Further precision increase eliminates DC offset

input_adc.cpp

 #include "utility/pdb.h"
 #include "utility/dspinst.h"
 
-#define COEF_HPF_DCBLOCK    (1048300<<4)  // DC Removal filter coefficient in S7.24
+#define COEF_HPF_DCBLOCK    (1048300<<10)  // DC Removal filter coefficient in S1.30
 
 DMAMEM static uint16_t analog_rx_buffer[AUDIO_BLOCK_SAMPLES];
 audio_block_t * AudioInputAnalog::block_left = NULL;
     // Probably not useful to spin cycles here stabilizing
     // since DC blocking is similar to te external analog filters
     tmp = (uint16_t) analogRead(pin);
-    tmp = ( ((int32_t) tmp) << 8);
+    tmp = ( ((int32_t) tmp) << 14);
     hpf_x1 = tmp;   // With constant DC level x1 would be x0
     hpf_y1 = 0;     // Output will settle here when stable
 
     //   y = a*(x[n] - x[n-1] + y[n-1])
     // The coefficient "a" is as follows:
     //  a = UNITY*e^(-2*pi*fc/fs)
-    //  UNITY = 2^20
-    //  fc = 2
-    //  fs = 44100
+    //  fc = 2 @ fs = 44100
     //
 	p = out_left->data;
 	end = p + AUDIO_BLOCK_SAMPLES;
 	do {
 		tmp = (uint16_t)(*p);
-        tmp = ( ((int32_t) tmp) << 8);
+        tmp = ( ((int32_t) tmp) << 14);
         int32_t acc = hpf_y1 - hpf_x1;
         acc += tmp;
-        hpf_y1 = FRACMUL_SHL(acc, COEF_HPF_DCBLOCK, 7);
+        hpf_y1 = FRACMUL_SHL(acc, COEF_HPF_DCBLOCK, 1);
         hpf_x1 = tmp;
-		s = signed_saturate_rshift(hpf_y1, 16, 8);
+		s = signed_saturate_rshift(hpf_y1, 16, 14);
 		*p++ = s;
 	} while (p < end);
 

input_adcs.cpp

 
 #if defined(__MK20DX256__) || defined(__MK64FX512__) || defined(__MK66FX1M0__)
 
-#define COEF_HPF_DCBLOCK    (1048300<<4)  // DC Removal filter coefficient in S7.24
+#define COEF_HPF_DCBLOCK    (1048300<<10)  // DC Removal filter coefficient in S1.30
 
 DMAMEM static uint16_t left_buffer[AUDIO_BLOCK_SAMPLES];
 DMAMEM static uint16_t right_buffer[AUDIO_BLOCK_SAMPLES];
     // Probably not useful to spin cycles here stabilizing
     // since DC blocking is similar to te external analog filters
     tmp = (uint16_t) analogRead(pin0);
-    tmp = ( ((int32_t) tmp) << 8);
+    tmp = ( ((int32_t) tmp) << 14);
     hpf_x1[0] = tmp;   // With constant DC level x1 would be x0
     hpf_y1[0] = 0;     // Output will settle here when stable
 
     tmp = (uint16_t) analogReadADC1(pin1);
-    tmp = ( ((int32_t) tmp) << 8);
+    tmp = ( ((int32_t) tmp) << 14);
     hpf_x1[1] = tmp;   // With constant DC level x1 would be x0
     hpf_y1[1] = 0;     // Output will settle here when stable
 
     //   y = a*(x[n] - x[n-1] + y[n-1])
     // The coefficient "a" is as follows:
     //  a = UNITY*e^(-2*pi*fc/fs)
-    //  UNITY = 2^20
-    //  fc = 2
-    //  fs = 44100
+    //  fc = 2 @ fs = 44100
     //
 
     // DC removal, LEFT
     end = p + AUDIO_BLOCK_SAMPLES;
     do {
         tmp = (uint16_t)(*p);
-        tmp = ( ((int32_t) tmp) << 8);
+        tmp = ( ((int32_t) tmp) << 14);
         int32_t acc = hpf_y1[0] - hpf_x1[0];
         acc += tmp;
-        hpf_y1[0] = FRACMUL_SHL(acc, COEF_HPF_DCBLOCK, 7);
+        hpf_y1[0] = FRACMUL_SHL(acc, COEF_HPF_DCBLOCK, 1);
         hpf_x1[0] = tmp;
-        s = signed_saturate_rshift(hpf_y1[0], 16, 8);
+        s = signed_saturate_rshift(hpf_y1[0], 16, 14);
         *p++ = s;
     } while (p < end);
 
     end = p + AUDIO_BLOCK_SAMPLES;
     do {
         tmp = (uint16_t)(*p);
-        tmp = ( ((int32_t) tmp) << 8);
+        tmp = ( ((int32_t) tmp) << 14);
         int32_t acc = hpf_y1[1] - hpf_x1[1];
         acc += tmp;
-        hpf_y1[1]= FRACMUL_SHL(acc, COEF_HPF_DCBLOCK, 7);
+        hpf_y1[1]= FRACMUL_SHL(acc, COEF_HPF_DCBLOCK, 1);
         hpf_x1[1] = tmp;
-        s = signed_saturate_rshift(hpf_y1[1], 16, 8);
+        s = signed_saturate_rshift(hpf_y1[1], 16, 14);
         *p++ = s;
     } while (p < end);