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Designing A Pocket Equalizer For Headphone Listening

(A HeadWize Design Series Paper)

In audiophile circles, equalizers are prescribed only as a last resort to correct imbalances
in recordings and room acoustics for loudspeaker playback. The case for using equalizers
with headphones is simply this: headphones by their very nature change the tonal
balance of music. The acoustic shaping that results from sound interacting with a
listener's head and outer ears varies from person to person and is missing in
headphones, which play directly to the eardrums. Therefore, not only does music in
headphones have a different tonal balance than was intended, but each listener hears a
slightly different presentation. If the listening system includes an headphone virtualizer
(such as Dolby Headphone), equalization may help restore the imaging if the
headphones have a different frequency response than what the virtualizer expects.

While equalizers for home stereos and studios are available in range of prices and
features, are there any options for portable users? Koss Corporation makes an
inexpensive, small 3-band equalizer (EQ-30 and EQ-50) for headphones that runs off 2-
AAA batteries, but it has received less than favorable reviews (poor filter action and poor
headphone drive). Some portable stereos have built-in equalizers, which are mostly
inferior gimmicks. These products obviously are not designed with audiophile concerns in
mind. The best way to get a high quality portable equalizer in these times of audiophile
minimalism is to build one. This article is a collection of opamp-based equalizer designs
that are all suitable for portable use (except for one vacuum tube circuit, included for the
benefit of glass audio fans). Appendix 1 describes a DIY project: building a mini tilt
equalizer/amplifier for headphones. Appendix 2 shows how to use OrCAD circuit
simulation software to model and customize equalizers.

EQUALIZATION AND HEADPHONES

In addition to correcting tonal imbalances in headphones and recordings, equalizers can,
to a certain extent, alter the perceived spatial characteristics of headphones. Headphone
acoustic simulators, such as virtualizers, electronically process stereo and multi-channel
signals so that they image outside the listener's head - as though they were being
projected by real loudspeakers. When an acoustic simulator is not available, equalization
can mimic some aspects of normal hearing. For example, Ron Cole and John Sunier at
the Binaural Source recommend a "biophonic" curve (shown above) as a guide to setting
equalizers to correct for ear canal resonance and other differences in the spectrum
between speaker listening and headphone listening. (See Taking Sound in Another
Direction for more information.) The curve suggests boosting three center points and is
the basis for this article's focus on 3-band equalizers.

Another reason for equalizing headphone sound has to do with human loudness
perception. When listening at safe volume levels, headphone sound may not be as
satisfying because the perception of loudness is frequency and volume dependent. As
shown in the Fletcher-Munson curves above, human hearing does not become "flat" until
the SPLs are at the threshold of pain. Since the curves change shape at different
loudness levels, accurate loudness compensation would have to be dynamically adjusted
to avoid overcompensation - an unlikely feature in a rack-mounted commercial
equalizer, let alone in a hand-built pocket EQ. Nevertheless, a small treble and bass
boost can improve the perceived frequency balance of low volume headphone sound.
 PASSIVE 2-BAND BAXANDALL

The passive circuit in figure 1 is a variation of the famous Baxandall circuit. It features a
smoothly increasing ±6 dB/octave slope of boost or cut. The bass and treble filters have
a shelving response, although the "shelves" are outside the audible range with these
component values. The filter equations in figure 1a predict the threshold and shelving
frequencies. When the bass control is rotated for maximum boost, the wiper shorts out
the .033uf capacitor. R3 and C4 form a frequency dependent voltage divider that
determines the shelf frequency of the boost. When the treble control is set for maximum
boost, the wiper bypasses the 10K resistor. C1 and R2 form a high pass filter.

In addition to simplicity, the circuit uses commonly available parts. Even the 100K log
taper pots, which are usually hard to find, are sold by Radio Shack as part number 271-
1732 (in a stacked configuration for stereo). At the midway point, the pot will have
about 10K on one side and 90K on the other. For the treble control, the side with 10K
parallels R2. For the bass control, the side with 10K parallels C4. Note that this circuit
could be divided for an individual bass or treble control (R5 may then be omitted - it
helps isolate the bass from the treble circuit when the two are put together).

The passive Baxandall must see a low impedance source and drive a high impedance
load to avoid loading the network and affecting the response curves. Low impedance
sources include the output of a portable stereo or preamp. It can drive a headphone
amplifier or high impedance headphones (possibly with a series resistor to increase the
load impedance), but typical low power portables may not be able to drive headphones
to adequate volume. Since passive networks are not amplifiers, the gain is simulated by
dropping the "flat" response 20dB below the input, for which a portable stereo may not
be able to compensate.

The active version of the 2-band Baxandall shown in figure 1b incorporates the
frequency shaping circuit into the feedback loop of the opamp. The feedback network is
almost identical to the passive Baxandall. The pots are now linear taper, and the double
capacitors in the bass and treble sections have been simplified to one per section. The
input stage serves as both an impedance buffer and as a phase correcter, so that the
output of the EQ is in phase with the input. The equations for this circuit are shown in
figure 1c. Unlike the passive version, this equalizer has true voltage gain: ±10 (±20dB).

ACTIVE 3-BAND BAXANDALL

A 2-band equalizer is limited in its ability to correct trouble spots in headphones. The
circuit in figure 2 is an active version of the Baxandall with an adjustable resonant filter
acting as a midrange control. Again, the inverting buffer at the input ensures that the
equalization stage sees a low source impedance and that the output phase of the
equalizer is non-inverting. The total gain/cut for any band is ±20dB.

As with the 2-band version, the treble and bass filters are shelving equalizers (figure 4) -
although the "shelves" occur outside the audible range with these component values. A
better shelving characteristic can be obtained by moving the shelving frequencies closer
together. With small modifications to the center frequencies, this circuit could generate a
broad facsimile of the biophonic curve. One important characteristic of the biophonic
curve is the "dip" at 7.5kHz to simulate the ear canal resonance of normal hearing. If
diffuse-field headphones are used, they probably already have a response curve with the
ear canal resonance compensation, and then the EQ will help adjust it for the individual
listener.

The treble and mid bands then are the most critical. They must be spaced far enough
apart to simulate the response dip at 7.5kHz. Moving the treble shelving frequency
higher to 20kHz will help. Because the operating range of the midrange resonant filter
overlaps with those of the bass and treble controls, no simple equations can describe
how changes in component values will exactly affect the response curves. Applying the
equations for the 2-band version to the 3-band, they appear to predict the bass and
treble crossover frequencies on the 3-band, but not the gain for the treble control.
The midrange control parameters must be set by experimentation. These following
changes are designed to have minimal impact on the bass and treble responses:

 R4 affects the overall midrange gain/cut. Increasing R4 will reduce gain and shift
the midrange center frequency higher. Decreasing R4 will raise the gain and set
the center frequency lower.
 Changing C4 and C5 will shift the midrange center frequency without affecting the
bass or treble. To shift the center frequency higher (lower), decrease (increase)
C4 and C5 in proportion such that C5 = 5C4. The frequency shift is proportional
to the shift in the value of C5: f1/f = C51/C5 - where f1 and C51 are the new
values.

This design has the virtue of being a single IC project (with the quad version of the
opamp), uses easier-to-find linear-taper pots, and is free of inductors. R1 and R2 are
chosen for unity gain, but the circuit can also function as a headphone amplifier, if the
R2/R1 feedback ratio is made greater than 1. For example, setting R1 to 10K ohms (and
C1 to 10uF) would give the equalizer a "flat" gain of 10. Then a volume control in front
of the inverter would be a good idea. In selecting the opamp, it should have a very high
slew rate for high bandwidth and be able to run off standard battery voltages.

Most quality, FET-input opamps will work (such as the LF353 from National
Semiconductor) when run from two 9-volt batteries or even one 9V battery configured as
a virtually grounded dual supply (see figure 10 below). The Burr-Brown OPA132 or
OPA134 is especially attractive, with a 20uV/sec slew rate, no phase inversion and a
power supply as low as ±2.5VDC. For the active equalizers circuit shown in this article,
the recommended battery supply is ±9V. Smaller power supplies will likely cause an
equalizer to clip frequently because of the high gains involved.

BANDPASS EQUALIZERS

Although the Baxandall circuits are good for general purposes, there are discriminating
headphone connoisseurs who insist on doing the least sonic harm when applying
equalization. In particular, the shelving characteristic of bass and treble bands can make
recordings sound bass-heavy or excessively noisy (noise is less of a problem when
listening to digital recordings). Parametric equalizers have a bandwidth control (also
called a Q control) for each band to further narrow the response of a resonant filter.
Since each band has a minimum of 3 adjustments (gain, frequency and bandwidth), a 3-
band parametric would have at least 9 controls and a barrel of parts and is impractical to
house in a truly pocket-sized enclosure.

The bandpass filters of an octave equalizer can be preset with a high Q. A 3-band
graphic equalizer with high Q bands is rare, because 3 bands usually do not offer enough
flexibility when the filter action is narrow. However, there are situations where a 3-band
graphic equalizer might be adequate, such as when the EQ is application specific and the
center frequencies are known in advance. Again, the number of parts to build a high Q
3-band equalizer is probably more than can fit into a small enclosure (although there are
"graphic EQs on a chip" that might work).

For the purposes of improving headphone listening, such complexity is not mandatory.
The effect of a resonant filter with standard 6 dB/octave slopes (figure 5) is broad only
at maximum gain. Thus, the action of the filter can be confined by not setting the gain to
extremes.
The Wien bridge equalizer in figure 6 uses dual resonant filters to isolate the effect of the
bass and treble controls to the center frequencies. A voltage divider at the input sets the
signal level into the filter which is then varied by feeding back the bandpass response
into the differential input of the opamp. The per-band gain is ±9 dB. Each bandpass
consists of a pair of RC filters, so the slope of boost or cut is 6 dB/octave on either side.
Even when the controls are set at maximum gain or cut (figure 7), the filters have little
or no effect on the low, middle and high portions of the audio spectrum.

Going back to the active Baxandall, that circuit's midrange resonant filter could be
duplicated for the bass and treble bands to target the audio spectrum more specifically.
To avoid the problem of overlapping bandpasses complicating the design of a single EQ
stage, each band should have its own EQ gain block. Unfortunately, this strategy
increases the number ICs (2 quad and 1 dual IC) but the whole circuit should still fit into
a pocket-sized enclosure (although the pocket may be a little bit on the large side).
The 3-band resonant equalizer in figure 8 ensures that the EQ stages see a low source
impedance by buffering the input signal via a non-inverting voltage follower. (To make
this circuit an amplifier/EQ, change the voltage follower to a non-inverting amplifier.)
Each EQ stage (figure 9) is configured with a low, mid or high center frequency and a
±10 dB boost/cut. The outputs from these EQs are then mixed back at unity gain for an
overall gain of 3. This mixing stage is another point where gain can be added to the
system. Since each EQ stage and the summing stage are inverting, the equalizer output
has the correct phase.

Each EQ stage is a combination of low frequency and high frequency shelving filters. In
fact, C1 and C2 can be alternately switched out to restore the circuit's low or high
frequency shelving characteristic. The center frequency and gain of a resonant filter are
set by overlapping the responses of the low and high pass filters. To design the filter,
first calculate the C1/C2 ratio by solving equation 1 (figure 9) for a given maximum
voltage gain (say 3 or about 10dB). Solve for C1 as a proportion of C2 and substitute
this equivalent for C1 in equation 2. Then choose a center frequency and solve for C2 in
microfarads. The shown C values set the center frequencies suggested by the biophonic
curve. BEFORE BUILDING WITH THESE C VALUES, check out the sound with an existing
equalizer.

Note that equation 1 for calculating gain does not work if the design attempts to attain
the same level of gain as a single-sided shelving EQ. For example, the maximum gain of
this circuit configured as a shelving EQ (by removing one of the capacitors) is
approximately (R2 || R3)/(R1 || R2) which comes to about 10 or 20dB. However, solving
for C1/C2 by plugging "A = 10" into equation 1 results in a negative ratio for C1/C2.
These equations limit the maximum gain to about 8 or 18dBs - which is plenty for most
applications.

TILT EQUALIZER

The equalizer circuits shown so far are of the traditional type: they divide the audio
spectrum into bands with a separate gain control for each band. The audiophile
community has decried the use of such EQs, because they are difficult to set properly
without introducing coloration into the audio signal - especially in the midrange. In the
early 1970s, Quad Ltd. believed that the proper role of equalization was to fix subtle
tonal flaws in the audio system. They developed a "tilt" tone control, which first
appeared on their model 34 preamplifier. Unlike the Baxandall controls, the tilt control
"tilts" the frequency content of the audio signal by simultaneously boosting the treble
and cutting the bass frequencies or vice-versa. The effect is subtle because the control
has a maximum boost/cut of 3dB.

Figure 10
The tilt EQ can be very beneficial for correcting tonal flaws in headphone sound, such as
excessive brightness or darkness, without being too sonically obtrusive. The above
schematic (figure 11) is from a preamplifier design by Reg Williamson and Alan Watling.
It is a tilt control with a center frequency of 900Hz and a maximum boost/cut of 6dB.
The circuit produces a shelving characteristic on either side on the center frequency (see
the graph below). When the pot is in the center position, the EQ's response is flat (the
bypass switch takes the EQ filter out of the audio path entirely). Turning the control to
the left (counter-clockwise) lightens the sound; to the right (clockwise) darkens it. Note
that at the extreme settings, the tilt EQ does result in a phase inversion.

Figure 11

TUBE-BASED EQUALIZERS

Although vacuum tube EQs can hardly be "pocket-sized," they are prized by tube audio
aficionados and sound professionals especially. All of the active EQs described so far use
solid state opamps. Although not well known, vacuum tube opamps such as the
Philbricks were available back in the Golden Age of tubes, when analog computers
required their precision. Designing an Opamp Headphone Amplifier includes an
introduction to building tube opamps (theory and schematics). These devices will work
very well with the resonant filter EQ designs shown in figure 10. In some cases,
however, it may be necessary to scale the feedback resistors up or down to meet circuit
preferences.
 Figure 12

The circuit in figure 9a is a biophonic EQ that contains a simple tube opamp made of an
input differential pair and a 12AT7 follower (the 33K resistors and the pots form the
feedback network). Frequency shaping is accomplished with 3 triode-based resonant
filters that vary the amp gain (each band can be adjusted ±12dB). The center
frequencies can be approximated from equation 2 in figure 9 with these modifications:
C1 = C2 and R1 = 10K. Only 6 tubes are needed: 5 12AX7 and 1 12AT7. The filament
voltages are either 6.3VAC at 4A or 12VAC at 2A (twist the filament wires for lowest
hum). The dual 150VDC supply should be able to supply 60mA and does not need to be
regulated. The THD is less than 0.001% at 1V RMS.

Appendix 1: Build a Pocket Tilt Equalizer/Headphone Amplifier

Of all the circuits listed in the article, the tilt equalizer may be the most acceptable to
audiophiles in terms of the simple circuit topology and the subtlety and effectiveness of
its action. Tom Edney built a pocket version of the tilt equalizer, which he modified to
drive his low-impedance Grado headphones. This section describes Edney's modifications
to the tilt EQ circuit and provides construction details.

Edney built the tilt EQ to compensate for the brightness in his Grado SR-60 headphones.
He writes:

A couple of months ago when I got a pair of Grado SR-60s (my first headphones
costing more than $30), I was immediately struck by how bright they are. I'd
heard that Grados were bright when I had been shopping for headphones, but not
being an audiophile, I didn't really understand the significance of "brightness". I
tried burning them in for about 60 hrs, but they showed only slight improvement.
The more I listened to them, the more displeased I was. The bass was lacking,
and the treble was way to prominent for my taste. This was aggravated by the
fact that I listen at very low volume, and as the Fletcher-Munson Loudness
Curves diagram indicates, the loss of low frequencies is affected more strongly
than other portions of the audio spectrum at low volumes.

So because of the combination of bright headphones, low volume, and personal

preference, I had very fatiguing headphones. I could only go for about 30-60
minutes at a stretch before I had to stop using them. I considered returning the
headphones, but first I looked around on the HeadWize site for an answer, and
found it in the Tilt Equalizer.
 Figure A1

Figure A1 is the circuit that Edney used. It has a direct bypass, and R3 is omitted.
Because Edney's headphones are low impedance Grados, they need more current drive
than the OPA134 can output when the EQ's bass response is boosted. Thus, the output
of the EQ section is buffered with a Burr-Brown BUF634 to provide high current drive for
low impedance headphones. The feedback capacitors were changed to 2200pF to move
the center frequency of the EQ up to about 2.3kHz. Edney based this change on listening
tests with his Windows Media Player equalizer. He says:

Before building the Tilt EQ, I tested the idea using Windows Media Player with its
10-band equalizer. I played around a bit with the sliders until I got the sound I
wanted, and saw that they formed an almost perfectly straight negative slope
crossing at the 7th band from the left. Bingo! I knew then that the Tilt EQ was
what I needed. Although I hadn't worked on a DIY electronics project since high
school (over 20 years), the circuit diagram looked simple enough that I decided
to give it a shot.

The [original] Tilt EQ circuit seems to underboost the bass (and over-reduces
treble) compared to my settings on the Windows Media Player EQ. I wondered if
that was because the Tilt EQ's pivot point is at 900 Hz, and the 7th band on the
software EQ is about 2 KHz. I just tried replacing the 5600pF caps (C3/C4) with
2200pF caps, which should put the pivot point at around 2KHz. At first there
didn't seem to be a huge difference, but after a while I found I could dial in a
satisfactory sound with the 2200pF, where I couldn't with the 5600pF. Then I
took the CD I was listening to and put it into my computer to see how my
Windows Media Player EQ settings sounded. They were identical. Now the ratio of
bass to treble seems higher, which is what I was looking for.
 Figure A2

Edney increased the gain of the first stage to 1.5, so that the bypass volume level would
be the same as the flat EQ volume level, when connected to his Casio PCDP. Normally,
gain matching is not required as all the stages of the original EQ have unity gain, and
the output of the EQ should be at the same level as the input. However, when he hooked
the tilt EQ to another source, the volume levels became unmatched again.

Figure A2 is an alternate circuit which functions as both an equalizer and headphone

amplifier. The bypass only switches out the EQ network, so the volume level is the same
regardless of the bypass setting (R3 is required). The input stage has been configured
for a gain of 2. The value of the volume control should not be higher than 10K ohms
(audio taper). Otherwise, the pot's impedance could load the feedback network and
change the gain of the first stage. Since the EQ is powered by two 9V batteries (see
below), the gain of the first stage should be no higher than about 4 to avoid clipping
when the EQ's tilt is at full bass or treble boost.

Figure A3

Figure A3 is the power supply. The two 9V batteries in series connect to an RC network
to create a ±9V output with a virtual ground. A single 9V battery will not provide enough
headroom for the EQ's output at full bass or treble tilt. (For more information about
virtually-grounded supplies, see Designing an Opamp Headphone Amplifier.) Edney
decoupled the power supply pins on each of the ICs with a pair of 10uF electrolytic and
0.01uF ceramic disc capacitors.
 CONSTRUCTION

The tilt control pot is a dual, linear type. Low value capacitors, such as the 2200pF, can
be hard to find in the preferred audiophile film types. Silver mica (or dipped mica)
capacitors commonly have this value, but can be expensive; however, many surplus
electronic outlets sell them at very reasonable prices (for example, All Electronics has
dipped micas for about $0.35 each). If film or silva mica types are not available or are
not affordable, NPO-type ceramic capacitors can be used. NPO ceramic capacitors are
more stable than other ceramic types.

The PACTEC enclosure holds two 9V batteries. The bypass toggle switch is mounted on
one side of the enclosure. The illuminated power switch (with green LED) is mounted on
the other side. The circuit board is a protoboard from Radio Shack. Edney listed the
steps that he followed to build the equalizer:

 I successfully breadboarded one channel without the BUF. Then soldered it to the
circuit board. Audio worked and so did the pot.
 Then I breadboarded a BUF634 to the one channel with success (good audio, pot
worked). I did not solder the BUF to the circuit board at that point.
 I soldered the second channel to the circuit board, no BUF. That channel worked,
and then I checked it with a breadboarded BUF. It worked fine.
 I added both BUFs to the circuit board.

I built the whole thing with solid wire I'd had on hand, but while trying to fit the
EQ into the enclosure I broke off a number of wires. I had to completely rewire
the EQ with stranded wire to keep this from happening. Obviously a mistake only
a rookie would make.

The dual 50K pot was almost impossible to find, especially in the small 16mm
configuration. I finally found it at Main Electronics in Vancouver, B.C. (part
number 08-1755). Excellent, fast service. I mounted it with the middle of the
rotation at 9 o'clock, so that it would work much like the picture of the Quad Ltd.
Tilt EQ shown in the article. Since my knob is round, I considered adding a
cosmetic straight line (instead of just the little arrow) to the knob to more
intuitively reflect the slope of the tilt. I may still do that.

My biggest problem was that my headphone jacks were *mono*, not stereo.
Because they were switched they had three leads, and I trusted the guy at the
electronics supply store, when he told me they were stereo. I was getting no
audio, apart from a clicking sound in the left ear at the rate of about 2 per
second. Anyway, I've got stereo jacks now and verified that it's all functioning
properly. The darn thing works!

The Results

Because Edney's EQ does not have a volume control, the audio source must have its own
volume control. He had no trouble using the headphone output of his Casio PCDP, but
the EQ generated hiss when connected to the headphone jack of his receiver. He says:

There is no discernable loss in sound quality with the Tilt EQ. I find that for most
CDs I set the Tilt EQ to about 10 o'clock (9 o'clock is flat/no change). This small
change warms up the audio just enough to lose the fatiguing brightness of the
Grados.

Since I'm not using a headphone amp, I plug the EQ into the headphone jack on
my PCDP rather than the line-out so I can use the built-in volume control of the
PCDP. That is what I matched the volume to. I'm very pleased with the results. I
really like being able to dial in the brightness in response to the music, the
headphones, and the audio source (PCDP, Walkman, portable radio).

I consider this EQ to be just barely pocket size. It is quite portable and is

perfectly suited to my desk at work, but for use in a pocket, I would make at
least two changes:

1. try to make it even smaller (using the quad version of the opamp,
eliminate the BUF634 if possible, and use smaller power and bypass
switches or eliminate the bypass altogether),
2. put both the input and the output jacks on the front next to the knob. This
would fit better in a pocket than having the input coming in on the side of
the unit the way I've done it.

Someday soon I hope to change the bypass so that it includes the input stage, as
the original article intended. The true bypass seemed at the time to be the best
way since it completely removed the EQ, but gain mismatch is a real problem
when I switch audio sources.

Appendix 2: Simulating a Tilt Equalizer in OrCAD PSpice

Circuit simulation software can be invaluable in designing or customizing equalizer

circuits. With complex EQ networks, this process is, at best, a hit-or-miss proposition for
hobbyists who do not have audio test equipment like a signal generator and oscilloscope,
and is still not easy for those who do. EQ networks can quickly become too complex to
describe with simple equations. In multi-band equalizers, the action of each band can
influence the responses of the other bands. Again, there may be no simple equations to
represent the overall response of a multi-band equalizer. While not always accurate,
predictions of the frequency response from simulations can guide the design process.

This section discusses how to use OrCAD Lite circuit simulation software to simulate the
performance of the tilt equalizer. OrCAD Lite is free and the CD can be ordered from
Cadence Systems. At the time of this writing, OrCAD Lite 9.2 is the latest version. OrCAD
Lite 9.1 can be downloaded from the Cadence website (a very large download at over
20M) and should work as well. There are 4 programs in OrCAD suite: Capture, Capture
CIS, PSpice and Layout. The minimum installation to run the amplifier simulations is
Capture (the schematic drawing program) and PSpice (the circuit simulation program).

Download Simulation Files for Tilt Equalizer

Download OrCAD Burr-Brown Simulation Libraries

After downloading tilteq_sim.zip and orcad_bb.zip, create a project directory and unzip
the contents of the tilteq_sim.zip archive into that directory. Then extract the contents of
the orcad_bb.zip archive into the <install path>\OrcadLite\Capture\Library\PSpice
directory. The files burr_brn.olb and burr_brn.lib are libraries containing simulation
models for several popular Burr Brown opamps, including the OPA134 used in this
equalizer.

Note: The Burr-Brown libraries contain some very large models (such as the BUF634)
that will not run in OrCAD Lite.

The two basic types of simulation included are frequency response (AC sweep) and time
domain. The time domain analysis shows the shape of the output waveform and can be
used to determine the harmonic distortion of the circuit's output or to visually inspect
the waveform for anomalies like clipping. They both run from the same schematic, but
the input sources are different. For the frequency response simulation, the audio input is
a VAC (AC voltage source). The time domain simulation requires a VSIN (sine wave
generator) input. Before running a simulation, make sure that the correct AC source is
connected to the amp's input on the schematic.
The following instructions for using the simulation files are not a complete tutorial for
OrCAD. The OrCAD HELP files and online manuals include tutorials for those who want to
learn more about OrCAD.

Frequency Response (AC Sweep) Analysis

1. Run OrCAD Capture and open the project file "tilt_eq.opj".

2. In the Project Manager window, expand the "PSPICE Resources|Simulation
Profiles" folder. Right click on "Schematic1-freq_resp" and select "Make Active."
3. In the Project Manager window, expand the "Design
Resources|.\tilt_eq.dsn|SCHEMATIC1" folder and double click on "PAGE1".
4. On the schematic, make sure that the input of the amp is connected to the V3 AC
voltage source. If it is connected to V4, drag the connection to V3. By default, V3
is set to 0.5V.
5. To add the Burr-Brown library to the Capture: click the Place Part toolbar button (
 ). The Place Part dialog appears. Click the Add Library button. Navigate to the
burr_brn.olb file and click Open. Make sure that the analog.olb and source.olb
libraries are also listed in the dialog. Click the Cancel button to close the Place
Part dialog.
6. From the menu, select PSpice|Edit Simulation Profile. The Simulation Settings
dialog appears. The settings should be as follows:

Analysis Type: AC Sweep/Noise

AC Sweep Type: Logarithmic (Decade), Start Freq = 10, End Freq = 100K,
Points/Decade = 100

7. To add the Burr-Brown library to PSpice: Click the "Libraries" tab. Click the
Browse button and navigate to the the burr_brn.lib file. Click the Add To Design
button. If the nom.lib file is not already listed in the dialog list, add it now. Then
close the Simulation Settings dialog.
8. To display the input and output frequency responses on a single graph, voltage
probes must be placed on the input and output points of the schematic. The
probes should already exist on the schematic. If not, here's how to add them:
Click the Voltage/Level Marker ( ) on the toolbar and place a marker at the
junction of R3a, R4a and C3. Place another marker just above RLoad.
9. OrCAD does not have a functional model of a potentiometer. R5a and R5b
represent a 50K-ohm pot. When R5a = R5b = 25K ohms, the pot's wiper is at the
center. To "rotate" the pot to a clockwise or counter-clockwise position, make R5a
<< R5b or R5a >> R5b, but in all cases, the sum of these resistors must total
50K. For example, in the schematic shown above, the equalizer is set for a bass
tilt with R5a = 1K and R5b = 49K.
10. To run the frequency response simulation, click the Run PSpice button on the
toolbar ( ). When the simulation finishes, the PSpice graphing window
appears. The input and output curves should be in different colors with a key at
the bottom of the graph.
 11. The horizontal axis of the graph does not have adequate markings to determine
center frequency of the tilt by eye. To display the PSpice cursor, select
Trace|Cursor|Display from the PSpice menu. A vertical cursor line appears on the
graph, and the Probe Cursor window appears over the graph. Drag the cursor to
the point where the output curve (shown here in green) intersects with the input
curve (shown here in red). Then, the exact coordinates of the intersection are
shown in the Probe Cursor window on the A1 line. The first number (2.3101K) is
the center frequency. The second number (500mV) is the corresponding voltage
for that frequency.

Time Domain (Transient) Analysis

1. On the Capture schematic, make sure that the input of the amp is connected to
the V2 sinewave source (the default values are: VAMPL=0.5, Freq. = 1K, VOFF =
0). If it is connected to V3, drag the connection to V2.
2. In the Project Manager window, expand the "PSPICE Resources|Simulation
Profiles" folder. Right click on "Schematic1-transient" and select "Make Active"
3. From the menu, select PSpice|Edit Simulation Profile. The Simulation Settings
dialog appears. The settings should be as follows:

Analysis Type: Time Domain(Transient)

Transient Options: Run to time = 10ms, Start saving data after = 0ms,
Max. step size = 0.001ms
 4. To display the input and output waveforms on a single graph, voltage probes
must be placed on the input and output points of the schematic. If the probes are
not already on the schematic, follow the procedure in step 8 of the Frequency
Response Analysis to add them.
5. To run the time domain simulation, click the Run PSpice button on the toolbar (
). When the simulation finishes, the PSpice graphing window appears. The
input and output curves should be in different colors with a key at the bottom of
the graph.
6. To determine the harmonic distortion at 1KHz (the sine wave frequency),
harmonics in the output waveform must be separated out through a Fourier

Transform. In the PSpice window, press the FFT toolbar button ( ). The PSpice
graph changes to show the harmonics for the input and output waveforms. The
input and output curves should be in different colors with a key at the bottom of
the graph.
7. The fundamental frequency at 1KHz will have the largest spike. The other
harmonics are too small to be seen at the default magnification. In the PSpice

window, press the Zoom Area toolbar button ( ) and drag a small rectangle in
the lower left corner of the FFT graph. The graph now displays a magnified view
of the selected area. Continue zooming in until the harmonic spikes at 2KHz,
3KHz, etc. are visible.
8. Harmonic spikes should exist for the output waveform only. The input is an ideal
sine wave generator and has no distortion. To calculate total harmonic distortion,
add up the spike values (voltages) at frequencies above 1KHz and divide by the
voltage at 1KHz (the fundamental).

Additional Simulation Tips

 To change the value of any component on a schematic in the Capture program,

double-click on the value and enter a new value at the prompt.

Note: Simulations only approximate the performance of a circuit. The actual

performance may vary considerably from the simulation as determined by a number of
factors, including the accuracy of the component models, and layout and construction
techniques.

References:

Berlin, Howard, Design of Operational Amplifier Circuits (1984).

Berglund, Rickard, "Ultra-Low-Distortion Graphic EQ," Glass Audio, 6/96, p. 40.

Burr-Brown Applications Notes (OPA132, OPA134).

Graf, Rudolph Encyclopedia of Electronic Circuits, Vols 1-6 (1985-1996).

Horowitz, P. and Hill, W. The Art of Electronics (1989).

Jung, W. Audio IC OpAmp Applications (1988).

Meiksen, Z. H. Complete Guide to Active Filter Design, Opamps, and Passive

Components (1990).
 National Semiconductor Application Notes (LF353, LM349).

National Semiconductor Audio/Radio Handbook (1986).

Texas Instruments Application Notes (TLE2426).

Williamson, Reg and Watling, Alan, "A New Control Preamp," Audio Amateur,
4/91, p. 10.
For the latest updates, see the Project Addendum.

c. 1998, 2000, 2001, 2002 Chu Moy.

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