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Active Filters

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0% found this document useful (0 votes)
13 views20 pages

Active Filters

Active filters

Uploaded by

Saad 1
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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- - - - - E x p e r i e n c e i n y o u r o w n r o o m t h e m a g i c a l n a t u r e o f

s t e r e o s o u n d - - - - -

| Build-Your-Own | Main Panel | Dipole Woofer | Crossover/EQ | Supplies |


| System Test | Design Models | Prototypes | Active Filters | Surround | FAQ |

Active Filters
Here is a catalog of line-level circuits that I have found useful for building active
loudspeakers. Many other topologies are possible, but one should always analyze
What's new
a circuit's signal handling capability and its contribution to overall system noise
before choosing it. A CAD software package such as CircuitMaker is most
convenient for analyzing and designing active filters. LspCAD software allows you
to see how an active filter changes the measured frequency response of a driver
LX - Store and lets you optimize it to a target response. All the line level filters below are
included in LspCAD standard and professional versions. Component values for all
the filters below and for a dual power supply can be determined from a circuit
design spreadsheet contributed by Bernhard Faulhaber. It covers more cases
than the earlier spreadsheet by Alister Sibbald.
BLACKLIST
1- Buffer stage
2- 12 dB/oct Linkwitz-Riley crossover
3- 24 dB/oct Linkwitz-Riley crossover
4- Delay correction
Conversations 5- Shelving lowpass & passive circuit
with Fitz 6- Shelving highpass & passive circuit
7- Notch filter
8- 6 dB/oct dipole equalization
9- 12 dB/oct highpass equalization ("Linkwitz Transform", Biquad)
OPLUG 10 - Variable gain & fixed attenuation
Forum 11 - Line driver
12 - Power supply
13 - Printed circuit boards
14 - Literature

Basics

The Magic in 2- -----------------------------------------------------------------


Channel Sound

Issues in speaker
design
1 - Buffer stage
Dipole models

Active filters

Amplifiers etc

Microphone

FAQ's

Loudspeakers

Crossovers A buffer as the first stage of an active crossover/equalizer provides the necessary
low source impedance to the following filter networks. The buffer also provides a
Room acoustics high impedance load to the preamplifier output circuit and the option of a
highpass filter for dc blocking. (w-xo-lp2.gif, pmtm-eq1.gif, 38xo_eq.gif) Top
Stereo Recording
and Rendering

Audio production
2 - 12 dB/oct Linkwitz-Riley crossover
Conclusions

Projects

Your own desig

LXmini

LXmini+2

LXstudio

LX521.4

LX521
reference

ORION
challenge
ORION-3.4 The two outputs from the LR2 crossover filter are 180 degrees out of phase at all
frequencies, which requires to use one of the drivers with reversed polarity, so
PLUTO-2.1
that the two acoustic outputs add in phase. At the crossover frequency the filter
WATSON-SEL outputs are 6 dB down.
The acoustic frequency and polar response is controlled by the electrical filters
PLUTO+
and the response of the mounted drivers. The electrical filter will not give the
subwoofer
desired results, if there is insufficient overlap and flatness of the driver frequency
THOR response and when they are offset from each other. This can be corrected in
subwoofer many cases with the addition of a phase shift correcting network. I consider the
crossover marginally useful, because the 12 dB/oct roll-off of the highpass filter
PHOENIX
dipole speaker
below the crossover frequency does not reduce the excursions of a driver's cone
when flat frequency response is obtained. My earlier assumption that the group
Three-Box active delay of a 4th order LR4 crossover at low frequencies would introduce audible
system (1978) distortion was not correct. Therefore I recommend not to use the LR2 crossover.
Reference
(38xo_eq1.gif, FAQ19, xo12-24b.gif)
earphones
The LR2 circuit uses the Sallen-Key active filter topology to implement the 2nd
Surround order transfer function. The response is defined by w0 and Q0 which sets the
sound
location of a pole pair in the complex frequency s-plane and by an additional two
Loudspeaker zeros at s = 0 for the highpass filter. In the case of the LR2 filters Q0 = 0.5, and Q0
& Room = 0.71 for each of the two cascaded 2nd order filters that form the LR4 filter. The
frequency response is obtained by setting s = jw and solving the transfer function
for magnitude and phase. The formulas below can be used to design filters with
different values for w0 or Q0, or to analyze a given circuit for its w0 and Q0 values.
Resources

Publications

Sound recordings

Links

Other designs

My current setup

About me

Site map

HOME

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Processes
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My Daughter
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What's new

LX - Store Any order Linkwitz-Riley filters can be implemented by a cascade of 2nd order
Sallen-Key filters. The Q0 values for each stage are listed in the table below. The
component values of each stage for a given crossover frequency f0 can be
calculated by using Q0 and selecting a convenient value for C2 or R2 in the
Conversations formulas above.
with Fitz
LR2 LR4 LR6 LR8 LR10
Q0 of stage
0.5 0.71 0.5 0.54 0.5
1
OPLUG Q0 of stage
Forum 0.71 1.0 1.34 0.62
2
Q0 of stage
1.0 0.54 1.62
3
Q0 of stage
1.34 0.62
4
Q0 of stage
1.62
5
dB/octave
12 24 36 48 60
slope

Crossover filters of higher order than LR4 are probably not useful, because of an
increasing peak in group delay around f0.
Top

3 - 24 dB/oct Linkwitz-Riley crossover


The 24 dB/oct LR4 crossover filter provides outputs which are 360 degrees offset
in phase at all frequencies. At the transition frequency Fp the response is 6 dB
down. The electrical network will only give the targeted exact acoustic filter
response, if the drivers are flat and have wide overlap. This is seldom the case.
The steep filter slopes make the combined acoustic response less sensitive to
magnitude errors in the driver responses, but phase shift errors usually have to be
corrected with an additional allpass network. (xo12-24b.gif, 38xo_eq1.gif,
models.htm#E) Top

Russ Riley and Siegfried Linkwitz, September 2006, Douglas City, CA

In the sixties, early seventies, I worked with Russ Riley at Hewlett-Packard's Palo Alto R&D
laboratory for the development of RF and Microwave test equipment. Like many other engineers
we had "G-Jobs", building such things as electronic ignitions for our VW bugs and vans, FM
receivers, phase-locked pulse width FM demodulators, short-wave receivers, audio pre- and
power amplifiers, third octave audio analyzers, headphone equalizers, and of course,
loudspeakers. After measuring the acoustic and electrical responses of commercial speakers
we equalized them and tried to understand why they were designed with strange looking driver
layouts, used large baffles, were stuffed with a variety of internal damping materials and used
various box stiffening and damping techniques. Eventually we completely redesigned them and
built our own speakers. Russ and his wife, Vicky, an accomplished organist, always had the
most critical and reliable ears. He was an ingenious design engineer, a strong contributor, who
inspired and challenged many of us on our HP and unofficial design projects.
Russ retired after over 40 years in R&D for HP/Agilent and now lives with his wife in a remote
mountain valley, in a genuine log cabin, amongst pear, plum and walnut trees, berry bushes,
chicken and deer, the sounds of a large creek, and the pine and fir trees that climb up the
slopes. He died peacefully in his log cabin on December 6, 2010.

4 - Delay correction

A first order allpass filter section with flat amplitude response but phase shift that
changes from 0 degrees to -180 degrees, or -180 degrees to -360 degrees, is
often used to correct phase response differences between drivers. Multiple
sections may delay the tweeter output and compensate for the driver being
mounted forward of the midrange. Active crossover circuits that do not include
phase correction circuitry are only marginally useable. (allpass.gif, allpass2.gif,
models.htm#E, 38xo_eq1.gif) Top

5 - Shelving lowpass

This type of circuit is useful to bring up the low frequency response in order to
compensate for the high frequency boost from front panel edge diffraction. It can
also serve to equalize the low frequency roll-off from an open baffle speaker.
(shlv-lpf.gif, 38xo_eq1.gif) Top

A passive RC version of the shelving lowpass is shown below.

6 - Shelving highpass

A circuit used to boost high frequencies or to smooth the transition between a


floor mounted woofer and a free standing midrange. (shlv-hpf.gif, 38xo_eq1.gif,
models.htm#F) Top

A passive RC version of the shelving highpass is shown below.


7 - Notch filter

Notch filters are used to introduce dips in the frequency response in order to
cancel driver or room resonances. The three circuits above have the same
response. A) is difficult to realize because of the large inductor. B) is used to
remove the peak in the 6 dB/oct dipole response. C) gives convenient component
values for room EQ below 100 Hz. (room EQ, inductr1.gif, inductr2.gif,
38xo_eq1.gif ) Top
8 - 6 dB/oct dipole equalization
Equalization of the dipole frequency response roll-off usually requires not only a 6
dB/oct boost towards low frequencies, but also removal of a peak in the
response. (Models A2) The three circuits differ in their ability to remove such
peak.

A) The shelving lowpass filter cannot correct for a peak.


B) The bridged-T based circuit is limited in the shape of curves that can be
realized. It has also higher gain for opamp noise than signal at high frequencies.
C) The shelving lowpass with added notch filter is the most flexible circuit.
(models.htm#D) Top

9 - 12 dB/oct highpass equalization ("Linkwitz Transform", Biquad)


A majority of drivers exhibit second order highpass behavior because they consist
of mechanical mass-compliance-damping systems. They are described by a pair
of zeroes at the s-plane origin and a pair of complex poles with a location defined
by Fs and Qt. The circuit above allows to place a pair of complex zeroes (Fz, Qz)
on top of the pole pair to exactly compensate their effect. A new pair of poles (Fp,
Qp) can then be placed at a lower or a higher frequency to obtain a different, more
desirable frequency response.
This allows to extend the response of a closed box woofer to lower frequencies,
in the above circuit example from 55 Hz to 19 Hz, provided the driver has
adequate volume displacement capability and power handling. The equalizer
frequency response is shown below, correcting for a woofer with peaked
response (Qp = 1.21) and early roll-off (Fp = 55 Hz), to obtain a response that is 6
dB down at 19 Hz and with Q = 0.5 .

The associated phase and group delay responses are shown below.
Not only is the frequency response extended, but the time response is also
improved, as indicated by the reduced overshoot and ringing of the lower cut-off
highpass filter step response.

It can be seen from the s-plane description of the transfer functions that the
complex poles of the driver in the box are canceled by a set of complex zeros in
the equalizer. The specified real axis poles of the equalizer, together with the
driver zeros at the s-plane origin, determine the overall loudspeaker response in
frequency and time.
The equalizer action is difficult to visualize in the time domain, because the driver
output waveform is the convolution of the input signal s(t) with the impulse
response of the equalizer h1(t), which in turn must be convolved with the impulse
response h2(t) of the driver. Convolution is a process whereby the current value of
the time response is determined by the time weighted integral over past behavior.
Below are the responses of driver, equalizer and driver-equalizer combination, if
the input signal s(t) is an impulse.

More illustrative are the responses to a 4-cycle, rectangular envelope 70 Hz


toneburst s(t). For example, the driver output is the convolution of the burst s(t)
with the driver's impulse response h2(t). Note that the driver phase leads the input
signal, as would be expected for a highpass response. Upon turn-off of the input
burst at 57.14 ms the driver response rings towards zero, governed by Fp = 55 Hz
and Qp = 1.21.
The equalizer output response lags its burst input. This signal will force upon the
driver a response correction so that it is no longer dominated by Fp = 55 Hz and
Qp = 1.21. The equalizer output signal is convolved with the impulse response
h2(t) of the driver to obtain the desired equalized driver output. Now, the decay of
the driver output follows the 2nd order highpass filter response determined by Qp
= 0.5 and Fp = 19 Hz of the equalizer, after the excitation has stopped.
Of course, none of the driver mechanical parameters like mass, compliance and
damping have been changed in the process of equalization, only the input signal
to the driver has been modified.

The above circuit can also be used to correct the low frequency roll-off of a
tweeter so that the equalized tweeter becomes a filter section in an exact LR4
acoustic highpass. (f0Q0fpQp.gif, pz-eql.xls, f0Q0.gif, FAQ15, sb80-3wy.htm,
sb186-48.gif , sb186-50.gif)

The 'CFL Linkwitz Transform Designer with Monte Carlo Sensitivity Ananlysis' by
Charlie Laub makes component value selection easy and shows the effect of
component tolerances upon the frequency response. Keep in mind that the LT is
based on a measurement of driver parameters Fs and Qt. Only the small signal
parameters are easy to define. Fs and Qt change with increasing signal level and
to varying degree for different drivers. This makes the equalization imprecise, but
it remains effective in practice.
Top

10 - Variable gain & fixed attenuation

A major advantage of line-level active crossovers is the efficiency with which


drivers of different sensitivity can be combined in a speaker system. The three
circuits use linear taper potentiometers but obtain a gain variation that is
approximately linear in dB. Circuits B and C assume a 10k ohm load such as the
input impedance of the power amplifier. Circuit A is optimal between filter stages
because of its low output impedance. The placement of the variable gain stage in
the filter chain must be carefully considered, because it affects noise
performance and signal handling. (gain-adj.gif, attnrout.gif, 38xo_eq1.gif) Top

Occasionally a fixed attenuation of A dB or a is needed for the input voltage V2 of


a circuit stage with input impedance R3 when driven from an operational amplifier
with output voltage V1. In the example below a 3 dB (a=1.41) attenuation is
desired. The load Rin that is seen by the opamp should be about 2000 ohm. The
following amplifier stage has an input impedance of 10k ohm.
For designing an attenuator with specified output impedance Rout see:
attnrout.gif

11 - Line driver

The output stage of the filter must be capable of driving cables, which typically
have a capacitances in the order of 150 pF per meter length, without going into
oscillation. A 196 ohm resistor maintains a resistive load component and tying
output to negative input for out-of-band frequencies (>100 kHz) reduces loop
gain. All of the above circuits can drive cables if operational amplifiers such as
the OPA2134 or OPA2604 are used. In most cases it is not necessary to have a
separate line driver.

Performance of active circuits should always be checked for inter-stage clipping,


and for oscillation with a wideband (>10 MHz) oscilloscope. Top

12 - Power supply

I recommend to leave the effort of building a regulated power supply to one of the
many vendors that offer wallplug and tabletop models. An output specification of
+/-12 V to +/-15 V DC at >250 mA and with <1% ripple and noise will suffice. Often
such supplies can be found at electronic surplus stores. Top

13 - Printed circuit boards WM1 and MT1

To simplify the construction of active line-level equalizers and crossovers I offer


three printed circuit boards, ORION/ASP, WM1 and MT1. The circuit traces are laid
out to allow for a variety of filter designs. It is up to the user to determine the
actual circuit configuration and component values. Then the necessary
components and jumpers are loaded at the appropriate locations on the board to
obtain the desired filter response. I will provide specific information for
assembling the PHOENIX crossover/equalizer on the ORION/ASP pcb and a
Linkwitz Transform on the WM1 pcb.

WM1 is designed to implement the functionality of circuits 1, 5, 6, 7, 8, 9 or 10 and


various combinations of these. The circuit board provides two of the topologies
below for two channels of equalization or for a more elaborate single channel
response correction.

The WM1 board can be used for:

Equalization of an existing speaker with passive crossovers, baffle step


correction and extension of the low frequency response.
Pole-zero equalization of a closed box woofer and a LR2 crossover lowpass
filter. Variable gain.
Pole-zero equalization of a midrange and a LR2 crossover highpass filter.
Dipole woofer equalization with notch and variable gain. LR2 crossover
lowpass.
Dipole woofer equalization for low Qts drivers.
Low frequency, individual channel and overall response equalization of
multi-way speakers, so long as elements of this topology allow you to
generate the response you need.
Equalization of add-on woofer , FAQ10, FAQ15

MT1 is designed to implement the functionality of circuits 1, 2, 3, 4, 5, 10 or 11


and various combinations of these. On the circuit board are two of the
topologies below.

The MT1 board can be used to construct:

A 2-way speaker with crossovers of order 1, 2, 3, or 4. The tweeter channel


has variable gain and delay circuitry to align the tweeter's acoustic center
with the woofer. The input buffer stage can provide 4p to 2p polar response
(baffle step) correction.
The tweeter and midrange channels of a 3-way system. The midrange
highpass filter of the woofer to mid crossover would have to be provided by
the WM1 board.
The tweeter and upper midrange or upper midrange and lower midrange
channels of a 4-way system.
A great variety of active multi-channel line level filters in combination with
the WM1 board.
Crossover for add-on woofer, FAQ10, FAQ15

The circuit boards are practical tools to experiment with and to learn about active
electronics. You will find that active loudspeaker systems give you the freedom to
match drivers of greatly different sensitivities, are easier to design, and can give
greater accuracy of sound reproduction, than is possible with passive, high-level
crossovers and filters.

See the Circuit Board page for ordering information. Top

14 - Literature

Much useful information can be obtained from application notes of the various
opamp manufacturers. If you need a refresher or an introduction to circuits, then
read:

[1] Martin Hartley Jones, A practical introduction to electronic circuits,


Cambridge University Press, 1995. It is a well illustrated, easy to read, yet
technically solid text. It covers a broad range of devices - from tubes to ICs - and
many basic circuit functions.

The following books cover a range of concepts and go into depth on specific,
relevant topics to strengthen understanding of electronic circuits and electro-
acoustic models.

[2] Herman J. Blinchikoff & Anatol I. Zverev, Filtering in the Time and Frequency
Domains, John Wiley, 1976. A broad and fundamental look at filters.
[3] Arthur B. Williams & Fred J. Taylor, Electronic Filter Design Handbook,
McGraw-Hill, 1995. Design and analysis formulas for all types of filters.
[4] Jasper J. Goedbloed, Electromagnetic Compatibility, Prentice Hall,1990.
Fundamental concepts and practices for dealing with radio frequency
interference.
[5] Henry W. Ott, Noise Reduction Techniques in Electronic Systems, John Wiley,
1976. Practical steps to combat RFI.
[6] Manfred Zollner & Eberhard Zwicker, Elektroakustik, Springer, 1998. The most
comprehensive and solid engineering level presentation of electro-acoustic
transducers and related subjects.
In German, no comparable English language text available, to my knowledge.
[7] Walter G. Jung, editor, Op Amp Applications, Analog Devices, 2002. Everything
you ever wanted to know about using operational amplifiers, and not just at audio
frequencies.
Top

------------------------------------------------------------------

| Build-Your-Own | Main Panel | Dipole Woofer | Crossover/EQ | Supplies |


| System Test | Design Models | Prototypes | Active Filters | Surround | FAQ |
What you hear is not the air pressure
variation in itself
but what has drawn your attention
in the streams of superimposed air
pressure variations
at your eardrums
An acoustic event has dimensions of Time,
Tone, Loudness and Space
Have they been recorded and rendered
sensibly?

___________________________________________________________
Last revised: 02/15/2023 - © 1999-2019 LINKWITZ LAB, All Rights Reserved

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