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Why do Helix Cabs and IRs Need Low and Hi Cuts?

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I am probably misunderstanding how Impulse Responses work so help me out here.  I thought that an Impulse Response would be convolved with an input signal/sound and produce an output matching the source (cabinet).  So for example if you feed into it a signal that has frequencies beyond what it is capable of reproducing then the output would not have those frequencies.  If that is the case why would there need to be a low and hi cut on the helix cabs and IRs?  Wouldn't those frequencies you are cutting not be present in the output because that's what the cab/IR already does?

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In a perfect world, we would not need adjustments on anything. It would just sound "perfect". Sadly we don't live in a perfect world (in any respect), and all of our ears hear differently.

 

There will be a "more detailed" explanation coming of how IR's actually work in Helix, but when that doesn't work for ya please see above.  ;)

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Because cabs in a room Rumble. Reflections bounce and give off high frequencies. Microphones pick all this up

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I'm no convolution expert and I'd love to have someone who is more versed in this technology give us an idea of what actually causes this effect.  But to clarify a bit, as others have noted a speaker cabinet IR is more than just the sound of the cabinet.  It's also the effect of the room, the physics of the environment and the effect of the amp producing the signal that plays into what eventually becomes the IR.  In researching this situation I've come to understand also that one thing that gets in the way of a perfect representation of an IR is the differences in the impedance load between the signal chain that produced the IR and the one that's reproducing it.  This is likely the thing that comes into play particularly when using something like a higher end FRFR type amplified speaker which is specifically designed to try and maintain a flat load across the full sonic range.

 

I've also come to understand that some of this may also be just in the way we hear things.  For example, depending on my position relative to my Yamaha DXR12, there can be a significant difference in what I'm hearing due to the design of the speaker.  Powered speakers like this are designed primarily for projecting sound across long distances as compared to traditional cabinets.  Therefore the closer I stand to the speaker the more I hear of he compression driver rather than the full mix of the compression driver and the main speaker.  Standing further away and off axis typically changes this response considerably.  That makes sense when you look at how these speakers are meant to be applied as PA speakers.  No one puts PA speakers right in the face of the audience because they would be too harsh.  When you think of how people use these speakers by placing them right at their feet facing directly into their face it's likely to sound much harsher then it would a few feet away.  This is one of the reasons that in live performance I place my DXR12 a few feet behind me as I would a traditional amp cabinet which results in a much more realistic representation with less harshness.

 

These are my conclusions based on my experiences but I'd love to hear a more technical explanation if there is one.

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essentially as yer man above suggests, it's to further tune the output to be more pleasing to the ear.

basically at the bare bones of it, an IR block in Helix is just a complicated EQ and compression process that conforms an input signal to a stored reference value.
now because your IR is a full range reference of the frequency response curve of a given speaker, and that speaker will be able during a sine sweep to produce low sub tones and sizzly highs as part of it's sound, the algorithm doing the processing is gonna add a bunch of subby tones and a bunch of sizzly high end that you dont always necessarily want to hear so it's a good idea to have the option to roll those off right in the impulse block so that you aren't feeding those frequencies you dont want into your next block or to your output.

you should always keep in mind that you are essentially trying to manipulate the sound in the same way a studio engineer might when recording a guitar and cutting the super lows and super highs on the track EQ is very common practice when recording guitars and other instruments in the studio.

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you should always keep in mind that you are essentially trying to manipulate the sound in the same way a studio engineer might when recording a guitar and cutting the super lows and super highs on the track EQ is very common practice when recording guitars and other instruments in the studio.

It is very common in the studio environment, matter of fact it is common in the studio environment to rolling everything to mono ("side level eq roll off") at around 250hz, and below. This free's up a lot of room/data in the low end that you will never hear, but gives more room for the mix to breathe. Most audio below 250hz starts to lose its directional bias, even more the further low you go. (this is why home theaters can have the sub in the corner of the room, and still have it fill the whole room) Making the stereo information in that spectrum flubby, unuseful, eating up headroom, while providing very little in the way of benefit. So it is best to just roll it off to mono.

 

I achieve this by using Fabfilter Pro-Q2 on my master mixer channel (arguable the best software EQ available) put it in mid/side mode.  Then I put a low cut at around 250hz to roll off the "side" signal, so everything below it stays "mid" signal (mono)  I even then around the 30hz mark roll of that "mid" signal just to get rid of a little of that rumble clustering from all the instruments adding unwanted noise to that area. (tightens up the sound)

 

You would think lobbing off frequency spectrums would make the sound thin, and tiny. But even though it seems counter-intuitive, in the mixing world you will find the opposite to be true quite often.

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So some of you are saying that an IR of a cabinet WILL produce frequencies that a real world cabinet would not produce and that's why we need to use the low and high cuts.  Others of you are saying the IR produces the SAME frequencies that a real world cabinet does BUT not all of those are pleasing or beneficial and so we cut them in helix the same way a sound guy would cut them at the board. 

 

The latter makes more sense to me from my (limited) understanding of how impulse responses work.  I was not aware that those low and high cuts at the board were a standard thing.  Cutting the lows makes sense from a mix standpoint to make room in the mix.  But I was under the impression that the sound coming out of the cab didn't have frequencies above a certain range because of the speakers used for guitars (not full range).  If that's the case I don't understand why it's standard practice to again cut those frequencies at the board (or in our case on the helix cab or IR). 

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never tried that EQ before. added to my ever growing list of "oooh a new plugin to play with" :)

It isn't cheap, so try to catch it on sale, Audiodeluxe.com has sales sometimes. 

 

Fabfilter also has an ongoing loyalty discount system. Meaning the more Fabfilter plugins you own, the more discount you get when ordering directly through them. Those discounts also stack with any other sales (including holiday sales) which are coming fast.  ;)

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So some of you are saying that an IR of a cabinet WILL produce frequencies that a real world cabinet would not produce and that's why we need to use the low and high cuts.  Others of you are saying the IR produces the SAME frequencies that a real world cabinet does BUT not all of those are pleasing or beneficial and so we cut them in helix the same way a sound guy would cut them at the board. 

 

The latter makes more sense to me from my (limited) understanding of how impulse responses work.  I was not aware that those low and high cuts at the board were a standard thing.  Cutting the lows makes sense from a mix standpoint to make room in the mix.  But I was under the impression that the sound coming out of the cab didn't have frequencies above a certain range because of the speakers used for guitars (not full range).  If that's the case I don't understand why it's standard practice to again cut those frequencies at the board (or in our case on the helix cab or IR). 

 

Well part of what you say is true in that guitar speakers have a more limited range, kind of.

 

For example, if you look at the specs for a Celestion Creamback it lists the response range as 75 - 5000hz.  That more the working range than the technical range if you look at the frequency response graph for the speaker which really goes across the whole spectrum but responds differently throughout the range.

 

graph.gif

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My experience is that the built in cab blocks generally require EQing to even be usable (to me), particularly with distorted tones. However, the IRs I have sound just fine without any cuts, though I generally find that adding a high cut gets rid of some of the fizziness when playing through a PA speaker. That doesn't surprise me a lot. If you've ever stuck your ear right up against a real cab (not recommended), the sound is way different than what you hear in the room. For my own comparisons, I compared what I was getting from the Helix to what I recorded of the same amp through a similar cab and messed with IRs until I got very close. What I have now is actually a little better overall than the real thing, but that could just be down to mic placement. At any rate, I'd normally EQ that recorded tone similar to what I do with the IRs now anyway. The high cut doesn't make a huge difference until it gets fairly aggressive, generally down around 10 kHz or lower is where I can start to actually hear any real difference. For the built-in cab blocks, I notice it before that, which says to me that there is a lot more unnecessary frequency information in those blocks.

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If that's the case I don't understand why it's standard practice to again cut those frequencies at the board (or in our case on the helix cab or IR). 

 

The reason is mainly that what the speaker is producing in those ranges outside it's specification is generally considered to be useless noise rather than "tone".

 

We're simply cutting out the parts that are not useful or pleasing so that we may more clearly here the part we like. :)

 

When you look at the response curve of a speaker it will always show you 20hz to 20khz but the speaker itself may only be specified to operate between say 200hz and 6khz. it might be easier said that should be read as "the bit of the spectrum that sounds nice in this speaker is 200 - 6000hz", if that makes sense...

 

Our IRs manipulate the incoming signal across the entire 20-20k range but if the IR is that of a speaker that is only good between 200-6000hz we only want to really keep that bit of the spectrum and maybe a little above it to preserve the presence of the sound.

 

As a live engineer, for me, it is automatic to high pass a guitar at 120hz without even hearing it. Sometimes i'll shift that back to 100 or forward to 250 to taste. If the room is large a low pass may not be needed but in a smaller or particularly reflective room (stone walls, arched roofs) a low pass at 10k is not uncommon on a mic'ed guitar because those high frequencies already have enough energy in them to propagate and reflect violently throughout the room.

 

You might actually be amazed by some of the mental EQ curves that get used in a live situation to make something sound good. I'm mixing tonight, if i remember I'll snap some pics of what rig was getting used and what the curve was like just for fun.

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So some of you are saying that an IR of a cabinet WILL produce frequencies that a real world cabinet would not produce and that's why we need to use the low and high cuts...

The IR's aren't really to blame for the need for hi/low cuts...any FRFR monitoring method (studio reference headphones, FRFR speaker(s), studio monitors, etc) is...Unless you're running through a "real" guitar amp/cab rig, it's the FRFR speakers that are producing frequencies outside the range that a guitar cabinet typically spits out. If you were to run through a "real" amp and cabinet, you would find less of a need to apply those cuts, as the physical limitations of the cabinet would be doing it for you.

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Its because the speaker/cab impulse responses are created by close mic-ing the speaker. This is to isolate the sound of the speaker from room reflections that would otherwise be doubled when you play through the cabinet IR. Stick your head up real close to your guitar speaker and you'll easily see why you need to high and low cut. Its not a natural place to listen to a speaker. But it provides the isolation and gives a lot of high end that you can cut if you need to. This is  a good compromise that provides some additional flexibility for tone shaping with mic selection and post - cab EQ. This is typical to what you would do with a mic'd cabinet live or in the studio - both usually require some post EQ. Its better to have too much and have to cut than too little and have no options. Or another way to look at it is that it's easier to EQ away what you don't need then it is to create something that isn't there.

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 If you were to run through a "real" amp and cabinet, you would find less of a need to apply those cuts, as the physical limitations of the cabinet would be doing it for you. 

 

 

 

And we have a winner! 

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The IR's aren't really to blame for the need for hi/low cuts...any FRFR monitoring method (studio reference headphones, FRFR speaker(s), studio monitors, etc) is...Unless you're running through a "real" guitar amp/cab rig, it's the FRFR speakers that are producing frequencies outside the range that a guitar cabinet typically spits out. If you were to run through a "real" amp and cabinet, you would find less of a need to apply those cuts, as the physical limitations of the cabinet would be doing it for you.

An IR reproduces the time varying frequency response of a speaker, cab, and mic combination through convolution. A well captured IR should be very close to the original response of the speaker as seen from the position of the chosen mic. If you run through a good FRFR, then you should hear an accurate representation of that same frequency response. So it is generally not the case that an IR and FRFR is directly the reason for needing high cuts.

 

Rather there are two indirect reasons. First, what the mic hears is not what you would hear from that guitar cabinet in the room. That’s simply because the mic is in a very different position than you are in the room. You’re a long ways away, very off axis, and getting a lot of room reflections. That significantly changes the response you hear, but not the response of the speaker or the IR that models it. Put your ear in the same position as that mic and you’ll quickly know why high cuts are needed.

 

Now its certainly possible to capture an IR at the normal human listening position. This would indeed capture the sound of the amp in the room. But when you put that through a FRFR, you get the amp in the room twice since the FRFR is also in a room. This will generally not sound vary natural. But it is possible to blend and IR that captures a room mic into your IR. That could reduce the need for high cuts.

 

The second reason high cuts are often needed is because a FRFR usually has much wider dispersion than a typical guitar speaker. So the amp in the room sound from a FRFR at the same listening position will sound brighter simply because what high frequencies that are being accurately generated by the IR are spread out more.

 

Finally, if you close mic a speaker cabinet in an isolation booth, and capture an IR of that same speaker and mic, they will both sound pretty much the same reproduced through the same FRFR. Its likely both will require high cut in a mix.

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An IR reproduces the time varying frequency response of a speaker, cab, and mic combination through convolution. A well captured IR should be very close to the original response of the speaker as seen from the position of the chosen mic. If you run through a good FRFR, then you should hear an accurate representation of that same frequency response. So it is generally not the case that an IR and FRFR is directly the reason for needing high cuts.

 

Rather there are two indirect reasons. First, what the mic hears is not what you would hear from that guitar cabinet in the room. That’s simply because the mic is in a very different position than you are in the room. You’re a long ways away, very off axis, and getting a lot of room reflections. That significantly changes the response you hear, but not the response of the speaker or the IR that models it. Put your ear in the same position as that mic and you’ll quickly know why high cuts are needed.

 

Now its certainly possible to capture an IR at the normal human listening position. This would indeed capture the sound of the amp in the room. But when you put that through a FRFR, you get the amp in the room twice since the FRFR is also in a room. This will generally not sound vary natural. But it is possible to blend and IR that captures a room mic into your IR. That could reduce the need for high cuts.

 

The second reason high cuts are often needed is because a FRFR usually has much wider dispersion than a typical guitar speaker. So the amp in the room sound from a FRFR at the same listening position will sound brighter simply because what high frequencies that are being accurately generated by the IR are spread out more.

 

Finally, if you close mic a speaker cabinet in an isolation booth, and capture an IR of that same speaker and mic, they will both sound pretty much the same reproduced through the same FRFR. Its likely both will require high cut in a mix.

I'm not gonna dispute any point in particular because this would quickly devolve into a semantic argument, as we're essentially saying the same thing. The physics and technical wizardry that allow these devices to do what they do doesn't concern me...but the practical application thereof, does. And to that end, if one were to take the "real" guitar amp of their choice and run the line out into a powered FRFR speaker, it will sound like a giant pile of fizzy $hit, despite the complete absence of an IR, and no matter how far off axis your head is. The same amp with the same settings through a guitar cabinet, will sound normal. Both scenarios are equally true for a modeler. So why do we need high cuts? FRFR...that's why.

 

Furthermore, since no one sits and plays with their head an inch from the speaker cone, all the talk about the faithful reproduction of captured sounds, dispersion, reflections, etc... while it might be technically "correct", it's really neither here nor there. At the end of the day, if you choose to run FRFR, you'll need high cuts...and if you play through a guitar cabinet, you won't. I've run modelers both ways, and that's how it shakes out.

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I'm not gonna dispute any point in particular because this would quickly devolve into a semantic argument, as we're essentially saying the same thing. The physics and technical wizardry that allow these devices to do what they do doesn't concern me...but the practical application thereof, does. And to that end, if one were to take the "real" guitar amp of their choice and run the line out into a powered FRFR speaker, it will sound like a giant pile of fizzy $hit, despite the complete absence of an IR, and no matter how far off axis your head is. The same amp with the same settings through a guitar cabinet, will sound normal. Both scenarios are equally true for a modeler. So why do we need high cuts? FRFR...that's why.

 

Furthermore, since no one sits and plays with their head an inch from the speaker cone, all the talk about the faithful reproduction of captured sounds, dispersion, reflections, etc... while it might be technically "correct", it's really neither here nor there. At the end of the day, if you choose to run FRFR, you'll need high cuts...and if you play through a guitar cabinet, you won't. I've run modelers both ways, and that's how it shakes out.

But that guitar cab that you mic up will probably need high cuts if you put it through the PA, because proximity and the speaker response... FRFR just puts out what you put into it, its not magically adding some high end into the sound that isn't there. I'd be pretty happy if more guitar players would at least check their tones with their head at a close distance and on axis from the speaker, it wold save a lot of us from crappy guitar tones when we go to see bands live. What your rig sounds like 10' and front and 3' off axis is not an accurate representation of what the rest of us hear, especially if we're in front of a stage and the cab is at head level. Guitarist think they're swimming in tonal bliss while everyone else in the room is quickly acquiring tinnitus. 

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But that guitar cab that you mic up will probably need high cuts if you put it through the PA, because proximity and the speaker response... FRFR just puts out what you put into it, its not magically adding some high end into the sound that isn't there. I'd be pretty happy if more guitar players would at least check their tones with their head at a close distance and on axis from the speaker, it wold save a lot of us from crappy guitar tones when we go to see bands live. What your rig sounds like 10' and front and 3' off axis is not an accurate representation of what the rest of us hear, especially if we're in front of a stage and the cab is at head level. Guitarist think they're swimming in tonal bliss while everyone else in the room is quickly acquiring tinnitus.

Yes, that's true...but it's also what sound guys are for. There is never going to be an all-encompassing solution to this, except not standing right in front the stage...I don't. It's the worst place to be at a show, for exactly that reason. But what good would it do to tweak sounds with my head mashed up against the grill of a cabinet if its not going to stay there? I need to hear myself through the din on stage too...burying myself in muddy tone to the point where I can't hear what I'm doing out of concern for those who insist on being 6" from the stage, will result in a $hitty performance. If the guy at the board does his job (and isn't deaf ;) ), then it's a non-issue for the other 98% of the people in the room anyway...

 

Besides, I don't have a cabinet on stage anymore. Just an L2T pointing up at me, and Helix going straight to the PA. Nobody's getting tinnitus from my rig, lol...

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But that guitar cab that you mic up will probably need high cuts if you put it through the PA, because proximity and the speaker response... FRFR just puts out what you put into it, its not magically adding some high end into the sound that isn't there. I'd be pretty happy if more guitar players would at least check their tones with their head at a close distance and on axis from the speaker, it wold save a lot of us from crappy guitar tones when we go to see bands live. What your rig sounds like 10' and front and 3' off axis is not an accurate representation of what the rest of us hear, especially if we're in front of a stage and the cab is at head level. Guitarist think they're swimming in tonal bliss while everyone else in the room is quickly acquiring tinnitus. 

 

When I was using a real amp/cab, this is exactly how I would check my tone. Since I was using a 2x12, it pretty much always sound dull and generally not great at ear level, but it sounded fantastic through the PA. To the argument that "that's what sound guys are for," no. That's what your EQ is for. No one but you is going to hear your amp at the position you hear it. If you are miked up, they are hearing it through the mic, and yes, a sound guy could EQ it, but you've also got to think about the people up front who are going to hear the unfiltered sound of the cab blasting at their ears. If you are not being miked, then there is no sound guy to apply EQ to your awful tone. So, do you EQ to work for just you or do you EQ to work for everyone else in the venue? Thinking it is up to the sound guy to fix your crappy tone is a good way to irritate every sound guy you work with. Which, in turn, will make you sound awful because they aren't going to want to help you out.

 

Edited because apparently the forum likes to change it to "lollipop off every sound guy." Which is amusing but also frustrating.

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When I was using a real amp/cab, this is exactly how I would check my tone. Since I was using a 2x12, it pretty much always sound dull and generally not great at ear level, but it sounded fantastic through the PA. To the argument that "that's what sound guys are for," no. That's what your EQ is for. No one but you is going to hear your amp at the position you hear it. If you are miked up, they are hearing it through the mic, and yes, a sound guy could EQ it, but you've also got to think about the people up front who are going to hear the unfiltered sound of the cab blasting at their ears. If you are not being miked, then there is no sound guy to apply EQ to your awful tone. So, do you EQ to work for just you or do you EQ to work for everyone else in the venue? Thinking it is up to the sound guy to fix your crappy tone is a good way to irritate every sound guy you work with. Which, in turn, will make you sound awful because they aren't going to want to help you out.

 

So in other words your argument is EQ everything for the 0.6% of the audience who happen to be standing directly in front of your cabinet? Do what you like, but all that accomplishes, especially when you're NOT being mic-ed, is that you'll sound wonderful for those select few, and your tone will be a $hit-awful, muddy mess for everyone else in the room.

 

And for what it's worth, I never suggested just throwing any old tone at the sound man, and expecting him to create miracles from a pile of crap...

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My personal opinion with the Helix is that if you depend on the sound man to EQ your sound, you haven't done a very good job of dialing in your patch.  That wouldn't necessarily be true in the older days of actual cabinets and mics.  But in these days with full control over every single element that can affect your tone right there in your hands you should be able to provide a sound that needs no channel EQ adjustments.  If there's a problem with the acoustics of the room that needs adjustment, that affects ALL of the channels equally, not just the guitar on one single channel.  So that needs to be addressed through the overall final graphic or parametric EQ on the board...not channel by channel.

 

For the last few years with both the HD500X and the Helix I've told all of the sound men to leave my channel flat, and it's worked out beautifully.

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I set my PA channel flat. No cut or boost.

And when I used mic'ed up amps it was the same way throughout my career.

 

People thinking that a soundman is making huge eq cuts are just wrong.

 

There is a reason that some guitarists have a desirable tone and many don't.

 

And it's not the soundman crafting a guitar tone. Lol

 

You run a room analyzer for your PA and get it as flat as possible in every room.

 

Then you stick a sm57 right up against the speaker in a 4x12 cab about halfway between center and edge and that's all you need to do.

 

Some here can claim otherwise...but many of us have had years of touring and know better.

 

With the Helix...we shouldn't have to make these cuts. But we do. And once you do, and get it right...it's awesome.

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So in other words your argument is EQ everything for the 0.6% of the audience who happen to be standing directly in front of your cabinet? Do what you like, but all that accomplishes, especially when you're NOT being mic-ed, is that you'll sound wonderful for those select few, and your tone will be a $hit-awful, muddy mess for everyone else in the room.

 

And for what it's worth, I never suggested just throwing any old tone at the sound man, and expecting him to create miracles from a pile of crap...

 

No, my argument is to EQ for the audience, who are ALL standing in front of your cabinet. You (the performer) are the only one who is not. By the time the sound gets to the audience, there isn't much difference between off axis and not. They do not need to be standing directly in front of the cab to hear the same tone.

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just to fill out some of the commentary

 

When you push a low power speaker hard it will flex along the cone as the spider and coil (the narrow bit) tries to move faster than the outer edge of the cone can keep up with. These ripples produce frequencies that are much higher than the input signal.  Its actually why something like a 30watt celestion sounds better on some guitar sounds than say, a JBL 12".
Anyway, this, plus odd vibrations cuased by grill cloth flapping etc, will make a bunch of unpleasant fizzy sounds that dont travel much betond the cabinet,(high freqs absorb faster) .  A few of the lower ones reach our ears and make the speaker sound "nice". 
A microphone put up against the cab will capture some of those sounds...

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It is very common in the studio environment, matter of fact it is common in the studio environment to rolling everything to mono ("side level eq roll off") at around 250hz, and below. This free's up a lot of room/data in the low end that you will never hear, but gives more room for the mix to breathe. Most audio below 250hz starts to lose its directional bias, even more the further low you go. (this is why home theaters can have the sub in the corner of the room, and still have it fill the whole room) Making the stereo information in that spectrum flubby, unuseful, eating up headroom, while providing very little in the way of benefit. So it is best to just roll it off to mono.

 

That's an interesting concept I will have to try.  I'll put a crossover split after the IR in path 1 set to 250k and send to 2A/2B.  I'll keep the higher stuff on 2A stereo and the low stuff on 2B mono and see how it comes out...

 

EDIT - made this patch to explore these techniques and a few others.  Check it out if you like...

http://line6.com/customtone/tone/3280575/

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That's an interesting concept I will have to try.  I'll put a crossover split after the IR in path 1 set to 250k and send to 2A/2B.  I'll keep the higher stuff on 2A stereo and the low stuff on 2B mono and see how it comes out...

Should work decently enough. Since L6 is adding in some kind of stereo separation block, putting that on the higher frequency chain can help as well. Though I will state I usually don't put a high-shelf for the side signal (boosting of stereo signal only, in the high frequency) until around 730hz - to - 765hz. (depending on the song).

 

Stereo separation is a kind of different route, but can achieve the similar results, albeit to a much more limited degree. So trial and error on that. May have to adjust the 250hz cross-over if you wanted to try that, perhaps a little higher than 250hz. 

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So in other words your argument is EQ everything for the 0.6% of the audience who happen to be standing directly in front of your cabinet? Do what you like, but all that accomplishes, especially when you're NOT being mic-ed, is that you'll sound wonderful for those select few, and your tone will be a $hit-awful, muddy mess for everyone else in the room.

 

And for what it's worth, I never suggested just throwing any old tone at the sound man, and expecting him to create miracles from a pile of crap...

 

I would reread that post. I got exactly the opposite of what you gleaned from it.

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Because what your ear hears six feet away from a cabinet is VERY different from what a mic hears at 1" away from the driver. This can't be stressed enough: the line outs of any full modeling signal path (that is, preamp > power amp > cabinet > mic) represents a recorded tone, NOT an amp-in-the-room tone.

You can, of course, utilize only a portion of the modeling signal path and use it with a real cabinet to scratch that amp-in-the-room tone. Modeling is never an all or nothing proposition.

When recording and mixing with other instruments, the engineer will often high cut and low cut the guitar tracks.

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I would reread that post. I got exactly the opposite of what you gleaned from it.

"but you've also got to think about the people up front who are going to hear the unfiltered sound of the cab blasting at their ears"

 

I really have no desire to revive this discussion, but just for the sake of closure, the quote above was really the part I was responding to...I can re-read that all day and it won't make any more sense to me than it did the first time.

 

It goes without saying that you'd want things to sound good over as large an area as possible, and to advocate otherwise would be inviting the men with the white coats and butterfly nets to take me away. This is especially challenging if you're not mic-ed, as a guitar cab won't disperse the way a PA does... and in such a scenario, why I should be overly concerned about a few square feet of real estate right up front, is beyond me. If that makes me nuts, then so be it...

 

But as I said somewhere above, it's all academic these days anyway. I haven't had a cabinet on stage in years at this point.

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So in other words your argument is EQ everything for the 0.6% of the audience who happen to be standing directly in front of your cabinet? Do what you like, but all that accomplishes, especially when you're NOT being mic-ed, is that you'll sound wonderful for those select few, and your tone will be a $hit-awful, muddy mess for everyone else in the room.

 

And for what it's worth, I never suggested just throwing any old tone at the sound man, and expecting him to create miracles from a pile of crap...

If you're EQ'ing your tone (using a real cabinet) to sound good where your head is standing (assuming you ~10' in front of the cab and a few feet above the speakers direction) your EQ will most likely peel paint directly in front of the stage and be boomy and muddy at the back of the room, so you're not considering any of the audience, you're only considering yourself. If you're playing places where you're not mic'ed, your best bet is to fire your cab at the back of the stage and EQ it to sound good out front, so you're not cutting any heads or relying on directionality for your tone.

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"but you've also got to think about the people up front who are going to hear the unfiltered sound of the cab blasting at their ears"

 

I really have no desire to revive this discussion, but just for the sake of closure, the quote above was really the part I was responding to...I can re-read that all day and it won't make any more sense to me than it did the first time.

 

It goes without saying that you'd want things to sound good over as large an area as possible, and to advocate otherwise would be inviting the men with the white coats and butterfly nets to take me away. This is especially challenging if you're not mic-ed, as a guitar cab won't disperse the way a PA does... and in such a scenario, why I should be overly concerned about a few square feet of real estate right up front, is beyond me. If that makes me nuts, then so be it...

 

But as I said somewhere above, it's all academic these days anyway. I haven't had a cabinet on stage in years at this point.

 

For the sake of my closure, context is everything and the context of that sentence with the rest of the post is different than just taking that one sentence out of context and deriving a conclusion from it. Are you a politician or in the media? ;) :D

 

He was refering to the large group of people near the front of the stage where the amps own EQ adjustments would make a difference since they would get more sound from the amp than the PA. Not just a couple of people directly in front. And he was also referring to the situation where the amp isn't miced at all. Like in a relatively small club.

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So if I've understood everything correctly there are 2 types of sounds we're talking about:

1) the sound of the amp/cab somewhere out in the room and

2) the sound of that same amp/cab but 1 inch away. 

 

In situation 1 the unpleasant sounds don't have enough power to reach everyone and so aren't heard. 

In situation 2 we then have 2 applications:

  A) live situation  (mic and then PA)

  B ) Helix (impulse response - going either to a recording or to the PA)

  In both of these situations we have low and high cuts that have to be applied as they are now part of the signal.

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He was refering to the large group of people near the front of the stage where the amps own EQ adjustments would make a difference since they would get more sound from the amp than the PA. Not just a couple of people directly in front. And he was also referring to the situation where the amp isn't miced at all. Like in a relatively small club.

Well I humbly apologize for my lack of context then... but he can't have it both ways. Without being mic-ed, you're either gonna (pardon the pun) fret over the ears of the folks in the front row, or EQ for the largest number of ears in the room, most of which are elsewhere. Can't do both...that's what I was driving at, and its precisely what I had addressed both times: A cab, no mic, and the bleeding ears of the people down in front. However many there may are, they're in the minority.

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Well I humbly apologize for my lack of context then... but he can't have it both ways. Without being mic-ed, you're either gonna (pardon the pun) fret over the ears of the folks in the front row, or EQ for the largest number of ears in the room, most of which are elsewhere. Can't do both...that's what I was driving at, and its precisely what I had addressed both times: A cab, no mic, and the bleeding ears of the people down in front. However many there may are, they're in the minority.

 

I thought he was refering to EQing for the most people if he wasn't miced adn not just the front row. Oh well. I guess it doesn't matter. In regard to the both of you and my interpretation of you guys, I agree with both of your points. I think we're just interpreting the post differently. Not a big deal.

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