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EQs increase the overall level when cutting out frequencies... WHY!
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11th July 2008
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EQs increase the overall level when cutting out frequencies... WHY!

Hi everybody,

this has been bugging me out for the longest time:

How come, when I cut out certain frequencies of a sound with an EQ, the overall level of the sound increases too?

In theory this doesn't make sense to me: when you cut out (take away) frequencies, then I assume that the sound should be decreased in volume?

I experience this alot (cut out frequencies = increase in volume), especially when I lowpass a sound but not only.

1- Is this normal?

2- Does it occur only in the digital world?

3- Only with specific eq types (no matter digital or analog)

4- Can you recommend an eq plugin that doesn't do this...

Thank you all for your answers as this issue is very confusing for me
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Great question, I've noticed this too- dragging down a frequency can all of a sudden make the EQ output clip... but I'm LOWERING the volume of a bandwidth?
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All eq's do this. Not just digital ones. It depends on what frequency. When you attenuate a frequency, that automatically makes other frequencies more prominent. Basically you are getting rid of the mud. The less mud the more loud. For example, if you filter out some of the low bass 30 hz or so in a track, then you will get a much hotter signal, and will be able to increase loudness by a substantial amount also the overall level will be hotter by just applying the high pass. It's not a big deal. Basically you are making room for the frequencies that aren't heard as well. Just turn down the output on the eq. Somebody else can describe what I am saying in more technical terms...but this is whats going on. You turn something down, something else becomes louder as a result. Also try using a more narrow Q when cutting. Use wider Q for adding gain.
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Do you mean that the resulting signal sounds louder or that it starts to clip? The first one should not normally happen. It can be that you are overloading your monitoring system with a lot of low frequency content and when filtering them out the system has more power to use for the audible frequencies. If you mean clipping, then it's just that loudness doesn't have much to do with peak levels. EQ cuts can boost the peak levels but should not increase the RMS (perceived) value. This is why you shouldn't use EQ cuts after normalizing (you could get digital clips).
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Could it have something to do with harmonic dominance? I dont know.
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Quote:
Originally Posted by jdtrbn View Post
This is why you shouldn't use EQ cuts after normalizing (you could get digital clips).
You shouldn't normalize at all. tutt

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Quote:
Originally Posted by jdtrbn View Post
Do you mean that the resulting signal sounds louder or that it starts to clip? The first one should not normally happen. It can be that you are overloading your monitoring system with a lot of low frequency content and when filtering them out the system has more power to use for the audible frequencies. If you mean clipping, then it's just that loudness doesn't have much to do with peak levels. EQ cuts can boost the peak levels but should not increase the RMS (perceived) value. This is why you shouldn't use EQ cuts after normalizing (you could get digital clips).
Thanks... Let me clarify:

When I cut certain frequencies with an eq plugin it starts to clip (to go into the red?).

But doesn't the fact that it goes "into the red" mean that the sound is also louder? I thought the channel meters where supposed to show the loudness level...

In any case my rule of thumb when mixing being "to not go into the red both on individual channels and on the master bus", this issue remains problematic.

I'm still confused... Maybe it's because I never really understood the real difference between RMS/clipping/peak levels...

It looks like I have to do my homework on these technichal notions before I can really understand why when I cut out certain frequencies of a sound, its relative channel meter on my daw goes into the red (which I assumed to be something that should not happen... but maybe I'm wrong here too)
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Which EQ are you using? The Waves RenEQ does this alot, many others don't do it quite as much.

It's to do with the narrow Q, I suspect. Most EQ's tends to boost a tad bit on the freq's around a cut - that is if you're doing a cut at 210 hz, you'll get a slight boost in the 210-230 hz area. There's an expression for it but I can't seem to remember it.

Edit: I just did an example in Logic with a relatively narrow Q. Had I narrowed the Q even more, the boost would've increased. Picture is attached.

Edit again: And yes, even on a 32-bit float system, there's no reason to go into the red if it's avoidable. When I mix, I usually have my mix peak at -6 dB so the Mastering Engineer has room for working. If I need to show roughs to clients, I just boost up against 0,0 dB on the SSL BusComp I have on my 2bus when mixing and put on a Waves L2 limiter that attentuates 1-3 dB on peaks.
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EQs increase the overall level when cutting out frequencies... WHY!-logic_eq.jpg  

Last edited by T. Gundersen; 11th July 2008 at 02:30 PM.. Reason: Added picture
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"Most EQ's tends to boost a tad bit on the freq's around a cut - that is if you're doing a cut at 210 hz, you'll get a slight boost in the 210-230 hz area. There's an expression for it but I can't seem to remember it."

Ah thanks T! Yes I have noticed this phenomenon too. If someone could come up with the exact expression it would definitely be appreciated.

But is it really the only explanation for the situation I have described?

I'm asking because when using apEQ (that's the one I use along with Waves classic line of EQs) the graphic doesn't show such an eq curve (eg. cut + little boost in the frequencies next to the ones I have cut), yet it still goes into the red.

It's really disturbing...

I really would like to know if there is a reliable eq plugin that will just cut certain frequencies without boosting any others (which assumedly leads to clipping)!!! if that's even possible from a technical viewpoint and desirable soundwise.
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Even though a non-graphic EQ doesn't show what I show on my picture, it still does the same.

If you need an EQ that don't play tricks on you, have a look at Flux ePure. All my colleagues tells me it's a blessing, but I haven't worked with it alot myself (plan on having it though). However it's not cheap..
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technically,its called "apparant level".
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Yes - that was also bugging me out.
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Bob Katz addresses this in Mastering Audio. Imagine a waveform, say 100hz, full scale, so it's hitting 0db on the meters. Now, if you boost a high frequency, you're adding another waveform with a shorter wavelength. This waveform interferes with the lower frequency waveform, basically cutting the top and bottom of the waveform off and replacing it with a smaller faster one. All of a sudden, the overall level appears to have lowered, and you could boost the level a bit, without visibly clipping. The important thing to notice, though, is that your are creating what's called an 'intersample' clip. When you put this on a CD (or even play it back on your monitors, your d/a converter will attempt to create a waveform that is louder than 0dbfs, and while some d/a can do it, many consumer level d/a's can't, so the music will sound clipped and distorted even at low levels.

My explanation is probably a little obtuse, Bob covers it much better! Basically, the high frequencies when boosted caused the overall peak level (not the RMS) to drop slightly. So, if you cut the highs, the overl peak level would rise slightly.
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Hello John. Thank you for your post.

However the situation I was describing is the opposite one:

I cut (not boost) certain frequencies in a sound (let's say: lowpass @ 150) and yet all of a sudden, the channel meter is not anymore at 0db as it was before eqing, but it goes into the red! Keep in mind that I cut (take away) frequencies, which should "logically" (in my own limited knowledge based logic) result in a decrease of the sound on the channel meter... or at most it should result in the channel meter staying at 0db... I don't know.

T. Gundersen provided the beginning of an answer, when he said that certain eqs (I have also noticed this) boost frequencies AROUND frequencies that you cut...

However I would like to know

1) if this is the only valid explanation: at least it makes sense...

2) if there is an eq plugin that doesn't do this: or a plugin that does this as little as possible... T. suggested flux epure... I'll habe to check it.

3) is there a software maybe that could help me see what exactly eq plugins do to a sound when eqing... this could come in handy for plugins that do not have a proper graphic representation of their eq curves...

4) only a digital (vs. analog) issue?
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I'd say you'd be wasting your energy analysing how the different EQ's work - instead you could just try a few good ones, test them on your own material, and get back to making music. I'd try these first: Flux ePure, Sonalksis SV517 Mk2 and the Sonnox EQ.

Of course if you enjoy testing out gear etc., by all means experiment with the measuring etc. Bring back results to the board.

Ps. It's not because it's digital..
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(yeah I'm that kind of guy... remember the kid who would always sit in the front row in class and ask TONS of questions? that was me... to a certain extent: I prefered to sit in the back, where the heater was at)

T, let's put it like this: (here comes the eq/car analogy)

You're driving a car... okay? (I never drove one.... ANYWAY)

Now, you know that there's one pedal to make your car go faster, the other one to make it go slower... I hope I make sense until now.

Wouldn't it bug you out, if you knew that - in certain situations - if you push the pedal that usually makes your car go slower, it will result in your car going faster...

Damn just imagine that. Well, that's how I feel how eqs and cutting frequencies right.

I just need to know, as I'm really striving for perfection...
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An increase in level when you cut isn't necessarily a deficiency in the design of the EQ. Most high-pass filters create a boost at the cut off frequency. Many classic EQ designs (as a previous poster mentioned) boost adjacent frequencies when you cut, or cut adjacent frequencies when you boost. SSL is a good example of this. Furthermore, shelving filters, from my experience, are more likely to exhibit this behavior. These phenomenon have a lot to do with various EQs' character.

If you like the EQ you're using that's doing this, just turn down the output gain. That said, if you're finding this is a consistent problem, you're tracking levels are probably a bit too hot. Just leave yourself some headroom when you record, and you'll be fine.
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Quote:
Originally Posted by Autotune Prophet View Post
Hello John. Thank you for your post.

However the situation I was describing is the opposite one:

I cut (not boost) certain frequencies in a sound (let's say: lowpass @ 150) and yet all of a sudden, the channel meter is not anymore at 0db as it was before eqing, but it goes into the red! Keep in mind that I cut (take away) frequencies, which should "logically" (in my own limited knowledge based logic) result in a decrease of the sound on the channel meter... or at most it should result in the channel meter staying at 0db... I don't know.
Sorry if my post was confusing. I gave the example of adding highs to get a lower level, because that's more common. If you cut high frequencies, you're allowing the lower frequency waveform to rise to it's full peak to peak waveform. The high frequencies cause the waveform to have a lower peak level, make sense? So, when you cut those highs, the low frequency waveforms can go higher, if that makes sense. The waveform isn't 'louder' exactly, but it's positive peak to negative peak distance is larger.

The frequency bump phenomenon is also a possible culprit, depending on how steep the cut is. A steep cut will often have a bigger bump just below the cutoff frequency.

Basically, you should always have a little room below 0dbfs, even with a mastered product. Bob Katz says you should have a minimum of 0.3db headroom, to account for the intersample phenomenon. (just to clarify, I'm talking about peak levels here, not RMS.)
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Thanks John. It makes sense now.

However, the problem seems to be even bigger as what I thought in the beginning... (thank you so much for the link ryst!!! extremely interesting stuff)

It is as if I had only touched the tip of the iceberg with my question.

I also encourage everyone to read the link provided by ryst

Here's an extract... I'm astonished although I have probably only understood half of what's really discussed there:

Quote:
Originally Posted by Paul Frindle @ recforums.prosoundweb.com
Actually (at the risk of putting the cat amongst the pigeons) I can suggest a simple experiment people can do themselves to illustrate this in action in the most graphic way, which should dispel any lingering doubt that it's important.

The aim is to show that what looks like a legal 'signal' way below any red light in your system can still represent something that cannot pass even remotely correctly out of your digital mixer at full level. And also to illustrate how this may affect your sound quality in practice when mixing ITB. It's a kind of worst case scenario - but it illustrates the problem.

You need a W/S like ProTools, a signal generator plug-in that has a good filter section that actually goes flat to 20KHz and rolls off at 24dB/oct or so.

- In Pro tools get a mono channel up,

- stick the PT generator plug-in at the beginning of the channel and set it for sine at say 1-2KHz.

- Follow this with a good filter plug-in set for the max slope at 20KHz. (For example the Oxford EQ plug-in has 36dB/oct at 20KHz and illustrates this well - any other good HF filter should work as well).

- As an initial test set the channel fader at 0dB and note that the PT meters shows the sinewave signal at -6dBr and that putting the filter plug-in in and out using bypass has no effect.

- OK now switch the signal genny to white noise and note that the level on PT is still -6dBr.

- Now un-bypass the filter plug-in and watch the signal level rise dramatically!! In the case of the Oxford 36dB/oct filter the meter level will rise a full 5dBr to nearly flat out.

Ok so what's happening - how is this possible? Well the digital genny plug-in produces sinewaves correctly - but when in noise setting it is just a random number generator driving the output. So although when set to -6dB peak value no sample ever gets to be greater than 50% modulation - a reconstruction of the undecoded SAMPLE VALUES produces nearly full level SIGNAL. Reconstruction means filtering and so the filter plug-in is acting like a partial reconstruction filter (much like a DAC) - which in turn is now feeding a more legitimate SIGNAL which the sample value only meter can read more correctly.

Ok now if this SAMPLE train is passing out of your DAC it too is being reconstructed correctly - so this -6dBr noise from the genny would a produce nearly full modulation SIGNAL if you fed this to the DAC directly - even though no sample gets to be bigger than 50% and no reading say's it's bigger than -6dBr.

If your filter is a good one you should be able to switch it in and out and hear no difference in the sound of the signal from your DAC - despite the PT meter reading wildly different. The filter has neither added nor taken anything significant out of the intended audio signal - but you have nearly doubled the sample values within the PT channel!

Ok, now wind the genny level up to say -2 or -3dB (still less than only 75% full level) and do the same thing. What happens? Well it now clips when the filter is in (samples bigger than flat out) - now the sound definitely changes when you switch the filter in and out - because it is mathematically limited and in error when the filter is in - cos it cannot pass through TDM slot at the output of the filter!!

That is what would be happening in your DAC, it would saturate if you sent this at only 3dB setting on the genny - reading -3dBr within the mixer itself, straight to the output!!

Ok now what does this mean for a mix? Well with all those mixed signals, cymbal crashes, HF EQ and limiting etc.. how close do you imagine the output signal can get to being a bit like white noise in places within a real production - even if none of the contributing channels hit the red light? Is this not the exact register of what we term as 'air' and 'resolution'? And people are aiming at max possible mix output levels on meters that do not show SIGNAL.

So why does an OTB mixer apparently sound better than an ITB mixer when you are modulating your digits close to 0dBr (sample value) all over the place? Well all those DACs (flawed as they may be) are acting to legitimately reconstruct your programme - before - you mix them all together and produce too many illegal signals that cannot pass out of your digital mixer! Paradoxically, the loss of sound quality due to all those converters is not as bad as the illegal signals created within the digital mixer by the 'too hot' signals you are trying in vain to pass out of the system.

It is not a summing issue at all (the one thing digits CAN do is add up almost perfectly). It's an illegal output problem caused by the fact that there are no meters that display actual SIGNAL in your whole mixing environment - you simply never see it happening.

So - go back and get your fav test mix back up on your W/S, re-mix the whole thing making sure that at every place in all chains (including between all plug-ins) never gets bigger than -6dBr. Make sure your final output after any limiting etc also never peaks beyond -6dBr. Now do the comparison between this ITB mix and a similar OTB mix. You might have a big surprise
Does anybody dare to say something on this
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That's an interesting post, Paul Frindle is an extremely well regarded individual!

I think it's just one more example of why it's important to maintain a significant amount of headroom throughout your signal path in the digital domain. At 24 bit (and 32-64 bit floating point in the box,) there's no reason to have signal hitting anywhere near clipping. I admit that I sometimes have to go back and lower faders, etc, but it's best to avoid any possible errors. Sometimes you can hear the difference, sometimes less so, but there are some processes that really benefit from having lots of headroom, compression and eq being two (for technical reasons that Frindle or others have explained elsewhere.)
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But isn't Paul Frindle (with comments such as "Now un-bypass the filter plug-in and watch the signal level rise dramatically!! In the case of the Oxford 36dB/oct filter the meter level will rise a full 5dBr to nearly flat out") suggesting that DAW level meters are not telling you exactly what's going on (level-wise) in your mix?

That's at least what I understood...

Let me be clear:

Is there a difference between the level meters of a console and that of a daw? VU meter vs. daw meter?

Will using a plugin such as PSP Vintagemeter permit me to bypass the problem of daw meters as discussed by Paul Frindle?
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I don't know about the PSP meter, you need what's called a 'Reconstruction Meter.'

Try the SSL X-ISM meter if you want to experiment. Here's a pretty good explanation:

Solid State Logic X-ISM Peak Meter at Kaos Audio

What Paul Frindle is talking about in his example is kind of specific to white noise, and while it's possible to have those 5dBr boosts, it's more likely they'll be smaller than that. It mostly applies to situations where there's a lot of energy near the filter point, like a mix with a lot of energy up near 20k. When it gets filtered by the DAC in a CD player for instance, the lower frequencies can peak above 0dBFS, which in some players will result in distortion and clipping.

There's no reason to be anywhere near these levels when mixing. A qualified mastering engineer can get your mix up near 'commercial cd levels' while keeping an eye out for these distortion issues.
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Quote:
Originally Posted by John Suitcase View Post
There's no reason to be anywhere near these levels when mixing. A qualified mastering engineer can get your mix up near 'commercial cd levels' while keeping an eye out for these distortion issues.
Exactly. Recording and mixing levels don't need to be hot. There are still a ton of people that don't understand or practice this. Most of my sessions I get from clients to mix still have a lot of levels WAY TOO HOT.
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Quote:
Originally Posted by John Suitcase View Post
I don't know about the PSP meter, you need what's called a 'Reconstruction Meter.'

Try the SSL X-ISM meter if you want to experiment. Here's a pretty good explanation:

Solid State Logic X-ISM Peak Meter at Kaos Audio

What Paul Frindle is talking about in his example is kind of specific to white noise, and while it's possible to have those 5dBr boosts, it's more likely they'll be smaller than that. It mostly applies to situations where there's a lot of energy near the filter point, like a mix with a lot of energy up near 20k. When it gets filtered by the DAC in a CD player for instance, the lower frequencies can peak above 0dBFS, which in some players will result in distortion and clipping.

There's no reason to be anywhere near these levels when mixing. A qualified mastering engineer can get your mix up near 'commercial cd levels' while keeping an eye out for these distortion issues.

The reason this happens is that the EQ is changing the shape of the waveform and so the peak level it reaches is higher. This is due to relative freq response changes and phase shift - and is not a fault of either the EQ (analogue or digital) or your system :-)

All wave forms except sine waves (however complex) can be thought of as a proportional mixture of lots of other frequencies (harmonics) and if you change that relationship - even by reducing one of them - it can cause a higher peak value.

One of the most obvious is a square wave - which is a proportional mixture of all odd harmonics, 1 3 5 7 ..... etc. If you reduce one of those harmonics the top of the square wave will no longer be flat and will get bigger in peak value.

You can try this out on any workstation. Get a low freq square wave from a generator plug-in and look at the waveform on your recording display. Then insert an EQ after the generator and start cutting the LF - and watch the peak levels rise as you reduce the level at some freqs. Record a bit of it and see what happens on the waveform display when zoomed in.

The worst possible case is if all LF is removed - and this will cause around 6dB of extra peak level!

Of course your programme does not hopefully consist of square waves - but it is harmonically rich (i.e. not a pure sine wave tone), so the same applies :-)

I hope this helps :-)


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

BTW I should add that this is why if you are trying to reach max modulation and loudness on your buss output all EQ must be done before your buss limiter - and never afterwards.

This is because a (conventional) limiter will attempt to squash down all program waveforms to exactly match peak levels on your meter - and changing it in any way afterwards will result in higher peak levels that will clip. And the only way to reduce the peak errors (or extra distortion) is then to reduce the level - and that reduces the loudness as well.

Not to turn this into a plug for our product - one of the main reasons the DSM plug can achieve louder results with somewhat less distortion over a wider range, is that it works dynamically with the freq content of the program in a complex fashion, before it applies it's limiting processing. In effect it's a multi-dimensional limiter - as opposed to a conventional single dimension 'value only' limiter.
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So. Are you guys saying that daw meters are less accurate than analog desk meters?
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So. Are you guys saying that daw meters are less accurate than analog desk meters?
Heh. Define "accurate"...
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So. Are you guys saying that daw meters are less accurate than analog desk meters?
Well, now that Paul is in this discussion, he can give a better explanation than I could!

My understanding is that DAW meters give an accurate peak reading (value-wise), but the D/A converter can create a waveform that has a higher peak, because of what Paul is talking about with regards to eq. So, the 'weird' effect of a signal level going up because of filtering is neither digital nor analogue, but it's a concern in the digital world because of the clipping that can occur, even when the meters in the DAW show that it's not clipping.
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No...I mean DEFINE what you mean by "accurate".

As in, "translates to real-world RMS reliably", or "detects peaks to a hundredth of a dB", or "finds and alerts the user to intersample peaks", or "all of the above".

Accuracy in meters is, oddly enough, highly subjective.
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