Adding AudioFilterIIR_F32.h (not well tested!)

AudioFilterIIR_F32.h

@@ -0,0 +1,100 @@
+/*
+ * AudioFilterFIR_F32
+ * 
+ * Created: Chip Audette (OpenAudio) Feb 2017
+ *    - Building from AudioFilterFIR from Teensy Audio Library (AudioFilterFIR credited to Pete (El Supremo))
+ * 
+ */
+
+#ifndef _filter_iir_f32
+#define _filter_iir_f32
+
+#include "Arduino.h"
+#include "AudioStream_F32.h"
+#include "arm_math.h"
+
+// Indicates that the code should just pass through the audio
+// without any filtering (as opposed to doing nothing at all)
+#define IIR_F32_PASSTHRU ((const float32_t *) 1)
+
+#define IIR_MAX_STAGES 1  //meaningless right now
+
+class AudioFilterIIR_F32 : public AudioStream_F32
+{
+  public:
+    AudioFilterIIR_F32(void): AudioStream_F32(1,inputQueueArray), coeff_p(FIR_F32_PASSTHRU) {
+    }
+    void begin(const float32_t *cp, int n_stages) {
+      coeff_p = cp;
+      // Initialize FIR instance (ARM DSP Math Library)
+      if (coeff_p && (coeff_p != IIR_F32_PASSTHRU) && n_stages <= IIR_MAX_STAGES) {
+        //https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html
+        arm_biquad_cascade_df1_init_f32(&iir_inst, n_stages, (float32_t *)coeff_p,  &StateF32[0]);
+      }
+    }
+    void end(void) {
+      coeff_p = NULL;
+    }
+    
+    void setBlockDC(void) {
+      //https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
+      //Use matlab to compute the coeff for HP at 40Hz: [b,a]=butter(2,40/(44100/2),'high'); %assumes fs_Hz = 44100
+      float32_t b[] = {8.173653471988667e-01,  -1.634730694397733e+00,   8.173653471988667e-01};  //from Matlab
+      float32_t a[] = { 1.000000000000000e+00,   -1.601092394183619e+00,  6.683689946118476e-01};  //from Matlab
+      setFilterCoeff_Matlab(b, a);
+    }
+    
+    void setFilterCoeff_Matlab(float32_t b[], float32_t a[]) { //one stage of N=2 IIR
+      //https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html#ga8e73b69a788e681a61bccc8959d823c5
+      //Use matlab to compute the coeff, such as: [b,a]=butter(2,20/(44100/2),'high'); %assumes fs_Hz = 44100
+      hp_coeff[0] = b[0];   hp_coeff[1] = b[1];  hp_coeff[2] = b[2]; //here are the matlab "b" coefficients
+      hp_coeff[3] = -a[1];  hp_coeff[4] = -a[2];  //the DSP needs the "a" terms to have opposite sign vs Matlab 
+      uint8_t n_stages = 1;
+      arm_biquad_cascade_df1_init_f32(&iir_inst, n_stages, hp_coeff,  &StateF32[0]);     
+    }
+    
+    virtual void update(void);
+   
+  private:
+    audio_block_f32_t *inputQueueArray[1];
+    float32_t hp_coeff[5 * 1] = {1.0, 0.0, 0.0, 0.0, 0.0}; //no filtering. actual filter coeff set later
+  
+    // pointer to current coefficients or NULL or FIR_PASSTHRU
+    const float32_t *coeff_p;
+  
+    // ARM DSP Math library filter instance
+    arm_biquad_casd_df1_inst_f32 iir_inst;
+    float32_t StateF32[4*IIR_MAX_STAGES];
+};
+
+
+void AudioFilterIIR_F32::update(void)
+{
+  audio_block_f32_t *block, *b_new;
+
+  block = AudioStream_F32::receiveWritable_f32();
+  if (!block) return;
+
+  // If there's no coefficient table, give up.  
+  if (coeff_p == NULL) {
+    AudioStream_F32::release(block);
+    return;
+  }
+
+  // do passthru
+  if (coeff_p == IIR_F32_PASSTHRU) {
+    // Just passthrough
+    AudioStream_F32::transmit(block);
+    AudioStream_F32::release(block);
+    return;
+  }
+
+  // do IIR
+  arm_biquad_cascade_df1_f32(&iir_inst, block->data, block->data, block->length);
+  AudioStream_F32::transmit(block); // send the IIR output
+  AudioStream_F32::release(block);
+}
+
+#endif
+
+