<h2> MUS250a,  lab for January 26, 2005 - Max Mathews</H2>

<p>Due Wednesday, Feb. 9</p>

<p>The <A href="opamp-circuits">information sheets passed out in class today</a> contain diagrams
for the op amp and filter circuits, many of which were demonstrated in
class today.  The list given below starts with simple filter circuits
and proceeds to more complicated circuits.  In the lab, make and
evaluate one filter circuit, the virtual ground circuit, and one
op-amp circuit.  The evaluation should show that the circuit actually
works.  

<p>Different circuits will require different evaluations.  Most
evaluations will involve putting an input into the circuit and
observing the output from the circuit.

<p>To receive credit for this lab, show all three working circuits to
a TA and turn in a lab writeup.  You may do any number of additional
Op-amp circuits for extra credit.


<h4>Input</h4>

<p>The input may be an audio signal from a signal generator or a
sensor signal from an FSR.  The simple pd patch <tt>signal-generator</tt> can
make a sine wave or white noise.

<p> To get the sound output from a Linux machine onto your protoboard
we have made a collection of cables that go from 1/4" unbalanced on
one end to a pair of bare wires on the other end.  Plug the 1/4"
unbalanced end into the back of the "omni i/o" audio interface, to the
output labelled "L monitor out."  Now you can plug one of the bare
wires into ground on your protoboard and the other into the input of
whatever circuit you want to test.

<h4>Output</h4>

<p>Ideally you should look at the output of your circuit on an
oscilloscope.  Since we have only two scopes you will have to take
turns.

<p>Another option would be to look at the output on a multimeter;
this will tell you the overall amplitude but will not give any sense
of the shape of the signal.  

<p>You can also wire the output of your filter to the speaker on your
board; this will be too quiet to hear the sine wave without
amplfication (which you can do with an op-amp for extra credit), but
it should be good enough to hear the effect of your filter on white
noise.

<h3>Filter circuits:</h3>

<ul>
<li>Audio low pass filter<br>
	requires audio inputs and outputs you can hear </li>

<li>Audio high pass filter<br>
	requires audio inputs and outputs you can hear</li>

<li>Sensor super low pass filter<br>

       	this filter is so slow you can watch the response to a step input 
	 on a multimeter.  The step input is obtained by pushing a button
	switch in the circuit.  What happens if you remove the resistor Rx?</li>
</ul>

<h3>Virtual Ground circuit</h3>

<ul>
<li>Circuit to make a virtual ground<br>

	You will need the virtual ground for all of the op-amp circuits below.
	If V++ is 13v, the virtual ground voltage should be 6.5 volts.
</li>
</ul>


<h3>Op-amp circuits</h3>

<ul>
<li>Simple square wave oscillator

  <ul>
  <li>select R and C to get about a 100Hz frequency</li>
  <li>observe the waveform on a scope. What kind of waveform is it? </li>
  <li>Listen to the waveform</li>
  <li>substitute an FSR for R. squeeze the FSR and listen to the result</li>
  </ul>
</li>

<li>Test circuit for positive input amplifier
<ul>
  <li>use the circuit with the LF353 and the LM6132  op amps</li>
  <li>observe the dc response by turning the pot</li>
  <li>what is the gain of the circuit?</li>

  <li>What happens to the output when you turn the pot to zero with
  the LF353 and with the LM6132?</li>
</ul>
</li>

<li>Fast rise, slow decay filter for pizzo  sensor drum stick

<ul><li>
This circuit converts the audio frequency pulse from a snare drumstick
to amplitude envelope similar to a piano timbre envelope. The envelope
can be sampled by the atmel a-d converters.  Use the stick as a
controller.  Use a scope to look at the signal out of the first op amp
and the signal out of the second op amp.  A diode is used to measure
the envelope of the audio.
</li></ul>

<li>Rubber band FSR therimin

<ul>

<li>
Two FSR's are used to control the frequency and amplitude of this
sound wave generator.  The use of rubber bands gives a force feedback
to the player which may make the instrument easier to play.</li>

<li>FSR 2 controls the frequency of the square wave oscillator.  FSR 2 is a
volume control & timbre control. The capacitor C1 in combination with
the FSR 2 makes a variable low pass filter that increases the high
frequency content of the sound as it gets louder.  The right hand two
op amps add second low pass filter which smooths sharp corners in the
waveform.  Try playing the therimin.  Try removing C1 in which case
only the amplitude of the waveform will be controlled by FSR 2.  Try
removing C2 and see what this does to the timbre.
</li>
</ul>


</li>
</ul>

<h3>Filtering in Pd</h3>

<p>For extra credit, learn about Pd's audio filter objects (hip~, lop~, bp~,
biquad~, and vcf~) and make a cool patch that uses them.  

</li>