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Preventing Internal Clipping And Achieving Unity Gain

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To achieve unity gain without any internal clipping, you must first make sure the input is not clipping.  Use the Guitar Input pad if needed.  Once you are sure the input isn't clipping, you should check for internal clipping after placing each successive model in a signal path.  There doesn't appear to be a way to do that internally, so you need to use the meters on an external device, like a DAW though USB.  I'd guess peaks around -6dBfs for clean tones would be a good bet.  Peaks should average between -12 & -6dbFs.  The bit resolution at -6dBfs out of 24 bits exceeds the S/N ratio of the HD device's analog circuitry by ~30dBfs.  Generally set high gain tone levels so the peaks are ~3 - 6dBfs lower than your more dynamic clean tones so the loudness levels are close.  Ultimately there will be some readjusting as you experiment further, but at least you won't ever get digital clipping if the clean tones have ~3 - 6dBfs headroom.  Keep in mind that centering both output mixer pans will increase the output signal, particularly with a single amp path setup, possibly clipping any models placed after it or the internal D/A converter.  Continue to refer to the external device meters to retain unity gain with peaks no greater than about -6dB.  I can't instruct on how to operate or interface for said external device -- each case is different.  Refer to the "Quirks. A" section of this guide for further info on this subject: http://foobazaar.com/podhd/toneGuide/quickGuide

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It would be nice if Line6 added a clip indicator that would flag clipping anywhere in the path.  Clipping really ruins the modeling.

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Digital technology requires us to stop the old habit of just cranking everything up and letting it rip!  You really have to pay attention to the details and actually build a tone thoughtfully and creatively...  I love it!

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It would be nice if Line6 added a clip indicator that would flag clipping anywhere in the path.  Clipping really ruins the modeling.

More than nice to include, it seems a shameful oversight on their part.  What were they thinking?!  Let's see...the main thing that need be avoided in digital modeling is digital clipping, so lets not bother including any internal meters, so it can't be guarded against and guitar players will have all kinds of level and clipping problems, further besmirching digital modeling devices...and after 2 years in and several updates, we still won't include any meters.  That sounds like a good plan.  Lets go with that.

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It might be hard to tell with distortion presets, They just might sound off, and you wouldn't know what exactly is clipping.  Besides, it's good to have an idea of how much headroom you have for really hard strumming on your loudest PUP's, and to check unity gain throughout the signal path.  You could easily get some build up along the path and run out of headroom at the end of the path if you just use your ears.  Then you might have to readjust everything -- still guessing to some extent.  Ears are certainly our best judge of tone, but not of peak levels.  Implementing even just one meter b4 the D/A would be very useful. 

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You just need one meter at the end of the chain.  You can check the output of each effect individually.  There is no real common sense when it comes to peaks.  They can be unpredictable.  Our ears can't really determine peak levels less than ~100ms in duration.  Guitar transients are much shorter than that.  Even a 10ms digital clip can alter the attack character of the instrument.  You might not even know it's happening, but it can definitely affect audience perception of the performance.  Micro digital clipping can leave people feeling uneasy and cause ear fatigue.  Rupert Neve pointed out that just 3% of 3rd and only 0.1% of 7th harmonic distortion can significantly alter our perception of a performance.  I don't see why you wouldn't at least welcome the option of a meter?  It's kind of hard to tell, but I think you might be getting some micro clipping in your presets.  That's partially why I mentioned it.  I hear clipping in so many digital recordings and product demos all over the place.  It's a shame that more effort isn't made to ensure against it.  People are more concerned with maximum S/N, but it's not worth the trade off -- especially considering that a little random noise often has a tone smoothing, therefore soothing, affect.

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Yes, I hear digital clipping, but many people can't discern it, especially if buried in amp distortion.  They only know that something sounds offensive.  I'll reiterate that as a human being, I can't consciously discern micro-clipping, but my ears will register it and my perception and mood may well be affected by it.  The only way to get maximum S/N without any digital clipping is with properly calibrated peak meters. 

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With digital electronics you want the largest signal you can pass through it without clipping anywhere.  Smaller and you are giving away resolution since you are not using all of the available bits.  It is easy detect clipping internally so having a clipping indicator makes sense.  If you are running clean sounds it's easy to hear clipping but if you are running already distorted effects it's not as obvious.  After all most overdrive effects are modeling some form of clipping anyway.  But the models are not going to produce the desired results if the DSP is clipping anywhere in the chain.

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Yes.  Studies referenced by Rupert Neve have shown the levels of harmonic distortion created by digital clipping to increase ear fatigue and affect mood.  They were tests done with op amp crossover distortion which produce the same levels of distortion.  Some people are of course more sensitive than others, but the results were pretty consistent.  We are still instinctual creatures to some extent.  When we hear an "alarming" sound under certain circumstances, the ear closest to the sound still moves slightly toward the sound in a futile attempt to increase clarity.  We can not supersede the instinctual responses we have to certain types of sounds -- unless we evolve past them at some point.

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I also want to add that comparing modelers to real amps is a very elusive thing.  This may seem pedestrian to some.  First, you'd have to take into account that, at least historically, no two tube amps sound quite the same to begin with, unless maybe they are carefully wired with all matched parts.  Just moving wires around inside can cause tone changes from EMI or whatever else.  There are also slight variances in tubes and bigger components like transformers.  Still, very minor knob adjustments can make a substantial difference in tone and response.  They might be calibrated slightly differently.  It's really impossible to exactly match the adjustments on a modeler to those made on a real amp.   Many also don't consider the fact that with a modeler we are usually listening to an amp/cab setup mic'd in a studio from the perspective of an audio engineer in the control room.  There is no way that will sound the same as listening to an amp directly in the same room.  Inaccurate monitors will also color the sound.  At the very least, they should be full range and turned up to level of the specific amp tone they are being compared to so the loudness curve is the same.  There is also the physical factor of the bass notes thumping against your body.  If your monitors can't deliver that, it won't feel the same. 

 

I suppose that modeling increases in complexity starting from the preamp, then the amp, and  to the speaker cab.  You could say mic modeling is perhaps almost as complex as cab modeling because of the physical element involved, but the resonance contribution of a mic to the overall sound pales in comparison to the complexity of the speaker/cab resonances.  At this point I believe most experts have uncovered enough of what goes on in an amp to create models virtually equal in complexity to the real things.  I think speaker cab modeling is the final frontier due to the fact that there is so much physically occurring.  If anything, it's probably the lack of the same kind of speaker/cab response that breaks the deal for some people.  There are technologies that can recreate it accurately, but they require loads of processing power. 

 

The response of most modern speaker/cab simulations seems sufficient regarding resonances.  The dynamic gain compression factor missing from the Pod HD models could be made up for with certain post FX, like the Vintage Mic Pre.  Seems to me though that the missing speaker gain compression factor could be dialed in with the amp DEP knobs.  Only thing I know of that might be slightly different is the amp gain variances that occur as the speaker impedance changes at different frequencies.  Amp gain increases along with speaker resonances.  It's a fairly minor point, but those harmonic variances could be that missing element in the "realism" factor.  The Speaker Cabs are part of the amp block, so it's possible the harmonic variances have been factored into the models, but Line 6 hasn't mentioned they are interactive in any manner, so I doubt they are.  Personally, I'm not sure I'd actually want gain increases at the speaker resonance points.  It probably sounds sweeter without them.  Not every inherent function in a guitar amp is necessarily desirable.  FI, I see no point in modeling tube microphonics.

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I should also mention PUP height as another major factor. To most it sems an obvious point, but adjusting PUP height can be a tricky thing. I defer to late guitar PUP designer Bill Lawrence on the matter. His height adjustment method can be found in the "3. The distance between pickup and string" section on this page: http://billlawrence.com/Pages/Pickupology/Introduction.htm

 

Lotta good info on his site. Height should be checked with strings depressed at the top fret. When he mentioned adjusting at stage volume, he refereed not only to averaging out the loudness curve, but to the affect gain has on the sound of the PUP's. For him, stage volume might mean a Fender Twin with the input gain on ~3. As the gain is raised higher, your higher inductance bridge PUP generally won't sound as much louder than your neck PUP as it sounded with the input gain at 3. At very high gain levels, the neck PUP might even sound louder than the bridge. Not to worry though, they will still sound the same when combined.

 

What's really important is that you don't clip the Pod HD input, the string to string balance is good, and that they have the tone you are looking for. With the IN-Z of a Pod HD set at 1M, the tone will be slightly brighter than an average tube amp that has some Miller C. For that matter, you might want to set the IN-Z to 230k before setting the PUP heights. Then check how the high E sounds back at 1M. Of course you just use your ears to judge, but you'll definitely know it's too close when the attack is overwhelming.  You might want to use the 230k setting for your presets if it passes highs to  your liking.  Just make sure you pick at the same location on the string as you check.  I find that the high E attack sounds tighter and fatter when I pick ~1/2" in front of the middle PUP poles on my Strat type than at ~1" where I normally pick -- just something to be aware of.

 

After setting the heights, I usually fine tune them for the combined tones. It's easier to hear the subtle high end shift on headphones -- that's assuming your PUP's have any highs. If you have some quality full range headphones, you might want to just do the whole process on them. It's easy to accidentally set the PUP's too close and cause input clipping. I finally got a digital caliper to more accurately check. A Nickel is 1.92mm thick. My low E string heights average a little closer than recommended at ~2.7mm. I prefer the extra low string warmth and punch.

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Hardly believe I forgot to mention this and it may seem obvious to some: as a general rule, you can use some kind of limiting FX after your clean amp tones to trim ~6dB off the transients without it being obvious.  Not only will it guard against clipping, but you could gain about that much back on the Pod output S/N ratio as the overall loudness is raised up ~6dB compared with the noise.  You might not gain much loudness with high gain tones, but it might help keep resonances under control.  The point is to set the threshold so it doesn't compress the average loudness...just the peaks.  It really needs to completely limit the peaks and not let through the first few ms like compressors do.  I think the Vintage Mic Pre is the only one that can do something close to that because it's not a compressor   Such "brickwall" compression requires some latency to "read ahead" and catch the peaks before they are limited. 

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Maybe the few people I hear saying that can't get a good tone from PODHD are just clipping to some degree.  If there was an easy way to let people know that something is clipping then they would know what to fix before giving up and taking it back or selling it.  It tells us when we're out of DSP, it'd be nice if it said something is clipping. 

 

But I'm asking this question about a company that built an EQ that used Percent instead of HZ...  Maybe that just answered my question.

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POD Xt had a clipping message...

True.I had a XT before my HD, was my first pedal.Same with the JTV, same with the HD...1 step forward and 2 back.,...

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Not having a clipping meter hardly qualifies as 2 steps back, but it would be useful to incorporate one.  Say folks, can we please not start to waiver OT?  I'd like to keep this thread focused.  Thanks for all the input so far.

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Don't know if this is on topic but I always have to watch how I use the compressors in the pod. The cause a tremendous amount of clipping even at minimal levels. I like to use one if I have to incorporate a Gate in the chain so I can level the signal back out. But I have to be real cautious. Now I've heard of people using the compressors at what I consider crazy levels with no problems. Same with default settings but my experience is if I am clipping 90% of the time it's coming from a compressor in the chain. And I've never been able to use one at default settings.

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That's a good example.  If you set the threshold on a compressor very low, it will boost the signal a lot when you aren't playing.  It might be boosted over 10dB depending on the threshold setting.  Since the signal is boosted so much, and nothing but a read ahead "brickwall" limiter (which can't be used in realtime) can catch 100% of the initial transient, the first transient after you haven't played for a second or so will pass through and possibly clip something after it.  You should only use low threshold settings before an amp that breaks up somewhat smoothly.  You definitely can't rely on a compressor for protection from D/A clipping after the amp, so you should only use one after the amp to even out the signal a little via ducking the higher transients.  You may need to use a much higher threshold (or lower drive knob setting) on the compressor to avoid any post amp clipping.  You also may need to keep your amp outputs or any post amp outputs lower than expected, because the initial  transients will still get through no matter what.

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