Fix indentation and move filtercoefficient to #define constant instead of int32_t variable

input_adc.cpp

 #include "utility/pdb.h"
 #include "utility/dspinst.h"
 
+#define COEF_HPF_DCBLOCK    1048300  // DC Removal filter coefficient in S12.19
+
 DMAMEM static uint16_t analog_rx_buffer[AUDIO_BLOCK_SAMPLES];
 audio_block_t * AudioInputAnalog::block_left = NULL;
 uint16_t AudioInputAnalog::block_offset = 0;
 int32_t AudioInputAnalog::hpf_y1 = 0;
 int32_t AudioInputAnalog::hpf_x1 = 0;
-int32_t AudioInputAnalog::a = 1048300;
 
 bool AudioInputAnalog::update_responsibility = false;
 DMAChannel AudioInputAnalog::dma(false);
 void AudioInputAnalog::init(uint8_t pin)
 {
     int32_t tmp;
-    a = 1048300;  // DC Removal filter coefficient in S12.19
 
 	// Configure the ADC and run at least one software-triggered
 	// conversion.  This completes the self calibration stuff and
         int32_t acc = tmp;
         acc += hpf_y1;
         acc -= hpf_x1;
-        hpf_y1 = FRACMUL_SHL(acc, a, 11);
+        hpf_y1 = FRACMUL_SHL(acc, COEF_HPF_DCBLOCK, 11);
         hpf_x1 = tmp;
 		s = signed_saturate_rshift(hpf_y1, 16, 4);
 		*p++ = s;

input_adc.h

         static uint16_t block_offset;
         static int32_t hpf_y1;
         static int32_t hpf_x1;
-        static int32_t a;
 
         static bool update_responsibility;
-    	static DMAChannel dma;
-    	static void isr(void);
-    	static void init(uint8_t pin);
+        static DMAChannel dma;
+        static void isr(void);
+        static void init(uint8_t pin);
 
 };
 

input_adcs.cpp

 
 #if defined(__MK20DX256__) || defined(__MK64FX512__) || defined(__MK66FX1M0__)
 
+#define COEF_HPF_DCBLOCK    1048300  // DC Removal filter coefficient in S12.19
+
 DMAMEM static uint16_t left_buffer[AUDIO_BLOCK_SAMPLES];
 DMAMEM static uint16_t right_buffer[AUDIO_BLOCK_SAMPLES];
 audio_block_t * AudioInputAnalogStereo::block_left = NULL;
 uint16_t AudioInputAnalogStereo::offset_right = 0;
 int32_t AudioInputAnalogStereo::hpf_y1[2] = { 0, 0 };
 int32_t AudioInputAnalogStereo::hpf_x1[2] = { 0, 0 };
-int32_t AudioInputAnalogStereo::a = 1048300;
 bool AudioInputAnalogStereo::update_responsibility = false;
 DMAChannel AudioInputAnalogStereo::dma0(false);
 DMAChannel AudioInputAnalogStereo::dma1(false);
         int32_t acc = tmp;
         acc += hpf_y1[0];
         acc -= hpf_x1[0];
-        hpf_y1[0] = FRACMUL_SHL(acc, a, 11);
+        hpf_y1[0] = FRACMUL_SHL(acc, COEF_HPF_DCBLOCK, 11);
         hpf_x1[0] = tmp;
         s = signed_saturate_rshift(hpf_y1[0], 16, 4);
         *p++ = s;
         int32_t acc = tmp;
         acc += hpf_y1[1];
         acc -= hpf_x1[1];
-        hpf_y1[1]= FRACMUL_SHL(acc, a, 11);
+        hpf_y1[1]= FRACMUL_SHL(acc, COEF_HPF_DCBLOCK, 11);
         hpf_x1[1] = tmp;
         s = signed_saturate_rshift(hpf_y1[1], 16, 4);
         *p++ = s;

input_adcs.h

 {
 public:
         AudioInputAnalogStereo() : AudioStream(0, NULL) {
-		init(A2, A3);
-	}
+        init(A2, A3);
+    }
         AudioInputAnalogStereo(uint8_t pin0, uint8_t pin1) : AudioStream(0, NULL) {
-		init(pin0, pin1);
-	}
+        init(pin0, pin1);
+    }
         virtual void update(void);
 private:
         static audio_block_t *block_left;
         static uint16_t offset_right;
         static int32_t hpf_y1[2];
         static int32_t hpf_x1[2];
-        static int32_t a;
 
         static bool update_responsibility;
-    	static DMAChannel dma0;
-    	static DMAChannel dma1;
-    	static void isr0(void);
-    	static void isr1(void);
-    	static void init(uint8_t pin0, uint8_t pin1);
+        static DMAChannel dma0;
+        static DMAChannel dma1;
+        static void isr0(void);
+        static void isr1(void);
+        static void init(uint8_t pin0, uint8_t pin1);
 };
 
 #endif