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gckelloch

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Everything posted by gckelloch

  1. I guess whatever works works -- keep in mind that if you roll off highs with one effect and then boost them back up with another, you are boosting noise and loosing some signal definition. Such is the premise of using a low capacitance cable with passive PUP's to pass highs to the amp in the first place...but yes, rolling off highs before the amp produces less preamp distortion hash. Amp section distortion character can be controlled with the tone stack knobs -- if the DEP Master knob is up high enough. Boosting the midrange and cutting treble and presence can produce a nice creamy sustain. Not sure that's what you want for this type of sound, but it's something to consider.
  2. We might be hijacking the thread, but there are good suggestions here. If you run stereo FX after the amp, you would need 2 XLR's to FOH to get the stereo signal. You only get the summed stereo signal when you use 1 1/4" main out jack. I did live sound for a while back in the 80's. We never ran stereo mixes back then, because you can't get a good mix for various audience positions. It's hard enough get a decent eq for various positions without having to consider stereo as well. FI, if you run a stereo delay, the dry signal won't be as loud for people on one side, so the guitar part could sound off time. It seems like M/S mixes could be made use of live, but I wouldn't worry about all that. If you run 2 XLR's to FOH, the HE can sum them to mono or whatever is normally done at the venue.
  3. If a reset doesn't fix it, it sounds like maybe a shorting switch. There's a video on how to maintain the switches: http://www.youtube.com/watch?v=S-JUStkzZ0I#t=2 Might as well clean them all, and stretch the springs a little, or replace them if you can. Not sure I would oil the switches as he does at the end. I'd be concerned the oil might seep down and gum up the contacts. The pot/switch cleaner spray might be safer for that. It might be better just to skip that step. Check the contacts on the boards around the switches before putting it back together to makes sure nothing is shorting there. I used Q-tips and alcohol to clean off contacts. You could try that on the cable contacts on the footswitch circuit boards as well. Touch an outlet ground right before taking off the boards so a static discharge doesn't burn out a chip or something.
  4. Hmm...that's a good idea. Glad I thought of it! :lol: Thanks boyce. There are SPL meter apps for cell phones. Unity gain can be checked through an audio system, but internal clipping can not. Monitoring should be reasonably loud for better meter accuracy. I don't think you need to turn off each previous effect when checking each successive one. Once you've checked the previous effect, all that matters is whats after it. I finally read through your original post perapera. It all makes sense now. Mono FX sum the 2 same signals to +6dB, and drop the output back -6dB, so they won't initially sum +6dB if one output is set differently. To achieve unity gain with unique inputs on either signal path, I guess the first mono FX block output on either pre path should actually be boosted +6dB. I just checked the peaks on some of my guitar recordings. It's not uncommon for the top ~6dB of the transients to fall within a few ms. If that were clipped, it may be too brief to be perceived at all if masked under various gain inducing FX. However, a much longer portion of the signal following the few ms clip is altered as an A/D or D/A circuit recovers. For that matter, internal digital signal clipping of a few ms before an amp may not be an issue, because there is no real circuitry involved to recover from it. Still, a few ms clip after the amps could definately cause ear fatigue and irritation in the listener.
  5. I agree that is probably a matter of tweaking. Chorus is probably one of the easier things to model accurately. The delay and modulation aspect of the sound is elementary. Modern simulations of "bucket brigade" filtering should be nearly identical to the actual pedals, but it is the more complex aspect to simulate.
  6. Yeah, I don't see how there could be a phase issue, but there could definitely be an internal clipping issue. I wouldn't be surprised if max input b4 A/D clipping is 6dB lower than the internal clipping of FX, I wouldn't use "same" input settings at all. I wonder if you can drive the amp model inputs more than the FX without clipping? If this hasn't been covered already, I'd also keep the amp block Channel Outputs down, and avoid boosting the Mixer block levels or centering the pans if running a mono summing FX post amps to ensure against post FX or D/A clipping. Line 6 should really clear all that up in the documentation.
  7. It sounds like a good plan, radatats. A couple of things in your second paragraph are a little unclear. You should probably over-clarify some of the things in the last half just for good measure. It doesn't hurt to overstate in some ways. Also, when connecting the HD500X 1/4" outs to a guitar amp FX return, which is essentially a lo-Z line level in, I think you want to set the 1/4" out switch next to the exp pedal to line, not amp. The amp option is for sending to the Hi-Z guitar input on a guitar amp.
  8. Yeah, he might be clipping the Pod input with the MXR+. You can run a distortion pedal in a Pod FX loop as long as it's placed b4 the amp model, but I guess the HD300 doesn't have an FX loop, so forget that. If you must use the MXR+, I think you need to turn it's output down fairly low. My experience with running an old delay pedal to a guitar amp (way back when I used guitar amps) was that the pedal output should be below halfway, maybe ~1/3 up. You want it to have about the same output level as your guitar going straight in. If the pedal is "true bypass", you can compare the level when it's clicked off. If it's not, you need to compare it with the level of the guitar plugged straight into the Pod. Clipping the pod input would sound atrocious. I imagine it's an op amp preamp. They break up very harshly -- like digital clipping. You really need to leave ~6dBfs headroom throughout the entire signal path to ensure against clipping when you play loudly high up on the neck. You should use accurate computer software meters through USB to achieve unity gain after each successive effect. Aim for average peaks between -12 & -6dBfs and you should be OK. You might sometimes get -2dBfs peaks, but it probably won't ever clip. You might be able to use an effect to drive an amp block much higher without clipping it. That question remains unanswered. Then it's just a mater of where you set the Master Output knob going to your Amps. I think you can leave it up all the way if you use the correct output switch and internal output type settings.
  9. Yeah, that is explained in the manuals. I think you'd want a line level out if you are sending the main outs to the FX returns on your guitar amps. You really should check out this online guide for some crucial setup tips: http://foobazaar.com/podhd/toneGuide/ I'd skip the FX loop and the MXR+. I don't think you really gain anything but more confusion with those factors in the mix. Just use the amps as your stage monitors. Maybe just run the MXR+ in the HD300 fx loop if it has one. There must be something similar to the MXR+ in the HD300. Maybe a compressor into a distortion, or two distortions in series. Something involving the Screamer, Overdrive and/or Classic Distortion might be what you are after.
  10. Can the Smart Harmonizer deal with chords? If not and he wants to play chords, I think he wants the Pitch Glide set to an octave or maybe a 5th.
  11. Hehe, where'd you get that idea MIKEY! ;) Yeah, that might not be quite what he is talking about, but it's a good place to start. If you just use an octave pitch, you don't need to worry about the key you're in. If you run a smart harmonizer after distortion, it might get confused as to where the pitch is and it might spaz out whenever you play chords. The "Pitch Glide" pedal might be what you are after. Just set it to an octave. If you set it up an octave, you might want o use an eq to roll off the highs above ~4.5kHz so it doesn't sound so much like a sped up recording of a guitar, like those old Les Paul recordings did.
  12. I suppose it's possible that the actual HD models are higher resolution than the ones in the units, but I doubt that. They probably created models in accordance with the affordable DSP power of the day, which is probably ~10x or more than what it was when the first POD series came out. Back in ~2000 I bought a Creative Labs SB Live PC soundcard for (~$100?). It had a SOTA 1GHz EMU brand DSP on board. I don't think this type of modeling is quite as power hungry as Liquid Audio or Nebula tech, but it might be. Line 6 could pack 8 1GHz DSP chips into a Pod HD pedal. They probably aren't very expensive in bulk. Even 4 1GHz chips is a massive amount of DSP -- more than enough to run some Nebula level models. Makes sense, Kronda. I'd find it hard to believe if they aren't modeling more amps as we debate -- unless not enough people are requesting them. I wouldn't be surprised if they model some of the new Bogner boutique amps. A Two Rock would be nice at some point, and maybe one of the other Metal amps those guys are always prattling on about. :P A Fender Super Reverb or the like would be pretty cool. Those are gaining popularity in the burgeoning surf-related genres. I don't see them increasing audio channels on the units, but I guess it's possible over USB. Dual cabs and/or mics per amp would be a real boon, but most people might usually just use dual amps anyway. Configuring 4 mics on a dual amp cab setup might result in a financial drain on the mental health system.
  13. 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.
  14. 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.
  15. The material in the core of the coil can also create eddy currents which can be utilized to dip harsher sounding frequencies in the critical ~2-4kHz hearing range. Soft and sweet sounding vintage PUP's can often have that feature designed in, but they can have high end losses as well. The wider magnetic aperture of the Lace Sensors softens the attack -- as is the case with most humbuckers...but the attack can sound too hard again if the PUP resonant peak is in the ~2.5 - 3.5kHz range, or from there up to ~5kHz when located at the bridge. With passive PUP's, the resonant peak is normally tuned via pot values and the cabling capacitance before the first preamp stage. Low C cable flattens the frequency response and extends the highs. There are ideal C ranges for certain PUP H values, but I find ~200pF works for several H value ranges. If you have too much highs for a given sound, you can use the HD500 IN-Z per preset to roll them off a bit.
  16. I surely have. I often prefer running modulation FX before distortion pedals in my modeler presets. It's more old school, I guess. I run some after distortion pedals. There are no set rules, but it is helpful to know how each effect really shapes the sound to know how to use it better. Audio engineers do have some standard practices they use in many scenarios, and the deeper understanding of the gear helps them create vibrant new sounds more quickly without unintentionally compromising something in that sound somewhere in the signal path.
  17. I can see how the 6V6 section the Divided amp could sound like a 5150. Class A driven 6V6 tubes generally have a sort of sweet dry papery distortion sound that I associate with a 5150, while EL84's sound more wet with a defined edge on top (probably from the 3rd harmonic). Knowing a bit about how the tubes and the operating class of the various amp sections contribute to the sound can be helpful: Whether the amp generates much 3rd harmonic distortion (pentodes do); class A or AB operation (cancels generated even harmonics); how tight or smooth the breakup is; how much amp distortion can be achieved with the Master DEP. Virtually all the amp characteristics can be adjusted with the DEP knobs -- essentially mimicking any amp. It then comes down to how the preamp contributes to the sound. If the preamp gain is set low enough so it doesn't distort, it won't contribute much to the sound. For that matter, it seems like mixing different Pres and Amps would open up some possibilities as well. Is it possible to run a Pre before an Amp? If so, to run pres before dual amps? As far as understanding tone stacks goes, you can look up tone stacks for common amps to see how they are configured and learn how to set them relatively flat. You should also learn about how your guitar circuitry creates filter shapes as well, and whether your PUP's can even deliver upper mids and highs if you are trying to make use of those frequencies in your tone. Many PUP's suffer from eddy current core losses that cancel out upper frequencies. The cable and pot values all interact to form a low pass filter. High inductance PUP's don't generally deliver upper mids much above ~3kHz. A decent level of ~2.5kHz is generally fine for most rock-based music, but you won't get any real detailed articulation. ~4.5kHz and ~2-4H PUP's are needed for that. ~6-8kHz can add a sweet sparkle that can be really nice for anything from country, jazz, indie pop, new wave, new age, Nic Cage, etc. Above that range goes into acoustic guitar tone territory. Very low inductance PUP's below 2H like a classic Tele neck (without a brass cover) or a Filtertron are needed for that range. You'd need a speaker that can deliver those highs as well.
  18. Remember though, a mic'd amp cab goes into a console channel or dedicated mic pre, and then historically to tape for some tape "glue" saturation. If you run it post amp, you should probably make use of the lpf knob to trim the high end harmonics. The hpf can also be used to clean up the mud when it's driven hard. The idea is to use it as a natural sounding limiter/enhancer, rather than use a compressor that processes the dynamic envelope in an artificial manner. Tape saturation would be a nice option, but it's often as DSP intensive as an amp model if it's realistic. Harsh clipping in the Vintage Pre could be from overloading the input. Keep the amp Channel Output down around 50%. It can go up higher for clean tones. Check out my recent posts for more on this matter.
  19. Makes sense Andy, but I wouldn't use the amp DEP Master for level adjustment. It affects the amp model response and distortion. Use the amp Channel Output for level. RIP: James Honeyman Scott. I'm a big early Pretenders fan.
  20. I'm not a metal player, but I can see how that would work. Still, I've always wondered why anyone would want to use an IC-based pedal in front of a finely tuned tube amp -- seems counter productive in a way. There must be a better way to soften/tighten the response without sacrificing sensitivity. I remember really liking the rack mount Chandler Tube Driver from the 80's. The contour knob (not present on the floor unit) achieved that kind of softening, while still retaining the dynamic sensitivity of the tubes -- all of which can be modeled. The TS thing sort of snowballed into a legend. Also, I'd want to make sure I wasn't clipping the amp model input. Not sure if that can occur, as can with FX pedals. It's something to consider regarding fizz.
  21. I'm pretty sure the amp models are not compromised when more FX are run. I don't think these units are programed that way. They run all models at full quality at all times, unless there are quality settings for each model...and I don't think there are. When the DSP load is exceeded, no more FX can be added. Some PC amp modelers have various quality settings that do compromise the models to accommodate CPU power.
  22. 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.
  23. That's a good point. The DEP parameters make it possible to simulate other amps and, to some extent, speaker cabs. If anything, they just need some cabs with JBL DxxxF speakers for full range (up to ~8kHz) sweet woody clean tones. The 4x12 Hiway cab Fane speakers have extended highs, but they are very different sounding than the JBL's. One thing they maybe should consider including in the speaker DEP is a resonance tuning feature to raise or lower the overall frequency of the resonances, as well as the resonance levels. It could be useful, but might create more confusion for plebs. The varying resonance traits of all the different cabs might make that feature obsolete, especially considering many cabs have the same or very similar speakers.
  24. I can see how it wouldn't be necessary until a next generation modeling archetype. If you know that for sure, do you know how much spare ROM is on board for new models? Deano, what do you mean by this? "...I'm almost convinced that the algorithms for all these amp models would sound so much better if they didn't include everything and the kitchen sink in these POD HDs".
  25. 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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