Removed the asynchronous loading of HTML files via JavaScript, because this will be blocked on real offline use (file:// protocol).

Removed the extern files, except the JSON file with the node descriptions, this will be loaded in offline mode too.

gui/red/main.js

 
 	$(function() {
 		$(".palette-spinner").show();
+		$.ajaxSetup({beforeSend: function(xhr){
+			if (xhr.overrideMimeType) {
+				xhr.overrideMimeType("application/json");
+			}
+		}});
 		$.getJSON( "resources/nodes_def.json", function( data ) {
 			var nodes = data["nodes"];
 			$.each(nodes, function(key, val) {

gui/red/ui/palette.js

 		}
 	}
 
-	function setTooltipContent(empty, key, elem) {
-		$.get( "resources/help/" + key + ".html", function( data ) {
+	function setTooltipContent(prefix, key, elem) {
+		// server test switched off - test purposes only
+		var patt = new RegExp(/^[http|https]/);
+		var server = false && patt.test(location.protocol);
+
+		var options = {
+			title: elem.type,
+			placement: "right",
+			trigger: "hover",
+			delay: { show: 750, hide: 50 },
+			html: true,
+			container:'body',
+			content : ""
+		};
+
+		if (!server) {
+			data = $("script[data-help-name|='" + key + "']").html();
 			var firstP = $("<div/>").append(data).children("p").first().html();
-			$(elem).popover({
-				title: elem.type,
-				placement: "right",
-				trigger: "hover",
-				delay: { show: 750, hide: 50 },
-				html: true,
-				container:'body',
-				content: firstP
+			options.content = firstP;
+			$(elem).popover(options);
+		} else {
+			$.get( "resources/help/" + key + ".html", function( data ) {
+				var firstP = $("<div/>").append(data).children("p").first().html();
+				options.content = firstP;
+				$(elem).popover(options);
 			});
-		});
+		}
 	}
 	
 	function removeNodeType(type) {

gui/red/ui/tab-info.js

 		this.setHelpContent(table, node.type);
 	}
 
-	function setHelpContent(empty, key) {
-		$.get( "resources/help/" + key + ".html", function( data ) {
-			$("#tab-info").html(empty + '<div class="node-help">' + data + '</div>');
-		}).fail(function () {
-			$("#tab-info").html(empty);
-		});
+	function setHelpContent(prefix, key) {
+		// server test switched off - test purposes only
+		var patt = new RegExp(/^[http|https]/);
+		var server = false && patt.test(location.protocol);
+
+
+		prefix = prefix == "" ? "<h3>" + key + "</h3>" : prefix;
+		if (!server) {
+			data = $("script[data-help-name|='" + key + "']").html();
+			$("#tab-info").html(prefix + '<div class="node-help">' + data + '</div>');
+		} else {
+			$.get( "resources/help/" + key + ".html", function( data ) {
+				$("#tab-info").html(prefix + '<h2>' + key + '</h2><div class="node-help">' + data + '</div>');
+			}).fail(function () {
+				$("#tab-info").html(prefix);
+			});
+		}
 	}
 	
 	return {

gui/resources/form/AudioAnalyzeFFT1024.html

-
-	<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>

gui/resources/form/AudioAnalyzeFFT256.html

-
-	<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>

gui/resources/form/AudioAnalyzePeak.html

-
-	<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>

gui/resources/form/AudioAnalyzePrint.html

-
-	<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>

gui/resources/form/AudioAnalyzeToneDetect.html

-
-	<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>

gui/resources/form/AudioControlSGTL5000.html

-
-	<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>

gui/resources/form/AudioControlWM8731.html

-
-	<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>

gui/resources/form/AudioControlWM8731master.html

-
-	<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>

gui/resources/form/AudioEffectBitcrusher.html

-
-    <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>

gui/resources/form/AudioEffectChorus.html

-
-	<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>

gui/resources/form/AudioEffectDelay.html

-
-	<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>

gui/resources/form/AudioEffectDelayExternal.html

-
-	<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>

gui/resources/form/AudioEffectEnvelope.html

-
-	<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>

gui/resources/form/AudioEffectFade.html

-
-	<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>

gui/resources/form/AudioEffectFlange.html

-
-	<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>

gui/resources/form/AudioEffectMultiply.html

-
-	<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>

gui/resources/form/AudioFilterBiquad.html

-
-	<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>

gui/resources/form/AudioFilterFIR.html

-
-	<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>

gui/resources/form/AudioInputAnalog.html

-
-	<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>

gui/resources/form/AudioInputI2S.html

-
-	<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>

gui/resources/form/AudioInputI2Sslave.html

-
-	<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>

gui/resources/form/AudioMixer4.html

-
-	<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>

gui/resources/form/AudioOutputAnalog.html

-
-	<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>

gui/resources/form/AudioOutputI2S.html

-
-	<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>

gui/resources/form/AudioOutputI2Sslave.html

-
-	<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>

gui/resources/form/AudioOutputPWM.html

-
-	<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>

gui/resources/form/AudioOutputSPDIF.html

-
-	<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>

gui/resources/form/AudioPlayMemory.html

-
-	<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>

gui/resources/form/AudioPlayQueue.html

-
-	<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>

gui/resources/form/AudioPlaySdRaw.html

-
-	<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>

gui/resources/form/AudioPlaySdWav.html

-
-	<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>

gui/resources/form/AudioPlaySerialflashRaw.html

-
-	<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>

gui/resources/form/AudioRecordQueue.html

-
-	<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>

gui/resources/form/AudioSynthNoisePink.html

-
-	<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>

gui/resources/form/AudioSynthNoiseWhite.html

-
-	<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>

gui/resources/form/AudioSynthToneSweep.html

-
-	<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>

gui/resources/form/AudioSynthWaveform.html

-
-	<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>

gui/resources/form/AudioSynthWaveformDc.html

-
-	<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>

gui/resources/form/AudioSynthWaveformSine.html

-
-	<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>

gui/resources/form/AudioSynthWaveformSineModulated.html

-
-	<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>

gui/resources/form/export-clipboard-dialog.html

-
-	<div class="form-row">
-		<label for="node-input-export" style="display: block; width:100%;"><i class="icon-share"></i> Source Code:</label>
-		<textarea readonly style="font-family: monospace; font-size: 12px; background:rgb(226, 229, 255); padding-left: 0.5em;" class="input-block-level" id="node-input-export" rows="12"></textarea>
-	</div>
-	<div class="form-tips">
-		Select the text above and copy to the clipboard with Ctrl-A Ctrl-C.
-	</div>

gui/resources/form/export-library-dialog.html

-
-	<div class="form-row">
-		<label for="node-input-filename" ><i class="icon-tag"></i> Filename:</label>
-		<input type="text" id="node-input-filename" placeholder="Filename">
-	</div>

gui/resources/form/import-dialog.html

-
-	<div class="form-row">
-		<label for="node-input-export"><i class="icon-share"></i>Nodes:</label>
-		<textarea style="font-family: monospace; font-size: 12px; background:rgb(226, 229, 255); padding-left: 0.5em;" class="input-block-level" id="node-input-import" rows="5" placeholder="Paste nodes here, or lookup in the library"></textarea>
-	</div>

gui/resources/help/AudioAnalyzeFFT1024.html

-
-	<h3>Summary</h3>
-	<p>Compute a 1024 point Fast Fourier Transform (FFT) frequency analysis,
-		with real value (magnitude) output.  The frequency resolution is
-		43 Hz, useful detailed for audio visualization.</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 convert to frequency bins</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>available</span>();</p>
-	<p class=desc>Returns true each time the FFT analysis produces new output data.
-	</p>
-	<p class=func><span class=keyword>read</span>(binNumber);</p>
-	<p class=desc>Read a single frequency bin, from 0 to 511.  The result is scaled
-		so 1.0 represents a full scale sine wave.
-	</p>
-	<p class=func><span class=keyword>read</span>(firstBin, lastBin);</p>
-	<p class=desc>Read several frequency bins, returning their sum.  The higher
-		audio octaves are represented by many bins, which are typically read
-		as a group for audio visualization.
-	</p>
-	<p class=func><span class=keyword>averageTogether</span>(number);</p>
-	<p class=desc>This function does nothing.  The 1024 point FFT always
-		updates at approximately 86 times per second.
-	</p>
-	<p class=func><span class=keyword>windowFunction</span>(window);</p>
-	<p class=desc>Set the window function to be used.  AudioWindowHanning1024
-		is the default.  Windowing may be disabled by NULL, but windowing
-		should be used for all non-periodic (music) signals, and all periodic
-		signals that are not exact integer division of the sample rate.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; FFT
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; SpectrumAnalyzerBasic
-	</p>
-	<h3>Notes</h3>
-	<p>The raw 16 bit output data bins may be access with myFFT.output[num], where
-		num is 0 to 511.</p>
-	<p>TODO: caveats about spectral leakage vs frequency precision for arbitrary signals</p>
-	<p>Window Types:
-		<ul>
-		<li><span class=literal>AudioWindowHanning1024</span> (default)</li>
-		<li><span class=literal>AudioWindowBartlett1024</span></li>
-		<li><span class=literal>AudioWindowBlackman1024</span></li>
-		<li><span class=literal>AudioWindowFlattop1024</span></li>
-		<li><span class=literal>AudioWindowBlackmanHarris1024</span></li>
-		<li><span class=literal>AudioWindowNuttall1024</span></li>
-		<li><span class=literal>AudioWindowBlackmanNuttall1024</span></li>
-		<li><span class=literal>AudioWindowWelch1024</span></li>
-		<li><span class=literal>AudioWindowHamming1024</span></li>
-		<li><span class=literal>AudioWindowCosine1024</span></li>
-		<li><span class=literal>AudioWindowTukey1024</span></li>
-		</ul>
-	</p>
-	<p>1024 point FFT has a peak CPU usage of approx 52% on Teensy 3.1.
-		Average usage is much lower.  Future versions might distribute the
-		load more evenly over time....
-	</p>

gui/resources/help/AudioAnalyzeFFT256.html

-
-	<h3>Summary</h3>
-	<p>Compute a 256 point Fast Fourier Transform (FFT) frequency analysis,
-		with real value (magnitude) output.  The frequency resolution is
-		172 Hz, useful for simple audio visualization.</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 convert to frequency bins</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>available</span>();</p>
-	<p class=desc>Returns true each time the FFT analysis produces new output data.
-	</p>
-	<p class=func><span class=keyword>read</span>(binNumber);</p>
-	<p class=desc>Read a single frequency bin, from 0 to 127.  The result is scaled
-		so 1.0 represents a full scale sine wave.
-	</p>
-	<p class=func><span class=keyword>read</span>(firstBin, lastBin);</p>
-	<p class=desc>Read several frequency bins, returning their sum.  The higher
-		audio octaves are represented by many bins, which are typically read
-		as a group for audio visualization.
-	</p>
-	<p class=func><span class=keyword>averageTogether</span>(number);</p>
-	<p class=desc>New data is produced very radidly, approximately 344 times
-		per second.  Multiple outputs can be averaged together, so available()
-		returns true at a slower rate.
-	</p>
-	<p class=func><span class=keyword>windowFunction</span>(window);</p>
-	<p class=desc>Set the window function to be used.  AudioWindowHanning256
-		is the default.  Windowing may be disabled by NULL, but windowing
-		should be used for all non-periodic (music) signals, and all periodic
-		signals that are not exact integer division of the sample rate.
-	</p>
-
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; MemoryAndCpuUsage
-	</p>
-	<h3>Notes</h3>
-	<p>The raw 16 bit output data bins may be access with myFFT.output[num], where
-		num is 0 to 127.</p>
-	<p>TODO: caveats about spectral leakage vs frequency precision for arbitrary signals</p>
-	<p>Window Types:
-		<ul>
-		<li><span class=literal>AudioWindowHanning256</span> (default)</li>
-		<li><span class=literal>AudioWindowBartlett256</span></li>
-		<li><span class=literal>AudioWindowBlackman256</span></li>
-		<li><span class=literal>AudioWindowFlattop256</span></li>
-		<li><span class=literal>AudioWindowBlackmanHarris256</span></li>
-		<li><span class=literal>AudioWindowNuttall256</span></li>
-		<li><span class=literal>AudioWindowBlackmanNuttall256</span></li>
-		<li><span class=literal>AudioWindowWelch256</span></li>
-		<li><span class=literal>AudioWindowHamming256</span></li>
-		<li><span class=literal>AudioWindowCosine256</span></li>
-		<li><span class=literal>AudioWindowTukey256</span></li>
-		</ul>
-	</p>

gui/resources/help/AudioAnalyzePeak.html

-
-	<h3>Summary</h3>
-	<p>Track the signal peak amplitude.  Very useful for simple
-		audio level response projects, and general troubleshooting.</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>available</span>();</p>
-	<p class=desc>Returns true each time new peak data is available.
-	</p>
-	<p class=func><span class=keyword>read</span>();</p>
-	<p class=desc>Read the highest peak value since the last read.
-		Return is from 0.0 to 1.0.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; PeakMeterMono
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; PeakMeterStereo
-	</p>
-	<h3>Notes</h3>
-	<p></p>

gui/resources/help/AudioAnalyzePrint.html

-
-	<h3>Summary</h3>
-	<p>Print raw audio data to the Arduino Serial Monitor.  This
-		object creates massive output quickly, and should not normally be used.</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 print</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>name</span>(string);</p>
-	<p class=desc>blah blah blah blah
-	</p>
-	<p class=func><span class=keyword>trigger</span>();</p>
-	<p class=desc>blah blah blah blah
-	</p>
-	<p class=func><span class=keyword>trigger</span>(level, edge);</p>
-	<p class=desc>blah blah blah blah
-	</p>
-	<p class=func><span class=keyword>delay</span>(samples);</p>
-	<p class=desc>blah blah blah blah
-	</p>
-	<p class=func><span class=keyword>length</span>(samples);</p>
-	<p class=desc>blah blah blah blah
-	</p>
-	<!--<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; 
-	</p>-->
-	<h3>Notes</h3>
-	<p>This object doesn't work very well and probably should not be used.</p>

gui/resources/help/AudioAnalyzeToneDetect.html

-
-	<h3>Summary</h3>
-	<p>Detect the level of a single tone</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>frequency</span>(freq);</p>
-	<p class=desc>Set the frequency to detect.  The default detection time
-		will be 10 cycles of this frequency.
-	</p>
-	<p class=func><span class=keyword>frequency</span>(freq, cycles);</p>
-	<p class=desc>Set the frequency to detect, and the number of cycles.
-		Longer detection time (more cycles) will give higher precision,
-		but of course slower response.
-	</p>
-	<p class=func><span class=keyword>available</span>();</p>
-	<p class=desc>Returns true (non-zero) each time a detection interval
-		(number of cycles) completed and a new level is detected.
-	</p>
-	<p class=func><span class=keyword>read</span>();</p>
-	<p class=desc>Read the detected signal level.  Range is 0 to 1.0.
-	</p>
-	<p class=func><span class=keyword>threshold</span>(level);</p>
-	<p class=desc>Set a detection threshold, where the bool test operation
-		will return true if at or above this level, or false when below.
-	</p>
-	<p class=func>(bool)</p>
-	<p class=desc>By testing the object as a boolean value, you can respond
-		to detection of a tone.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; DialTone_Serial
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; DialTone_7segment
-	</p>
-	<h3>Notes</h3>
-	<p>Low frequency detection has trouble with numerical precision.
-		Works really well for all 8 DTMF frequencies, but fails for
-		detecting "sub audible tones" used in some control applications.</p>
-	<p>The (bool) test continues to return true until the next detection
-		interval (the configured number of cycles).  This behavior may
-		change in future versions, for a single true each time the signal
-		is detected, and then false for the remainder of that interval.</p>

gui/resources/help/AudioControlSGTL5000.html

-
-	<h3>Summary</h3>
-	<p>Control the SGTL5000 chip on the
-		<a href="http://www.pjrc.com/store/teensy3_audio.html" target="_blank">audio shield</a>.
-		SGTL5000 is always used in slave mode, where Teensy controls
-		all I2S timing.
-	</p>
-	<p align=center><img src="img/sgtl5000closeup.jpg"></p>
-	<h3>Audio Connections</h3>
-	<p>This object has no audio inputs or outputs.  Separate i2s objects
-		are used to send and receive audio data.  I2S master mode objects
-		must be used, because this object configures the SGTL5000 in slave
-		mode, where it depends on Teensy to provide all I2S clocks.
-		This object controls
-		how the SGTL5000 will use those I2S audio streams.</p>
-
-	<h3>Functions</h3>
-	<p>These are the most commonly used SGTL5000 functions.</p>
-	<p class=func><span class=keyword>enable</span>();</p>
-	<p class=desc>Start the SGTL5000.  This function should be called first.
-	</p>
-	<p class=func><span class=keyword>volume</span>(level);</p>
-	<p class=desc>Set the headphone volume level.  Range is 0 to 1.0, but
-		0.8 corresponds to the maximum undistorted output for a full scale
-		signal.  Usually 0.5 is a comfortable listening level.  The line
-		level outputs are <em>not</em> changed by this function.
-	</p>
-	<p class=func><span class=keyword>inputSelect</span>(input);</p>
-	<p class=desc>Select which input to use: AUDIO_INPUT_LINEIN or AUDIO_INPUT_MIC.
-	</p>
-	<p class=func><span class=keyword>micGain</span>(dB);</p>
-	<p class=desc>When using the microphone input, set the amplifier gain.
-		The input number is in decibels, from 0 to 63.
-	</p>
-
-	<h3>Signal Levels</h3>
-
-	<p>The default signal levels should be used for most applications,
-		but these functions allow you to customize the analog signals.</p>
-
-	<p class=func><span class=keyword>muteHeadphone</span>();</p>
-	<p class=desc>Silence the headphone output.
-	</p>
-	<p class=func><span class=keyword>unmuteHeadphone</span>();</p>
-	<p class=desc>Turn the headphone output on.
-	</p>
-	<p class=func><span class=keyword>muteLineout</span>();</p>
-	<p class=desc>Silence the line level outputs.
-	</p>
-	<p class=func><span class=keyword>unmuteLineout</span>();</p>
-	<p class=desc>Turn the line level outputs on.
-	</p>
-	<p class=func><span class=keyword>lineInLevel</span>(both);</p>
-	<p class=desc style="padding-bottom:0.2em;">Adjust the sensitivity of the line-level inputs.
-		Fifteen settings are possible:
-	</p>
-<pre class="desc">
- 0: 3.12 Volts p-p
- 1: 2.63 Volts p-p
- 2: 2.22 Volts p-p
- 3: 1.87 Volts p-p
- 4: 1.58 Volts p-p
- 5: 1.33 Volts p-p  (default)
- 6: 1.11 Volts p-p
- 7: 0.94 Volts p-p
- 8: 0.79 Volts p-p
- 9: 0.67 Volts p-p
-10: 0.56 Volts p-p
-11: 0.48 Volts p-p
-12: 0.40 Volts p-p
-13: 0.34 Volts p-p
-14: 0.29 Volts p-p
-15: 0.24 Volts p-p
-</pre>
-	<p class=func><span class=keyword>lineInLevel</span>(left, right);</p>
-	<p class=desc>Adjust the sensitivity of the line-level inputs, with different
-		settings for left and right.  The same 15 settings are available.
-	</p>
-	<p class=func><span class=keyword>lineOutLevel</span>(both);</p>
-	<p class=desc style="padding-bottom:0.2em;">Adjust the line level output
-		voltage range.  The following settings are possible:
-	</p>
-<pre class="desc">
-13: 3.16 Volts p-p
-14: 2.98 Volts p-p
-15: 2.83 Volts p-p
-16: 2.67 Volts p-p
-17: 2.53 Volts p-p
-18: 2.39 Volts p-p
-19: 2.26 Volts p-p
-20: 2.14 Volts p-p
-21: 2.02 Volts p-p
-22: 1.91 Volts p-p
-23: 1.80 Volts p-p
-24: 1.71 Volts p-p
-25: 1.62 Volts p-p
-26: 1.53 Volts p-p
-27: 1.44 Volts p-p
-28: 1.37 Volts p-p
-29: 1.29 Volts p-p  (default)
-30: 1.22 Volts p-p
-31: 1.16 Volts p-p
-</pre>
-	<p class=func><span class=keyword>lineOutLevel</span>(left, right);</p>
-	<p class=desc>Adjust the line level outout voltage range, with separate
-		settings for left and right.  The same settings (13 to 31) are available.
-	</p>
-
-
-	<h3>Signal Conditioning</h3>
-
-	<p>Usually these digital signal conditioning features should be left at their
-		default settings.
-	</p>
-
-	<p class=func><span class=keyword>adcHighPassFilterFreeze</span>();</p>
-	<p class=desc>By default, the analog input (either line-level inputs or mic)
-		is high-pass filtered, to remove any DC component.  This function
-		freezes the filter, so the current DC component is still substracted, but
-		the filter stops tracking any DC or low frequency changes.
-	</p>
-	<p class=func><span class=keyword>adcHighPassFilterDisable</span>();</p>
-	<p class=desc>Completely disable the analog input filter.  DC and sub-audible
-		low frequencies are allowed to enter the digital signal.
-	</p>
-	<p class=func><span class=keyword>adcHighPassFilterEnable</span>();</p>
-	<p class=desc>Turn the DC-blocking filter back on, if disabled, or
-		allows it to resume tracking DC and low frequency changes, if
-		previously frozen. This is the default setting.
-	</p>
-	<p class=func><span class=keyword>dacVolume</span>(both);</p>
-	<p class=desc>Normally output volume should be used with volume(), which
-		changes the analog gain in the headphone amplifier.  This function
-		on the other hand controls digital attenuation before conversion to analog, which
-		reduces resolution, but allows another fine control of output
-		signal level.  The ranges is 0 to 1.0, with the default (no digital attenuation)
-		at 1.0.
-	</p>
-	<p  class=desc>dacVolume uses zero-crossing detect to avoid clicks, and ramping is handled by
-	the chip so that a new volume may be set directly in a single call.
-	</p>
-	<p class=func><span class=keyword>dacVolume</span>(left, right);</p>
-	<p class=desc>Adjust the digital output volume separately on left and
-		right channels.
-	</p>
-
-
-	<h3>Audio Processor</h3>
-
-	<p>The optional digital audio processor is capable of implementing
-		one or more of: automatic volume control, surround sound control,
-		bass enhancement, and tonal adjustments (either a
-		simple tone control, or a parametric equalizer, or a graphic equalizer),
-		in that order.
-	</p>
-	<p>These signal processing features are implemented in the SGTL5000 chip,
-		so they do not consume CPU time on Teensy. However, the order of
-		these processes is fixed in the hardware.
-	</p>
-	<p>It is good practice to mute the outputs before enabling or disabling
-	the Audio Processor, to avoid clicks or thumps.
-	</p>
-
-	<p class=func><span class=keyword>audioPreProcessorEnable</span>();</p>
-	<p class=desc>Enable the audio processor to pre-process the input
-		(from either line-level inputs or microphone) before it's sent
-		to Teensy by I2S.
-	</p>
-	<p class=func><span class=keyword>audioPostProcessorEnable</span>();</p>
-	<p class=desc>Enable the audio processor to post-process Teensy's
-		I2S output before it's turned into analog signals for the
-		headphones and/or line level outputs.
-	</p>
-	<p class=func><span class=keyword>audioProcessorDisable</span>();</p>
-	<p class=desc>Disable the audio processor.
-	</p>
-	<p class=func><span class=keyword>autoVolumeControl</span>(maxGain, response, hardLimit, threshold, attack, decay);</p>
-	<p class=desc>Configures the auto volume control, which is implemented as a compressor/expander
-	or hard limiter. 	<em>maxGain</em> is the maximum gain that can be applied for expanding, and
-	can take one of three values: 0 (0dB), 1 (6.0dB) and 2 (12dB). Values greater than 2 are treated
-	as 2. <em>response</em> controls the integration time for the compressor and can take
-	four values: 0 (0ms), 1 (25ms), 2 (50ms) or 3 (100ms). Larger values average the volume
-	over a longer time, allowing short-term peaks through.
-	</p>
-	<p class=desc>If <em>hardLimit</em> is 0, a 'soft
-	knee' compressor is used to progressively compress louder values which are near to or above the
-	threashold (the louder they are, the greater the compression). If it is 1, a hard compressor
-	is used (all values above the threashold are the same loudness). The <em>threashold</em> is specified
-	as a float in the range 0dBFS to -96dBFS, where -18dBFS is a typical value.
-	<em>attack</em> is a float controlling the rate of decrease in gain when the signal is over
-	threashold, in dB/s. <em>decay</em> controls how fast gain is restored once the level
-	drops below threashold, again in dB/s. It is typically set to a longer value than attack.
-	</p>
-	<p class=func><span class=keyword>autoVolumeEnable</span>();</p>
-	<p class=desc>Enables auto volume control, using the previously specified settings.
-	</p>
-	<p class=func><span class=keyword>autoVolumeDisable</span>();</p>
-	<p class=desc>Disables auto volume control.
-	</p>
-
-	<p class=func><span class=keyword>surroundSoundEnable</span>();</p>
-	<p class=desc>Enable virtual surround processing, to give a broader and
-	deeper stereo image (even with mono input).
-	</p>
-	<p class=func><span class=keyword>surroundSoundDisable</span>();</p>
-	<p class=desc>Disable virtual surround processing. Before disabling, ramp up
-	the width to maximum to avoid pops.
-	</p>
-	<p class=func><span class=keyword>surroundSound</span>(width);</p>
-	<p class=desc>Configures virtual surround width from 0 (mono) to 7 (widest).
-	</p>
-	<p class=func><span class=keyword>surroundSound</span>(width, select);</p>
-	<p class=desc>Configures virtual surround width from 0 (mono) to 7 (widest).
-	<em>select</em> may be set to 1 (disable), 2 (mono input) or 3 (stereo input).
-	</p>
-
-	<p class=func><span class=keyword>enhanceBassEnable</span>();</p>
-	<p class=desc>Enable bass enhancement. A mono, low-pass filtered copy of
-	the original stereo signal has bass levels boosted and is then mixed back into
-	the stereo signal, which is then optionally high pass filtered (to remove
-	inaudible subsonic frequencies).
-	</p>
-	<p class=func><span class=keyword>enhanceBassDisable</span>();</p>
-	<p class=desc>Disable bass enhancement. Before disabling, ramp down the bass
-	enhancement level to zero.
-	</p>
-	<p class=func><span class=keyword>enhanceBass</span>(lr_lev, bass_lev);</p>
-	<p class=desc>Configures the bass enhancement by setting the levels of the
-	original stereo signal and the bass-enhanced mono level which will be mixed together.
-	There is no high-pass filter.
-	</p>
-	<p  class=desc>When changing bass level, call this function repeatedly to ramp up or down the bass in
-	steps of 0.5dB, to avoid pops.
-	</p>
-	<p class=func><span class=keyword>enhanceBass</span>(lr_lev, bass_lev, hpf_bypass, cutoff);</p>
-	<p class=desc>Configures the bass enhancement by setting the levels of the
-	original stereo signal and the bass-enhanced mono level which will be mixed together.
-	The high-pass filter may be enabled (0) or bypassed (1). The cutoff frequency is specified
-	as follows:
-	</p>
-	<pre class="desc">
-value  frequency
- 0      80Hz
- 1     100Hz
- 2     125Hz
- 3     150Hz
- 4     175Hz
- 5     200Hz
- 6     225Hz
-</pre>
-	<p  class=desc>When changing bass level, call this function repeatedly to ramp up or down the bass in
-	steps of 0.5dB, to avoid pops.
-	</p>
-
-	<p class=func><span class=keyword>eqSelect</span>(n);</p>
-	<p class=desc>Selects the type of frequency control, where <em>n</em> is
-	one of</p>
-	<p class=desc><b>FLAT_FREQUENCY (0)</b><br>
-    Equalizers and tone controls disabled, flat frequency response.</p>
-    <p class=desc><b>PARAMETRIC_EQUALIZER (1)</b><br>
-    Enables the 7-band parametric equalizer, thus disabling the
-	tone controls and graphic equalizer.</p>
-    <p class=desc><b>TONE_CONTROLS (2)</b><br>
-    Enables bass and treble tone controls, disabling the parametric
-	equalization and graphic equalizer.</p>
-    <p class=desc><b>GRAPHIC_EQUALIZER (3)</b><br>
-    Enables the five-band graphic equalizer, disabling the parametric
-	equalization and tone controls.</p>
-
-
-	<p class=func><span class=keyword>eqBands</span>(bass, treble);</p>
-	<p class=desc>Configures bass and treble tone controls, which are
-	implemented as one second order low pass filter (bass) in parallel with
-	one second order high pass filter (treble).
-	</p>
-	<p class=desc>When changing bass or treble level, call this function repeatedly to ramp
-	up or down the level in steps of 0.04 (=0.5dB) or so, to avoid pops.
-	</p>
-	<p class=func><span class=keyword>eqBands</span>(bass, mid_bass, midrange, mid_treble, treble);</p>
-	<p class=desc>Configures the graphic equalizer. It is implemented by five parallel,
-	second order biquad filters with fixed frequencies of 115Hz, 330Hz, 990Hz, 3kHz,
-	and 9.9kHz. Each band has a range of adjustment from 1.00 (+12dB) to -1.00 (-11.75dB).
-	</p>
-	<p class=func><span class=keyword>eqBand</span>(bandNum, n);</p>
-	<p class=desc>Configures the gain or cut on one band in the graphic equalizer.
-	<em>bandnum</em> can range from 1 to 5; <em>n</em> is a float in the range 1.00 to -1.00.
-	</p>
-	<p  class=desc>When changing a band, call this function repeatedly to ramp up the gain in steps of 0.5dB,
-	to avoid pops.
-	</p>
-
-	<p class=func><span class=keyword>eqFilter</span>(filterNum, filterParameters);</p>
-	<p class=desc>Configurs the parametric equalizer. The number of filters (1 to 7)
-	is specified along with  a pointer to an array of filter coefficients.
-	The parametric equalizer is implemented using 7 cascaded, second order bi-quad
-	filters whose frequencies, gain, and Q may be freely configured, but each filter
-	can only be specified as a set of filter coefficients.
-	</p>
-	<p class=func><span class=keyword>eqFilterCount</span>(n);</p>
-	<p class=desc>Enables zero or more of the already enabled parametric filters.
-	</p>
-
-	<h3>Examples</h3>
-	<p>Nearly all of the library's examples use this object.  These
-		examples demonstrate its special features.
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; PassThroughStereo
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; SGTL5000 &gt; dap_bass_enhance
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; SGTL5000 &gt; dap_avc_agc
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; SGTL5000 &gt; balanceDAC
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; SGTL5000 &gt; balanceHP
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; SGTL5000 &gt; CalcBiquadToneControlDAP
-	</p>
-	<h3>Notes</h3>
-	<p>TODO: add example with rock/classical/speech presets, where rock uses bass boost
-	and surround enhancement while speech uses bandpass filtering and auto volume control
-	compression.
-	</p>
-	<p>TODO: add example with two analogRead pots for bass and treble to demonstrate ramping.
-	</p>

gui/resources/help/AudioControlWM8731.html

-
-	<h3>Summary</h3>
-	<p>Control a WM8731 chip in slave mode, where it receives all clocks from Teensy</p>
-	<h3>Audio Connections</h3>
-	<p>This object has no audio inputs or outputs.  Separate i2s objects
-		are used to send and receive audio data.  I2S master mode objects
-		must be used, since this control object configures the WM8731 into
-		slave mode.
-	</p>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>enable</span>();</p>
-	<p class=desc>blah blah blah blah
-	</p>
-	<p class=func><span class=keyword>disable</span>();</p>
-	<p class=desc>not implemented
-	</p>
-	<p class=func><span class=keyword>volume</span>(level);</p>
-	<p class=desc>blah blah blah blah
-	</p>
-	<p class=func><span class=keyword>inputLevel</span>(level);</p>
-	<p class=desc>not implemented
-	</p>
-	<p class=func><span class=keyword>inputSelect</span>(input);</p>
-	<p class=desc>not implemented
-	</p>
-	<!--<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt;
-	</p>-->
-	<h3>Notes</h3>
-	<p></p>

gui/resources/help/AudioControlWM8731master.html

-
-	<h3>Summary</h3>
-	<p>Control a WM8731 chip in master mode, where it controls all I2S timing.</p>
-	<h3>Audio Connections</h3>
-	<p>This object has no audio inputs or outputs.  Separate i2s objects
-		are used to send and receive audio data.  I2S slave mode objects
-		must be used, since this control object configures the WM8731 into
-		master mode.
-	</p>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>enable</span>();</p>
-	<p class=desc>blah blah blah blah
-	</p>
-	<p class=func><span class=keyword>disable</span>();</p>
-	<p class=desc>not implemented
-	</p>
-	<p class=func><span class=keyword>volume</span>(level);</p>
-	<p class=desc>blah blah blah blah
-	</p>
-	<p class=func><span class=keyword>inputLevel</span>(level);</p>
-	<p class=desc>not implemented
-	</p>
-	<p class=func><span class=keyword>inputSelect</span>(input);</p>
-	<p class=desc>not implemented
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; WM8731MikroSine
-	</p>
-	<h3>Notes</h3>
-	<p></p>

gui/resources/help/AudioEffectBitcrusher.html

-
-    <h3>Summary</h3>
-    <p>Reduce the samplerate and/or bitdepth of a source signal, resulting in
-		a distorted sound.</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 Input</td></tr>
-        <tr class=odd><td align=center>Out 0</td><td>Signal Output</td></tr>
-    </table>
-    <h3>Parameters</h3>
-    <p class=func><span class=keyword>bits</span>(xcrushBits);</p>
-    <p class=desc>xcrushBits sets the bitdepth, from 1 to 16. A Value of 16
-    does not crush the bitdepth, and is effectively a passthru for this part
-    of the function.</p>
-
-    <p class=func><span class=keyword>sampleRate</span>(xsampleRate);</p>
-
-    <p class=desc>xsampleRate sets the frequency, from 1 to 44100Hz, however it
-    works in integer steps so you will only really get a handful of results from
-    the many samplerates you can pass. 44100 is passthru.</p>
-
-    <p class=desc>set xbitDepth to 16 and xsampleRate to 44100 to pass audio
-    	through without any Bitcrush effect.</p>
-    <h3>Examples</h3>
-    <p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Bitcrusher
-    </p>
-    <h3>Notes</h3>
-    <p>Needs a lot of improvement. Options for anti-aliasing would be nice in
-    the future, but for now, it's rough, it's dirty and it sounds a bit like
-    Nine Inch Nails.
-    </p>
-    <p><a href="http://www.pjrc.com/teensy/td_libs_AudioProcessorUsage.html" target="_blank">AudioNoInterrupts()</a>
-        should be used when changing
-        settings on multiple objects, so all changes always take effect
-        at the same moment.
-    </p>

gui/resources/help/AudioEffectChorus.html

-
-<h3>Summary</h3>
-	<p>The chorus effect simulates the richness of several nearly-identical
-	sound sources (like the way a choir sounds different to a single singer).
-	It does this by sampling from a delay line, so each voice is actually
-	the same but at a slightly different point in time. This is a type of
-	comb filtering.</p>
-	<p>Chorus combines one or more samples ranging from the most recent
-	sample back to about 50ms ago. The additional samples are evenly spread
-	through the supplied delay line, and there is no modulation.</p>
-	<p>If the number of voices is specified as 2, then the
-	effect combines the current sample and the oldest sample (the last one
-	in the delay line). If the number of voices is 3 then the effect combines
-	the most recent sample, the oldest sample and the sample in the middle of
-	the delay line.</p>
-	<p>For two voices the effect can be represented as:<br/>
-	result = (sample(0) + sample(dt))/2<br/>
-	where sample(0) represents the current sample and sample(dt)
-	is the sample in the delay line from dt milliseconds ago.</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 Input</td></tr>
-		<tr class="odd"><td align="center">Out 0</td><td>Chorused Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>begin</span>(delayBuffer, length, n_chorus);</p>
-	<p class=desc>Create a chorus by specifying the address of the delayline, the
-	total number of samples in the delay line (often done as an integer multiple of
-	AUDIO_BLOCK_SAMPLES) and the number of voices in the chorus <em>including</em>
-	the original voice (so, 2 and up to get a chorus effect, although you can
-	specify 1 if you want).
-	</p>
-	<p class=func><span class=keyword>modify</span>(n_chorus);</p>
-	<p class=desc>Alters the number of voices in a running chorus (previously started with begin).
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Chorus
-	</p>
-	<h3>Notes</h3>
-	<p>The longer the length of the chorus, the more memory blocks are used.</p>

gui/resources/help/AudioEffectDelay.html

-
-	<h3>Summary</h3>
-	<div>
-	<p>Delay a signal.  Up to 8 separate delay taps can be used.</p>
-	<p align=center><img src="img/delay.png"><br><small>1 kHz burst, delayed 5.2 ms.</small></p>
-	</div>
-	<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 Input</td></tr>
-		<tr class=odd><td align=center>Out 0</td><td>Delay Tap #1</td></tr>
-		<tr class=odd><td align=center>Out 1</td><td>Delay Tap #2</td></tr>
-		<tr class=odd><td align=center>Out 2</td><td>Delay Tap #3</td></tr>
-		<tr class=odd><td align=center>Out 3</td><td>Delay Tap #4</td></tr>
-		<tr class=odd><td align=center>Out 4</td><td>Delay Tap #5</td></tr>
-		<tr class=odd><td align=center>Out 5</td><td>Delay Tap #6</td></tr>
-		<tr class=odd><td align=center>Out 6</td><td>Delay Tap #7</td></tr>
-		<tr class=odd><td align=center>Out 7</td><td>Delay Tap #8</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>delay</span>(channel, milliseconds);</p>
-	<p class=desc>Set output channel (0 to 7) to delay the signals by
-		milliseconds.  The maximum delay is approx 333 ms.  The actual delay
-		is rounded to the nearest sample.  Each channel can be configured for
-		any delay.  There is no requirement to configure the "taps" in increasing
-		delay order.
-	</p>
-	<p class=func><span class=keyword>disable</span>(channel);</p>
-	<p class=desc>Disable a channel.  The output of this channel becomes
-		silent.  If this channel is the longest delay, memory usage is
-		automatically reduced to accomodate only the remaining channels used.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Delay
-	</p>
-	<h3>Notes</h3>
-	<p>Memory for the delayed signal is take from the memory pool allocated by
-		<a href="http://www.pjrc.com/teensy/td_libs_AudioConnection.html" target="_blank">AudioMemory()</a>.
-		Each block allows about 3 milliseconds of delay, so AudioMemory
-		should be increased to allow for the longest delay tap.
-	</p>

gui/resources/help/AudioEffectDelayExternal.html

-
-	<h3>Summary</h3>
-	<div>
-	<p>Delay a signal, using external memory for longer delay times!  Up to 8 separate delay taps can be used.</p>
-	<p align=center><img src="img/delay.png"><br><small>1 kHz burst, delayed 5.2 ms.</small></p>
-	</div>
-	<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 Input</td></tr>
-		<tr class=odd><td align=center>Out 0</td><td>Delay Tap #1</td></tr>
-		<tr class=odd><td align=center>Out 1</td><td>Delay Tap #2</td></tr>
-		<tr class=odd><td align=center>Out 2</td><td>Delay Tap #3</td></tr>
-		<tr class=odd><td align=center>Out 3</td><td>Delay Tap #4</td></tr>
-		<tr class=odd><td align=center>Out 4</td><td>Delay Tap #5</td></tr>
-		<tr class=odd><td align=center>Out 5</td><td>Delay Tap #6</td></tr>
-		<tr class=odd><td align=center>Out 6</td><td>Delay Tap #7</td></tr>
-		<tr class=odd><td align=center>Out 7</td><td>Delay Tap #8</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>delay</span>(channel, milliseconds);</p>
-	<p class=desc>Set output channel (0 to 7) to delay the signals by
-		milliseconds.  The maximum delay is approx 333 ms.  The actual delay
-		is rounded to the nearest sample.  Each channel can be configured for
-		any delay.  There is no requirement to configure the "taps" in increasing
-		delay order.
-	</p>
-	<p class=func><span class=keyword>disable</span>(channel);</p>
-	<p class=desc>Disable a channel.  The output of this channel becomes
-		silent.  If this channel is the longest delay, memory usage is
-		automatically reduced to accomodate only the remaining channels used.
-	</p>
-	<h3>Hardware</h3>
-	<p>By default, or when <span class=literal>AUDIO_MEMORY_23LC1024</span> is used (see below),
-		 a single 23LC1024 RAM chip is used, with these pins:
-    <table class=doc align=center cellpadding=3>
-        <tr class=top><th>Pin</th><th>Signal</th></tr>
-        <tr class=odd><td align=center>6</td><td>CS</td></tr>
-        <tr class=odd><td align=center>7</td><td>MOSI</td></tr>
-        <tr class=odd><td align=center>12</td><td>MISO</td></tr>
-        <tr class=odd><td align=center>14</td><td>SCK</td></tr>
-    </table>
-	</p>
-	<p>When <span class=literal>AUDIO_MEMORY_MEMORYBOARD</span> is used, up to six
-		23LC1024 chips are used.
-	</p>
-	<p align=center><img src="img/memoryboard.jpg"><br><small><a href="https://oshpark.com/shared_projects/KZt5PaU7" target="_blank">Memoryboard 4</a></small></p>
-	<p>
-    <table class=doc align=center cellpadding=3>
-        <tr class=top><th>Pin</th><th>Signal</th></tr>
-        <tr class=odd><td align=center>2</td><td>CS0 (encoded)</td></tr>
-        <tr class=odd><td align=center>3</td><td>CS1 (encoded)</td></tr>
-        <tr class=odd><td align=center>4</td><td>CS2 (encoded)</td></tr>
-        <tr class=odd><td align=center>7</td><td>MOSI</td></tr>
-        <tr class=odd><td align=center>12</td><td>MISO</td></tr>
-        <tr class=odd><td align=center>14</td><td>SCK</td></tr>
-    </table>
-	</p>
-	<p>
-	If fewer than 6 chips are soldered, the optional parameter for maximum delay
-	must be used.  See below for details.  Each chip provides 1485 ms of delay
-	memory, so the total of all objects using AUDIO_MEMORY_MEMORYBOARD must not
-	exceed the amount of memory physically present.
-	</p>
-	<h3>Examples</h3>
-	<p>
-	<a href="https://www.youtube.com/watch?v=d80d1HWy5_s" target="_blank">Demo Video</a> (YouTube)
-	</p>
-	<!-- <p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Delay
-	</p> -->
-	<p>
-	<a href="https://forum.pjrc.com/threads/29276-Limits-of-delay-effect-in-audio-library?p=79436&viewfull=1#post79436" target="_blank">Forum Conversaton</a> (with sample code)
-	</p>
-	<h3>Notes</h3>
-	<p>External RAM allows for longer delays without consuming
-		limited internal RAM.  However, SPI communication is required,
-		which consumes much more CPU time.  The
-		<a href="http://www.pjrc.com/teensy/td_libs_AudioProcessorUsage.html">AudioProcessorUsageMax</a>
-		function may be used to monitor how much CPU time is consumed.
-	</p>
-	<p>You may specify the type of hardware to be used by editing the code.  AUDIO_MEMORY_23LC1024
-		specifies a single 23LC1024 chip.  AUDIO_MEMORY_MEMORYBOARD allows using up to 6 of these
-		chips.
-	</p>
-	<p class=desc><span class=keyword>AudioEffectDelayExternal</span>  delayExt1(<span class=literal>AUDIO_MEMORY_23LC1024</span>);
-	</p>
-	<p>You may also create more than one delay using the same hardware, where the memory is partitioned
-		by specifying a maximum delay in milliseconds.  This can be useful if you wish to delay both
-		channels of a stereo signal.
-
-	<p class=desc><span class=keyword>AudioEffectDelayExternal</span>  delayExt1(<span class=literal>AUDIO_MEMORY_23LC1024</span>, 700);<br><span class=keyword>AudioEffectDelayExternal</span>  delayExt2(<span class=literal>AUDIO_MEMORY_23LC1024</span>, 700);
-	</p>

gui/resources/help/AudioEffectEnvelope.html

-
-	<h3>Summary</h3>
-	<div>
-	<p>Modify a signal with a DAHDSR (Delay Attack Hold Decay Sustain
-		Release) envelope.
-	</p>
-	<p align=center><img src="img/dahdsr.png"></p>
-	</div>
-	<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 Input</td></tr>
-		<tr class=odd><td align=center>Out 0</td><td>Signal with Envelope Applied</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>noteOn</span>();</p>
-	<p class=desc>Begin the delay to attack, or the attack phase is
-		delay is zero.
-	</p>
-	<p class=func><span class=keyword>noteOff</span>();</p>
-	<p class=desc>Begin the release phase.
-	</p>
-	<p class=func><span class=keyword>delay</span>(milliseconds);</p>
-	<p class=desc>Set the delay from noteOn to the attach phase.  The
-		default is zero, for no delay.
-	</p>
-	<p class=func><span class=keyword>attack</span>(milliseconds);</p>
-	<p class=desc>Set the attack time.  The default is 1.5 milliseconds.
-	</p>
-	<p class=func><span class=keyword>hold</span>(milliseconds);</p>
-	<p class=desc>Set the hold time.  The default is 0.5 milliseconds.
-	</p>
-	<p class=func><span class=keyword>decay</span>(milliseconds);</p>
-	<p class=desc>Set the decay time.  The default is 15 milliseconds.
-	</p>
-	<p class=func><span class=keyword>sustain</span>(level);</p>
-	<p class=desc>Set the sustain level.  The range is 0 to 1.0.  The
-		gain will be maintained at this level after the decay phase,
-		until noteOff() is called.
-	</p>
-	<p class=func><span class=keyword>release</span>(milliseconds);</p>
-	<p class=desc>Set the release time.  The default is 30 millisecond.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Synthesis &gt; PlaySynthMusic
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Synthesis &gt; pulseWidth
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; MemoryAndCpuUsage
-	</p>
-	<h3>Notes</h3>
-	<p>To achieve the more common ADSR shape, simply
-        set delay and hold to zero.</p>
-	<p>The recommended range for each of the 5 timing inputs is 0 to 50
-		milliseconds.  Up to 200 ms can be used, with somewhat reduced
-		accuracy</p>

gui/resources/help/AudioEffectFade.html

-
-	<h3>Summary</h3>
-	<p>Gradually increase or decrease audio level.</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 Input</td></tr>
-		<tr class=odd><td align=center>Out 0</td><td>Signal Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>fadeIn</span>(milliseconds);</p>
-	<p class=desc>Begin increasing the audio level, to reach 1.0 (input passed
-		directly to the output) after "milliseconds" time.
-	</p>
-	<p class=func><span class=keyword>fadeOut</span>(milliseconds);</p>
-	<p class=desc>Begin decreasing the audio level, to reach 0 (no output)
-		after "milliseconds" time.
-	</p>
-	<!--<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; 
-	</p>-->
-	<h3>Notes</h3>
-	<p>Cross fading can be built with 2 fade objects fed into a mixer.
-		When one fade object is off (fully faded out) and the other on
-		(fully faded in), if both are started at the same moment for the
-		same time duration, their signal gains always add to 1.0.  This
-		allows 2 fade objects to work together for a smooth transition
-		between a pair of signals.
-	</p>
-	<p><a href="http://www.pjrc.com/teensy/td_libs_AudioProcessorUsage.html" target="_blank">AudioNoInterrupts()</a>
-		should be used when changing
-		settings on multiple objects, so all changes always take effect
-		at the same moment.
-	</p>

gui/resources/help/AudioEffectFlange.html

-
-<h3>Summary</h3>
-	<p>Originally, flanging was produced by playing the same signal on two synchronized
-	reel-to-reel tape recorders and making one of the reels slow down and speed up by
-	pressing on the flange of the reel (hence the name). This is a type of
-	comb filtering, and produces a harmonically-related series of peaks and notches
-	in the audio spectrum.</p>
-	<p>This flanger uses a delay line, combining the original voice with only one sample from the delay
-	line, but the position of that sample varies sinusoidally.</p>
-	<p>The effect can be represented as:<br>
-	result = sample(0) + sample(dt + depth*sin(2*PI*Fe))</p>
-	<p>The value of the sine function is always a number from -1 to +1 and
-	so the result of depth*(sin(Fe)) is always a number from -depth to +depth.
-	Thus, the delayed sample will be selected from the range (dt-depth) to
-	(dt+depth). This selection will vary at whatever rate is specified as the
-	frequency of the effect, Fe. Typically a low frequency (a few Hertz) is used.
-	<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 Input</td></tr>
-		<tr class="odd"><td align="center">Out 0</td><td>Flanged Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>begin</span>(delayBuffer, length, offset, depth, delayRate);</p>
-	<p class=desc>Create a flanger by specifying the address of the delayline, the
-	total number of samples in the delay line (often done as an integer multiple of
-	AUDIO_BLOCK_SAMPLES), the offset (how far back the flanged sample is from the original voice),
-	the modulation depth (larger values give a greater variation) and the modulation
-	frequency, in Hertz.
-	</p>
-	<p class=func><span class=keyword>modify</span>(offset, depth, delayRate);</p>
-	<p class=desc>Alters the parameters in a running flanger (previously started with begin).
-	</p>
-
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Flange
-	</p>
-	<h3>Notes</h3>
-	<p>The longer the length of the delay buffer, the more memory blocks are used.</p>
-	<p>Try these settings:<br>
-#define FLANGE_DELAY_LENGTH (2*AUDIO_BLOCK_SAMPLES)<br>
-and<br>
-int s_idx = 2*FLANGE_DELAY_LENGTH/4;<br>
-int s_depth = FLANGE_DELAY_LENGTH/4;<br>
-double s_freq = 3;</p>
-<p>The flange effect can also produce a chorus-like effect if a longer
-delay line is used with a slower modulation rate, for example try:<br>
-#define FLANGE_DELAY_LENGTH (12*AUDIO_BLOCK_SAMPLES)<br>
-and<br>
-int s_idx = 3*FLANGE_DELAY_LENGTH/4;<br>
-int s_depth = FLANGE_DELAY_LENGTH/8;<br>
-double s_freq = .0625;</p>

gui/resources/help/AudioEffectMultiply.html

-
-	<h3>Summary</h3>
-	<div>
-	<p>Multiply two signals together, useful for amplitude modulation
-		or "voltage controlled amplification".
-	</p>
-	<p align=center><img src="img/multiply.png"><br><small>56 Hz and 1 kHz sine waves multiplied.</small></p>
-	</div>
-	<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 Input</td></tr>
-		<tr class=odd><td align=center>In 1</td><td>Signal Input</td></tr>
-		<tr class=odd><td align=center>Out 0</td><td>Signal with Envelope Applied</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p>There are no functions to call from the Arduino sketch.
-		This object simply multiplies the 2 signals to create
-		a continuous output
-	</p>
-	<!--<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; 
-	</p>-->
-	<h3>Notes</h3>
-	<p>
-    </p>

gui/resources/help/AudioFilterBiquad.html

-
-	<h3>Summary</h3>
-	<div>
-	<p>Biquadratic cascaded filter, useful for all sorts of filtering.
-		Up to 4 stages may be cascaded.
-	</p>
-	<p align=center><img src="img/biquad.png"></p>
-	</div>
-	<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 be filtered</td></tr>
-		<tr class=odd><td align=center>Out 0</td><td>Filtered Signal Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>setLowpass</span>(stage, frequency, Q);</p>
-	<p class=desc>Configure one stage of the filter (0 to 3) with low pass
-		response, with the specified corner frequency and Q shape.  If Q is
-		higher that 0.7071, be careful of filter gain (see below).
-	</p>
-	<p class=func><span class=keyword>setHighpass</span>(stage, frequency, Q);</p>
-	<p class=desc>Configure one stage of the filter (0 to 3) with high pass
-		response, with the specified corner frequency and Q shape.  If Q is
-		higher that 0.7071, be careful of filter gain (see below).
-	</p>
-	<p class=func><span class=keyword>setBandpass</span>(stage, frequency, Q);</p>
-	<p class=desc>Configure one stage of the filter (0 to 3) with band pass
-		response.  The filter has unity gain at the specified frequency.  Q
-		controls the width of frequencies allowed to pass.
-	</p>
-	<p class=func><span class=keyword>setNotch</span>(stage, frequency, Q);</p>
-	<p class=desc>Configure one stage of the filter (0 to 3) with band reject (notch)
-		response.  Q controls the width of rejected frequencies.
-	</p>
-	<p class=func><span class=keyword>setCoefficients</span>(stage, array[5]);</p>
-	<p class=desc>Configure one stage of the filter (0 to 3) with an arbitrary
-		filter response.  The array of coefficients is in order: B0, B1, B2, A1, A2.
-		Each coefficient must be less than 2.0 and greater than -2.0.  The array
-		should be type double.  Alternately, it may be type int, where 1.0 is
-		represented with 1073741824 (2<sup>30</sup>).
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Filter
-	</p>
-	<h3>Notes</h3>
-	<p>Filters can with gain must have their input signals attenuated, so the
-		signal does not exceed 1.0.
-	</p>
-	<p>This object implements up to 4 cascaded stages.  Unconfigured stages will
-		not pass any signal.
-	</p>
-	<p>Biquad filters with low corner frequency (under about 400 Hz) can run into
-		trouble with limited numerical precision, causing the filter to perform
-		poorly.  For very low corner frequency, the State Variable (Chamberlin)
-		filter should be used.
-	</p>

gui/resources/help/AudioFilterFIR.html

-
-	<h3>Summary</h3>
-	<div>
-	<p>Finite impulse response filter, useful for all sorts of filtering.
-	</p>
-	<p align=center><img src="img/fir_filter.png"></p>
-	</div>
-	<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 be filtered</td></tr>
-		<tr class=odd><td align=center>Out 0</td><td>Filtered Signal Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>begin</span>(array, length);</p>
-	<p class=desc>Initialize the filter.  The array must be 16 bit integers (the
-		filter's impulse response), and
-		length indicates the number of points in the array.  Array may also be
-		FIR_PASSTHRU (length = 0), to directly pass the input to output without
-		filtering.
-	</p>
-	<p class=func><span class=keyword>end</span>();</p>
-	<p class=desc>Turn the filter off.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Filter_FIR
-	</p>
-	<h3>Notes</h3>
-	<p>FIR filters requires more CPU time than Biquad (IIR), but they can
-		implement filters with better phase response.
-	</p>
-	<p>A 100 point filter requires 9% CPU time on Teensy 3.1.  The maximum
-		supported filter length is 200 points.
-	</p>
-	<p>The free
-		<a href="http://t-filter.appspot.com/fir/index.html" target="_blank"> TFilter Design Tool</a>
-		can be used to create the impulse response array.  Be sure to set the sampling
-		frequency to 44117 HZ (it defaults to only 2000 Hz) and the output type to "int" (16 bit).
-	</p>
-	<p>
-		If you use TFilter Design's "C/C++ array" option, it's output has "int" definition, which
-		is 32 bits on Teensy 3.1.  Edit "int" to "short" for an array of 16 bit numbers,
-		and add "const" to avoid consuming extra RAM.
-	</p>

gui/resources/help/AudioFilterStateVariable.html

-
-	<h3>Summary</h3>
-	<p>A State Variable (Chamberlin) Filter with 12 dB/octave roll-off,
-		adjustable resonance, and optional signal control of corner
-		frequency.</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 Filter</td></tr>
-		<tr class=odd><td align=center>In 1</td><td>Frequency Control</td></tr>
-		<tr class=odd><td align=center>Out 0</td><td>Low Pass Output</td></tr>
-		<tr class=odd><td align=center>Out 1</td><td>Band Pass Output</td></tr>
-		<tr class=odd><td align=center>Out 2</td><td>High Pass Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>frequency</span>(freq);</p>
-	<p class=desc>Set the filter's corner frequency.  When a signal is
-		connected to the control input, the filter will implement this
-		frequency when the signal is zero.
-	</p>
-	<p class=func><span class=keyword>resonance</span>(Q);</p>
-	<p class=desc>Set the filter's resonance.  Q ranges from 0.7 to 5.0.
-		Resonance greater than 0.707 will amplify the signal near the
-		corner frequency.  You must attenuate the signal before input
-		to this filter, to prevent clipping.
-	</p>
-	<p class=func><span class=keyword>octaveControl</span>(octaves);</p>
-	<p class=desc>Set how much (in octaves) the control signal can alter
-		the filter's corner freqency.  Range is 0 to 7 octaves.  For
-		example, when set to 2.5, a full scale positive signal (1.0) will
-		shift the filter frequency up 2.5 octaves, and a full scale negative
-		signal will shift it down 2.5 octaves.
-	</p>
-	<!--<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; 
-	</p>-->
-	<h3>Notes</h3>
-	<p>
-		When controlled by a signal, the equation for the filter
-		frequency is:
-	</p>
-	<p>
-		F = Fcenter * 2^<sup>(signal * octaves)</sup>
-		<br><small>If anyone knows how to do HTML equations, please
-		help me improve this.....</small>
-	</p>
-	<p>When operating with signal control of corner frequency, this
-		object uses approximately 4% of the CPU time on Teensy 3.1.
-	</p>

gui/resources/help/AudioInputAnalog.html

-
-	<h3>Summary</h3>
-	<p>Receive audio using the built-in analog to digital converter.</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>Out 0</td><td>Audio Channel</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p>This object has no functions to call from the Arduino sketch.  It
-		simply streams data from the ADC to its output port.</p>
-	<h3>Hardware</h3>
-	<p>Pin A2 is used for audio input.  This circuitry is recommended.</p>
-	<p align=center><img src="img/adccircuit.png"></p>
-	<p>Signal range is 0 to 1.2V</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; PassThroughMono
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; PeakMeterMono
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; DialTone_7segment
-	</p>
-	<h3>Notes</h3>
-	<p>A different pin may be used, but adding it as an parameter
-		to the AudioInputAnalog object definition.
-	</p>
-	<p>For example, to use pin A3:
-	</p>
-	<p class=desc><span class=keyword>AudioInputAnalog</span>  adc1(<span class=literal>A3</span>);
-	</p>
-
-	<p>Noise due to high source impedance, which allows rapidly switching digital signals
-		to capacitively couple... avoiding higher analog impedance is the solution.</p>
-	<p>Power Supply rejection issue with simple DC bias (bigger capacitor may be needed if 3.3V has low frequency noise)</p>
-	<p>Algorithm for automatic DC bias tracking</p>
-	<p>TODO: actual noise measurements with different input circuitry
-		(it's not as quiet as the audio shield)</p>

gui/resources/help/AudioInputI2S.html

-
-	<h3>Summary</h3>
-	<p>Receive 16 bit stereo audio from the
-	<a href="http://www.pjrc.com/store/teensy3_audio.html" target="_blank">audio shield</a>
-		or another I2S device, using I2S master mode.</p>
-	<p align=center><img src="img/audioshield_inputs.jpg"></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>Out 0</td><td>Left Channel</td></tr>
-		<tr class=odd><td align=center>Out 1</td><td>Right Channel</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p>This object has no functions to call from the Arduino sketch.  It
-		simply streams data from the I2S hardware to its 2 output ports.</p>
-	<h3>Hardware</h3>
-	<p align=center><img src="img/audioshield_backside.jpg"></p>
-	<p>The I2S signals are used in "master" mode, where Teensy creates
-		all 3 clock signals and controls all data timing.</p>
-	<table class=doc align=center cellpadding=3>
-		<tr class=top><th>Pin</th><th>Signal</th><th>Direction</th></tr>
-		<tr class=odd><td align=center>9</td><td>BCLK</td><td>Output</td></tr>
-		<tr class=odd><td align=center>11</td><td>MCLK</td><td>Output</td></tr>
-		<tr class=odd><td align=center>13</td><td>RX</td><td>Input</td></tr>
-		<tr class=odd><td align=center>23</td><td>LRCLK</td><td>Output</td></tr>
-	</table>
-	<p>Audio from
-		master mode I2S may be used in the same project as ADC, DAC and
-		PWM signals, because all remain in sync to Teensy's timing</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; PassThroughStereo
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Recorder
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; PeakMeterStereo
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; FFT
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; SpectrumAnalyzerBasic
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Chorus
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Flange
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Filter
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Filter_FIR
-	</p>
-	<h3>Notes</h3>
-	<p>Normally, this object is used with the Audio Shield, which
-		is controlled separately by the "sgtl5000" object.</p>
-	<p>Only one I2S input and one I2S output object may be used.  Master
-		and slave modes may not be mixed (both must be of the same type).
-	</p>
-	<p>I2S master objects can be used together with non-I2S input and output
-		objects, for simultaneous audio streaming on different hardware.</p>

gui/resources/help/AudioInputI2Sslave.html

-
-	<h3>Summary</h3>
-	<p>Receive 16 bit stereo audio from an I2S device using I2S slave mode.</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>Out 0</td><td>Left Channel</td></tr>
-		<tr class=odd><td align=center>Out 1</td><td>Right Channel</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p>This object has no functions to call from the Arduino sketch.  It
-		simply streams data from the I2S hardware to its 2 output ports.</p>
-	<h3>Hardware</h3>
-	<p>The I2S signals are used in "slave" mode, where the I2S device controls
-		data timing.</p>
-	<table class=doc align=center cellpadding=3>
-		<tr class=top><th>Pin</th><th>Signal</th><th>Direction</th></tr>
-		<tr class=odd><td align=center>9</td><td>BCLK</td><td>Input</td></tr>
-		<tr class=odd><td align=center>13</td><td>RX</td><td>Input</td></tr>
-		<tr class=odd><td align=center>23</td><td>LRCLK</td><td>Input</td></tr>
-	</table>
-	<!--<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; 
-	</p>-->
-	<h3>Notes</h3>
-	<p>Slave mode I2S <b>should not used in the same project as ADC, DAC and
-		PWM</b> signals.  Differences in timing between the I2S device and
-		Teensy's clock can cause occasional audio glitches when I2S slave mode
-		is used together with other input or output objects based on Teensy's
-		timing.</p>
-	<p>Only one I2S input and one I2S output object may be used.  Master
-		and slave modes may not be mixed (both must be of the same type).
-	</p>

gui/resources/help/AudioMixer4.html

-
-	<h3>Summary</h3>
-	<p>Combine up to 4 audio signals together, each with adjustable gain.
-		All channels support signal attenuation or amplification.</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>Input signal #1</td></tr>
-		<tr class=odd><td align=center>In 1</td><td>Input signal #2</td></tr>
-		<tr class=odd><td align=center>In 2</td><td>Input signal #3</td></tr>
-		<tr class=odd><td align=center>In 3</td><td>Input signal #4</td></tr>
-		<tr class=odd><td align=center>Out 0</td><td>Sum of all inputs</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>gain</span>(channel, level);</p>
-	<p class=desc>Adjust the amplification or attenuation.  "channel" must
-		be 0 to 3.  "level" may be any floating point number from 0 to 32767.
-		1.0 passes the signal through directly.  Level of 0 shuts the channel
-		off completely.  Between 0 to 1.0 attenuates the signal, and above
-		1.0 amplifies it.  All 4 channels have separate settings.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; SamplePlayer
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Synthesis &gt; PlaySynthMusic
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; SpectrumAnalyzerBasic
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; DialTone_Serial
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; MemoryAndCpuUsage
-	</p>
-	<h3>Notes</h3>
-	<p>Signal clipping can occur when any channel has gain greater than 1.0,
-		or when multiple signals add together to greater than 1.0.</p>
-	<p>More than 4 channels may be combined by connecting multiple mixers
-		in tandem.  For example, a 16 channel mixer may be built using 5
-		mixers, where the fifth mixer combines the outputs of the first 4.
-	</p>

gui/resources/help/AudioOutputAnalog.html

-
-	<h3>Summary</h3>
-	<p>Transmit 12 bit audio using Teensy 3.1's built-in digital to analog converter.</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>Audio Channel</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p>This object has no functions to call from the Arduino sketch.  It
-		simply streams data from the ADC to its output port.</p>
-	<h3>Hardware</h3>
-	<p align=center><img src="img/dacpin.jpg"></p>
-	<p>Signal range is 0 to 1.2V</p>
-	<p>Most applications require at least a 10&micro;F DC-blocking capacitor.</p>
-	<p>TODO: photo of Teensy 3.1 with 10&micro;F capacitor and 3.5mm jack.</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; PassThroughMono
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; SamplePlayer
-	</p>
-	<h3>Notes</h3>
-	<p>The output rate is 44.1 kHz (no oversampling).  Ultrasonic noise present if
-		not filtered.  This may not
-		be an issue for many uses, but care should be used if amplified and driven
-		to high power tweeters.</p>

gui/resources/help/AudioOutputI2S.html

-
-	<h3>Summary</h3>
-	<p>Transmit 16 bit stereo audio to the
-	<a href="http://www.pjrc.com/store/teensy3_audio.html" target="_blank">audio shield</a>
-		or another I2S device, using I2S master mode.</p>
-	<p align=center><img src="img/audioshield_outputs.jpg"></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>Left Channel</td></tr>
-		<tr class=odd><td align=center>In 1</td><td>Right Channel</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p>This object has no functions to call from the Arduino sketch.  It
-		simply streams data from its 2 input ports to the I2S hardware.</p>
-	<h3>Hardware</h3>
-	<p align=center><img src="img/audioshield_backside.jpg"></p>
-	<p>The I2S signals are used in "master" mode, where Teensy creates
-		all 3 clock signals and controls all data timing.</p>
-	<table class=doc align=center cellpadding=3>
-		<tr class=top><th>Pin</th><th>Signal</th><th>Direction</th></tr>
-		<tr class=odd><td align=center>9</td><td>BCLK</td><td>Output</td></tr>
-		<tr class=odd><td align=center>11</td><td>MCLK</td><td>Output</td></tr>
-		<tr class=odd><td align=center>22</td><td>TX</td><td>Output</td></tr>
-		<tr class=odd><td align=center>23</td><td>LRCLK</td><td>Output</td></tr>
-	</table>
-	<p>Audio from
-		master mode I2S may be used in the same project as ADC, DAC and
-		PWM signals, because all remain in sync to Teensy's timing</p>
-	<h3>Examples</h3>
-	<p>Nearly all the examples use this object.  Here are some of the highlights:</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; PassThroughStereo
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; SamplePlayer
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Recorder
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; WavFilePlayer
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Chorus
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Synthesis &gt; PlaySynthMusic
-	</p>
-	<h3>Notes</h3>
-	<p>Normally, this object is used with the Audio Shield, which
-		is controlled separately by the "sgtl5000" object.</p>
-	<p>Only one I2S input and one I2S output object may be used.  Master
-		and slave modes may not be mixed (both must be of the same type).
-	</p>

gui/resources/help/AudioOutputI2Sslave.html

-
-	<h3>Summary</h3>
-	<p>Transmit 16 bit stereo audio to an I2S device using I2S slave mode.</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>Left Channel</td></tr>
-		<tr class=odd><td align=center>In 1</td><td>Right Channel</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p>This object has no functions to call from the Arduino sketch.  It
-		simply streams data from its 2 input ports to the I2S hardware.</p>
-	<h3>Hardware</h3>
-	<p>The I2S signals are used in "slave" mode, where the I2S device controls
-		data timing.</p>
-	<table class=doc align=center cellpadding=3>
-		<tr class=top><th>Pin</th><th>Signal</th><th>Direction</th></tr>
-		<tr class=odd><td align=center>9</td><td>BCLK</td><td>Input</td></tr>
-		<tr class=odd><td align=center>22</td><td>TX</td><td>Output</td></tr>
-		<tr class=odd><td align=center>23</td><td>LRCLK</td><td>Input</td></tr>
-	</table>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; WM8731MikroSine
-	</p>
-	<h3>Notes</h3>
-	<p>Slave mode I2S <b>should not used in the same project as ADC, DAC and
-		PWM</b> signals.  Differences in timing between the I2S device and
-		Teensy's clock can cause occasional audio glitches when I2S slave mode
-		is used together with other input or output objects based on Teensy's
-		timing.</p>
-	<p>Only one I2S input and one I2S output object may be used.  Master
-		and slave modes may not be mixed (both must be of the same type).
-	</p>

gui/resources/help/AudioOutputPWM.html

-
-	<h3>Summary</h3>
-	<p>Transmit audio using Teensy 3.1's PWM pins.  Two pins are
-		used for coarse and fine pulses, to be combined by scaled
-		resistors.</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>Audio Channel</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p>This object has no functions to call from the Arduino sketch.  It
-		simply streams data from the its input port to the PWM pins.</p>
-	<h3>Hardware</h3>
-	<p>The following circuit is recommended.</p>
-	<p align=center><img src="img/pwmdualcircuit.jpg"></p>
-	<p>Signal range is approx 1.55 Vp-p.</p>
-	<p>These resistor values assume approx 20 ohms output impedance
-		on the digital pins.  The 127K resistor may be adjusted or
-		trimmed for variation in output drive and tolerance on the
-		475 ohm resistor.</p>
-	<p>A plastic film (Polypropylene, Polyethylene, Polyester, etc) or
-		C0G/NPO ceramic capacitor should be used for filtering.  Low
-		quality ceramic (X7R, Y5V, Z5U, etc) can cause signal distortion.</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; PassThroughMono
-	</p>
-	<h3>Notes</h3>
-	<p>This object only works properly when Tools > CPU_Speed is set to
-		48 or 96 MHz.  Other speeds aren't supported and will likely fail
-		in strange ways.</p>
-	<p>The PWM carrier frequency is 88.2 kHz.  The suggested circuit
-		will only slightly filter the carrier.  Extra filtering will be
-		required for a clean signal without the ultrasonic PWM carrier.
-		</p>
-	<p>Analog signals created by filtering PWM waveforms use the digital
-		power supply as their reference voltage.  Any noise on the digital
-		power line can directly couple to the output signal.  The built-in DAC or
-		<a href="http://www.pjrc.com/store/teensy3_audio.html" target="_blank">audio shield</a>
-		should be used when higher quality signals are needed.</p>

gui/resources/help/AudioOutputSPDIF.html

-
-	<h3>Summary</h3>
-	<p>Transmit 16 bit stereo audio as Digital S/PDIF.</p>
-	<p align=center><img src="img/spdif_proto.jpg"></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>Left Channel</td></tr>
-		<tr class=odd><td align=center>In 1</td><td>Right Channel</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p>This object has no functions to call from the Arduino sketch.  It
-		simply streams data from its 2 input ports S/PDIF encoded digital
-		audio on pin 22.</p>
-	<h3>Hardware</h3>
-	<p>The S/PDIF output signal can be used to drive an optical TOSLINK
-		cable, or a standard (usually orange) RCA jack.</p>
-	<table class=doc align=center cellpadding=3>
-		<tr class=top><th>Pin</th><th>Signal</th><th>Direction</th></tr>
-		<tr class=odd><td align=center>22</td><td>S/PDIF</td><td>Output</td></tr>
-	</table>
-	<p>For optical TOSLINK output, this
-		<a href="https://www.oshpark.com/shared_projects/KcDBKHta" target="_blank">OSH Park board</a>
-		can be used with the inexpensive Everlight PLT133/T6A connector, available
-		at Digikey, 1080-1434-ND.
-		</p>
-	<h3>Examples</h3>
-	<p>The AudioOutputSPDIF object can be used in place of the AudioOutputI2S object,
-	<p>used in nearly all the examples.  The WavFilePlayer shows how to substitute
-		output objects for different hardware types.
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; WavFilePlayer
-	</p>
-	<h3>Credits</h3>
-	<p><a href="https://github.com/FrankBoesing" target="_blank">Frank Boesing</a>
-		developed the AudioOutputSPDIF code.  The original
-		<a href="https://forum.pjrc.com/threads/28639-S-pdif" target="_blank">forum disussion</a>
-		included valuable input and code from "kpc".
-	<h3>Notes</h3>
-	<p>S/PDIF output uses the I2S hardware.  This object can not be used
-		together with any of the I2S objects, because it requires the I2S
-		hardware with different internal settings.</p>
-	</p>

gui/resources/help/AudioPlayMemory.html

-
-	<h3>Summary</h3>
-	<p>Play a short sound clip, stored directly in memory.
-		Data files are created with the
-		<a href="https://github.com/PaulStoffregen/Audio/tree/master/examples/SamplePlayer/wav2sketch" target="_blank">wav2sketch program</a>,
-		and copied to the sketch folder to become part of your sketch.</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>Out 0</td><td>Sound Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>play</span>(data);</p>
-	<p class=desc>Begin playing a sound clip.  If already playing, the
-		currently playing clip is stopped and this new data begins
-		playing from the beginning.
-	</p>
-	<p class=func><span class=keyword>stop</span>();</p>
-	<p class=desc>Stop playing.  If not playing, this function has no effect.
-	</p>
-	<p class=func><span class=keyword>isPlaying</span>();</p>
-	<p class=desc>Return true (non-zero) if playing, or false (zero)
-		when not playing.
-	</p>
-	<p class=func><span class=keyword>positionMillis</span>();</p>
-	<p class=desc>While playing, return the current time offset, in
-		milliseconds.  When not playing, the return from this function
-		is undefined.
-	</p>
-	<p class=func><span class=keyword>lengthMillis</span>();</p>
-	<p class=desc>Return the total length of the current sound clip,
-		in milliseconds.  When not playing, the return from this function
-		is undefined.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; SamplePlayer
-	</p>
-	<h3>Notes</h3>
-	<p>TODO: supported sample rates: 11.025, 22.05, 44.1</p>
-	<p>TODO: ulaw vs uncompressed encoding</p>
-	<p>Polyphonic playback can be built by creating multiple
-		objects, with their output combined by mixers.</p>

gui/resources/help/AudioPlayQueue.html

-
-	<h3>Summary</h3>
-	<p>Play audio data provided by the Arduino sketch.  This object provides
-		functions to allow the sketch code to push data into the audio system.</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>Out 0</td><td>Sound Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>play</span>(int16);</p>
-	<p class=desc>not yet implemented
-	</p>
-	<p class=func><span class=keyword>play</span>(int16[], length);</p>
-	<p class=desc>not yet implemented
-	</p>
-	<p class=func><span class=keyword>getBuffer</span>();</p>
-	<p class=desc>Returns a pointer to an array of 128 int16.  This buffer
-		is within the audio library memory pool, providing the most efficient
-		way to input data to the audio system.  The buffer is likely to be
-		populated by previously used data, so the entire 128 words should be
-		written before calling playBuffer().  Only a single buffer should be
-		requested at a time.  This function may return NULL if no memory is
-		available.
-	</p>
-	<p class=func><span class=keyword>playBuffer</span>();</p>
-	<p class=desc>Transmit the buffer previously obtained from getBuffer().
-	</p>
-	<h3>Examples</h3>
-		<p><a href="http://community.arm.com/groups/embedded/blog/2014/05/23/led-video-panel-at-maker-faire-2014" target="_blank">4320 LED Video+Sound Project</a>
-		</p>
-	<!--<p class=exam>File &gt; Examples &gt; Audio &gt; 
-	</p>-->
-	<h3>Notes</h3>
-	<p>TODO: many caveats....</p>
-	<p>
-	</p>

gui/resources/help/AudioPlaySdRaw.html

-
-	<h3>Summary</h3>
-	<p>Play a RAW data file, stored on a SD card.  RAW format is simpler
-		than WAV and begins playing immediately, without parsing WAV file
-		header info.</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>Out 0</td><td>Sound Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>play</span>(filename);</p>
-	<p class=desc>Begin playing a RAW data file.  If a file is already playing,
-		it is stopped and this file starts playing from the beginning.
-	</p>
-	<p class=func><span class=keyword>stop</span>();</p>
-	<p class=desc>Stop playing.  If not playing, this function has no effect.
-	</p>
-	<p class=func><span class=keyword>isPlaying</span>();</p>
-	<p class=desc>Return true (non-zero) if playing, or false (zero)
-		when not playing.
-	</p>
-	<p class=func><span class=keyword>positionMillis</span>();</p>
-	<p class=desc>While playing, return the current time offset, in
-		milliseconds.  When not playing, the return from this function
-		is undefined.
-	</p>
-	<p class=func><span class=keyword>lengthMillis</span>();</p>
-	<p class=desc>Return the total length of the current sound clip,
-		in milliseconds.  When not playing, the return from this function
-		is undefined.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Recorder
-	</p>
-	<h3>Notes</h3>
-	<p>The data file must be RAW 16 bit signed integers in LSB-first format.
-	</p>
-	<p>While playing, the audio library accesses the SD card automatically.
-		If card access is required, you must
-		<a href="http://www.pjrc.com/teensy/td_libs_AudioProcessorUsage.html" target="_blank">AudioNoInterrupts()</a>
-		to prevent the library from accessing the SD card while you use it.
-		Disabling the audio library interrupt for too long may cause audible
-		dropouts or glitches.
-	</p>

gui/resources/help/AudioPlaySdWav.html

-
-	<h3>Summary</h3>
-	<p>Play a WAV file, stored on a SD card.</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>Out 0</td><td>Left Channel Output</td></tr>
-		<tr class=odd><td align=center>Out 1</td><td>Right Channel Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>play</span>(filename);</p>
-	<p class=desc>Begin playing a WAV file.  If a file is already playing,
-		it is stopped and this file starts playing from the beginning.
-	</p>
-	<p class=func><span class=keyword>stop</span>();</p>
-	<p class=desc>Stop playing.  If not playing, this function has no effect.
-	</p>
-	<p class=func><span class=keyword>isPlaying</span>();</p>
-	<p class=desc>Return true (non-zero) if playing, or false (zero)
-		when not playing.  See the note below about delayed start.
-	</p>
-	<p class=func><span class=keyword>positionMillis</span>();</p>
-	<p class=desc>While playing, return the current time offset, in
-		milliseconds.  When not playing, the return from this function
-		is undefined.
-	</p>
-	<p class=func><span class=keyword>lengthMillis</span>();</p>
-	<p class=desc>Return the total length of the current sound clip,
-		in milliseconds.  When not playing, the return from this function
-		is undefined.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; WavFilePlayer
-	</p>
-	<h3>Notes</h3>
-	<p>Only 16 bit PCM, 44100 Hz WAV files are supported.  When mono
-		files are played, both output ports transmit a copy of the
-		single sound.  Of course, stereo WAV files play with the left
-		channel on port 0 and the right channel on port 1.
-	</p>
-	<p>A brief delay after calling play() will usually occur before
-		isPlaying() returns true and positionMillis() returns valid
-		time offset.  WAV files have a header at the beginning of the
-		file, which the audio library must read and parse before
-		playing can begin.
-	</p>
-	<p>While playing, the audio library accesses the SD card automatically.
-		If card access is required, you must
-		<a href="http://www.pjrc.com/teensy/td_libs_AudioProcessorUsage.html" target="_blank">use AudioNoInterrupts()</a>
-		to prevent the library from accessing the SD card while you use it.
-		Disabling the audio library interrupt for too long may cause audible
-		dropouts or glitches.
-	</p>

gui/resources/help/AudioPlaySerialflashRaw.html

-
-	<h3>Summary</h3>
-	<div>
-	<p>Play a RAW data file, stored on a Serial Flash chip.  These chips
-		are far more efficient than SD cards, allowing many files to be
-		played simultaneously by copies of this object.
-		</p>
-	<p align=center><img src="img/w25q128fv.jpg"><br><small>W25Q128FV Serial Flash</small></p>
-	</div>
-	<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>Out 0</td><td>Sound Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>play</span>(filename);</p>
-	<p class=desc>Begin playing a RAW data file.  If a file is already playing,
-		it is stopped and this file starts playing from the beginning.
-	</p>
-	<p class=func><span class=keyword>stop</span>();</p>
-	<p class=desc>Stop playing.  If not playing, this function has no effect.
-	</p>
-	<p class=func><span class=keyword>isPlaying</span>();</p>
-	<p class=desc>Return true (non-zero) if playing, or false (zero)
-		when not playing.
-	</p>
-	<p class=func><span class=keyword>positionMillis</span>();</p>
-	<p class=desc>While playing, return the current time offset, in
-		milliseconds.  When not playing, the return from this function
-		is undefined.
-	</p>
-	<p class=func><span class=keyword>lengthMillis</span>();</p>
-	<p class=desc>Return the total length of the current sound clip,
-		in milliseconds.  When not playing, the return from this function
-		is undefined.
-	</p>
-	<h3>Examples</h3>
-	<!--<p class=exam>File &gt; Examples &gt; Audio &gt; Recorder-->
-	<p class=exam>TODO: play example needed....
-	</p>
-	<p class=exam>File &gt; Examples &gt; SerialFlash &gt; CopyFromSD
-	</p>
-	<h3>Notes</h3>
-	<p>The data file must be RAW 16 bit signed integers in LSB-first format.
-	</p>
-	<p>The <a href="https://github.com/PaulStoffregen/SerialFlash" target="_blank">SerialFlash library</a>
-		is used to access the flash chip.  You can also use SerialFlash's functions
-		to access the stored files, or add data to the flash chip.
-	</p>
-	<p>File names are case sensitive with SerialFlash.  If your sound does
-		not play, use <b>File &gt; Examples &gt; SerialFlash &gt; ListFiles</b> to
-		check the exact file names stored in the flash memory chip.

gui/resources/help/AudioRecordQueue.html

-
-	<h3>Summary</h3>
-	<p>Record audio data by sending to the Arduino sketch.  This object allows
-		sketch code to receive audio packets.</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>Sound To Access</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>begin</span>();</p>
-	<p class=desc>Begin capturing incoming audio to the queue.  After calling
-		begin, readBuffer() and freeBuffer(), or clear() must be used frequently
-		to prevent the queue from filling up.
-	</p>
-	<p class=func><span class=keyword>available</span>();</p>
-	<p class=desc>Returns the number of audio packets available to read.
-	</p>
-	<p class=func><span class=keyword>readBuffer</span>();</p>
-	<p class=desc>Read a single audio packet.  A pointer to a 128 sample
-		array of 16 bit integers is returned.  NULL is returned if no packets
-		are available.
-	</p>
-	<p class=func><span class=keyword>freeBuffer</span>();</p>
-	<p class=desc>Release the memory from the previously read packet returned
-		from readBuffer().  Only a single packet at a time may be read, and
-		each packet must be freed with this function, to return the memory to
-		the audio library.
-	</p>
-	<p class=func><span class=keyword>clear</span>();</p>
-	<p class=desc>Discard all audio held in the queue.
-	</p>
-	<p class=func><span class=keyword>end</span>();</p>
-	<p class=desc>Stop capturing incoming audio into the queue.  Data already
-		captured remains in the queue and may be read with readBuffer().
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Recorder
-	</p>
-	<h3>Notes</h3>
-	<p>
-		Up to 52 packets may be queued by this object, which allows approximately
-		150 ms of audio to be held in the queue, to allow time for the Arduino
-		sketch to write data to media or do other high-latency tasks.
-
-The actual packets are taken
-		from the pool created by AudioMemory().
-	</p>

gui/resources/help/AudioSynthNoisePink.html

-
-	<h3>Summary</h3>
-	<div>
-	<p>Create pink noise, using Stefan Stenzel's "New Shade Of Pink" algorithm.
-		</p>
-	<!--<p align=center><img src="img/whitenoise.png"></p>-->
-	</div>
-	<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>Out 0</td><td>Pink Noise</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>amplitude</span>(level);</p>
-	<p class=desc>Set the output peak level, from 0 (off) to 1.0.
-		The default is off.  Noise is generated only after setting
-		to a non-zero level.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; MemoryAndCpuUsage
-	</p>
-	<h3>Notes</h3>
-	<p>Setting the amplitude to zero causes this object to stop using
-		CPU time.  CPU usage is approx 3% on Teensy 3.1.
-	</p>
-	<p>Stefan Stenzel's
-		<a href="http://stenzel.waldorfmusic.de/post/pink/" target="_blank">New Shade Of Pink</a>
-		algorithm.  Stefan's terms of use are "Use for any purpose. If used
-		in a commercial product, you should give me one."
-	</p>

gui/resources/help/AudioSynthNoiseWhite.html

-
-	<h3>Summary</h3>
-	<div>
-	<p>Create white noise.
-		</p>
-	<p align=center><img src="img/whitenoise.png"></p>
-	</div>
-	<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>Out 0</td><td>White Noise</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>amplitude</span>(level);</p>
-	<p class=desc>Set the output peak level, from 0 (off) to 1.0.
-		The default is off.  Noise is generated only after setting
-		to a non-zero level.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; 
-	</p>
-	<h3>Notes</h3>
-	<p>Setting the amplitude to zero causes this object to stop using
-		CPU time to generate random numbers.
-	</p>

gui/resources/help/AudioSynthToneSweep.html

-
-	<h3>Summary</h3>
-	<p>Create a continuously varying (in frequency) sine wave</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>Out 0</td><td>Continuously varying tone</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>play</span>(level, lowFreq, highFreq, time);</p>
-	<p class=desc>Start generating frequency sweep output.  The time is specified
-		in milliseconds.  Level is 0 to 1.0.
-	</p>
-	<p class=func><span class=keyword>isPlaying</span>();</p>
-	<p class=desc>Returns true (non-zero) while the output is active.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; ToneSweep
-	</p>
-	<h3>Notes</h3>
-	<p>Uses excessive CPU time</p>

gui/resources/help/AudioSynthWaveform.html

-
-	<h3>Summary</h3>
-	<p>Create a waveform: sine, sawtooth, square, triangle, pulse or arbitrary.</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>Out 0</td><td>Waveform Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>begin</span>(waveform);</p>
-	<p class=desc>Configure the waveform type to create.
-	</p>
-	<p class=func><span class=keyword>begin</span>(level, frequency, waveform);</p>
-	<p class=desc>Output a waveform, and set the amplitude and frequency.
-	</p>
-	<p class=func><span class=keyword>frequency</span>(freq);</p>
-	<p class=desc>Change the frequency.
-	</p>
-	<p class=func><span class=keyword>amplitude</span>(level);</p>
-	<p class=desc>Change the amplitude.  Set to 0 to turn the signal off.
-	</p>
-	<p class=func><span class=keyword>phase</span>(angle);</p>
-	<p class=desc>
-		Cause the generated waveform to jump to a specific point within
-		its cycle.  Angle is from 0 to 360 degrees.  When multiple objects
-		are configured,
-		<a href="http://www.pjrc.com/teensy/td_libs_AudioProcessorUsage.html" target="_blank">AudioNoInterrupts()</a>
-		should be used to guarantee all new settings take effect together.
-	</p>
-	<p class=func><span class=keyword>pulseWidth</span>(amount);</p>
-	<p class=desc>Change the width (duty cycle) of the pulse.</p>
-	<p class=func><span class=keyword>arbitraryWaveform</span>(array, maxFreq);</p>
-	<p class=desc>
-		Configure the waveform to be used with WAVEFORM_ARBITRARY.  Array
-		must be an array of 256 samples.  Currently, the data is used
-		without any filtering, which can cause aliasing with frequencies
-		above 172 Hz.  For higher frequency output, you must bandwidth
-		limit your waveform data.  Someday, "maxFreq" will be used to
-		do this automatically.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Synthesis &gt; PlaySynthMusic
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Synthesis &gt; pulseWidth
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; WM8731MikroSine
-	</p>
-	<h3>Notes</h3>
-	<p>Supported Waveforms:<br>
-		<ul>
-		<li><span class=literal>WAVEFORM_SINE</span></li>
-		<li><span class=literal>WAVEFORM_SAWTOOTH</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>
-		</ul>
-	</p>

gui/resources/help/AudioSynthWaveformDc.html

-
-	<h3>Summary</h3>
-	<p>Create constant (DC) signal, useful for control of objects that take
-		a modulation or control input signal.  This constant level can be
-		used to modify other waveforms using mixer or multiplier objects</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>Out 0</td><td>Output constant DC level</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>amplitude</span>(level);</p>
-	<p class=desc>Set the output.  Level is -1.0 to 1.0.  The output is
-		changed immediately.
-	</p>
-	<p class=func><span class=keyword>amplitude</span>(level, milliseconds);</p>
-	<p class=desc>Set the output.  Level is -1.0 to 1.0.  The output is
-		gradually changed over a "milliseconds" time period.  Any time may
-		be specified, but periods longer than 1 second may be automatically
-		shortened for small level changes, due to numerical precision limits.
-	</p>
-	<!--<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; 
-	</p>-->
-	<h3>Notes</h3>
-	<p>Of course, the term "DC", for Direct Current, doesn't properly apply
-		to a pure digital stream of numerical values.  But the term is widely
-		understood in audio applications, so hopefully it's not too confusing?</p>

gui/resources/help/AudioSynthWaveformSine.html

-
-	<h3>Summary</h3>
-	<p>Create a sine wave signal</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>Out 0</td><td>Sine Wave Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>amplitude</span>(level);</p>
-	<p class=desc>Set the amplitude, from 0 to 1.0.
-	</p>
-	<p class=func><span class=keyword>frequency</span>(freq);</p>
-	<p class=desc>Set the frequency, from 0 to 22000.  Very low values may
-		be used to create a LFO (Low Frequency Oscillator) for objects
-		with modulation signal inputs.
-	</p>
-	<p class=func><span class=keyword>phase</span>(angle);</p>
-	<p class=desc>
-		Cause the generated waveform to jump to a specific point within
-		its cycle.  Angle is from 0 to 360 degrees.  When multiple objects
-		are configured,
-		<a href="http://www.pjrc.com/teensy/td_libs_AudioProcessorUsage.html" target="_blank">AudioNoInterrupts()</a>
-		should be used to guarantee all new settings take effect together.
-	</p>
-	<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; MemoryAndCpuUsage
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; DialTone_Serial
-	</p>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; FFT
-	</p>
-	<h3>Notes</h3>
-	<p></p>

gui/resources/help/AudioSynthWaveformSineModulated.html

-
-	<h3>Summary</h3>
-	<p>Create a modulated sine wave, using any audio signal to continuously
-		modulate the sine wave frequency.</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>Modulation Signal</td></tr>
-		<tr class=odd><td align=center>Out 0</td><td>Sine Wave Output</td></tr>
-	</table>
-	<h3>Functions</h3>
-	<p class=func><span class=keyword>amplitude</span>(level);</p>
-	<p class=desc>Set the amplitude, from 0 to 1.0.
-	</p>
-	<p class=func><span class=keyword>frequency</span>(freq);</p>
-	<p class=desc>Set the center frequency, from 0 to 11000.  The output will
-		be this center frequency when the input modulation signal is zero.
-		Modulation input 1.0 causes the frequency to double, and input -1.0
-		causes zero Hz (DC) output.  For less modulation, attenuate the input
-		signal (perhaps with a mixer object) before it arrives here.
-	</p>
-	<p class=func><span class=keyword>phase</span>(angle);</p>
-	<p class=desc>
-		Cause the generated waveform to jump to a specific point within
-		its cycle.  Angle is from 0 to 360 degrees.  When multiple objects
-		are configured,
-		<a href="http://www.pjrc.com/teensy/td_libs_AudioProcessorUsage.html" target="_blank">AudioNoInterrupts()</a>
-		should be used to guarantee all new settings take effect together.
-	</p>
-	<!--<h3>Examples</h3>
-	<p class=exam>File &gt; Examples &gt; Audio &gt; 
-	</p>-->
-	<h3>Notes</h3>
-	<p></p>