Merge remote-tracking branch 'refs/remotes/PaulStoffregen/master'

Audio.h

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

analyze_notefreq.cpp

+/* Audio Library Note Frequency Detection & Guitar/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_notefreq.h"
+#include "utility/dspinst.h"
+#include "arm_math.h"
+
+#define HALF_BLOCKS AUDIO_GUITARTUNER_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 AudioAnalyzeNoteFrequency::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_GUITARTUNER_BLOCKS ) {
+        if ( next_buffer ) {
+            if ( !first_run && process_buffer ) process( );
+            for ( int i = 0; i < AUDIO_GUITARTUNER_BLOCKS; i++ ) copy_buffer( AudioBuffer+( i * 0x80 ), blocklist1[i]->data );
+            for ( int i = 0; i < AUDIO_GUITARTUNER_BLOCKS; i++ ) release(blocklist1[i] );
+        } else {
+            if ( !first_run && process_buffer ) process( );
+            for ( int i = 0; i < AUDIO_GUITARTUNER_BLOCKS; i++ ) copy_buffer( AudioBuffer+( i * 0x80 ), blocklist2[i]->data );
+            for ( int i = 0; i < AUDIO_GUITARTUNER_BLOCKS; i++ ) release( blocklist2[i] );
+        }
+        process_buffer = true;
+        first_run = false;
+        state = 0;
+        //digitalWriteFast(LED_BUILTIN, !digitalReadFast(LED_BUILTIN));
+    }
+}
+
+FASTRUN void AudioAnalyzeNoteFrequency::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 AudioAnalyzeNoteFrequency::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 AudioAnalyzeNoteFrequency::begin( 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 AudioAnalyzeNoteFrequency::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 AudioAnalyzeNoteFrequency::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 AudioAnalyzeNoteFrequency::probability( void ) {
+    __disable_irq( );
+    float p = periodicity;
+    __enable_irq( );
+    return p;
+}
+
+/**
+ *  Initialise parameters.
+ *
+ *  @param thresh    Allowed uncertainty
+ */
+void AudioAnalyzeNoteFrequency::threshold( float p ) {
+    __disable_irq( );
+    yin_threshold = p;
+    __enable_irq( );
+}

analyze_notefreq.h

+/* Audio Library Note Frequency Detection & Guitar/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 AudioAnalyzeNoteFrequency_h_
+#define AudioAnalyzeNoteFrequency_h_
+
+#include "AudioStream.h"
+/***********************************************************************
+ *              Safe to adjust these values below                      *
+ *                                                                     *
+ *  This parameter defines the size of the buffer.                     *
+ *                                                                     *
+ *  1.  AUDIO_GUITARTUNER_BLOCKS -  Buffer size is 128 * AUDIO_BLOCKS. *
+ *                      The more AUDIO_GUITARTUNER_BLOCKS the lower    *
+ *                      the frequency you can detect. The default      *
+ *                      (24) is set to measure down to 29.14 Hz        *
+ *                      or B(flat)0.                                   *
+ *                                                                     *
+ ***********************************************************************/
+#define AUDIO_GUITARTUNER_BLOCKS  24
+/***********************************************************************/
+class AudioAnalyzeNoteFrequency : public AudioStream {
+public:
+    /**
+     *  constructor to setup Audio Library and initialize
+     *
+     *  @return none
+     */
+    AudioAnalyzeNoteFrequency( 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 begin( 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_GUITARTUNER_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_GUITARTUNER_BLOCKS];
+    audio_block_t *blocklist2[AUDIO_GUITARTUNER_BLOCKS];
+    audio_block_t *inputQueueArray[1];
+};
+#endif

examples/Analysis/NoteFrequency/NoteFrequency.ino

+/* Detect the frequency of music notes, by Colin Duffy
+
+   This example repeatedly plays a guitar note (output to the DAC pin)
+   and prints an analysis of the frequency to the Arduino Serial Monitor
+
+   https://forum.pjrc.com/threads/32252-Different-Range-FFT-Algorithm/page2
+   https://github.com/duff2013/AudioTuner
+*/
+/*
+ C     C#    D     Eb    E     F     F#    G     G#    A     Bb    B
+ 0 16.35 17.32 18.35 19.45 20.60 21.83 23.12 24.50 25.96 27.50 29.14 30.87
+ 1 32.70 34.65 36.71 38.89 41.20 43.65 46.25 49.00 51.91 55.00 58.27 61.74
+ 2 65.41 69.30 73.42 77.78 82.41 87.31 92.50 98.00 103.8 110.0 116.5 123.5
+ 3 130.8 138.6 146.8 155.6 164.8 174.6 185.0 196.0 207.7 220.0 233.1 246.9
+ 4 261.6 277.2 293.7 311.1 329.6 349.2 370.0 392.0 415.3 440.0 466.2 493.9
+ 5 523.3 554.4 587.3 622.3 659.3 698.5 740.0 784.0 830.6 880.0 932.3 987.8
+ 6 1047  1109  1175  1245  1319  1397  1480  1568  1661  1760  1865  1976
+ 7 2093  2217  2349  2489  2637  2794  2960  3136  3322  3520  3729  3951
+ 8 4186  4435  4699  4978  5274  5588  5920  6272  6645  7040  7459  7902
+ 
+ Guitar strings are E2=82.41Hz, A2=110Hz, D3=146.8Hz, G3=196Hz, B3=246.9Hz, E4=329.6Hz
+ 
+ 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 @ 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.
+ */
+#include <SerialFlash.h>
+#include <Audio.h>
+#include <Wire.h>
+#include <SPI.h>
+#include <SD.h>
+//---------------------------------------------------------------------------------------
+#include "e2_note.h"
+#include "a2_note.h"
+#include "d3_note.h"
+#include "g3_note.h"
+#include "b3_note.h"
+#include "e4_note.h"
+//---------------------------------------------------------------------------------------
+AudioAnalyzeNoteFrequency notefreq;
+AudioOutputAnalog         dac;
+AudioPlayMemory           wav_note;
+AudioMixer4               mixer;
+//---------------------------------------------------------------------------------------
+AudioConnection patchCord0(wav_note, 0, mixer, 0);
+AudioConnection patchCord1(mixer, 0, notefreq, 0);
+AudioConnection patchCord2(mixer, 0, dac, 0);
+//---------------------------------------------------------------------------------------
+IntervalTimer playNoteTimer;
+
+void playNote(void) {
+    if (!wav_note.isPlaying()) {
+        wav_note.play(e2_note);
+        //wav_note.play(a2_note);
+        //wav_note.play(d3_note);
+        //wav_note.play(g3_note);
+        //wav_note.play(b3_note);
+        //wav_note.play(e4_note);
+        digitalWriteFast(LED_BUILTIN, !digitalReadFast(LED_BUILTIN));
+    }
+}
+//---------------------------------------------------------------------------------------
+void setup() {
+    AudioMemory(30);
+    /*
+     *  Initialize the yin algorithm's absolute
+     *  threshold, this is good number.
+     */
+    notefreq.begin(.15);
+    pinMode(LED_BUILTIN, OUTPUT);
+    playNoteTimer.begin(playNote, 1000);
+}
+
+void loop() {
+    // read back fundamental frequency
+    if (notefreq.available()) {
+        float note = notefreq.read();
+        float prob = notefreq.probability();
+        Serial.printf("Note: %3.2f | Probability: %.2f\n", note, prob);
+    }
+}

examples/Analysis/NoteFrequency/a2_note.h

+#include "Arduino.h"
+extern const unsigned int a2_note[53971];

examples/Analysis/NoteFrequency/b3_note.h

+#include "Arduino.h"
+extern const unsigned int b3_note[53990];

examples/Analysis/NoteFrequency/d3_note.h

+#include "Arduino.h"
+extern const unsigned int d3_note[53974];

examples/Analysis/NoteFrequency/e2_note.h

+#include "Arduino.h"
+extern const unsigned int e2_note[53990];

examples/Analysis/NoteFrequency/e4_note.h

+#include "Arduino.h"
+extern const unsigned int e4_note[53990];

examples/Analysis/NoteFrequency/g3_note.h

+#include "Arduino.h"
+extern const unsigned int g3_note[53965];

examples/WavFilePlayer/WavFilePlayer.ino

 
 
 void loop() {
-  playFile("SDTEST1.WAV");
+  playFile("SDTEST1.WAV");  // filenames are always uppercase 8.3 format
   delay(500);
   playFile("SDTEST2.WAV");
   delay(500);

gui/index.html

 		{"type":"AudioAnalyzeFFT256","data":{"defaults":{"name":{"value":"new"}},"shortName":"fft256","inputs":1,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png"}},
 		{"type":"AudioAnalyzeFFT1024","data":{"defaults":{"name":{"value":"new"}},"shortName":"fft1024","inputs":1,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png"}},
 		{"type":"AudioAnalyzeToneDetect","data":{"defaults":{"name":{"value":"new"}},"shortName":"tone","inputs":1,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png"}},
+		{"type":"AudioAnalyzeNoteFrequency","data":{"defaults":{"name":{"value":"new"}},"shortName":"notefreq","inputs":1,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png"}},
 		{"type":"AudioAnalyzePrint","data":{"defaults":{"name":{"value":"new"}},"shortName":"print","inputs":1,"outputs":0,"category":"analyze-function","color":"#E6E0F8","icon":"arrow-in.png"}},
 		{"type":"AudioControlSGTL5000","data":{"defaults":{"name":{"value":"new"}},"shortName":"sgtl5000","inputs":0,"outputs":0,"category":"control-function","color":"#E6E0F8","icon":"arrow-in.png"}},
 		{"type":"AudioControlWM8731","data":{"defaults":{"name":{"value":"new"}},"shortName":"wm8731","inputs":0,"outputs":0,"category":"control-function","color":"#E6E0F8","icon":"arrow-in.png"}},
 	</p>
 	<p>I2S master objects can be used together with non-I2S input and output
 		objects, for simultaneous audio streaming on different hardware.</p>
-</script><
+</script>
 <script type="text/x-red" data-template-name="AudioInputI2S">
 	<div class="form-row">
 		<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
 		<ul>
 		<li><span class=literal>WAVEFORM_SINE</span></li>
 		<li><span class=literal>WAVEFORM_SAWTOOTH</span></li>
+		<li><span class=literal>WAVEFORM_SAWTOOTH_REVERSE</span></li>
 		<li><span class=literal>WAVEFORM_SQUARE</span></li>
 		<li><span class=literal>WAVEFORM_TRIANGLE</span></li>
 		<li><span class=literal>WAVEFORM_ARBITRARY</span></li>
 		<li><span class=literal>WAVEFORM_PULSE</span></li>
+		<li><span class=literal>WAVEFORM_SAMPLE_HOLD</span></li>
 		</ul>
 	</p>
 </script>
 	</div>
 </script>
 
+<script type="text/x-red" data-help-name="AudioAnalyzeNoteFrequency">
+	<h3>Summary</h3>
+	<p>Detect with fairly good accuracy the fundamental frequency f<sub>o</sub>
+		of musical notes, such as electric guitar and bass.</p>
+    <p>Written By Collin Duffy</p>
+	<h3>Audio Connections</h3>
+	<table class=doc align=center cellpadding=3>
+		<tr class=top><th>Port</th><th>Purpose</th></tr>
+		<tr class=odd><td align=center>In 0</td><td>Signal to analyze</td></tr>
+	</table>
+	<h3>Functions</h3>
+	<p class=func><span class=keyword>begin</span>(threshold);</p>
+	<p class=desc>Initialize and start detecting frequencies,
+		with an initial threshold (the amount of allowed uncertainty).
+	</p>
+	<p class=func><span class=keyword>available</span>();</p>
+	<p class=desc>Returns true (non-zero) when a valid
+		frequency is detected.
+	</p>
+	<p class=func><span class=keyword>read</span>();</p>
+	<p class=desc>Read the detected frequency.
+	</p>
+	<p class=func><span class=keyword>probability</span>();</p>
+	<p class=desc>Return the level of certainty, betweeo 0 to 1.0.
+	</p>
+	<p class=func><span class=keyword>threshold</span>(level);</p>
+	<p class=desc>Set the detection threshold, the amount of allowed uncertainty.
+	</p>
+	<h3>Examples</h3>
+	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; NoteFrequency
+	</p>
+	<h3>Notes</h3>
+	<p>The <a href="http://recherche.ircam.fr/equipes/pcm/cheveign/pss/2002_JASA_YIN.pdf">YIN algorithm</a> (PDF)
+		is used to detect frequencies, with many optimizations for
+		frequencies between 29-400Hz.  This algorithm can be somewhat
+		memory and processor hungry but will allow you to detect with
+		fairly good accuracy the fundamental frequencies from
+		electric guitars and basses.</p>
+	<p>Within the code, AUDIO_GUITARTUNER_BLOCKS
+		may be edited to control low frequency range.  The default
+		(24) allows measurement down to 29.14 Hz, or B(flat)0.</p>
+	<p>TODO: The usable upper range of this object is not well known.
+		Duff says "it should be good up to 1000Hz", but may have trouble
+		at 4 kHz.  Please <a href="https://forum.pjrc.com/threads/32252-Different-Range-FFT-Algorithm/page2">post feedback here</a>, ideally with audio clips for the NoteFrequency example.</p>
+	<p>This object was contributed by Collin Duffy from his
+		<a href="https://github.com/duff2013/AudioTuner">AudioTuner project</a>.
+		Additional details and documentation may be found there.</p>
+</script>
+<script type="text/x-red" data-template-name="AudioAnalyzeNoteFrequency">
+	<div class="form-row">
+		<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
+		<input type="text" id="node-input-name" placeholder="Name">
+	</div>
+</script>
+
 <script type="text/x-red" data-help-name="AudioAnalyzePrint">
 	<h3>Summary</h3>
 	<p>Print raw audio data to the Arduino Serial Monitor.  This

gui/red/nodes.js

 		return node_defs[type];
 	}
 	function selectNode(name) {
-	//	window.history.pushState(null, null, window.location.protocol + "//"
-	//		+ window.location.host + window.location.pathname + '?info=' + name);
+		// on Chrome this causes "Uncaught SecurityError" when used from file:
+		// but other than errors in the console, doesn't seem to harm anything
+		window.history.pushState(null, null, window.location.protocol + "//"
+			+ window.location.host + window.location.pathname + '?info=' + name);
 	}
 	function addNode(n) {
 		if (n._def.category == "config") {
 				}
 
 				// ... and it has to end with an semikolon ...
-				var pattSe = new RegExp(/.*;$/);
+				var pattSe = new RegExp(/.*;.*$/);
 				var pattCoord = new RegExp(/.*\/\/xy=\d+,\d+$/);
 				if (pattSe.test(line) || pattCoord.test(line)) {
 					var word = parts[1].trim();

gui/red/ui/view.js

 		}
 	}
 
+	function doSort (arr) {
+		arr.sort(function (a, b) {
+			var nameA = a.name ? a.name : a.id;
+			var nameB = b.name ? b.name : b.id;
+			return nameA.localeCompare(nameB, 'en', {numeric: 'true'});
+		});
+	}
+
+	function setNewCoords (lastX, lastY, arr) {
+		var x = lastX;
+		var y = lastY;
+		for (var i = 0; i < arr.length; i++) {
+			var node = arr[i];
+			var name = node.name ? node.name : node.id;
+			var def = node._def;
+			var dH = Math.max(RED.view.defaults.height, (Math.max(def.outputs, def.inputs) || 0) * 15);
+			x = lastX + Math.max(RED.view.defaults.width, RED.view.calculateTextWidth(name) + (def.inputs > 0 ? 7 : 0));
+			node.x = x;
+			node.y = y + dH/2;
+			y = y + dH + 15;
+			node.dirty = true;
+		}
+		return { x: x, y: y };
+	}
+
 	function arrangeAll() {
-		// TODO: arrange imported nodes without coordinates
+		var ioNoIn = [];
+		var ioInOut = [];
+		var ioMultiple = [];
+		var ioNoOut = [];
+		var ioCtrl = [];
+
+		RED.nodes.eachNode(function (node) {
+
+			if (node._def.inputs == 0 && node._def.outputs == 0) {
+				ioCtrl.push(node);
+			} else if (node._def.inputs == 0) {
+				ioNoIn.push(node);
+			} else if (node._def.outputs == 0) {
+				ioNoOut.push(node);
+			} else if (node._def.inputs == 1 && node._def.outputs == 1) {
+				ioInOut.push(node);
+			} else if (node._def.inputs > 1) {
+				ioMultiple.push(node);
+			}
+		});
+
+		var cols = new Array(ioNoIn, ioInOut, ioMultiple, ioNoOut, ioCtrl);
+		var lowestY = 0;
+
+		for (var i = 0; i < cols.length; i++) {
+			var dX = ((i < cols.length - 1) ?  i : 0) * (RED.view.defaults.width * 2) + (RED.view.defaults.width / 2) + 15;
+			var dY = ((i < cols.length - 1) ?  (RED.view.defaults.height / 4) : lowestY) + 15;
+			var startX = 0;
+			var startY = 0;
+
+			doSort(cols[i]);
+			var last = setNewCoords(startX + dX, startY + dY, cols[i]);
+			lowestY = Math.max(lowestY, last.y);
+			startX = ((i < cols.length - 1) ? last.x : 0) + (RED.view.defaults.width) * 4;
+			startY = lowestY + (RED.view.defaults.height * 1.5);
+		}
+		RED.storage.update();
+		redraw();
 	}
 
 	RED.keyboard.add(/* z */ 90,{ctrl:true},function(){RED.history.pop();});

keywords.txt

 AudioAnalyzePeak	KEYWORD2
 AudioAnalyzePrint	KEYWORD2
 AudioAnalyzeToneDetect	KEYWORD2
+AudioAnalyzeGuitarTuner	KEYWORD2
 AudioEffectChorus	KEYWORD2
 AudioEffectFade	KEYWORD2
 AudioEffectFlange	KEYWORD2
 sampleRate	KEYWORD2
 bits	KEYWORD2
 mute_PCM	KEYWORD2
+probability	KEYWORD2
 
 AudioMemoryUsage	KEYWORD2
 AudioMemoryUsageMax	KEYWORD2
 WAVEFORM_TRIANGLE	LITERAL1
 WAVEFORM_ARBITRARY	LITERAL1
 WAVEFORM_PULSE	LITERAL1
+WAVEFORM_SAWTOOTH_REVERSE	LITERAL1
+WAVEFORM_SAMPLE_HOLD	LITERAL1
 
 AUDIO_MEMORY_23LC1024	LITERAL1
 AUDIO_MEMORY_MEMORYBOARD	LITERAL1

synth_waveform.cpp

       }
       break;
 
+    case WAVEFORM_SAWTOOTH_REVERSE:
+      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
+        *bp++ = ((short)(tone_phase>>15)*tone_amp) >> 15;
+         // phase and incr are both unsigned 32-bit fractions
+         tone_phase -= tone_incr;
+      }
+      break;
+
     case WAVEFORM_TRIANGLE:
       for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
         if(tone_phase & 0x80000000) {
       }
       break;
       
+    case WAVEFORM_SAMPLE_HOLD:
+      for(int i = 0;i < AUDIO_BLOCK_SAMPLES;i++) {
+        if(tone_phase < tone_incr) {
+          sample = random(-tone_amp, tone_amp);
+        }
+        *bp++ = sample;
+        tone_phase += tone_incr;
+      }
+      break;
     }
     if (tone_offset) {
 	bp = block->data;

synth_waveform.h

 #define WAVEFORM_TRIANGLE  3
 #define WAVEFORM_ARBITRARY 4
 #define WAVEFORM_PULSE     5
+#define WAVEFORM_SAWTOOTH_REVERSE 6
+#define WAVEFORM_SAMPLE_HOLD 7
 
 // todo: remove these...
 #define TONE_TYPE_SINE     0
   short    tone_freq;
   uint32_t tone_phase;
   uint32_t tone_width;
+  // sample for SAMPLE_HOLD
+  short sample;
   // volatile prevents the compiler optimizing out the frequency function
   volatile uint32_t tone_incr;
   short    tone_type;