fft: Capture microphone's feed and detect fundamental frequency

Signed-off-by: Raphael Gault <[email protected]>

Author: raphaelgault

inc/MicroBitAudioProcessor.h

@@ -30,7 +30,8 @@ DEALINGS IN THE SOFTWARE.
 #ifndef MICROBIT_AUDIO_PROCESSOR_H
 #define MICROBIT_AUDIO_PROCESSOR_H
 
-#define AUDIO_SAMPLES_NUMBER    256
+#define MIC_SAMPLE_RATE         (11 * 1024)
+#define AUDIO_SAMPLES_NUMBER    1024 
 
 class MicroBitAudioProcessor : public DataSink
 {
@@ -40,13 +41,18 @@ class MicroBitAudioProcessor : public DataSink
     arm_rfft_fast_instance_f32 fft_instance;
     float *buf; 
     float *output; 
+    float *mag; 
     uint16_t position;
+    bool recording;
+    float rec[AUDIO_SAMPLES_NUMBER * 2];
 
     public:
     MicroBitAudioProcessor(DataSource& source); 
     ~MicroBitAudioProcessor(); 
     virtual int pullRequest();
     int setDivisor(int d);
+    void startRecording();
+    void stopRecording(MicroBit& uBit);
 };
 
 #endif

source/MicroBitAudioProcessor.cpp

@@ -30,15 +30,17 @@ MicroBitAudioProcessor::MicroBitAudioProcessor(DataSource& source) : audiostream
     audiostream.connect(*this);
     zeroOffset = 0;
     divisor = 1;
-    //arm_rfft_fast_init_f32(&fft_instance, AUDIO_SAMPLES_NUMBER);
+    arm_rfft_fast_init_f32(&fft_instance, AUDIO_SAMPLES_NUMBER);
 
     /* Double Buffering: We allocate twice the number of samples*/
     buf = (float *)malloc(sizeof(float) * AUDIO_SAMPLES_NUMBER * 2);
     output = (float *)malloc(sizeof(float) * AUDIO_SAMPLES_NUMBER);
+    mag = (float *)malloc(sizeof(float) * AUDIO_SAMPLES_NUMBER / 2);
 
     position = 0;
+    recording = false;
 
-    if (buf == NULL || output == NULL) {
+    if (buf == NULL || output == NULL || mag == NULL) {
         DMESG("DEVICE_NO_RESOURCES");
         target_panic(DEVICE_OOM);
     }
@@ -48,6 +50,7 @@ MicroBitAudioProcessor::~MicroBitAudioProcessor()
 {
     free(buf);
     free(output);
+    free(mag);
 }
 
 int MicroBitAudioProcessor::pullRequest()
@@ -56,40 +59,59 @@ int MicroBitAudioProcessor::pullRequest()
     int minimum = 0;
     int maximum = 0;
     int s;
-    int8_t result;
+    int result;
 
-    auto mic_samples = audiostream.pull().getBytes();
+    auto mic_samples = audiostream.pull();
+
+    if (!recording)
+        return DEVICE_OK;
 
     int16_t *data = (int16_t *) &mic_samples[0];
-    int samples = AUDIO_SAMPLES_NUMBER / 2;
+    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 = (int8_t)s;
+        //s = s - zeroOffset;
+        //s = s / divisor;
+        result = s;
 
         if (s < minimum)
             minimum = s;
 
         if (s > maximum)
             maximum = s;
 
+        //if (recording)
+        //    rec[position] = (float)result;
+
         data++;
         buf[position++] = (float)result;
 
+
         if (!(position % AUDIO_SAMPLES_NUMBER))
         {
-        /* We have AUDIO_SAMPLES_NUMBER samples, we can run the FFT on them */
-            uint8_t offset = position <= AUDIO_SAMPLES_NUMBER ? 0 : AUDIO_SAMPLES_NUMBER;
+            float maxValue = 0;
+            uint32_t index = 0;
+
+            /* We have AUDIO_SAMPLES_NUMBER samples, we can run the FFT on them */
+            uint16_t offset = position <= AUDIO_SAMPLES_NUMBER ? 0 : AUDIO_SAMPLES_NUMBER;
             if (offset != 0)
                 position = 0;
 
-            //arm_rfft_fast_f32(&fft_instance, buf + offset, output, 0);
-            DMESG("o[0]: %d", output[0]);
+            DMESG("Run FFT, %d", offset);
+            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);
+
+            uint32_t freq = ((uint32_t)MIC_SAMPLE_RATE / AUDIO_SAMPLES_NUMBER) * (index + 1);
+            DMESG("Freq: %d (max: %d.%d, Index: %d)",
+                  freq,
+                  (int)maxValue,
+                  ((int)(maxValue * 100) % 100),
+                  index);
         }
     }
 
@@ -103,3 +125,16 @@ int MicroBitAudioProcessor::setDivisor(int d)
     divisor = d;
     return DEVICE_OK;
 }
+
+
+void MicroBitAudioProcessor::startRecording()
+{
+    this->recording = true;
+    DMESG("START RECORDING");
+}
+
+void MicroBitAudioProcessor::stopRecording(MicroBit& uBit)
+{
+    this->recording = false;
+    DMESG("STOP RECORDING");
+}