Mic High Pass Filter

[Samuel Groner]

Hi

I keep using high pass filters on my guitar recordings; so I thought a passive mic level one would be nice, to be plugged between mic and preamp. Check this schemo: [removed]

Could some gentleman with a good hand in filter design throw in some values for C1, C2 and L? Would love to have values for Butterworth designs @ 100 Hz and 150 Hz. I think we could assume a source impedance of 150 ohm and a load of 1.5k. I added the 3k to prevent things going mad if I connect it to a pre with way higher input impedance.

I'll use this thing with dynamic/ribbon mics (currently mostly SM-57, M201TG and M130), so no need for phantom power.

Do I have to be prepared for resonances when connecting this to a transformer balanced pre?

Thanks!

Samuel

 

[chrissugar]

Samuel

That schem is too complicated. Remove the two C1 capacitors and the L inductor.
Leave only the C2 capacitors and the resistor. Use something like 2K for the resistor, and calculate the capacity of the two C2 with the next formula:
C=2 x 1/(2 x pi x F x R)
F is the cut off freq.
One more thing, if you use this high pass filter between microphone and preamp it will not pass Phantom power, so you will have to use it only with ribbons, dynamics, self powered condensers or tube mics with external PSU.

chrissugar

 

[pstamler]

[quote author="chrissugar"]Samuel

That schem is too complicated. Remove the two C1 capacitors and the L inductor.
Leave only the C2 capacitors and the resistor. Use something like 2K for the resistor, and calculate the capacity of the two C2 with the next formula:
C=2 x 1/(2 x pi x F x R)
F is the cut off freq.
[/quote]

That's fine if a 6dB/octave filter is what you want; the original design yields an 18dB/octave filter, which has its uses, especially when there's lots of low-frequency noise about.

Peace,
Paul

 

[chrissugar]

Yes Paul, you are absolutely right, but I proposed this filter, because I usually prefer the lower order filters. They have a gentle slope and better phase response from a musical point of view. Although, in some situations high order filter is the best solution to solve the problem.

chrissugar

 

[bcarso]

Samuel:

A pretty close approximation to Butterworth:

For 100Hz: C1 = 2.02uF, C2 = 3.88uF, L = 1.008H (this is a nice solution as the values are close to standard parts)

For 150Hz: C1 = 1.347uF, C2 = 2.587uF, L = 672mH (at least the L is near a stock value)

Assumptions: no series R in L; terminated in your 1.5k load with your safety 3.0k present, 150 ohm source Z.

Insertion loss in the passband is about 1.3dB.


Comment: these are big fat parts. It shows why active filters are popular... If the L's are of sufficiently high quality as well as the C's this will be a big lump in the line, or wherever it is.
Brad

 

[bcarso]

Comment: you may be looking for a suitable inductor for a while. In addition to wanting to have the series R small, the distributed C should also be small. A quick websearch found a 1H toroid with 121 ohms R and a self-resonant frequency of 12kHz, neither of which is acceptable.

Actually the equivalent C will be swamped out in this circuit, leaving one with the series R as the biggest deviation from ideality. The effect on the response is a greater droop in the transition band, which might be something that could be lived with.

 

[bcarso]

By raising C1 a bit, to about 2.5uF for the 100 Hz version, so can get nearly back to where you want to be in the transition band. The other effect of the 121 ohm series R is that at lower frequencies in the stopband the response transitions to a two-pole rolloff---but you are already down so far at this point that it shouldn't be a practical consideration.

 

[Samuel Groner]

Thanks for the comments!

Comment: you may be looking for a suitable inductor for a while.

What about a Wilco BSL105 (www.wilcocorp.com)? They do not say anything about distributed C, though. In fact, I e-mailed them one jear back about that (for another part type) and they just told me that they don't know. Pretty strange, or my Q was confusing...

Would a 2nd order design give lower inductor values? Or maybe I just go with a 1st order, as suggested by chrissugar.

Samuel

 

[bcarso]

That Wilco part looks pretty reasonable based on the series R. The 20% tolerance is a bit worrisome though.

You could ask them if they know what the self-resonant frequency is, although as I say the circuit application pretty well swamps out the distributed capacitance.

Going to first order is very simple and eliminates the inductor, but it may not give the low frequency rejection you want.

Second order with just the L and C doesn't change the required inductor size to speak of.

You could just buy a big cup core and wind your own. These come with bobbins and have an inductance per square of number of turns coefficient; for example the Fair-Rite pot core 5678362221 core would give you about 1 H with 391 turns, if I calculated correctly. The tolerance on the resulting inductance is 20% but you can add or subtract turns unless you are running out of room. You can also wind some initial number of turns, measure the L, and then calculate what the actual coefficient is for that sample. I haven't figured out what the largest gauge of wire is that would fit ~400 turns on that core but that's a good geometry exercise . The pot cores are self-shielding almost as much as toroids. Toroids are nice but are extremely tedious to wind unless you have a machine---especially for this kind of number of turns.

 

[chrissugar]

My sugestion is this, build one first order filter now because it is simple and cheap and later when you have the inductor build the high order one. Experiment for a while and you will find them useful for diferent situations.

chrissugar

 

[Samuel Groner]

I ordered two samples from Wilco, so this were zero-$ inductors! I'll go with 3rd order, 1st is really to low for my application, I tried it with my DAW.

Brad, could you teach me the lowpass-to-highpass transformation or point me to an easy to understand source? I know how to design passive lowpass filters, but never got this transformation...

Thanks!

Samuel

 

[bcarso]

Hmmmm---lemme check some refs and I will recommend some.

Typically, although I just fumbled through this design with informed simulation, by now virtually all passive filters are tabulated, at least in normalized form, somewhere. I suspect that is true of this one. The unequal source Z output Z makes it a little more unusual however.

Wilco is nice to us. I am not sure how well shielded their parts are, however. You may end up in the inductor fab activity in time.

Cheers,

Brad

 

[bcarso]

PS: One could readily do a phantom-power active filter too. It would never have as low intrinsic random noise as a good RLC, but it could be made fairly immune to mag field pickup.

 

[bcarso]

Samuel, after a night of some sleep I realized the lowpass-highpass interchange is simple. You want to replace C's with L's and vice versa. Each interchange should give you the same reactance magnitude as before. So 1/ωC = ωL is how you determine each value (ω = 2Πf). Thus an L of value 1/((ω^2)C) replaces that C, etc.

 

[Samuel Groner]

Thanks, Brad, for the explanation. I thought that it must be pretty easy. I've read parts of Zverev's "Handbook Of Filter Synthesis", but I must have missed the section on this topic...

How would you attempt the active version? Sallen & Key VCVS with some gain? A few transistors should do, I guess.

Samuel

 

[pstamler]

[quote author="Samuel Groner"]How would you attempt the active version? Sallen & Key VCVS with some gain? A few transistors should do, I guess.[/quote]

They would need to be very quiet; at least 6dB quieter than your preamp's input to avoid adding significant noise to the signal. This strikes me as difficult and overkill; I'd either go for a passive filter or design an active filter to go inside the preamp, after the gain stage(s).

Peace,
Paul

 

[bcarso]

Samuel, there is a (unbalanced) unity-gain topology with 3 equal Cs and a fairly big element spread in the three R's for Butterworth. It probably has a pretty high noise gain too. It also has an input impedance that is a function of frequency just like the LCR, and as PRR points out the nominal dynamic mic output Z gets inductive at high frequencies.

So to do a good design you would want to know what the real mic Z is over frequency and take that into account. I don't know if a typical handheld LCR meter would be safe to attach to the mic output or not.

Really, for the best noise performance the preamp ought to precede the filter, except that if you have truly horrendous low freq material I guess you could get into overload issues. But then it's time to do something acoustically. It may well be that the studio noise swamps out the electronic noise and it's not much of an issue.

A compromise would be a bit of gain/buffering followed by the active 3-pole highpass, which could use low impedances for good noise performance as it would no longer be loading down the mic.

There are some circuits that do a balanced Sallen-Key sort of approach and some others as well, saving a cap or two compared to two filters, but I would probably favor just adding an inverter stage for the other side of the balanced output, making sure the impedances of each output are equal.