Primitive filtering for audioscope x scaling.

Adding an inadequate window FIR filter for the x scaling on audioscope,
this also deals with falling off the end of the buffer.
Gadgetoid#14

Author: Kevin J Walters

effect/effect_audioscope_tuner.cpp

@@ -5,6 +5,7 @@
  */
 
 
+#include <array>
 #include <cmath>
 #include <cstdint>
 #include <string>
@@ -17,6 +18,65 @@
 #include "uad/display.hpp"
 
 
+static const std::array win5_32{3, 7, 12, 7, 3};  // TODO ponder symmetry
+static constexpr uint32_t win5_32_shift = 5;
+
+static inline int16_t window_win5_32(std::vector <int16_t> data1, int32_t idx) {
+    int32_t value = 0;
+
+    size_t half_win_len = std::size(win5_32) / 2;
+    int32_t d_idx = idx - half_win_len;
+    size_t data1_len = data1.size();
+    if (d_idx >= 0 && d_idx < data1_len - std::size(win5_32)) {
+        // More efficient version for most cases
+        for (const auto weight : win5_32) {
+            value += weight * int32_t(data1[d_idx++]);
+        }
+        value = value >> win5_32_shift;
+    } else {
+        int32_t weights = 0;
+        for (const auto weight : win5_32) {
+            if (d_idx >= 0 && d_idx < data1_len) {
+                value += weight * data1[d_idx];
+                weights += weight;
+            };
+            ++d_idx;
+        }
+        if (weights != 0) {
+            value /= weights;
+        }
+    }
+
+    return int16_t(value);
+}
+
+static inline int16_t window_win3_4(std::vector <int16_t> data1, int32_t idx) {
+
+    if (idx >= 1 && idx < data1.size() - 1) {
+        int32_t value = 0;
+        // These are 1/4 2/4 1/4 weights
+        value = data1[idx - 1 ] + (data1[idx] << 1) + data1[idx + 1];
+        value = value >> 2;
+        return int16_t(value);
+    } else {
+        return data1[idx];
+    }
+}
+
+static inline int16_t window(std::vector <int16_t> data1, int32_t window, int32_t idx) {
+    int32_t value = 0; 
+
+    if (window == 5) {
+        value = window_win5_32(data1, idx);
+    } else if (window == 3) {
+        value = window_win3_4(data1, idx);
+    } else {
+        value = data1[idx];        
+    }
+
+    return int16_t(value);
+}
+
 void EffectAudioscopeTuner::updateDisplay(void) {
     float frequency = tuner.latest_frequency;
     auto signal = tuner.getSamples();
@@ -35,11 +95,11 @@ void EffectAudioscopeTuner::updateDisplay(void) {
         }
     }
 
-    // TODO ensure tuner.sample_array access does not go beyond buffer
     // TODO add multiple fading passes
     for (uint8_t x=0; x < Display::WIDTH; x++) {
-        int32_t y = Y_MID_POS - int32_t(roundf(signal[waveform_idx] * y_scale));
-        waveform_idx += 10; // TODO proper (fast) reasmpling
+        int32_t value = window(signal, 5, waveform_idx);  // returns 0 beyond end of buffer
+        int32_t y = Y_MID_POS - int32_t(roundf(value * y_scale));
+        waveform_idx += 10;
         if (y >=0 && y < Display::HEIGHT) {
             display.set_pixel(x, y,
                               0u, 128u, 0u);