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Linear Phase EQ vs Channel EQ


emilenitrate

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Any other pointers regarding choosing between linear & non-linear phase eq are appreciated. The simpler the better, I still have no idea which to choose. - Emile

 

I really don't know how to answer those kinds of questions. What guitar should you use? What amp? Should the sound be clean? Or distorted? Should you scream the lyrics or should you have a breathy voice? It's all about taste, about using your ears and making decisions.

 

Try both, listen and pick the one you like for the task at hand. If you really don't know which one to pick, pick the Channel EQ: it sounds better - to me (maybe that's the simpler answer you were looking for? :lol: ).

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I understand what you're saying and of course it makes perfect sense. In the final analysis of course I'm going to a/b and try to decide what sounds best however...

 

I think what I'm looking for is for users to say somethings like, when you shelve up to 120 hz to reduce low end haze that gets in the way of bass & kick parts use non-linear because of blah blah blah. Or the opposite! I don't really know and more experienced mix oriented users can point me in some general directions that of course, I'll question and experiment for myself, but guidelines and common wisdom are usually good starting points.

 

To draw on the guitar analogy, real blackface Fender amps sound great when you turn the treble all the way up, the bass midway, and the volume past half, but maybe that's too bright for a Tele. It doesn't mean you have to do it, but it could be helpful to someone who's only used Line 6. Sometimes it helps to know what is that you're "supposed" to be hearing. That doesn't mean that it's the truth, but possibly one of several truths.

 

Actually, use the channel eq because it sounds better is concrete advice. Kind of a plain blanket statement but still worth keeping in mind when made by someone so familiar with the program. Thanks, I'm not done with this. - Emile

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In most cases you're better off using the Channel EQ. Use it for treating individual channels in the mix.

 

A quick tip: try using a 12 dB/Oct HPF (low cut) instead of the default 24 dB/Oct. You have to set your frequency (cut off point) a bit higher but it will generally sound better because the filter ringing will not be as obvious.

 

Once in a while when mixing or mastering you will experience a situation where activating a HPF changes the peak level on the channel or master bus dramatically. This happens due to a particular "unlucky" phase change/delay. Phase changes are normal when equalizing, in fact that's how an equalizer works in the first place. But in this instance it could be annoying since the peak level changes so much. Large peak changes could cause overloads, cause dynamic plug-ins to pump, and eat up your headroom.

 

Using a linear phase equalizer keeps the so-called group delays in control. The phase change that caused the peak to change dramatically before will no longer happen. In these cases you should think about switching to a linear phase equalizer.

 

I hear you say "why shouldn't I always use a linear phase equalizer then?". Because there's no such thing as a free lunch.

 

With a minimum phase equalizer (such as the Channel EQ) you get some post filter ringing, i.e. a small ringing effect after the sound. This is normal and part of the design. Try imagining a snare drum with a short tail added by the eq (an exaggerated example).

 

Now try imagining that same snare drum with the added tail in front of the sound. With a linear phase equalizer you get the filter ringing as ghost image before the transient. This can blur transients, making your music appear less punchy or precise. Also we are quite used to the sound of minimum phase equalizers which means that linear phase doesn't sound quite "right" to some people.

 

There's no right or wrong answer to your question but in general I would advice you to use the Channel EQ when mixing, and maybe use the linear phase equalizer during the mastering phase or whenever you experience a dramatic change in peak level when equalizing.

 

While I use hardware for equalizing in the mastering process, I often strike a balance. I might use a linear phase for the HPF and a minimum phase eq for the rest.

 

Since Logic allows you to switch between the Channel EQ and Linear Phase EQ on the fly (simply change the plug-in on the insert slot) you can do an A/B test quite easily. Your settings in the plug-in are retained.

Edited by lagerfeldt
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Yes, the linear phase will introduce much more latency.

 

However, if you're using full plug-in delay compensation in Logic Pro, it will not be an issue unless you're playing or recording instruments or integrating outboard in your setup.

 

sure but when having alot of parallel and serial routing with HW IOs etc each little latency inducer adds up to a large in the end... :)

 

Its just a thing to take in consideration aswell when going to use it on channeltracks. The standard eq is fairly RT 0 msec latency

 

The linearphase eq is pretty accurate for mastering, low cutting etc

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Emile,

Speaking of parallel processing, linear phase filtering is useful, (if not required) when EQ'ing the return of a parallel process. For example say you're sending a bit of the drum bus to a parallel comp. EQ'ing the return would require linear phase filters because a minimum phase filter will twist the signal phase, as lagerfeldt pointed out. Mixing this signal back together on the stereo bus with the original would result in some comb filtering of the drum sounds.

 

While that might be the sound you're after, you've now got a situation that's difficult to control and unpredictable.

 

In general I agree with David and lagerfeldt that minimum phase filtering is the most useful as a creative filtering tool. But there are times when they can be deadly to transients, especially with irresponsible settings. I started an article here that looks at some of the effects and differences of filtering musical signals. You may find it interesting. I'll finish it one of these days.

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Thanks to David, lagerfeldt, Septicsound and fader8 for attempting to answer my questions. You've all provided me with some insights and a general direction to start experimenting in (fader8's compression bus example). I've known that changing the plugin from channel to linear phase retains the settings but wasn't able to hear a difference. Now I have an idea of what to listen for. Thanks again. - Emile
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If you really don't know which one to pick, pick the Channel EQ: it sounds better - to me (maybe that's the simpler answer you were looking for? :lol: ).

 

It might be mean to say but in my experience the older people are (ergo the more exposure they had to analog equipment) the more they tend to lean towards the channel EQ.

 

If one is used to think digitally then colouring the sound and EQing the sound are not necessarily the same thing. I love how the lin. phase EQ does not add any color to the sound. (const group delay).

 

Think it is quite funny how we now in the digital age when one can actually do what mixing intended: Mix a signal without colouring it. No worries about how a console sounds or a EQ or so. And now suddenly everybody screams for the analogue flavour they've gotten so used. (Though I don't think that was ever the intention when the first consoles were build, more part of what was possible or a side effect with technology at that time)

 

In the end I think it's whatever u grow up with. I love line phase EQs, they sound so clear and defined. Add tube or tape sim afterwards if u want to an analogue flavour.

 

cheers

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Think it is quite funny how we now in the digital age when one can actually do what mixing intended: Mix a signal without colouring it.

If only that were true for filtering, but it's not. Digital filters must follow the same laws of physics that analog filters do. For linear phase filters particularly, there are scenarios where the pre-ring artifacts are completely unacceptable, softening attacks, etc.

 

 

And now suddenly everybody screams for the analogue flavour they've gotten so used. (Though I don't think that was ever the intention when the first consoles were build, more part of what was possible or a side effect with technology at that time)

 

In the end I think it's whatever u grow up with. I love line phase EQs, they sound so clear and defined. Add tube or tape sim afterwards if u want to an analogue flavour.

 

I see your point, but tube and tape sim is an altogether different effect than what is achieved by filtering. They add harmonics and intermod, but do not effect the signal in the time domain by shifting the phase at different frequencies, which is an entirely different form of timbre shift.

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Think it is quite funny how we now in the digital age when one can actually do what mixing intended: Mix a signal without colouring it.

If only that were true for filtering, but it's not. Digital filters must follow the same laws of physics that analog filters do. For linear phase filters particularly, there are scenarios where the pre-ring artifacts are completely unacceptable, softening attacks, etc.

 

 

And now suddenly everybody screams for the analogue flavour they've gotten so used. (Though I don't think that was ever the intention when the first consoles were build, more part of what was possible or a side effect with technology at that time)

 

In the end I think it's whatever u grow up with. I love line phase EQs, they sound so clear and defined. Add tube or tape sim afterwards if u want to an analogue flavour.

 

I see your point, but tube and tape sim is an altogether different effect than what is achieved by filtering. They add harmonics and intermod, but do not effect the signal in the time domain by shifting the phase at different frequencies, which is an entirely different form of timbre shift.

 

Alright that's true, should have said allpass or so. But as with all - the picture is not just black and white. I use Waves Rcomp quite a lot. I still think that it is very funny that now with digital technology we can do what mixing was intended for in the first place. But now we don't want to, ...

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If only that were true for filtering, but it's not. Digital filters must follow the same laws of physics that analog filters do.

 

Actually not. Digital filters are a very different story than analogue ones. It takes quite some effort to design digital filters to model the analogue ones half decent.

I spent some years of my professional carrier doing that. Fun but painful ;).

 

but yes - in the end - there are lots of common principles.

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It takes quite some effort to design digital filters to model the analogue ones half decent.

Agreed. Difficult to get all the aspects the same. And like you, sometimes I wonder if it's really worth going there. Point's moot I guess as there's no shortage of EQ plug-ins out there. Who'd have thought the market would be this saturated, eh? I'm sure there are still more to come.

 

Now if there was an EQ that had "ALL" the features "I" want . . . nah, an impossible dream. :(

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I see your point, but tube and tape sim is an altogether different effect than what is achieved by filtering. They add harmonics and intermod, but do not effect the signal in the time domain by shifting the phase at different frequencies, which is an entirely different form of timbre shift.

 

But real analog tape does affect signals in the time domain by smearing transients. This comes about simply because regions of the tape that contain sharp transients and are strongly magnetized bleed to nearby regions on the tape. So the transients are not as sharp. In the most extreme cases rolls of recorded tape that were left rolled up for a long time even had ghost audio in places from other layers of the tape.

 

The sound of that transient smearing . . . turning a thack in to a ke-thunk . . . is what I've always thought that tape sim plugs were missing.

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Time to team up with a group of developers :).

That would be fun! Hmmm . . my own personal EQ . . .

 

But real analog tape does affect signals in the time domain by smearing transients. This comes about simply because regions of the tape that contain sharp transients and are strongly magnetized bleed to nearby regions on the tape. So the transients are not as sharp. In the most extreme cases rolls of recorded tape that were left rolled up for a long time even had ghost audio in places from other layers of the tape.

That machine would have to be in pretty bad shape and running at 3-3/4ips for this to happen at anything other than VERY high frequencies. But a well tuned machine at 15 or 30 ips with good quality tape, properly biased and aligned, wouldn't exhibit much audible high frequency smearing.

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