	hb2 = hb1;
	hb1 = pre;
	ha1 = p_sample;
}

void AudioFilterSW::Processor::process_one_interp(float &p_sample) {
	float pre = p_sample;
	p_sample = (p_sample * coeffs.b0 + hb1 * coeffs.b1 + hb2 * coeffs.b2 + ha1 * coeffs.a1 + ha2 * coeffs.a2);
	ha2 = ha1;
	hb2 = hb1;
	hb1 = pre;
	ha1 = p_sample;

	coeffs.b0 += incr_coeffs.b0;
	coeffs.b1 += incr_coeffs.b1;
	coeffs.b2 += incr_coeffs.b2;
	coeffs.a1 += incr_coeffs.a1;
	coeffs.a2 += incr_coeffs.a2;
}

#endif // AUDIO_FILTER_SW_H
