type cleanup

noisemeter-device/spl-meter.cpp

@@ -19,9 +19,9 @@
 #define MIC_OFFSET_DB 0                 // Default offset (sine-wave RMS vs. dBFS). Modify this value for linear calibration
 
 // Customize these values from microphone datasheet
-#define MIC_SENSITIVITY -26    // dBFS value expected at MIC_REF_DB (Sensitivity value from datasheet)
-#define MIC_REF_DB 94.0        // Value at which point sensitivity is specified in datasheet (dB)
-#define MIC_BITS 24            // valid number of bits in I2S data
+#define MIC_SENSITIVITY -26.f // dBFS value expected at MIC_REF_DB (Sensitivity value from datasheet)
+#define MIC_REF_DB       94.f // Value at which point sensitivity is specified in datasheet (dB)
+#define MIC_BITS         24   // valid number of bits in I2S data
 #define MIC_CONVERT(s) (s >> (SAMPLE_BITS - MIC_BITS))
 
 //
@@ -32,13 +32,10 @@
 #define DMA_BANKS 32
 
 // Calculate reference amplitude value at compile time
-constexpr double MIC_REF_AMPL = pow(10, double(MIC_SENSITIVITY) / 20) * ((1 << (MIC_BITS - 1)) - 1);
+constexpr auto MIC_REF_AMPL = std::pow(10.f, MIC_SENSITIVITY / 20.f) * ((1 << (MIC_BITS - 1)) - 1);
 
 void SPLMeter::initMicrophone() noexcept
 {
-  // Setup I2S to sample mono channel for SAMPLE_RATE * SAMPLE_BITS
-  // NOTE: Recent update to Arduino_esp32 (1.0.2 -> 1.0.3)
-  //       seems to have swapped ONLY_LEFT and ONLY_RIGHT channels
   const i2s_config_t i2s_config = {
     mode: i2s_mode_t(I2S_MODE_MASTER | I2S_MODE_RX),
     sample_rate: SAMPLE_RATE,
@@ -78,20 +75,20 @@ std::optional<float> SPLMeter::readMicrophoneData() noexcept
   // Convert (including shifting) integer microphone values to floats,
   // using the same buffer (assumed sample size is same as size of float),
   // to save a bit of memory
-  auto int_samples = reinterpret_cast<SAMPLE_T*>(samples);
-
-  for (int i = 0; i < SAMPLES_SHORT; i++) samples[i] = MIC_CONVERT(int_samples[i]);
+  for (auto& s : samples)
+      s.f = MIC_CONVERT(s.i);
 
   // Apply equalization and calculate Z-weighted sum of squares,
   // writes filtered samples back to the same buffer.
-  const auto sum_sqr_SPL = MIC_EQUALIZER.filter(samples, samples, SAMPLES_SHORT);
+  auto fptr = &samples[0].f;
+  const auto sum_sqr_SPL = MIC_EQUALIZER.filter(fptr, fptr, samples.size());
 
   // Apply weighting and calucate weigthed sum of squares
-  const auto sum_sqr_weighted = WEIGHTING.filter(samples, samples, SAMPLES_SHORT);
+  const auto sum_sqr_weighted = WEIGHTING.filter(fptr, fptr, samples.size());
 
   // Calculate dB values relative to MIC_REF_AMPL and adjust for microphone reference
-  double short_RMS = sqrt(double(sum_sqr_SPL) / SAMPLES_SHORT);
-  double short_SPL_dB = MIC_OFFSET_DB + MIC_REF_DB + 20 * log10(short_RMS / MIC_REF_AMPL);
+  const auto short_RMS = std::sqrt(sum_sqr_SPL / samples.size());
+  const auto short_SPL_dB = MIC_OFFSET_DB + MIC_REF_DB + 20 * std::log10(short_RMS / MIC_REF_AMPL);
 
   // In case of acoustic overload or below noise floor measurement, report infinty Leq value
   if (short_SPL_dB > MIC_OVERLOAD_DB) {
@@ -102,15 +99,15 @@ std::optional<float> SPLMeter::readMicrophoneData() noexcept
 
   // Accumulate Leq sum
   Leq_sum_sqr += sum_sqr_weighted;
-  Leq_samples += SAMPLES_SHORT;
+  Leq_samples += samples.size();
 
   // When we gather enough samples, calculate new Leq value
   if (Leq_samples >= SAMPLE_RATE * LEQ_PERIOD) {
-    double Leq_RMS = sqrt(Leq_sum_sqr / Leq_samples);
+    const auto Leq_RMS = std::sqrt(Leq_sum_sqr / Leq_samples);
     Leq_sum_sqr = 0;
     Leq_samples = 0;
 
-    return MIC_OFFSET_DB + MIC_REF_DB + 20 * log10(Leq_RMS / MIC_REF_AMPL); // Leq dB
+    return MIC_OFFSET_DB + MIC_REF_DB + 20 * std::log10(Leq_RMS / MIC_REF_AMPL); // Leq dB
   } else {
     return {};
   }
@@ -122,10 +119,7 @@ void SPLMeter::i2sRead()
   //
   // Data is moved from DMA buffers to our 'samples' buffer by the driver ISR
   // and when there is requested ammount of data, task is unblocked
-  //
-  // Note: i2s_read does not care it is writing in float[] buffer, it will write
-  //       integer values to the given address, as received from the hardware peripheral.
   size_t bytes_read;
-  i2s_read(I2S_PORT, samples, sizeof(samples), &bytes_read, portMAX_DELAY);
+  i2s_read(I2S_PORT, samples.data(), samples.size() * sizeof(samples[0]), &bytes_read, portMAX_DELAY);
 }
 

noisemeter-device/spl-meter.h

@@ -1,22 +1,25 @@
 #ifndef SPL_METER_H
 #define SPL_METER_H
 
+#include <array>
 #include <cstdint>
 #include <optional>
 
 #define SAMPLE_RATE 48000  // Hz, fixed to design of IIR filters
 #define SAMPLE_BITS 32     // bits
-#define SAMPLE_T int32_t
 #define SAMPLES_SHORT (SAMPLE_RATE / 8)  // ~125ms
 
-#define MIC_OVERLOAD_DB 120.0  // dB - Acoustic overload point
-#define MIC_NOISE_DB 29        // dB - Noise floor
-
-static_assert(sizeof(float) == sizeof(SAMPLE_T));
-using SampleBuffer alignas(4) = float[SAMPLES_SHORT];
+#define MIC_OVERLOAD_DB 120.f // dB - Acoustic overload point
+#define MIC_NOISE_DB     29.f // dB - Noise floor
 
 class SPLMeter
 {
+    static_assert(sizeof(float) == sizeof(std::int32_t));
+    union sample_t {
+        std::int32_t i;
+        float f;
+    };
+
 public:
     // Prepares I2S Driver and mic config.
     void initMicrophone() noexcept;
@@ -26,11 +29,12 @@ class SPLMeter
 
 private:
     // Buffer for block of samples
-    SampleBuffer samples;
+    alignas(4)
+    std::array<sample_t, SAMPLES_SHORT> samples;
 
     // Sampling accumulators
     unsigned Leq_samples = 0;
-    double Leq_sum_sqr = 0;
+    float Leq_sum_sqr = 0;
 
     void i2sRead();
 };