Filter Design Wizard

I want to recommend Analog Devices online filter designer:

www.analog.com/designtools/en/filterwizard/

I wanted to make a high-pass filter for my bass guitar, 24dB/octave chebyshev filter, -3DB point at 30Hz.

Finished it a few hours ago and measured the performance with my scope and a frequency generator and it's spot on the simulation.

It also calculated E24 series resistor values to use with all 0.1uF capacitors to give the required response and a gain of 1. This is something you couldn't get remotely close to achieving using the tables in Horowitz and Hill.

Yep, I use that one quite a bit, but must say I almost prefer the Texas Instruments one - both very good .

**LINK**

I dabble a lot in building Ham Radios, and use op-amp filters in the audio chain, esp. for direct conversion receivers, where the filters should be low noise with LOTS of gain.

Use LT_Spice for simulation - very good.

However, the BEST filter design program ever was Texas Inst. Filter-Pro vers 2. That no longer exists -

Ver 3.1 is available, but it is so full of bugs - poor.

The replacement TI version is, like the AD one, all on line, which personally I hate. Means you have to register, have to be online, etc. Suppose that's the trend these days ( Fusion, etc..). There are MANY very useful little 'calculator' programs, esp in the RF and AF field, and most of them have moved to .php programs that have to run from the website, online. So, back to the good old calculator and brain cells most of the time..!

I suppose I also have an aversion to providing my personal info on website after website..

high-pass filter for my bass guitar - Sound like a contradiction in terms...

What PCB CAD do you use and where did you get the PCB made?

Joe

Edited By Joseph Noci 1 on 30/09/2019 07:51:14

Hi Joe,

I'm pretty sure you don't need to log in to use the Analog Devices calculator.

> high-pass filter for my bass guitar - Sound like a contradiction in terms...

The -3dB point is 30Hz, chebyshev chosen to give a sharp 'knee'. My 5-string bass has the lowest string tuned to low B which is about 31Hz. The most popular filter (Thumpinator) uses 25Hz, other people advocate as high as 50Hz, so you actually get the fundamentals of the low notes being cut quite hard.

It has two effects - it protects the amp/speakers from any low frequency transients such as handling noise (e.g. when you pull the string before releasing it, or when you mute over a pickup), the follow on from this is that by removing subsonics it increases headroom - this is probably a more meaningful effect with the higher cut-off, although many thumpinator users say it lets them drive an amp harder.

Some amps have an HPF built in.

I made the pcb in the kitchen, using a positive designed in the hobbyist (100mm square board limit) version of Eagle. A classic Autodesk product that is extremely capable but ridiculously klunky, but is free. It take longer to find the parts you need in the libraries or online than to route a board manually!

It's a right dog's breakfast to get it to fit around the jacks, fortunately with audio frequencies you can get away with a spaghetti layout! I've since rerouted it as one track had to be cut to get the board to fit one corner.

Interesting. Anyone know of a simple to use* online simulator/analyser that can cope with a 'Linkwitz transform' filter? I have a schematic with some funny component values - different from what the text-book formulae suggest. Not sure whether the differences are typos or intentional. The circuit's designer is not available for interrogation. I'm using Linux, hence the request for an online app.

* I'm far too lazy to learn how to use a Spice clone.

THIS link contains a downloadable spreadsheet.

Neil

Interesting concept, sort of like a preset left hand end of a graphic equaliser...

But these is a cost: " Getting this kind of response out of a sealed enclosure setup requires some tradeoffs.  You will be giving up some of the overall SPL producing capability of the driver because of the excursion overhead needed at the lower frequencies. "

Edited By Neil Wyatt on 30/09/2019 11:04:23

Nice Neil! Interesting how many 'enabling' hobbies we all seem to have. ( Not a fan of that word, though..)

I'm pretty sure you don't need to log in to use the Analog Devices calculator.

Not necessary to log in, but I had to register on site first to be able to use it - as it can store your files in their 'cloud' - (Yuk!) -

KiwiB - On the Linkwitz - here's a spreadsheet calculator - you probably have it if you are playing with this stuff..

**LINK**

But to simulate I have not seen anything specific to this application. However, I DO recommend LT_Spice - free and not difficult with many examples and many Tuts.

The spreadsheet helps with the data does do most of what you may need though?

Joe

Thanks Neil and Joe. I need to clarify. I think I need something that plots the frequency response, given the component values, rather than a component value calculator. The schematic does not give the intended filter parameters, just the component values, and these seem strange. From memory, pairs of Rs and Cs are supposed to have identical values, but not in this schematic, so I'm wondering why... Sorry to be vague, but electronics isn't my field, and I'm, perhaps rather cheekily, only posting because the original post piqued my interest.

Are you familiar with how you can take a sine wave, then add in its odd harmonics and the final waveform approaches a sine wave?

These filters use a similar principle, uncovered by Mr Butterworth, who invented the filter that bears his name. So rather than stacking identical filters, you create a series of 'detuned ', pairs of RC filters each of which averages out to the desired frequency. These add up to give the response you want.

You can aim for different trade offs between 'sharpness of the knee' and 'flatness of the plateau' by teh exxact values of the filter poles.

The maths is convoluted, which is why simulators can be used, typically choosing convenient, relatively tight tolerance capacitors then accepting odd resistor values.

Neil

Last time I looked at Fourier analysis adding in-phase odd harmonics of reducing amplitude to a sine wave approximated a square wave.

I had a quick go with the simulator; compared to some it is easy to use. I didn't register or login, but I didn't try and save anything. I have an AD login, so it's possible my computer remembered it anyway.

Warning: Skip this bit if maths isn't your thing!

It's interesting the way the simulator switches between filter characteristics without so much as a by your leave. Like all simulators they can be useful if one understands the theory, but if one doesn't then they can lead one into a cul de sac. Via Wikipedia I've just read the original paper by Butterworth from 1930. Of course that was long before opamps and RC filters, so everything is valves and L, C and R. The intent of the original paper was to design filters that were maximally flat in the passband without compromising the roll off. In other words the amplitude characteristic is monotonic and flat in the passband, which it appears previous filters were not. It's interesting to note that the original article only refers to second order sections. The concept of single real poles isn't mentioned, so the original Butterworth filters could only be even order.

In order to understand filters an appreciation of the complex s-plane, and poles and zeros is useful. On the s-plane the x-axis is sigma, a measure of how a signal decays and the y-axis is j times omega, complex frequency. A pole is a point of inifinite value and a zero is just that, zero. If a rubber sheet is stretched over the poles, and nailed down at the zeros, then a section along the y-axis will give the frequency response of the filter. There are some constraints, for stability all poles must be in the lefthand half, ie, sigma is negative. Poles always come in complex conjugate pairs. A special case is when omega is zero in which case the two poles are coincident, and real, on the x-axis.

The filter simulator starts off with Bessel filters, which are maximally flat in group delay, ie, signal distortion in time is minimised. This is at the expense of roll off. Pole position is determined from Bessel functions, hence the name. As more roll off is required the program switches to Butterworth which as stated is maximally flat in amplitude in the passband. So no amplitude distortion and faster roll off, but at the expense of non-linear group delay giving time distortion of the signal. A Butterworth filter only has poles, and those poles are equi-spaced, and lie on, a circle. As yet more roll off is required the filter characteristic becomes Chebyshev, which allows ripple in the passband in return for faster roll off while still being monotonic in the stop band. Technically these are type 1 Chebyshev fillters. Type 2 Chebyshev filters are flat in the pass band but have ripple in the stop band - never seen them used. A type 1 Chebyshev filter is also pole only, and the poles lie on an ellipse. In the limit as the ellipse becomes a circle the passband ripple reduces to zero and the filter becomes Butterworth. Although not used in the simulator the fastest roll off is with an elliptic filter that has poles on an ellipse giving ripple in the pass band and zeros on the y-axis giving ripple in the stop band.

It took me a while to find the design where the capacitor values are uniform; in the multi-feedback circuit. The Sallen-Key arrangement is the classic filter circuit mentioned in all text books. It's ok but can have problems implementing high Q sections. There are better, but more complex, circuits available such as the biquad.

The tolerance feature of the simulator was useful, and very quick. Not sure how it worked, seemed a bit fast for a proper Monte Carlo anaylsis?

I had to look up the Linkwitz circuit. Looks a bit odd with T-sections in the input and feedback paths. It's not high enough on my priority list to do a proper analysis. In the past I've used Tina from Texas Instruments as a simple circuit simulator. I expect it is based on Spice internally but has a graphical input interface. Ah, I see Kiwi Bloke is using Linux, in which case he's probably out of luck for a simple simulator.

I'm with Bob Pease on circuit simulators. They have their uses, but are only as good as the component models. Ultimately prototyping trumps the theory and simulation.

Andrew

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Thanks, Andrew ... I was about to start doubting my sanity

MichaelG.

Fairly obvious typo...

N.

QUCS is a relatively easy to use simulator for Linux. Alternatively I find LTSpice works well under Wine.

Russell

Elsie also runs under Wine on Linux.

Playing with this wideband Software Defined Radio, I found it overloaded at night due to very loud MW broadcast signals and powerful Long Wave time standard signals. I used Elsie to design a filter to attenuate the whole MW band and to notch out TDF (France) on 162kHz and BBC R4 on 198kHz, while leaving everything else open.

Unfortunately, the filter only revealed the next problem! That's the appalling radio racket created by internet users with ADSL connections, plus the generally high level of horrible noises from unsuppressed domestic equipment.

Not quite on topic because Elsie does Inductor-Capacitor rather than Resistor-capacitor filters, but some may be interested. These days we're spoilt for choice!

Dave

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Just as an aside: This is quite nicely done [in Javascript] **LINK**

http://195.134.76.37/applets/AppletFourier/Appl_Fourier2.html

MichaelG.

Thanks Russell, I'll look up QUCS. Thanks Andrew for your input - I wish I understood the theory...

Having somewhat hijacked this thread, I've felt obliged to try to follow some of the suggestions - make some effort to solve my own problem. Crikey! The maths is frightening, but not as opaque as the underlying concepts hinted at by Andrew J. OK, I think I should give up, my poor aged brain won't cope with going off in such a complex new direction. Current mental activity is centred on wondering how to manipulate a CertiFlat welding table kit: not disturbing the fit of the components whilst invering it for final TIG welding. It will weigh >100kg.

Before I give up on this filter thing entirely, however, the problem is (if I remember correctly) that the circuit contains various Sallen-Key-like filters, built around op-amps. The usual implementation of these seems to use two same-value Rs and two same-value Cs. It looks as though this may be because it simplifies the maths. The schematic uses different values for the Rs, and I wondered why. I can't remember whether the C values were identical, and now I can't remember where the wretched schematic is. The perils of working from memory...