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Removed redundant Design Tool files

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boblark 2 years ago
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  1. 71
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  2. 67
      docs/audio_html/AudioAnalyzeFFT256.html
  3. 57
      docs/audio_html/AudioAnalyzeNoteFrequency.html
  4. 38
      docs/audio_html/AudioAnalyzePeak.html
  5. 42
      docs/audio_html/AudioAnalyzePrint.html
  6. 32
      docs/audio_html/AudioAnalyzeRMS.html
  7. 56
      docs/audio_html/AudioAnalyzeToneDetect.html
  8. 61
      docs/audio_html/AudioControlAK4558.html
  9. 64
      docs/audio_html/AudioControlCS4272.html
  10. 350
      docs/audio_html/AudioControlSGTL5000.html
  11. 42
      docs/audio_html/AudioControlWM8731.html
  12. 40
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  13. 47
      docs/audio_html/AudioEffectBitcrusher.html
  14. 51
      docs/audio_html/AudioEffectChorus.html
  15. 49
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  16. 103
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  17. 64
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  18. 46
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  19. 62
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  20. 42
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  21. 36
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  22. 32
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  23. 88
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  24. 64
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  25. 61
      docs/audio_html/AudioFilterStateVariable.html
  26. 56
      docs/audio_html/AudioInputAnalog.html
  27. 58
      docs/audio_html/AudioInputAnalogStereo.html
  28. 66
      docs/audio_html/AudioInputI2S.html
  29. 49
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  30. 46
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  31. 41
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  32. 49
      docs/audio_html/AudioMixer4.html
  33. 54
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  34. 47
      docs/audio_html/AudioOutputAnalogStereo.html
  35. 59
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  36. 50
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  37. 44
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  38. 44
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  39. 50
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  40. 53
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  41. 41
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  42. 52
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  43. 49
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  44. 55
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  45. 70
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  46. 62
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  47. 58
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  48. 35
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  49. 40
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  50. 33
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  51. 44
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  52. 31
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  53. 70
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  54. 40
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  55. 44
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  56. 41
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  57. 46
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71
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<script type="text/x-red" data-help-name="AudioAnalyzeFFT1024">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</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 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>
</script>
<script type="text/x-red" data-template-name="AudioAnalyzeFFT1024">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioAnalyzeFFT256">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</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 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>
</script>
<script type="text/x-red" data-template-name="AudioAnalyzeFFT256">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioAnalyzeNoteFrequency">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Detect with fairly good accuracy the fundamental frequency f<sub>o</sub>
of musical notes, such as electric guitar and bass.</p>
</div>
<p>Written By Collin Duffy</p>
<h3>Audio Connections</h3>
<table class=doc align=center cellpadding=3>
<tr class=top><th>Port</th><th>Purpose</th></tr>
<tr class=odd><td align=center>In 0</td><td>Signal to analyze</td></tr>
</table>
<h3>Functions</h3>
<p class=func><span class=keyword>begin</span>(threshold);</p>
<p class=desc>Initialize and start detecting frequencies,
with an initial threshold (the amount of allowed uncertainty).
</p>
<p class=func><span class=keyword>available</span>();</p>
<p class=desc>Returns true (non-zero) when a valid
frequency is detected.
</p>
<p class=func><span class=keyword>read</span>();</p>
<p class=desc>Read the detected frequency.
</p>
<p class=func><span class=keyword>probability</span>();</p>
<p class=desc>Return the level of certainty, betweeo 0 to 1.0.
</p>
<p class=func><span class=keyword>threshold</span>(level);</p>
<p class=desc>Set the detection threshold, the amount of allowed uncertainty.
</p>
<h3>Examples</h3>
<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; NoteFrequency
</p>
<h3>Notes</h3>
<p>The <a href="http://recherche.ircam.fr/equipes/pcm/cheveign/pss/2002_JASA_YIN.pdf">YIN algorithm</a> (PDF)
is used to detect frequencies, with many optimizations for
frequencies between 29-400Hz. This algorithm can be somewhat
memory and processor hungry but will allow you to detect with
fairly good accuracy the fundamental frequencies from
electric guitars and basses.</p>
<p>Within the code, AUDIO_GUITARTUNER_BLOCKS
may be edited to control low frequency range. The default
(24) allows measurement down to 29.14 Hz, or B(flat)0.</p>
<p>TODO: The usable upper range of this object is not well known.
Duff says "it should be good up to 1000Hz", but may have trouble
at 4 kHz. Please <a href="https://forum.pjrc.com/threads/32252-Different-Range-FFT-Algorithm/page2">post feedback here</a>, ideally with audio clips for the NoteFrequency example.</p>
<p>This object was contributed by Collin Duffy from his
<a href="https://github.com/duff2013/AudioTuner">AudioTuner project</a>.
Additional details and documentation may be found there.</p>
</script>
<script type="text/x-red" data-template-name="AudioAnalyzeNoteFrequency">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioAnalyzePeak">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Track the signal peak amplitude. Very useful for simple
audio level response projects, and general troubleshooting.</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 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 amplitude value since the last read.
Return is from 0.0 to 1.0.
</p>
<p class=func><span class=keyword>readPeakToPeak</span>();</p>
<p class=desc>Read the highest peak-to-peak amplitude since the last read.
Return is from 0.0 to 2.0.
</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>
</script>
<script type="text/x-red" data-template-name="AudioAnalyzePeak">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioAnalyzePrint">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Print raw audio data to the Arduino Serial Monitor. This
object creates massive output quickly, and should not normally be used.</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 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>
</script>
<script type="text/x-red" data-template-name="AudioAnalyzePrint">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

32
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<script type="text/x-red" data-help-name="AudioAnalyzeRMS">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Track the signal RMS amplitude. Useful for
audio level response projects, and general troubleshooting.</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 analyze</td></tr>
</table>
<h3>Functions</h3>
<p class=func><span class=keyword>available</span>();</p>
<p class=desc>Returns true if new RMS data is available.
</p>
<p class=func><span class=keyword>read</span>();</p>
<p class=desc>Read the new RMS value.
Return is from 0.0 to 1.0.
</p>
<h3>Examples</h3>
<p class=exam>File &gt; Examples &gt; Audio &gt; Analysis &gt; PeakAndRMSMeterStereo</p>
</p>
<h3>Notes</h3>
<p></p>
</script>
<script type="text/x-red" data-template-name="AudioAnalyzeRMS">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

56
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<script type="text/x-red" data-help-name="AudioAnalyzeToneDetect">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Detect the level of a single tone</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 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>
</script>
<script type="text/x-red" data-template-name="AudioAnalyzeToneDetect">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioControlAK4558">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Control the AK4558 chip on the <a href="https://hackaday.io/project/8567-hifi-audio-codec-module" target="_blank">HiFi Audio CODEC Module</a>
in slave mode, where the Teensy controls all I2S timing.</p>
</div>
<h3>Audio Connections</h3>
<p>This object has no audio inputs or outputs. Separate I2S objects
are used to send and receive audio data.
</p>
<h3>Functions</h3>
<p class=func><span class=keyword>enable</span>();</p>
<p class=desc>Enables the CODEC to work with 44.1 KHz - 16 bit data. This function does not enable the ADC/DAC modules.
</p>
<p class=func><span class=keyword>enableIn</span>();</p>
<p class=desc>Enables the ADC module.
</p>
<p class=func><span class=keyword>enableOut</span>();</p>
<p class=desc>Enables the DAC module.
</p>
<p class=func><span class=keyword>disable</span>();</p>
<p class=desc>Disables the ADC and the DAC modules.
</p>
<p class=func><span class=keyword>disableIn</span>();</p>
<p class=desc>Disable the ADC module.
</p>
<p class=func><span class=keyword>disableOut</span>();</p>
<p class=desc>Disable the DAC module.
</p>
<p class=func><span class=keyword>volume</span>(level);</p>
<p class=desc>Accepts a float in range 0.0-1.0 and sets the line output volume accordingly.
</p>
<p class=func><span class=keyword>volumeLeft</span>(level);</p>
<p class=desc>Accepts a float in range 0.0-1.0 and sets the left line output volume accordingly.
</p>
<p class=func><span class=keyword>volumeRight</span>(level);</p>
<p class=desc>Accepts a float in range 0.0-1.0 and sets the right line output volume accordingly.
</p>
<p class=func><span class=keyword>inputLevel</span>(level);</p>
<p class=desc>NOT SUPPORTED BY THE AK4558
</p>
<p class=func><span class=keyword>inputSelect</span>(input);</p>
<p class=desc>not implemented yet
</p>
<h3>Examples</h3>
<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; AK4558 &gt; PassthroughTest
</p>
<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; AK4558 &gt; SineOutTest
</p>
<h3>Notes</h3>
<p>TODO: Implement inputSelect() function to enable mono left, mono right, stereo operation.</p>
<p>TODO: Implement ADC and DAC filters control.</p>
<p>TODO: Implement DAC level attenuator attack rate modifier.</p>
</script>
<script type="text/x-red" data-template-name="AudioControlAK4558">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioControlCS4272">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Control the CS4272 chip on the <a href="https://hackaday.io/project/5912-teensy-super-audio-board" target="_blank">Super Audio Board</a>.
</p>
<p>TODO: does this control object put the CS4272 into I2S master or slave mode</p>
</div>
<h3>Audio Connections</h3>
<p>This object has no audio inputs or outputs. Separate I2S objects
are used to send and receive audio data.
</p>
<h3>Functions</h3>
<p class=func><span class=keyword>enable</span>();</p>
<p class=desc>Enables the CODEC to work with 44.1 KHz - 16 bit data. This function does not enable the ADC/DAC modules.
</p>
<p class=func><span class=keyword>volume</span>(vol);</p>
<p class=desc>Set the volume level. Range is 0 to 1.0.
</p>
<p class=func><span class=keyword>volume</span>(left, right);</p>
<p class=desc>Set the volume level. Range is 0 to 1.0.
</p>
<p class=func><span class=keyword>dacVolume</span>(vol);</p>
<p class=desc>Set the volume level. Range is 0 to 1.0. TODO: what's the
distinction between volume() and dacVolume()?
</p>
<p class=func><span class=keyword>dacVolume</span>(left, right);</p>
<p class=desc>Set the volume level. Range is 0 to 1.0.
</p>
<p class=func><span class=keyword>muteOutput</span>();</p>
<p class=desc>TODO: description
</p>
<p class=func><span class=keyword>unmuteOutput</span>();</p>
<p class=desc>TODO: description
</p>
<p class=func><span class=keyword>muteInput</span>();</p>
<p class=desc>TODO: description
</p>
<p class=func><span class=keyword>unmuteInput</span>();</p>
<p class=desc>TODO: description
</p>
<p class=func><span class=keyword>enableDither</span>();</p>
<p class=desc>TODO: description
</p>
<p class=func><span class=keyword>disableDither</span>();</p>
<p class=desc>TODO: description
</p>
<h3>Hardware</h3>
<p>Pin 2 must be connected to the CS4272 reset. SDA &amp; SCL are used for all control.
</p>
<h3>Notes</h3>
</script>
<script type="text/x-red" data-template-name="AudioControlCS4272">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioControlSGTL5000">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</div>
<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 graceful
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>
<p class=func><span class=keyword>dacVolumeRamp</span>();</p>
<p class=desc>Enable graceful volume ramping. The dacVolume adjusts gradually using
an exponential curve. Pops or loud clicks are avoided when making large
changes in volume level.
</p>
<p class=func><span class=keyword>dacVolumeRampLinear</span>();</p>
<p class=desc>Enable faster volume ramping. A slight click may be heard during a
large volume change.
</p>
<p class=func><span class=keyword>dacVolumeRampDisable</span>();</p>
<p class=desc>Do not use any gradual ramping. The zero cross feature still helps
for small changes, but large volume changes may produce a pop or click.
</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>
<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; SGTL5000 &gt; VolumeRamp
</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>
</script>
<script type="text/x-red" data-template-name="AudioControlSGTL5000">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioControlWM8731">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Control a WM8731 chip in slave mode, where it receives all clocks from Teensy</p>
</div>
<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>
</script>
<script type="text/x-red" data-template-name="AudioControlWM8731">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
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</div>
</script>

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<script type="text/x-red" data-help-name="AudioControlWM8731master">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Control a WM8731 chip in master mode, where it controls all I2S timing.</p>
</div>
<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>
</script>
<script type="text/x-red" data-template-name="AudioControlWM8731master">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
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</div>
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<script type="text/x-red" data-help-name="AudioEffectBitcrusher">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Reduce the samplerate and/or bitdepth of a source signal, resulting in
a distorted sound.</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 Output</td></tr>
</table>
<h3>Functions</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>
</script>
<script type="text/x-red" data-template-name="AudioEffectBitcrusher">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioEffectChorus">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</div>
<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>
</script>
<script type="text/x-red" data-template-name="AudioEffectChorus">
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</div>
</script>

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<script type="text/x-red" data-help-name="AudioEffectDelay">
<h3>Summary</h3>
<div class=tooltipinfo>
<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 425 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>
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<script type="text/x-red" data-help-name="AudioEffectDelayExternal">
<h3>Summary</h3>
<div class=tooltipinfo>
<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 1.5 seconds for each 23LC1024 chip.
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>
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<script type="text/x-red" data-help-name="AudioEffectEnvelope">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
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<script type="text/x-red" data-help-name="AudioEffectFade">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Gradually increase or decrease audio level.</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 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>
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<script type="text/x-red" data-help-name="AudioEffectFlange">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</div>
<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>voices</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>
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<script type="text/x-red" data-help-name="AudioEffectMidSide">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Convert stereo signals to/from Mid-Side format.
Mid-Side encoding can be used to increase stereo width, make the lower
frequencies mono (to please your sub), or as the basis of audio compression.</p>
</div>
<h3>Audio Connections</h3>
<table class=doc align=center cellpadding=3>
<tr class=top><th>Port</th><th>While<br>Encoding</th><th>While<br>Decoding</th></tr>
<tr class=odd><td align=center>In 0</td><td>Left Input</td><td>Mid Output</td></tr>
<tr class=odd><td align=center>In 1</td><td>Right Input</td><td>Side Output</td></tr>
<tr class=odd><td align=center>Out 0</td><td>Mid Input</td><td>Left Output</td></tr>
<tr class=odd><td align=center>Out 1</td><td>Side Input</td><td>Right Output</td></tr>
</table>
<h3>Functions</h3>
<p class=func><span class=keyword>encode</span>();</p>
<p class=desc>Configure this object to encode from stereo to Mid-Side format.</p>
<p class=func><span class=keyword>decode</span>();</p>
<p class=desc>Configure this object to decode from Mid-Side format back to stereo signals.</p>
<h3>Examples</h3>
<p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Mid_Side</p>
<h3>Notes</h3>
<p>Many interesting stereo effects can be achieved by manipulating Mid-Side signals.</p>
<p>Normally a pair of these objects are used, one to encode, then additional
gain/attenuation or effects applied to the Mid-Side signals, and finally
decoding back to stereo signals</p>
<p>To prevent saturation, halving is done in the encoding, that is:</p>
<p>Mid = (left+right)/2</p>
<p>Side = (left-right)/2</p>
<p>And to decode:</p>
<p>Left = Mid+Side</p>
<p>Right = Mid-Side</p>
</script>
<script type="text/x-red" data-template-name="AudioEffectMidSide">
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<script type="text/x-red" data-help-name="AudioEffectMultiply">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</script>
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<script type="text/x-red" data-help-name="AudioEffectReverb">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Reverb with adjustable reverberation time. Contributed by Joao Rossi FIlho.
</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>Input</td></tr>
<tr class="odd"><td align="center">Out 0</td><td>Output</td></tr>
</table>
<h3>Functions</h3>
<p class=func><span class=keyword>reverbTime</span>(seconds);</p>
<p class=desc>Sets the amount of reverberation time.
</p>
<h3>Examples</h3>
<p><a href="https://twitter.com/joaorossifilho/status/779737126841753601">Video Demo</a>
</p>
<!--<p class=exam>File &gt; Examples &gt; Audio &gt; Effects &gt; Flange
</p>-->
<h3>Notes</h3>
<p>This effect may have distortion problems with the input signal is more than 0.5.</p>
</script>
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<script type="text/x-red" data-help-name="AudioFilterBiquad">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>setLowShelf</span>(stage, frequency, gain, slope);</p>
<p class=desc>Configure one stage of the filter (0 to 3) with low shelf response.
A low shelf filter attenuates or amplifies signals below the specified frequency.
Frequency controls the slope midpoint, gain is in dB and can be both
positive or negative. The slope parameter controls steepness of gain transition.
A slope of 1 yields maximum steepness without overshoot,
lower values yield a less steep slope. See the picture below for a visualization
of the slope parameter's effect.
Be careful with positive gains and slopes higher than 1 as they introduce gain
(see warning below).
</p>
</p>
<p class=func><span class=keyword>setHighShelf</span>(stage, frequency, gain, slope);</p>
<p class=desc>Configure one stage of the filter (0 to 3) with high shelf response.
A high shelf filter attenuates or amplifies signals above the specified frequency.
Frequency controls the slope midpoint, gain is in dB and can be both
positive or negative. The slope parameter controls steepness of gain transition.
A slope of 1 yields maximum steepness without overshoot,
lower values yield a less steep slope. See the picture below for a visualization
of the slope parameter's effect.
Be careful with positive gains and slopes higher than 1 as they introduce gain
(see warning below).
</p>
<p align=center><img src="img/shelf_filter.png"></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>
</script>
<script type="text/x-red" data-template-name="AudioFilterBiquad">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioFilterFIR">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>Known Issues</h3>
<p>Your filter's impulse response array must have an even length. If you have
add odd number of taps, you must add an extra zero to increase the length
to an even number.
</p>
<p>The minimum number of taps is 4. If you use less, add extra zeros to increase
the length to 4.
</p>
<p>The impulse response must be given in reverse order. Many filters have
symetrical impluse response, making this a non-issue. If your filter has
a non-symetrical response, make sure the data is in reverse time order.
</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.engineerjs.com/" 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>
</script>
<script type="text/x-red" data-template-name="AudioFilterFIR">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioFilterStateVariable">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>A State Variable (Chamberlin) Filter with 12 dB/octave roll-off,
adjustable resonance, and optional signal control of corner
frequency.</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 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>
</script>
<script type="text/x-red" data-template-name="AudioFilterFIR">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioInputAnalog">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Receive audio using the built-in analog to digital converter.</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>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>
<p>With a <a href="https://forum.pjrc.com/threads/40468-Help-with-Basic-Audio-Lib-results?p=126317&viewfull=1#post126317">small modification</a> Adafruit's <a href="https://www.adafruit.com/products/1063">MAX4466 microphone</a> can be used</p>
<p align=center><a href="https://forum.pjrc.com/threads/40468-Help-with-Basic-Audio-Lib-results?p=126317&viewfull=1#post126317"><img src="img/adccircuitmic.jpg" border=0></a></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>
<p class=exam>File &gt; Examples &gt; OctoWS2811 &gt; SpectrumAnalyzer
</p>
<h3>Notes</h3>
<p><b>analogRead() must not be used</b>, because AudioInputAnalog is regularly
accessing the ADC hardware. If both access the hardware at the same
moment, analogRead() can end up waiting forever, which effectively
crashes your program.
</p>
<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>
</script>
<script type="text/x-red" data-template-name="AudioInputAnalog">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioInputAnalogStereo">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Receive stereo audio using the built-in analog to digital converters.</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>Audio Channel (Left)</td></tr>
<tr class=odd><td align=center>Out 1</td><td>Audio Channel (Right)</td></tr>
</table>
<h3>Functions</h3>
<p>This object has no functions to call from the Arduino sketch. It
simply streams data from both ADCs to its output ports.</p>
<h3>Hardware</h3>
<p>By default, pins A2 & A3 are used for audio input. This circuitry is recommended.</p>
<p align=center><img src="img/adccircuit2.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>
<p class=exam>File &gt; Examples &gt; OctoWS2811 &gt; SpectrumAnalyzer
</p>
-->
<h3>Notes</h3>
<p><b>analogRead() must not be used</b>, because AudioInputAnalogStereo is regularly
accessing the ADC hardware. If both access the hardware at the same
moment, analogRead() can end up waiting forever, which effectively
crashes your program.
</p>
<p>A different pin may be used, but adding it as an parameter
to the AudioInputAnalog object definition.
</p>
<p>For example:
</p>
<p class=desc><span class=keyword>AudioInputAnalogStereo</span> adc1(<span class=literal>A3</span>, <span class=literal>A2</span>);
</p>
<p>TODO: add info here about which pins work for input 0 and 1.
</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>
</script>
<script type="text/x-red" data-template-name="AudioInputAnalogStereo">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioInputI2S">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</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>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>
</script>
<script type="text/x-red" data-template-name="AudioInputI2S">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioInputI2SQuad">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Receive 16 bit quad (4) channel audio from two
<a href="http://www.pjrc.com/store/teensy3_audio.html" target="_blank">audio shields</a>
or another I2S devices, using I2S master mode.</p>
<p align=center><img src="img/audioshield_quad_in.jpg"></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>Channel #1</td></tr>
<tr class=odd><td align=center>Out 1</td><td>Channel #2</td></tr>
<tr class=odd><td align=center>Out 2</td><td>Channel #3</td></tr>
<tr class=odd><td align=center>Out 3</td><td>Channel #4</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 4 output ports.</p>
<h3>Hardware</h3>
<p>See this Sparkfun blog for <a href="https://www.sparkfun.com/news/2055" target="_blank">how
to connect two audio adaptors for 4 channel audio</a>.
<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>Teensy<br>3.2 Pin</th><th>Teensy<br>3.5/3.6 Pin</th><th>Signal</th><th>Direction</th></tr>
<tr class=odd><td align=center>9</td><td align=center>9</td><td>BCLK</td><td>Output</td></tr>
<tr class=odd><td align=center>11</td><td align=center>11</td><td>MCLK</td><td>Output</td></tr>
<tr class=odd><td align=center>13</td><td align=center>13</td><td>RX</td><td>Input</td></tr>
<tr class=odd><td align=center>30</td><td align=center>38</td><td>RX</td><td>Input</td></tr>
<tr class=odd><td align=center>23</td><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; PassThroughQuad
</p>
<h3>Notes</h3>
<p>Normally, this object is used with two Audio Shield, which
are controlled separately by a pair "sgtl5000" object.</p>
</script>
<script type="text/x-red" data-template-name="AudioInputI2SQuad">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioInputI2Sslave">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Receive 16 bit stereo audio from an I2S device using I2S slave mode
(where the ADC or codec chip, not Teensy, controls audio timing).</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>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>
</script>
<script type="text/x-red" data-template-name="AudioInputI2Sslave">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioInputUSB">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Receive stereo audio from a PC or Mac. Teensy appears as a USB
sound device.</p>
<p align=center><img src="img/usbtype_audio_in.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>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 USB to its 2 output ports.</p>
<!--
<h3>Hardware</h3>
-->
<h3>Examples</h3>
<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; PassThroughUSB</p>
</p>
<h3>Notes</h3>
<p>Arduino's <b>Tools &gt; USB Type</b> menu must be set to <b>Audio</b>.
</p>
<p align=center><img src="img/usbtype_audio.png"></p>
<p>USB input &amp; output does not cause the Teensy Audio Library to
update. At least one non-USB input or output object must be
present for the entire library to update properly.</p>
<p>A known problem exists with USB audio from Macintosh computers.
An imperfect <a href="https://forum.pjrc.com/threads/34855-Distorted-audio-when-using-USB-input-on-Teensy-3-1?p=110392&viewfull=1#post110392">workaround
can be enabled by editing usb_audio.cpp</a>.
Find and uncomment "#define MACOSX_ADAPTIVE_LIMIT".</p>
</script>
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</script>

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<script type="text/x-red" data-help-name="AudioMixer4">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Combine up to 4 audio signals together, each with adjustable gain.
All channels support signal attenuation or amplification.</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>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>
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<script type="text/x-red" data-help-name="AudioOutputAnalog">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Transmit 12 bit audio using Teensy's built-in digital to analog converter.</p>
<p align=center><img src="img/dac_speaker.jpg"><br>
<small><a href="http://www.pjrc.com/store/prop_shield.html" target="_blank_">Prop Shield with 4&ohm; Speaker</a></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>Audio Channel</td></tr>
</table>
<h3>Functions</h3>
<p class=func><span class=keyword>analogReference</span>(ref);</p>
<p class=desc>Configure output voltage range:<br>
<span class=literal>INTERNAL</span> selects 1.2 volt peak-to-peak output.<br>
<span class=literal>EXTERNAL</span> selects 3.3 volt peak-to-peak output.
</p>
<h3>Hardware</h3>
<p align=center><img src="img/dacpin.jpg"></p>
<p>Signal range default is 0 to 1.2V</p>
<p>The output voltage has DC level. Some applications require a DC-blocking capacitor. If unsure, a 10&micro;F is usually a safe value to use. If an aluminum or tantalum capacitor is used, the positive terminal should connect to Teensy's DAC pin.</p>
<p>The DAC pin is used with the
<a href="http://www.pjrc.com/store/prop_shield.html" target="_blank_">Prop Shield</a>
to drive speakers.</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>
<p class=exam><a href="https://github.com/PaulStoffregen/TouchGuitar" target="_blank">TouchGuitar</a>
</p>
<p class=exam><a href="https://community.arm.com/groups/embedded/blog/2014/05/23/led-video-panel-at-maker-faire-2014" target="_blank">LED Video Board</a>
</p>
<p class=exam>File &gt; Examples &gt; OctoWS2811 &gt; VideoSDcard
</p>
<p class=exam>File &gt; Examples &gt; SerialFlash &gt; MP3Player
</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>
<p>When using 3.3V output, the power supply is used for the analog reference. Noise
present on the 3.3V power can couple to the DAC output signal.
</p>
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<script type="text/x-red" data-help-name="AudioOutputAnalogStereo">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Transmit 12 bit stereo audio using Teensy 3.5 or 3.6 built-in digital to analog converters.</p>
<!--<p align=center><img src="img/dac_speaker.jpg"><br>
<small><a href="http://www.pjrc.com/store/prop_shield.html" target="_blank_">Prop Shield with 4&ohm; Speaker</a></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>Audio Channel (Left)</td></tr>
<tr class=odd><td align=center>In 1</td><td>Audio Channel (Right)</td></tr>
</table>
<h3>Functions</h3>
<p class=func><span class=keyword>analogReference</span>(ref);</p>
<p class=desc>Configure output voltage range:<br>
<span class=literal>INTERNAL</span> selects 1.2 volt peak-to-peak output.<br>
<span class=literal>EXTERNAL</span> selects 3.3 volt peak-to-peak output.
</p>
<h3>Hardware</h3>
<p align=center><img src="img/dacpins.png"></p>
<p>Signal range default is 0 to 1.2V</p>
<p>The output voltage has DC level. Some applications require a DC-blocking capacitor. If unsure, a 10&micro;F is usually a safe value to use. If an aluminum or tantalum capacitor is used, the positive terminal should connect to Teensy's DAC pin.</p>
<p>The DAC pin is used with the
<a href="http://www.pjrc.com/store/prop_shield.html" target="_blank_">Prop Shield</a>
to drive speakers.</p>
<h3>Examples</h3>
<!--<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; PassThroughMono
</p>
<p class=exam>File &gt; Examples &gt; SerialFlash &gt; MP3Player
</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>
<p>When using 3.3V output, the power supply is used for the analog reference. Noise
present on the 3.3V power can couple to the DAC output signal.
</p>
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<script type="text/x-red" data-help-name="AudioOutputI2S">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</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>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>
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<script type="text/x-red" data-help-name="AudioOutputI2SQuad">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Transmit quad (4) channel 16 bit audio, using I2S master mode.</p>
<p align=center><img src="img/audioshield_quad_out.jpg"></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>Channel #1</td></tr>
<tr class=odd><td align=center>In 1</td><td>Channel #2</td></tr>
<tr class=odd><td align=center>In 2</td><td>Channel #3</td></tr>
<tr class=odd><td align=center>In 3</td><td>Channel #4</td></tr>
</table>
<h3>Functions</h3>
<p>This object has no functions to call from the Arduino sketch. It
simply streams data from its 4 input ports to the I2S hardware.</p>
<h3>Hardware</h3>
<p>See this Sparkfun blog for <a href="https://www.sparkfun.com/news/2055" target="_blank">how
to connect two audio adaptors for 4 channel audio</a>. More
<a href="https://forum.pjrc.com/threads/29373-Bit-bang-multiple-I2S-inputs-simultaneously?p=79606#post79606" target="_blank">details</a> are also available.
<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 (ch 1+2)</td><td>Output</td></tr>
<tr class=odd><td align=center>15</td><td>TX (ch 3+4)</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 class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; PassThroughQuad
</p>
<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; SGTL5000 &gt; QuadChannelOutput
</p>
<h3>Notes</h3>
<p>Normally, this object is used with two Audio Shields, which
are controlled separately by a pair of "sgtl5000" objects.</p>
</script>
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<script type="text/x-red" data-help-name="AudioOutputI2Sslave">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Transmit 16 bit stereo audio to an I2S device using I2S slave mode
(where the DAC or codec chip, not Teensy, controls audio timing).</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>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>
</script>
<script type="text/x-red" data-template-name="AudioOutputI2Sslave">
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<script type="text/x-red" data-help-name="AudioOutputPT8211">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Transmit 16 bit stereo audio to a low-cost PT8211 DAC chip. 4X oversampling
and filtering are automatically used to improve output quality.</p>
<p align=center><img src="img/pt8211.jpg"></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>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 a PT8211 chip. 4X
oversampling and filtering is automatically used to improve quality.</p>
<h3>Hardware</h3>
<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>BCK</td><td>Output</td></tr>
<tr class=odd><td align=center>22</td><td>DIN</td><td>Output</td></tr>
<tr class=odd><td align=center>23</td><td>WS</td><td>Output</td></tr>
</table>
<p>More information can be found in the PT8211 datasheet.
</p>
<h3>Examples</h3>
<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; PT8211Sine
</p>
<h3>Credits</h3>
<p>Frank Boesing and Benjamin developed this PT8211 object. Details can be
found on this
<a href="https://forum.pjrc.com/threads/29284-Dual-channel-16bit-dac-PT8211/page3" target="_blank">forum disussion</a>.
<h3>Notes</h3>
<p>
</p>
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</div>
</script>

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<script type="text/x-red" data-help-name="AudioOutputPWM">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</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>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 &gt; 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>
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<script type="text/x-red" data-help-name="AudioOutputSPDIF">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Transmit 16 bit stereo audio as Digital S/PDIF.</p>
<p align=center><img src="img/spdif_proto.jpg"></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>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>
</script>
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<script type="text/x-red" data-help-name="AudioOutputUSB">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Send stereo audio to a PC or Mac. Teensy appears as a USB
sound device.</p>
<p align=center><img src="img/usbtype_audio_out.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>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 from it's 2 input ports to the USB.</p>
<!--
<h3>Hardware</h3>
-->
<h3>Examples</h3>
<p class=exam>File &gt; Examples &gt; Audio &gt; HardwareTesting &gt; WavFilePlayerUSB</p>
</p>
<h3>Notes</h3>
<p>Arduino's <b>Tools &gt; USB Type</b> menu must be set to <b>Audio</b>.
</p>
<p align=center><img src="img/usbtype_audio.png"></p>
<p>USB input &amp; output does not cause the Teensy Audio Library to
update. At least one non-USB input or output object must be
present for the entire library to update properly.</p>
<p>A known problem exists with USB audio from Macintosh computers.
An imperfect <a href="https://forum.pjrc.com/threads/34855-Distorted-audio-when-using-USB-input-on-Teensy-3-1?p=110392&viewfull=1#post110392">workaround
can be enabled by editing usb_audio.cpp</a>.
Find and uncomment "#define MACOSX_ADAPTIVE_LIMIT".</p>
</script>
<script type="text/x-red" data-template-name="AudioOutputUSB">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioPlayMemory">
<h3>Summary</h3>
<div class=tooltipinfo>
<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/extras/wav2sketch" target="_blank">wav2sketch program</a>,
and copied to the sketch folder to become part of your sketch.</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>(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>
</script>
<script type="text/x-red" data-template-name="AudioPlayMemory">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
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</div>
</script>

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<script type="text/x-red" data-help-name="AudioPlayQueue">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</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>(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>
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</div>
</script>

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<script type="text/x-red" data-help-name="AudioPlaySdRaw">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</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>
<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>
</script>
<script type="text/x-red" data-template-name="AudioPlaySdRaw">
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</div>
</script>

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<script type="text/x-red" data-help-name="AudioPlaySdWav">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Play a WAV file, stored on a SD card.</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>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>
<p>An experimental SD library optimization exists, which can remove these
SD library restrictions. It also allows reliable playback of more
files at the same time. To enable this special code, find and edit
the SD_t3.h file within your Arduino folder. See the comments within
that file for details.
</p>
</script>
<script type="text/x-red" data-template-name="AudioPlaySdWav">
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</div>
</script>

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<script type="text/x-red" data-help-name="AudioPlaySerialflashRaw">
<h3>Summary</h3>
<div class=tooltipinfo>
<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.
</script>
<script type="text/x-red" data-template-name="AudioPlaySerialflashRaw">
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</div>
</script>

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<script type="text/x-red" data-help-name="AudioRecordQueue">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Record audio data by sending to the Arduino sketch. This object allows
sketch code to receive audio packets.</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>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>
</script>
<script type="text/x-red" data-template-name="AudioRecordQueue">
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<script type="text/x-red" data-help-name="AudioSynthKarplusStrong">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Synthesize a plucked string sound, such as a guitar string.
</p>
<p align=center><img src="img/touchguitar.jpg"></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>noteOn</span>(frequency, velocity);</p>
<p class=desc>Begin a new string note. Velocity can be from 0 to 1.0,
indicating how hard the string is plucked.
</p>
<p class=func><span class=keyword>noteOff</span>(velocity);</p>
<p class=desc>Stop the sound output.
</p>
<h3>Examples</h3>
<p class=exam>File &gt; Examples &gt; Synthesis &gt; Guitar
</p>
<p class=exam><a href="https://github.com/PaulStoffregen/TouchGuitar" target="_blank">TouchGuitar</a>
</p>
<h3>Notes</h3>
<p></p>
</script>
<script type="text/x-red" data-template-name="AudioSynthSimpleDrum">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
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</div>
</script>

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<script type="text/x-red" data-help-name="AudioSynthNoisePink">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</script>
<script type="text/x-red" data-template-name="AudioSynthNoisePink">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioSynthNoiseWhite">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
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<script type="text/x-red" data-help-name="AudioSynthSimpleDrum">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Generate a synthesised drum sound. Also useful for laser pistol and bursting
bubble sound effects.</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>Drum Tone Output</td></tr>
</table>
<h3>Functions</h3>
<p class=func><span class=keyword>noteOn</span>();</p>
<p class=desc>Trigger the drum.
</p>
<p class=func><span class=keyword>frequency</span>(frequency);</p>
<p class=desc>Set the base frequency of the drum.
</p>
<p class=func><span class=keyword>length</span>(milliseconds);</p>
<p class=desc>Set the duration of the envelope, in milliseconds.
</p>
<p class=func><span class=keyword>secondMix</span>(level);</p>
<p class=desc>Emulates a two-headed tom, by adding a second sine wave that is
harmonized a perfect fifth above
the base frequency. Using this involves a slight CPU penalty.
</p>
<p class=func><span class=keyword>pitchMod</span>(depth);</p>
<p class=desc>Set the depth of envelope of the pitch, by a maximum of two octaves.
Default is 0.5, with no modulation. Values above 0.5 cause the pitch to sweep
downwards, values lower than 0.5 cause the pitch to sweep upwards.
</p>
<h3>Examples</h3>
<p class=exam>File &gt; Examples &gt; Synthesis &gt; SimpleDrum
</p>
<h3>Notes</h3>
<p></p>
</script>
<script type="text/x-red" data-template-name="AudioSynthSimpleDrum">
<div class="form-row">
<label for="node-input-name"><i class="fa fa-tag"></i> Name</label>
<input type="text" id="node-input-name" placeholder="Name">
</div>
</script>

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<script type="text/x-red" data-help-name="AudioSynthToneSweep">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Create a continuously varying (in frequency) sine wave</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>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>
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<script type="text/x-red" data-help-name="AudioSynthWaveform">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Create a waveform: sine, sawtooth, square, triangle, pulse or arbitrary.</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>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_SAWTOOTH_REVERSE</span></li>
<li><span class=literal>WAVEFORM_SQUARE</span></li>
<li><span class=literal>WAVEFORM_TRIANGLE</span></li>
<li><span class=literal>WAVEFORM_ARBITRARY</span></li>
<li><span class=literal>WAVEFORM_PULSE</span></li>
<li><span class=literal>WAVEFORM_SAMPLE_HOLD</span></li>
</ul>
</p>
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<script type="text/x-red" data-help-name="AudioSynthWaveformDc">
<h3>Summary</h3>
<div class=tooltipinfo>
<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>
</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>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>
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<script type="text/x-red" data-help-name="AudioSynthWaveformSine">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Create a sine wave signal</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>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>
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<script type="text/x-red" data-help-name="AudioSynthWaveformSineHires">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Create a highly precise, low distortion sine wave signal.
Mainly useful for codec &amp; analog circuitry testing.</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>Sine Wave, upper bits</td></tr>
<tr class=odd><td align=center>Out 1</td><td>Sine Wave, lower bits</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>Notes</h3>
<p>An 11th order Taylor series approximation is used to generate
a very accurate sine wave. At least the upper 25 bits are believe
to be perfect. This is mainly intended for testing 24 bit codec chips!</p>
</script>
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<script type="text/x-red" data-help-name="AudioSynthWaveformSineModulated">
<h3>Summary</h3>
<div class=tooltipinfo>
<p>Create a modulated sine wave, using any audio signal to continuously
modulate the sine wave frequency.</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>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>
</script>
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