Updating fft branch in line with origin. Added ability to query MicroBitAudioProcessor and sample fft C++ Program

Author: JoshuaAHill

inc/MicroBitAudioProcessor.h

@@ -45,11 +45,13 @@ class MicroBitAudioProcessor : public DataSink
     uint16_t position;
     bool recording;
     float rec[AUDIO_SAMPLES_NUMBER * 2];
+    int lastFreq;
 
     public:
     MicroBitAudioProcessor(DataSource& source); 
     ~MicroBitAudioProcessor(); 
     virtual int pullRequest();
+    int getFrequency();
     int setDivisor(int d);
     void startRecording();
     void stopRecording(MicroBit& uBit);

samples/fft_example.cpp

@@ -0,0 +1,76 @@
+#include "MicroBit.h"
+#include "CodalDmesg.h"
+#include "MicroBitAudioProcessor.h"
+#include "StreamNormalizer.h"
+#include "Tests.h"
+
+static NRF52ADCChannel *mic = NULL;
+static StreamNormalizer *processor = NULL;
+static MicroBitAudioProcessor *fft = NULL;
+
+MicroBit uBit;
+
+/**
+  * fft_test function - creates an example MicroBitAudioProcessor and then queries it for results.
+  * Currently configured to use 1024 samples with 8bit signed data.
+  */
+void fft_test(){
+    uBit.display.print("L");
+     if (mic == NULL){
+        mic = uBit.adc.getChannel(uBit.io.microphone);
+        mic->setGain(7,0);
+    }
+
+    if (processor == NULL)
+        processor = new StreamNormalizer(mic->output, 1.0f, true, DATASTREAM_FORMAT_8BIT_SIGNED, 10);
+
+    if (fft == NULL)
+        fft = new MicroBitAudioProcessor(processor->output);
+
+    uBit.io.runmic.setDigitalValue(1);
+    uBit.io.runmic.setHighDrive(true);
+
+
+    //Start fft running
+    fft->startRecording();
+
+
+    while(1){
+        //TODO - de-noise : if last X samples are same - display ect.
+        //The output values depend on the input type (DATASTREAM_FORMAT_8BIT_SIGNED) and the size
+        //of the FFT - which is changed using the 'AUDIO_SAMPLES_NUMBER' in MicroBitAudioProcessor.h
+        //default is 1024
+        uBit.sleep(100);
+        int freq = fft->getFrequency();
+        DMESG("%s %d", "frequency: ", freq);
+        if(freq > 0)
+            uBit.display.print("?");
+        if(freq > 530)
+            uBit.display.print("C");
+        if(freq > 600)
+            uBit.display.print("D");
+        if(freq > 680)
+            uBit.display.print("E");
+        if(freq > 710)
+            uBit.display.print("F");
+        if(freq > 800)
+            uBit.display.print("G");
+        if(freq > 900)
+            uBit.display.print("A");
+        if(freq > 1010)
+            uBit.display.print("B");
+        if(freq > 1050)
+            uBit.display.print("?");
+    }
+}
+
+
+
+int 
+main()
+{
+
+    uBit.init();
+    fft_test();
+    
+}

source/MicroBitAudioProcessor.cpp

@@ -55,9 +55,7 @@ MicroBitAudioProcessor::~MicroBitAudioProcessor()
 
 int MicroBitAudioProcessor::pullRequest()
 {
-    int z = 0;
-    int minimum = 0;
-    int maximum = 0;
+
     int s;
     int result;
 
@@ -66,27 +64,17 @@ int MicroBitAudioProcessor::pullRequest()
     if (!recording)
         return DEVICE_OK;
 
+    //using 8 bits produces more accurate to input results (not 2x like using 16) but issue with 
+    //F and G both producing 363hz -> investigate futher with crossing 8 bit + differnet sample numbers
+    //int8_t *data = (int8_t *) &mic_samples[0];
     int16_t *data = (int16_t *) &mic_samples[0];
     int samples = mic_samples.length() / 2;
 
     for (int i=0; i < samples; i++)
     {
-        z += *data;
-
-        s = (int) *data;
-        //s = s - zeroOffset;
-        //s = s / divisor;
-        result = s;
-
-        if (s < minimum)
-            minimum = s;
-
-        if (s > maximum)
-            maximum = s;
-
-        //if (recording)
-        //    rec[position] = (float)result;
-
+        
+        result = (int) *data;
+        
         data++;
         buf[position++] = (float)result;
 
@@ -102,9 +90,14 @@ int MicroBitAudioProcessor::pullRequest()
                 position = 0;
 
             DMESG("Run FFT, %d", offset);
+            //auto a = system_timer_current_time();
             arm_rfft_fast_f32(&fft_instance, buf + offset, output, 0);
             arm_cmplx_mag_f32(output, mag, AUDIO_SAMPLES_NUMBER / 2);
             arm_max_f32(mag + 1, AUDIO_SAMPLES_NUMBER / 2 - 1, &maxValue, &index);
+            //auto b = system_timer_current_time();
+
+            //DMESG("Before FFT: %d", (int)a);
+            //DMESG("After FFT: %d (%d)", (int)b, (int)(b - a));
 
             uint32_t freq = ((uint32_t)MIC_SAMPLE_RATE / AUDIO_SAMPLES_NUMBER) * (index + 1);
             DMESG("Freq: %d (max: %d.%d, Index: %d)",
@@ -115,11 +108,13 @@ int MicroBitAudioProcessor::pullRequest()
         }
     }
 
-    z = z / samples;
-    zeroOffset = z;
     return DEVICE_OK;
 }
 
+int MicroBitAudioProcessor::getFrequency(){
+    return lastFreq;
+}
+
 int MicroBitAudioProcessor::setDivisor(int d)
 {
     divisor = d;