Import Duff's AudioTuner library

https://forum.pjrc.com/threads/32252-Different-Range-FFT-Algorithm?p=91895&viewfull=1#post91895

Audio.h

 #include "analyze_fft1024.h"
 #include "analyze_print.h"
 #include "analyze_tonedetect.h"
+#include "analyze_guitartuner.h"
 #include "analyze_peak.h"
 #include "control_sgtl5000.h"
 #include "control_wm8731.h"

analyze_guitartuner.cpp

+/* Audio Library Guitar and Bass Tuner
+ * Copyright (c) 2015, Colin Duffy
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice, development funding notice, and this permission
+ * notice shall be included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include "analyze_guitartuner.h"
+#include "utility/dspinst.h"
+#include "arm_math.h"
+
+#define HALF_BLOCKS AUDIO_BLOCKS * 64
+
+#define LOOP1(a)  a
+#define LOOP2(a)  a LOOP1(a)
+#define LOOP3(a)  a LOOP2(a)
+#define LOOP4(a)  a LOOP3(a)
+#define LOOP8(a)  a LOOP3(a) a LOOP3(a)
+#define LOOP16(a) a LOOP8(a) a LOOP2(a) a LOOP3(a)
+#define LOOP32(a)  a LOOP16(a) a LOOP8(a) a LOOP1(a) a LOOP3(a)
+#define LOOP64(a)  a LOOP32(a) a LOOP16(a) a LOOP8(a) a LOOP2(a) a LOOP1(a)
+#define UNROLL(n,a) LOOP##n(a)
+
+static void copy_buffer(void *destination, const void *source) {
+    const uint16_t *src = (const uint16_t *)source;
+    uint16_t *dst = (uint16_t *)destination;
+    for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) *dst++ = *src++;
+}
+
+void AudioTuner::update( void ) {
+    
+    audio_block_t *block;
+    
+    block = receiveReadOnly();
+    if (!block) return;
+    
+    if ( !enabled ) {
+        release( block );
+        return;
+    }
+    
+    digitalWriteFast(2, HIGH);
+    if ( next_buffer ) {
+        blocklist1[state++] = block;
+        if ( !first_run && process_buffer ) process( );
+    } else {
+        blocklist2[state++] = block;
+        if ( !first_run && process_buffer ) process( );
+    }
+    
+    if ( state >= AUDIO_BLOCKS ) {
+        if ( next_buffer ) {
+            if ( !first_run && process_buffer ) process( );
+            for ( int i = 0; i < AUDIO_BLOCKS; i++ ) copy_buffer( AudioBuffer+( i * 0x80 ), blocklist1[i]->data );
+            for ( int i = 0; i < AUDIO_BLOCKS; i++ ) release(blocklist1[i] );
+        } else {
+            if ( !first_run && process_buffer ) process( );
+            for ( int i = 0; i < AUDIO_BLOCKS; i++ ) copy_buffer( AudioBuffer+( i * 0x80 ), blocklist2[i]->data );
+            for ( int i = 0; i < AUDIO_BLOCKS; i++ ) release( blocklist2[i] );
+        }
+        process_buffer = true;
+        first_run = false;
+        state = 0;
+        //digitalWriteFast(LED_BUILTIN, !digitalReadFast(LED_BUILTIN));
+    }
+}
+
+FASTRUN void AudioTuner::process( void ) {
+    //digitalWriteFast(0, HIGH);
+    
+    const int16_t *p;
+    p = AudioBuffer;
+    
+    uint16_t cycles = 64;;
+    uint16_t tau = tau_global;
+    do {
+        uint16_t x   = 0;
+        int64_t  sum = 0;
+        //uint32_t res;
+        do {
+            /*int16_t current1, lag1, current2, lag2;
+             int32_t val1, val2;
+             lag1 = *( ( uint32_t * )p + ( x + tau ) );
+             current1 = *( ( uint32_t * )p + x );
+             x += 32;
+             lag2 = *( ( uint32_t * )p + ( x + tau ) );
+             current2 = *( ( uint32_t * )p + x );
+             val1 = __PKHBT(current1, current2, 0x10);
+             val2 = __PKHBT(lag1, lag2, 0x10);
+             res = __SSUB16( val1, val2 );
+             sum = __SMLALD(res, res, sum);
+             //sum = __SMLSLD(delta1, delta2, sum);*/
+            int16_t current, lag, delta;
+            //UNROLL(16,
+                   lag = *( ( int16_t * )p + ( x+tau ) );
+                   current = *( ( int16_t * )p+x );
+                   delta = ( current-lag );
+                   sum += delta * delta;
+#if F_CPU == 144000000
+                   x += 8;
+#elif F_CPU == 120000000
+                   x += 12;
+#elif F_CPU == 96000000
+                   x += 16;
+#elif F_CPU < 96000000
+                   x += 32;
+#endif
+                   //);
+        } while ( x <= HALF_BLOCKS );
+
+        running_sum += sum;
+        yin_buffer[yin_idx] = sum*tau;
+        rs_buffer[yin_idx] = running_sum;
+        yin_idx = ( ++yin_idx >= 5 ) ? 0 : yin_idx;
+        tau = estimate( yin_buffer, rs_buffer, yin_idx, tau );
+
+        if ( tau == 0 ) {
+            process_buffer  = false;
+            new_output      = true;
+            yin_idx         = 1;
+            running_sum     = 0;
+            tau_global      = 1;
+            //digitalWriteFast(2, LOW);
+            //digitalWriteFast(0, LOW);
+            return;
+        }
+    } while ( --cycles );
+    
+    if ( tau >= HALF_BLOCKS ) {
+        process_buffer  = false;
+        new_output      = false;
+        yin_idx         = 1;
+        running_sum     = 0;
+        tau_global      = 1;
+        //digitalWriteFast(0, LOW);
+        return;
+    }
+    tau_global = tau;
+    //digitalWriteFast(0, LOW);
+}
+
+/**
+ *  check the sampled data for fundmental frequency
+ *
+ *  @param yin  buffer to hold sum*tau value
+ *  @param rs   buffer to hold running sum for sampled window
+ *  @param head buffer index
+ *  @param tau  lag we are currently working on this gets incremented
+ *
+ *  @return tau
+ */
+uint16_t AudioTuner::estimate( int64_t *yin, int64_t *rs, uint16_t head, uint16_t tau ) {
+    const int64_t *y = ( int64_t * )yin;
+    const int64_t *r = ( int64_t * )rs;
+    uint16_t _tau, _head;
+    const float thresh = yin_threshold;
+    _tau = tau;
+    _head = head;
+    
+    if ( _tau > 4 ) {
+        
+        uint16_t idx0, idx1, idx2;
+        idx0 = _head;
+        idx1 = _head + 1;
+        idx1 = ( idx1 >= 5 ) ? 0 : idx1;
+        idx2 = head + 2;
+        idx2 = ( idx2 >= 5 ) ? 0 : idx2;
+        
+        float s0, s1, s2;
+        s0 = ( ( float )*( y+idx0 ) / *( r+idx0 ) );
+        s1 = ( ( float )*( y+idx1 ) / *( r+idx1 ) );
+        s2 = ( ( float )*( y+idx2 ) / *( r+idx2 ) );
+        
+        if ( s1 < thresh && s1 < s2 ) {
+            uint16_t period = _tau - 3;
+            periodicity = 1 - s1;
+            data = period + 0.5f * ( s0 - s2 ) / ( s0 - 2.0f * s1 + s2 );
+            return 0;
+        }
+    }
+    return _tau + 1;
+}
+
+/**
+ *  Initialise
+ *
+ *  @param threshold Allowed uncertainty
+ *  @param cpu_max   How much cpu usage before throttling
+ */
+void AudioTuner::initialize( float threshold ) {
+    __disable_irq( );
+    process_buffer = false;
+    yin_threshold  = threshold;
+    periodicity    = 0.0f;
+    next_buffer    = true;
+    running_sum    = 0;
+    tau_global     = 1;
+    first_run      = true;
+    yin_idx        = 1;
+    enabled        = true;
+    state          = 0;
+    data           = 0.0f;
+    __enable_irq( );
+}
+
+/**
+ *  available
+ *
+ *  @return true if data is ready else false
+ */
+bool AudioTuner::available( void ) {
+    __disable_irq( );
+    bool flag = new_output;
+    if ( flag ) new_output = false;
+    __enable_irq( );
+    return flag;
+}
+
+/**
+ *  read processes the data samples for the Yin algorithm.
+ *
+ *  @return frequency in hertz
+ */
+float AudioTuner::read( void ) {
+    __disable_irq( );
+    float d = data;
+    __enable_irq( );
+    return AUDIO_SAMPLE_RATE_EXACT / d;
+}
+
+/**
+ *  Periodicity of the sampled signal from Yin algorithm from read function.
+ *
+ *  @return periodicity
+ */
+float AudioTuner::probability( void ) {
+    __disable_irq( );
+    float p = periodicity;
+    __enable_irq( );
+    return p;
+}
+
+/**
+ *  Initialise parameters.
+ *
+ *  @param thresh    Allowed uncertainty
+ */
+void AudioTuner::threshold( float p ) {
+    __disable_irq( );
+    yin_threshold = p;
+    __enable_irq( );
+}

analyze_guitartuner.h

+/* Audio Library Guitar and Bass Tuner
+ * Copyright (c) 2015, Colin Duffy
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice, development funding notice, and this permission
+ * notice shall be included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#ifndef AudioTuner_h_
+#define AudioTuner_h_
+
+#include "AudioStream.h"
+/****************************************************************
+ *              Safe to adjust these values below               *
+ *                                                              *
+ *  This parameter defines the size of the buffer.              *
+ *                                                              *
+ *  1.  AUDIO_BLOCKS -  Buffer size is 128 * AUDIO_BLOCKS.      *
+ *                      The more AUDIO_BLOCKS the lower the     *
+ *                      frequency you can detect. The defualt   *
+ *                      (24) is set to measure down to 29.14    *
+ *                      Hz or B(flat)0.                         *
+ *                                                              *
+ ****************************************************************/
+#define AUDIO_BLOCKS  24
+/****************************************************************/
+class AudioTuner : public AudioStream {
+public:
+    /**
+     *  constructor to setup Audio Library and initialize
+     *
+     *  @return none
+     */
+    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
+     *
+     *  @return none
+     */
+    void initialize( float threshold );
+    
+    /**
+     *  sets threshold value
+     *
+     *  @param thresh
+     *  @return none
+     */
+    void threshold( float p );
+    
+    /**
+     *  triggers true when valid frequency is found
+     *
+     *  @return flag to indicate valid frequency is found
+     */
+    bool available( void );
+    /**
+     *  get frequency
+     *
+     *  @return frequency in hertz
+     */
+    float read( void );
+    
+    /**
+     *  get predicitity
+     *
+     *  @return probability of frequency found
+     */
+    float probability( void );
+    
+    /**
+     *  Audio Library calls this update function ~2.9ms
+     *
+     *  @return none
+     */
+    virtual void update( void );
+
+private:
+    /**
+     *  check the sampled data for fundamental frequency
+     *
+     *  @param yin  buffer to hold sum*tau value
+     *  @param rs   buffer to hold running sum for sampled window
+     *  @param head buffer index
+     *  @param tau  lag we are currently working on this gets incremented
+     *
+     *  @return tau
+     */
+    uint16_t estimate( int64_t *yin, int64_t *rs, uint16_t head, uint16_t tau );
+    
+    /**
+     *  process audio data
+     *
+     *  @return none
+     */
+    void process( void );
+    
+    /**
+     *  Variables
+     */
+    uint64_t running_sum;
+    uint16_t tau_global;
+    int64_t  rs_buffer[5], yin_buffer[5];
+    int16_t  AudioBuffer[AUDIO_BLOCKS*128] __attribute__ ( ( aligned ( 4 ) ) );
+    uint8_t  yin_idx, state;
+    float    periodicity, yin_threshold, cpu_usage_max, data;
+    bool     enabled, next_buffer, first_run;
+    volatile bool new_output, process_buffer;
+    audio_block_t *blocklist1[AUDIO_BLOCKS];
+    audio_block_t *blocklist2[AUDIO_BLOCKS];
+    audio_block_t *inputQueueArray[1];
+};
+#endif