update

AudioTuner.cpp

     end = p + AUDIO_BLOCK_SAMPLES;
     
     /*
-     * Set the number of cycles to processed per receiving block.
-     *
-     */
-    uint16_t cycles;
-    const uint16_t usage_max = cpu_usage_max;
-    if ( AudioProcessorUsage( ) > usage_max ) {
-#if NUM_SAMPLES >= 8192
-        cycles = tau_global + 2;
-#elif NUM_SAMPLES == 4096
-        cycles = tau_global + 4;
-#elif NUM_SAMPLES == 2048
-        cycles = tau_global + 8;
-#elif NUM_SAMPLES <= 1024
-        cycles = tau_global + 16;
-#endif
+     * Double buffering, one fills while the other is processed
+     * 2x the throughput.
+    */
+    uint16_t *dst;
+    bool next = next_buffer;
+    if ( next ) {
+        //digitalWriteFast(6, HIGH);
+        dst = ( uint16_t * )buffer;
     }
     else {
-#if NUM_SAMPLES >= 8192
-        cycles = tau_global + 8;
-#elif NUM_SAMPLES == 4096
-        cycles = tau_global + 16;
-#elif NUM_SAMPLES == 2048
-        cycles = tau_global + 32;
-#elif NUM_SAMPLES <= 1024
-        cycles = tau_global + 64;
-#endif
+       //digitalWriteFast(6, LOW);
+       dst = ( uint16_t * )buffer + NUM_SAMPLES;
     }
     
+    // gather data/and release block
     uint16_t count = count_global;
-    
-    /*
-     * Double buffering, one fill while the other is processed
-     * 2x the throughput.
-    */
-    uint16_t *dst;
-    if ( next_buffer ) dst = ( uint16_t * )buffer;
-    else dst = ( uint16_t * )buffer + NUM_SAMPLES;
-    
-    // gather data
     do {
         *( dst+count++ ) = *( uint16_t * )p;
         p += SAMPLE_RATE;
      */
     if ( count >= NUM_SAMPLES ) {
         //digitalWriteFast(2, !digitalReadFast(2));
+        __disable_irq();
         next_buffer = !next_buffer;
         process_buffer  = true;
         count_global    = 0;
         yin_idx         = 1;
         running_sum     = 0;
         count           = 0;
+        __enable_irq();
     }
     count_global = count;// update global count
     
+    /*
+     * Set the number of cycles to be processed per receiving block.
+     */
+    uint16_t cycles;
+    const uint16_t usage_max = cpu_usage_max;
+    if ( AudioProcessorUsage( ) > usage_max ) {
+#if NUM_SAMPLES >= 8192
+        cycles = tau_global + 2;
+#elif NUM_SAMPLES == 4096
+        cycles = tau_global + 4;
+#elif NUM_SAMPLES == 2048
+        cycles = tau_global + 8;
+#elif NUM_SAMPLES <= 1024
+        cycles = tau_global + 32;
+#endif
+    }
+    else {
+#if NUM_SAMPLES >= 8192
+        cycles = tau_global + 8;
+#elif NUM_SAMPLES == 4096
+        cycles = tau_global + 16;
+#elif NUM_SAMPLES == 2048
+        cycles = tau_global + 32;
+#elif NUM_SAMPLES <= 1024
+        cycles = tau_global + 64;
+#endif
+    }
+    
     if ( process_buffer ) {
         //digitalWriteFast(0, HIGH);
         uint16_t tau;
-        uint16_t  next;
         next = next_buffer;
         tau = tau_global;
         do {
             int64_t sum  = 0;
             const int16_t *end, *buf;
-            if ( next ) buf = buffer + NUM_SAMPLES;
-            else buf = buffer;
+            if ( next ) {
+                //digitalWriteFast(4, LOW);
+                buf = buffer + NUM_SAMPLES;
+            }
+            else {
+                //digitalWriteFast(4, HIGH);
+                buf = buffer;
+            }
             end = buf + HALF_BUFFER;
+            
+            // TODO: How to make faster?
             do {
                 int16_t current, lag, delta;
                 UNROLL( 8,
         idx2 = ( idx2 >= 5 ) ? 0 : idx2;
         
         float s0, s1, s2;
-        s0 = ( ( float )*( p+idx0 ) / r[idx0] );
-        s1 = ( ( float )*( p+idx1 ) / r[idx1] );
-        s2 = ( ( float )*( p+idx2 ) / r[idx2] );
+        s0 = ( ( float )*( p+idx0 ) / *( r+idx0 ) );
+        s1 = ( ( float )*( p+idx1 ) / *( r+idx1 ) );
+        s2 = ( ( float )*( p+idx2 ) / *( r+idx2 ) );
         
         if ( s1 < yin_threshold && s1 < s2 ) {
             uint16_t period = _tau - 3;
             return 0;
         }
         
-        if ( s1 > 2.4 ) return _tau + 2;
-        else return _tau + 1;
+        //if ( s1 > 2.4 ) return _tau + 2;
+        //else return _tau + 1;
     }
     return _tau + 1;
 }
  *  @param threshold Allowed uncertainty
  *  @param cpu_max   How much cpu usage before throttling
  */
-void AudioTuner::initialize( float threshold, uint8_t cpu_max ) {
+void AudioTuner::initialize( float threshold, float cpu_max ) {
     __disable_irq( );
-    cpu_usage_max = cpu_max;
+    cpu_usage_max = cpu_max*100;
     yin_threshold = threshold;
     process_buffer = false;
     periodicity    = 0.0f;
-    next_buffer    = 1;
+    next_buffer    = true;
     running_sum    = 0;
     count_global   = 0;
     yin_idx        = 1;
     __disable_irq( );
     float d = data;
     __enable_irq( );
-    d = SAMPLE_RATE_EXACT / d;
-    return d;
+    return SAMPLE_RATE_EXACT / d;
 }
 
 /**

AudioTuner.h

      *
      *  @return none
      */
-    AudioTuner( void ) : AudioStream( 1, inputQueueArray ), enabled( false ), new_output(false) {
-        digitalWriteFast(2, LOW);
-    }
+    AudioTuner( void ) : AudioStream( 1, inputQueueArray ), enabled( false ), new_output(false) {}
+    
     /**
      *  initialize variables and start conversion
      *
      *  @param threshold Allowed uncertainty
      *  @param cpu_max   How much cpu usage before throttling
      */
-    void initialize( float threshold, uint8_t cpu_max);
+    void initialize( float threshold, float cpu_max);
     
     /**
      *  sets threshold value
     uint16_t estimate( int64_t *yin, int64_t *rs, uint16_t head, uint16_t tau );
     
     int16_t  buffer[NUM_SAMPLES*2] __attribute__ ( ( aligned ( 4 ) ) );
-    float    periodicity, yin_threshold, data;
+    float    periodicity, yin_threshold, data, cpu_usage_max;
     int64_t  rs_buffer[5], yin_buffer[5];
     uint64_t running_sum;
-    uint16_t tau_global, count_global, tau_cycles, cpu_usage_max;
-    uint8_t  next_buffer, yin_idx;
-    bool     enabled, process_buffer;
+    uint16_t tau_global, count_global, tau_cycles;
+    uint8_t  yin_idx;
+    bool     enabled, process_buffer, next_buffer;
     volatile bool new_output;
     audio_block_t *inputQueueArray[1];
 };

README.md

 <p align="center">
-    <b>Guitar and Bass Tuner Library</b><br>
-    <b>Teensy 3.1/2 v2.1</b><br>
+    <b>Guitar and Bass Tuner Library v2.2</b><br>
+    <b>Teensy 3.1/2</b><br>
 </p>
 
 >Software algorithm ([YIN]) for guitar and bass tuning using a Teensy Audio Library. This audio object's algorithm can be some what memory and processor hungry but will allow you to detect with fairly good accuracy the fundamental frequencies f<sub>o</sub> from electric guitars and basses. 

examples/Sample_Guitar_Tunning_Notes/Sample_Guitar_Tunning_Notes.ino

  Bass strings are (5th string) B0=30.87Hz, (4th string) E1=41.20Hz, A1=55Hz, D2=73.42Hz, G2=98Hz
  
  This example tests the yin algorithm with actual notes from nylon string guitar recorded
- as wav format at 16B @ 44100smpls/sec. Since the decay of the notes will be longer than what
+ as wav format at 16B @ 44100 samples/sec. Since the decay of the notes will be longer than what
  the teensy can store in flash these notes are truncated to ~120,000B or about 1/2 of the whole
  signal.
  */
 void setup() {
     AudioMemory(4);
     /*
-     *  Intialize the yin algorithm's threshold
-     *  and percent of current cpu usage used
-     *  before slowing the algorithm down.
+     *  Initialize the yin algorithm's absolute
+     *  threshold, this is good number.
+     *
+     *  Percent of overall current cpu usage used
+     *  before making the search algorithm less
+     *  aggressive (0.0 - 1.0).
      */
-    tuner.initialize(.15f, 90);
+    tuner.initialize(.15, .99);
     pinMode(LED_BUILTIN, OUTPUT);
     playNoteTimer.begin(playNote, 1000);
 }
 
 void loop() {
-    // read back fundmental frequency
+    // read back fundamental frequency
     if (tuner.available()) {
         float note = tuner.read();
         float prob = tuner.probability();
-        Serial.printf("Note: %3.2f | Probility: %.2f\n", note, prob);
+        Serial.printf("Note: %3.2f | Probability: %.2f\n", note, prob);
     }
 }

examples/Simple_Sine/Handle_Commands.ino

         float t = p.toFloat();
         Serial.print("new frequency: ");
         Serial.println(t);
-        //AudioNoInterrupts();  // disable audio library momentarily
+        AudioNoInterrupts();  // disable audio library momentarily
         sine.frequency(p.toFloat());
-        //AudioInterrupts();    // enable, both tones will start together
+        AudioInterrupts();    // enable, both tones will start together
     }
     else if (p.startsWith("a ")) {
         p.trim();

examples/Simple_Sine/Simple_Sine.ino

 void setup() {
     AudioMemory(4);
     /*
-     *  Intialize the yin algorithm's threshold
-     *  and percent of current cpu usage used
-     *  before slowing the algorithm down.
+     *  Initialize the yin algorithm's absolute
+     *  threshold, this is good number.
+     *
+     *  Percent of overall current cpu usage used 
+     *  before making the search algorithm less
+     *  aggressive (0.0 - 1.0).
      */
-    tuner.initialize(.15f, 90);
+    tuner.initialize(.15, .99);
+    
     sine.frequency(30.87);
     sine.amplitude(1);
 }
 
 void loop() {
-    // read back fundmental frequency
+    // read back fundamental frequency
     if (tuner.available()) {
         float note = tuner.read();
         float prob = tuner.probability();
-        Serial.printf("Note: %3.2f | Probility: %.2f\n", note, prob);
+        Serial.printf("Note: %3.2f | Probability: %.2f\n", note, prob);
     }
     
     if (Serial.available()) {

library.properties

 name=AudioTuner
-version=2.1
+version=2.2
 author=Colin Duffy
 maintainer=Colin Duffy
 sentence=Yin algorithm

revision.md

+><b>Updated (10/12/15 v2.2)</b><br>
+* Fixed yin cpu usage throttling code in update function.<br>
+* Function initialize second param takes a float (0.0 - 1.0).<br>
+* Fix many spelling and grammar errors. :(<br>
+
 ><b>Updated (10/11/15 v2.1)</b><br>
 * Made yin implementation faster and more reliable.<br>
 * Improved user interface.<br>