Change Ladder filter API to be similar to other filters

filter_ladder.cpp

@@ -15,7 +15,7 @@
 #include <stdint.h>
 #define MOOG_PI 3.14159265358979323846264338327950288
 
-float HNMoog::LPF(float s, int i)
+float AudioFilterLadder::LPF(float s, int i)
 {
 	float ft = s * (1/1.3) + (0.3/1.3) * z0[i] - z1[i];
 	ft = ft * alpha + z1[i];
@@ -24,7 +24,7 @@ float HNMoog::LPF(float s, int i)
 	return ft;
 }
 
-void HNMoog::SetResonance(float res)
+void AudioFilterLadder::resonance(float res)
 {
 	// maps resonance = 0->1 to K = 0 -> 4
 	if (res > 1) {
@@ -35,13 +35,13 @@ void HNMoog::SetResonance(float res)
 	K = 4.0 * res;
 }
 
-void HNMoog::SetCutoff(float c)
+void AudioFilterLadder::frequency(float c)
 {
 	Fbase = c;
 	compute_coeffs(c);
 }
 
-void HNMoog::compute_coeffs(float c)
+void AudioFilterLadder::compute_coeffs(float c)
 {
 	if (c > 0.49 * AUDIO_SAMPLE_RATE_EXACT)
 		c = 0.49 * AUDIO_SAMPLE_RATE_EXACT;
@@ -53,13 +53,13 @@ void HNMoog::compute_coeffs(float c)
 	alpha = 0.9892f * wc - 0.4324f * wc2 + 0.1381f * wc * wc2 - 0.0202f * wc2 * wc2;
 }
 
-inline float fast_tanh(float x)
+static inline float fast_tanh(float x)
 {
 	float x2 = x * x;
 	return x * (27.0 + x2) / (27.0 + 9.0 * x2);
 }
 
-void HNMoog::update(void)
+void AudioFilterLadder::update(void)
 {
 	audio_block_t *blocka, *blockb;
 	float ftot, FCmod;

filter_ladder.h

@@ -15,21 +15,21 @@
 #include "Arduino.h"
 #include "AudioStream.h"
 
-class HNMoog : public AudioStream
+class AudioFilterLadder: public AudioStream
 {
 public:
-	HNMoog() : AudioStream(2,inputQueueArray) {};
-	void SetCutoff(float FC);
-	void SetResonance(float reson);
-	float LPF(float s, int i);
-	void compute_coeffs(float fc);
+	AudioFilterLadder() : AudioStream(2, inputQueueArray) {};
+	void frequency(float FC);
+	void resonance(float reson);
 	virtual void update(void);
 private:
+	float LPF(float s, int i);
+	void compute_coeffs(float fc);
 	float alpha = 1.0;
-	float beta[4] = {0};
-	float z0[4] = {0};
-	float z1[4] = {0};
-	float K;
+	float beta[4] = {0.0, 0.0, 0.0, 0.0};
+	float z0[4] = {0.0, 0.0, 0.0, 0.0};
+	float z1[4] = {0.0, 0.0, 0.0, 0.0};
+	float K = 1.0;
 	float Fbase = 1000;
 	float overdrive = 1.0;
 	audio_block_t *inputQueueArray[2];

keywords.txt

@@ -63,6 +63,7 @@ AudioEffectRectifier	KEYWORD2
 AudioFilterBiquad	KEYWORD2
 AudioFilterFIR	KEYWORD2
 AudioFilterStateVariable	KEYWORD2
+AudioFilterLadder	KEYWORD2
 AudioInputAnalog	KEYWORD2
 AudioInputAnalogStereo	KEYWORD2
 AudioMixer4	KEYWORD2