OK techies, what do you make of this explanation?

http://www.fabfilter.com/forum/2268/48db-hi-cut-slope-causes-boost-in-level?replies=1#comment7390
 
I can come close to thinking I can handle it by considering that, to my limited knowledge, at least some, EQs use a sort of parallel processing and then combine signals to get the final output.
 
Then I think, "hey it's only a lo-cut", and the explanation that the peaking is due to comb filtering seems like, well, that's what EQ is anyways... controlled filtering.
 
What do you think?
 
I'm off to work on the road but when I get back I'll try a spectral analysis to see what frequencies are doing the nasty.

Mike in your question to FabFilter you mention Hi Cut in the heading and Lo Cut in your actual question. Which is it?
 
Also if you mention Q then you cannot be talking about filters. Filters don't have a Q setting usually. So are you referring to a peak style EQ then.
 
Anyway I did some tests and set up a source of pink noise at a very precise level and metered the output with a VU meter. I tried several filters to start with, all at 48 dB/Oct slope at both ends of the spectrum. (I even set up a Cakewalk LP64 EQ and replicated the response and tried moving all the nodes and still got no upward level shift)  In all cases I got no upward level shift here at all anyway no matter what the cutoff freq was. (only a downward shiftt in level as to be expected as you move the cutoff freq further in etc)
 
I also tried this with a peak style EQ as well with a Q of 1.0 and also got no such level shift either. Only a drop in level as to be expected.
 
Sounds a bit strange to me. The fact is if you are taking stuff away the level should go down not up. That is what is happening here for me.  In all the years of mixing I have done and used steep filters at either end of the spectrum I have never seen any upward level shifts. (probably because I don't use any Fab Filter stuff maybe!)

Hi Jeff,
I just saw your message. I am working a gig the next 2 days but look forward to reading the details of the info you have shared.

Thank you!

The total amplitude of a signal is the result of the combined frequency parts. If you cut a frequency band with an EQ, especially a strong low end, you can create a peak where there was none, because the wave might have been inhibiting the total at a certain point. Think of a perfect sine of, say, 50 Hz that is part of a rich harmonic signal close to peaking. The 50 Hz wave has parts where it has positive values and parts where it has negative values. If the total signal is close to peaking on a place where the 50 Hz tone has a strong negative value, you can create a peak by cutting it. It occurs with other frequencies too but is usually clearest with strong lows.

Sorry if this is not what you meant, I remember being surprised by this realization when I read it. I think in a manual of a Waves plug.

Well, it may not help much but it is (or should be) well known among signal processing engineers that if you have a multiplicity of sinewaves, the phases of them can be chosen to minimize the peak of the entire signal. This is useful to keep a wideband signal within range of a DSP. So, if you move the frequency of any of these sinewaves, or change the phase of any of them, or REMOVE any of the sinewaves, the peak of the entire signal will go up. I think that is what you're seeing. You won't see this effect unless you have "many" sinewaves that happen to fall into certain phase relationships.

Strangely enough I posted a similar question about the Quadcurve EQ recently: http://forum.cakewalk.com...aspx?m=3086433&p=1

I researched this a little when I was writing the EQ section of my FX video and was looking into how a regular EQ filter works as opposed to a Linear Phase EQ .
 
A regular EQ will always impart a very slight delay on material and hence a slight phase shift which in turn leads to some comb filtering. This will be more noticeable around the centre frequency. Therefore there will always be a little wave distortion/boost around the cut off frequency of a lo cut filter. The steeper the cut the worse the phase shift, hence the distortion of the wave and boost BTW.
 
Having said that when cutting it's also less noticeable than if applying a boost to say a peak filter. I am no expert by any means but my understanding is while the artefacts introduced are there I would doubt very much if they are capable of introducing clipping unless of course the material is very, very close to clipping already.
 
A linear phase EQ uses a much larger audio buffer hence no phase shift but uses more processing power in the first place.
 
Edit for detail.

mike_mccue
Linear phase EQ produces "pre-ringing"... I wouldn't be surprised if an "unanticipated" peak was observed with a linear phase EQ filter.
 
I just got up, will have some coffee, and then describe the circumstances with more detail and try to answer some of the questions asked above.

Me too! Except I am enjoying my coffee as I read this.
 
Waiting...

I found the explanations about the phase interference very interesting and like Jeff, I had not noticed the peaking behavior before.
 
I am also familiar with the *idea* that band pass filters are said to not have "Q" factors. My college instructor from so many years ago use to say things like this to us.
 
Over the years I have learned that many Hi pass and Lo pass filters do have "Q" factors included as an adjustable parameter, and more recently, I have learned that the Q factor, in this context, effectively describes the knee, or corner of the transition to the slope.
 
In the example of the FabFilter Pro-Q2, there is a Q adjustment provided for all of the Lo-cut slope choices except for the 6dB/octave choice.
 
 
 
 
 
Regarding the implication of the interference filtering: I think I have an explanation for why the material I was working with made the issue so apparent.
 
I was working on a listening test that focused on the quietest portion of the trail off of a sustain to hear the effects of a reverb I work with. I was also making another test of listening for the effects of dither choices because I wanted to re-evaluate my opinions about the usefulness of dither.
 
 To that end I was working with a sample of a single hit of a crash cymbal. I made the sample with by combining tracks made with a pair of Coles 4038 overhead mics and a Schoeps MK41 SDC used as a close mic. The 4038s came in through a Chandler TG2, and the MK41 came in through a John Hardy M1. I explain this to point out that the sound of the cymbal was rich and detailed and the sustain was long and clear.
 
 After mixing the three mics into a stereo bus I added an instance of Exponential Audio's Phoenix reverb and set up a "plate" which I adjusted to use an uncharacteristic 10 second reverb time. This served to add detail and texture to the quietest parts of the cymbal's sustain.
 
 Then I added a limiter to the master bus and set it up so that it was easy to hear rather long sustain and used this to make new samples that I could test with my dither tests. (don't get distracted by a discussion of dither... it is inconsequential to this discussion about EQ) The limiter was set to look ahead and had an absolute ceiling of -1.0dBFS. I was not especially concerned with the aesthetic of how low loud the initial transient seemed, and I was not concerned with "rules" that discourage people from working so closely to 0dBFS.
 
 Finally I dropped an instance of FabFilter Pro-Q 2 on the master bus and set up a 48dB slope lo-cut at 30Hz with a Q of 1. That's when I noticed the peaking. Normally I would have dragged the instance of the EQ to precede the limiter and I would not have noticed the peaking so readily, but in this case I made my adjustments to the EQ and saw the very obvious extra peaking on the output meters before I placed the EQ  where I really wanted it in the fx chain.
 
 It was easy to see that with the EQ bypassed the peaking was matching the limiters -1.0dBFS output, but with the EQ engaged it was at +1.0dBFS. Wow.
 
 
 
 So... having explained all the circumstances, and having considered what has been explained by those who have responded, I am going to propose the possibility that the "in-harmonic" nature of a cymbal's "indefinite pitch percussion" voice, and its especially closely spaced and "randomly" mixed harmonic content, increases the possibility that subtracting with an EQ filter can result in interference filtering that produces increased peaking at the output.
 
 I find the idea fascinating and imagine I'll be thinking about it for quite a while.
 
 I prepared an animated gif that compares the playback of the cymbal crash with the EQ engaged and bypassed. The graph shows the "frozen" maximum peaks. You may be able to discern how many of the peaks along the spectrum, specifically in the mid range, seem to be slightly higher in level, and perhaps you may imagine how the small variances can sum to an increase that the broadband PPM meter reports as a difference of 2dBFS. 
 

 
Thanks to everyone for sharing ideas. I look forward to any further comments you may have.
 

I agree that the picture isn't very helpful, but that's because you're looking at a spectrum rather than a summed wave. You can at least see a pretty powerful low roll off. With a signal that close to 0dBfs it's not hard to see how that might increase peaking. I don't really know how to explain it any better than I already did, nor am I really an expert on the matter.

Ah, Sorry mis-understanding. I can see the differences in the mids, I just thought we were talking specifically about around the cut-off area.
 
I have no idea why there would be any change there. Of course with a snapshot picture it's almost impossible to tell what is going on but if the EQ is causing it, I'd suggest buying a better EQ or use the Prochannel.  <---- That is a joke!!
 
On a more serious note have you tried a different EQ to see if that is the issue?
 

Sanderxpander
This is normal behavior for an EQ, even if most of the time people don't realize it.

I assume that it is the comb filtering caused by the slight phase delay then?

When I create a full mix with everything in the mix I am still nowhere near 0 dB FS. The rms level of the full mix is still down at the reference level which may be -14 or -20 dB in my case. That is how the K system works. Tracks at the ref level, buses at the ref level, the final mix at the ref level.  (For me) anything other than that is just plain wrong.
 
Because I work this way the problems that have been raised here  (and many others)  never have any effect in reality. (they are not important!) That is why it is just a great concept.  It eliminates most problems before they arise.  The issue that Mike is talking about here is totally insignificant.  I am not saying they don't exist, but if you are clever you can avoid them alltogether.
 
It probably explains the reason I am seeing no level changes. I was doing all the tests at the chosen ref level I was using at the time.  I did find the test signal in Mike's case interesting (cymbals) something I would have not thought of.  But in reality cymbals are often very low in the mix while they still can be heard very clearly.  Also if I was using a HPF on cymbals I would probably be pushing the cutoff frequency a little higher as well.  In that situation levels are usually going down not up.

Excuse me if I'm misunderstanding because I don't use K-level metering, but in my understanding this becomes a problem even before hitting the main fader/meter, as long as your audio file is significantly hot and your EQ is pre-fader (which is basically always). Or do you always gain down when using K-level?

I just mean if you have a hot audio file, your EQ can still clip regardless of any meter or fader level.