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Posts posted by gckelloch

  1. Tried Thump and Decay instead of Low Freq Cutoff and couldn't tell where Thump and Decay were doing anything and it was way too bassy.  Maybe it's more pronounced with differant amps, mics and cabs.  I'm using the Angel Pre with the 4x12 XXL cab and 4038 ribbon mic.  Oh well, I'm real happy with what I have with the LF Cutoff at 140hz and Resonance at 40%.  Those two things really helped me dial in the tone I was looking for.  Always happy to hear and try suggestions though.

    It makes sense that there's mundo bass with that cab and mic. I think the 4038 has much more bass than say the 57. I notice you used the Pre rather than the amp model. The amp models offer a more spongy response and richer distortion. The amp DEP controls also become active for further shaping. You might prefer it with the amp DEP Master at ~8/10ths full power.  Amp distortion usually becomes evident at ~7/10ths full power. The tone often starts to fatten above that and then the high end harmonics really start to come out. You can tune the distortion character between the amp DEP Master and gain controls. Once you really grasp the other amp DEP controls, you might find them useful.  Bias X still escapes me.

  2. Very groovy indeed, sir.  :D  Once you really grasp how the parameters affect the tone and response, it begins to come together.  The Decay is probably the last thing to be concerned with.  It smooths over the sound, so there's less detail.

  3. I('d have to say the OP was correct.. he was obviously frustrated with eh HD.. As many of us are because it isn't straightforward to get the tone you want... way too much tweaking of controls that have no set meaning or description is required. It could be done so much simpler.. we don't need to have all the cab edit parameters.. a simple visual, mic you move around in front of a speaker to dial in the sound would be much more helpful, it VISUALLY tells you what you are doing, instead of trying to figure out confusing knobs that you'd never have in real life. Anyway.. instead of being snarky and bashing someone for being frustrated with an overly complicated device.. next time try being a little understanding instead of the lollipop int eh room.

    OK, but the device operates the way it operates.  No one forced him to buy it. Adapt to the way it works.  Trust me, moving mics around seriously complicates things and doesn't provide the same results as the Speaker Cab DEP parameters.  Set up the Amp DEP and fine tune with the Speaker DEP.  Not to worry, just leave the DEP's at default if you are confused.  They are ideally implemented AFAIC.  Deal with them after you grasp the way they work.

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  4. Two of the DEP controls I use the most is Low Freq Cutoff and Resonance.  The first one if obvious and I usually set it to around 140hz.  Resonance seems to be more of a High Frequency adjustment.  It combined with Presence seems to really help me dial in a tone.

    The DEP features you mention are for the Speaker Cab.  The Thump value adjusts the bass resonance, while the Resonance value adjusts the entire frequency range. 140Hz cutoff seems pretty high for guitar.  I'd think something down around ~80-100Hz might be more standard.  Maybe you have the Thump set too high?  Decay adjusts the speaker damping from tight to loose.  Loose speaker damping may result in some high end attenuation.

  5. I completely rewrote this entry since I first posted it. I think it's worth a re-read.


    An important factor to consider is that a PUP's Rdc value (often incorrectly labeled "DC R" or some variation) has nothing to do with its Z (impedance). A PUP can have a Z of 10x or more its Rdc value. A mixer I had once considered has 10M Hi-Z inputs. It makes much less difference on the tone above ~1M, because the pots factor into the total load, and they have usually have much lower values. As I mentioned b4, to get the flattest and most extended frequency response, it's best to use a low C (Capacitance) cable. ~200pF is a good value combined with the recommended pot values for the H value of the PUP's on a given guitar. It's best to use PUP H values within a ~1:2 ratio on the same guitar, so the pot values will work well for all PUP's. If you have appropriate pot values for your PUP's, you won't have sharp resonance peaks.

    It's not actually any more consistent to rely on the simulated "Miller Capacitance" affect of the Eleven Rack units, because different PUP's, pots and cabling will produce different results when combined with the Miller C...and that Miller C is different depending on the tube circuitry design. Some designs may have more C than others. Furthermore, the Eleven Rack examples of the PUP with the 1M load do not appear to include the pot values, which would put the load at ~110k for a guitar with 250k pots -- reducing the peak to the same level as the 1M+Cap example. That strikes me as intentionally misleading.


    For most guitars with ~2H-4H PUP's you want a pretty flat response on the bridge PUP out to ~4kHz. Presence peaks tend to sound too bright at the bridge unless up at ~6kHz+, which is sweet and sparkly through a Fender Twin, but can be fizzy with a lot of gain. High gain preamps often roll off the highs anyway, so it could be just fine. A ~3dB peak at ~4.3kHz or ~6kHz for the middle or neck PUP works well. A few other peak values will work, but picking articulation suffers as ~4kHz is lost, and peaks around the 3kHz range accentuate the dreaded "ice pick" tone. With ~8H+ PUP's, 500k pots and a ~200pF cable, the added Miller C won't have as much affect. The roll off would be down near 2kHz and wouldn't change much unless a lot more C is added. It's just the way that stuff works.


    Incidentally, rolling the tone knob down a bit smooths off the peak just like the 1M+cap. It just doesn't lower the peak location as much. Funny that the Eleven Rack example has the 1M+cap peak located at the ~3kHz "ice pick" zone. Did he even listen to the PUP example he gave? If you have a fairly flat extended response on your guitar PUP's, you can dial in a peak with a LP filter. Peaks in the ~1.5-2.5kHz range on the bridge can be useful for a sweet vocal lead to a defined mean rock/metal tone.

    I learned a lot from the late Bill Lawrence about all this stuff. His company, Wilde USA, produces the most consistent, efficient and refined PUP's on the market. Quite affordable as well. The appropriate series H value is available for each PUP model. It's definitely worth trying particularly the MicroCoils -- if nothing else than to hear what your guitar really sounds like, and maybe as references to judge other PUP's by. Nuff said on that. I won't push other product brands here.

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  6. Thanks Dino. :D  I'm not an electrical engineer, but from what I know, tubes squelch the highs when overdriven, which of course is programed into the modeling.  Wouldn't the amp model have said any tube traits designed in, attenuating any highs as the preamp would?  Perhaps the A/D front end is somehow configured to act like tubes when they aren't being driven? Couldn't such a thing be incorporated into an op amp design?  I'm not sure what is used as the analog guitar preamp circuitry.  Also, I think excess impedance loading limits dynamic response. It all sounds like something to look into.

  7. +1 on recording loops for tone editing.


    Thanks for the welcome, RIblues.  The Amp DEP parameters might actually do something more, or a little different than what I wrote, but it gives you the basic idea.  Bias X seems particularly hard to really discern, especially if it's adjusted with a whole lot of gain in the sound.  Although, you do need some amp gain to tell. I think the Bias X adjusts how abruptly to smoothly the sound breaks up -- which is normally an inherent trait of the tube type. So, it's like changing the tube types in a way.


    The Sag would normally be more a function of the amp rectifier and/or transformers and PSU. A very "saggy" PSU would actually damp the whole signal during the attack -- as was the case with the underpowered PSU on the early Marshall S100 amps. Not sure how the Sag is implemented here. It might actually damp the whole signal at extreme settings, or it might just serve to increase the transformer type S-curve, or something similar. Experiment.


    I forgot to mention that it's probably a good idea to use some type of limiter, preamp and/or tape saturation after the amps.  Not only will it guard against possibly clipping the output D/A, but it will sound more like a real recorded guitar sound.  The Vintage Pre might have some nice gain limiting traits. Realistic tape saturation is all the rage in modeling now...and for good reason.  You might be able to use the Tube Echo, Tape Echo, or Echo Platter as a tape saturation effect with the Delay Time and Feedback set to 0. I don't know if the Drive parameter creates smooth fat tape type saturation, or if it just drives the input circuitry. Guess it's all in what sounds good to you.  You might want to disengage the post limiting effect when you edit the amp tone.  I do that with modelers.


    This may seem obvious, but remember the Fletcher Munson loudness curve.  It's a good idea to figure out how to monitor at ~85dB RMS, so the curve flattens out a bit.  You can monitor at that level for ~8 hours.  It's harder on the ears through headphones, so be careful.  Stop when you start to feel ear irritation.  Find and post up a dB listening level/time graph and follow the recommendations.


  8. Sheesh...where do I begin? ;)


    Changing the input-Z will not make the tone from each output sound different. The "auto" setting will configure the Z for certain pedals only if they are first in the chain...and to 1M for all other models. Technically, lower than ~1M IN-Z actually compromises the guitar's dynamic range and attenuates the highs, but the pedals will sound more like the hardware versions with their "authentic' input Z setting.  So, there's no harm in leaving the Z to Auto.


    It doesn't sound like there is anything wrong with the "Fizz" version. It doesn't seem to be clipping. It just sounds like it has more upper harmonic content. That would indicate that the 1/4" output switch on HD500 face is not set to "line" like it should be when you are recording a line level signal. You should also make sure you are connected to the line ins on your Presonus A/I device from the main outs on the HD500, and not the Hi-Z guitar inputs. When reamping, you should connect from a line level out on the A/I unit to an FX return (set to line) on the HD500, and not to the HD500 Guitar or Aux Ins.


    Also, the amp is the least spongy with the Sag at 0. That seems a bit extreme. Somewhere around the middle would be more amp-like. Same deal with the Bias X, but regarding amp distortion: amp distortion is the hardest sounding and least dynamic when at 0. It makes sense to boost the mids on the amp with the 4x12" Hiway cab. You may want to back it down after setting the Master DEP up higher. The Dyn 57 mic has a lot of presence and not much bass, so you may want to back the treble and/or presence off a bit. You might try one of the ribbons mics, the U67 or Dyn 409 instead. Just something to be aware of. The 57 may be just fine. They quite often are.


    Regarding FX, you have the Output at 100 and the Drive at 0 on the Tube Screamer. I don't see a point in using the TS at all with those settings. TS's somewhat kill the dynamics and frequency range of the sound. They have their purposes, but I don't think it will help with the huge metal tone you seem to be going for. Moving along, the Gain on the last eq is at 0. I'd think that would drop the signal output a lot? Once you get your signal levels and amp set up right, you probably won't need the extra eqs.


    Not that it will hurt to leave it, but I'm not sure why to aim for a -12dBFS internal peak signal level? Pro digital recording level is actually -18dBFS. You could aim for -6dBFS peaks all the way through to the recording without any problems. I always record around there. Recording to 32bit Float has no real advantage, but it doesn't hurt. The 24bit A/D converter would clip way b4 the file does, so you'd still essentially have a clipped file if it did. 32bit Float is useful for post processing.


    For ~$380/pair shipped, I think the Equator D5 are the best "bang for buck" home studio monitors. Mix, SOS and EM mags all gave them stellar reviews. They are a ported coaxial design with internal DSP compensation. Bass extension, midrange clarity, and phase accuracy is outstanding. You'd really have to spend over a grand for anything more accurate. They are sold direct from Equator with a 60 day refund policy. I keep the input trim on my monitors down to -15dB, so I absolutely never distort the inputs. You either need to do that, or keep your A/I unit master output down ~halfway to be sure you don't get distortion. Break them in as with any monitors for a day facing each other with 180 degree channel inverted white noise up loud, but not distorting. Use them a lot in those 60 days to make sure they are OK.  Power switching amps are less tolerant in some ways of spec anomalies than analog amps.  If they were constructed OK, they should last many years if plugged into a surge protector.  Don't buy balanced cables unless you need some more quality cables -- you won't notice the noise difference with short cable runs. Run signal cables perpendicular to power cables for the lowest noise. Make sure you plug your gear into a surge protector, or even a SP/PC/UPS combo unit. I got a Belken combo unit for ~$70 several years back -- good investment. Use the same outlet for all your gear to avoid ground loop hums.


    Monitors should be elevated ~1' above the desk up to ear level when seated. I mounted ~1'x1' shelves of criss-crossed glued 5/8" thick finished poplar onto the back corners of my desk ~3' apart at the outer edges. I just use thick mouse pads under my monitors. I have a very complex shaped top floor carpeted bedroom area with a bed, thick folded linens and hanging clothes, and other absorbing junk, so there aren't obvious resonances. I hang a heavy blanket on the wall behind my head to kill high-end slap back. It all works fine. You could spend some money treating your room. It may not be necessary when close monitoring, unless you have significant echoes and resonances. A decent room is as least as important as decent monitors. You can check mixes with adequate studio headphones, but you should use a stereo monitor sim plugin if you plan to mix on them -- it has become a more common practice lately.


    Guess I got on a roll and had to cover everything under the sun. Hope it all helps.

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  9. I actually don't own a POD HD500, but I convinced my brother to buy one and have had the pleasure of experimenting with it a few times. I use PC amp modelers now and have used amp modelers since I bought a Boss GT-5 back in '96. If I wanted a hardware unit, the HD500/x would be my first choice. There is a steep learning curve, but as you gain clarity on the basic functioning, everything else will be much easier. If you don't get the basics down, you will remain confused and frustrated as you attempt to program new sounds. Others have already covered the GUI, so I won't go into it, but will concentrate on some basics, and how to get approach getting the sounds you want.


    An oft overlooked aspect of learning new digital gear is the internal level structure. It's really important to get a handle on how the volume levels function from input to output, and to leave ample headroom when driving external gear. You can probably leave the "Guitar In" pad switch off. Keep in mind it will affect the preset inputs if changed. The input level to the amp model is very important regarding attack character. You might want to run an eq b4 the amp as a "trim" adjustment b4 the amp. Seems to me that should be included at the beginning of the signal chain anyway. I find that the peaks of saturated sounds should be at least ~6dB below the full output level, so clean sounds will have enough loudness in comparison. I imagine it is quite possible to distort the HD500 output stage as well as any external gear it goes into. Check the internal HD500 software mixer levels, and I'd keep the Master hardware knob on ~12 o'clock so you don't overload external gear. It's important to note that setting both input paths to the same option and running mono FX will increase the input signal 6dB -- as will setting both software mixer knobs to center, potentially overloading the output. It's something to experiment with and set accordingly for each preset. It would be great if they implement M/S knobs in the software mixer so when you run two amps: one can be focused in the center of the mix, and the other at the sides.  If there is a stereo house mix, it's better for the off center audience members not to run paned stereo signals.


    One thing you need to really learn to listen for with amps is the response, or "feel", as well as the overall tone. The response has to do with how the tone is shaped over time. It's been the biggest complaint with modelers since the beginning. Good sounding tube guitar amps usually create some treble ducking when turned up loud and your picking attack is strong. Different tube types also have different gain envelope shapes from a smoother to harder breakup and varying degrees of generated even and odd harmonics largely depending on the operating class mode: A (even and odd) or AB (harder sounding odd only). 


    Beam tetrode power tubes like the KT66 used in the Marshall S100, JTM45 & Route 66 amps, the 6V6 in the Fender Deluxe, and the 6CA7 in Hiwatts generate much less 3rd order harmonics than pentode tubes like EL34 or EL84, but the 3rd harmonic in the guitar string will still be accentuated as any amp is pushed harder.  Still, they tend to break up sweeter, but require a negative-feedback "Presence" control from the transformer to damp intermodulation distortion.  Another thing that tube amps exhibit is a type of S-curve gain recovery created by the rectifier, output transformer and/or power supply. It's what gives it that certain sponginess that we call "feel".


    Someone mentioned trying the preamp versions of the amp models. Certainly, you can get some good tones with the preamps, but they won't have the sponginess and harmonic richness of the amp models with the amp DEP Master level set up high. They will however generate both even and odd harmonics, being that preamps are class A and don't cancel the generated even harmonics like class AB amps do. Nor will class A mode generate the hasher random harmonics that crossover distortion in class AB mode does. The guitar speaker should generally filter out some crossover harshness, so it's often not a problem -- too many factors to give a definitive answer. These factors can all be adjusted with the DEP settings. I think there should be some better descriptions of the DEP controls. Below is my interpretation of the factors they affect:

    Master: sponginess and harmonic richness

    SAG: sponginess

    Bias: boldness and generated even harmonics.

    Bias X: gain compression linearity = smoothness of breakup.


    You may want to adjust the Master DEP and/or the Channel Output after adjusting the Bias or Bias X, but the SAG is also part of the equation. Once you grasp what they all do, you can start to really make use of them.


    The Speaker Cab DEP descriptions seem adequate and are a really great addition. The Decay control is actually speaker damping. It can roll off the highs as well as affecting the tightness. You probably want to set up the Amp DEP first, then fine tune with the Speaker DEP. Strong cab resonances and loose speaker damping can mask the amp response/tone. What can I say, you'll get to know the cabs as you try them. Open backs generally have less bass and looser response than closed.


    It's difficult to describe the mics, but they all have frequency and subtle resonance characters. The SM57 is the "go to" guitar mic. It accentuates 4-6 kHz, so be aware of that. The ribbon mics are more neutral and nice to start with. Ribbons have fast transient response, so they can be a bit crisp on bright high gain tones. The U67 Tube condenser can smooth out those tones. The Dyn 409 is bass heavy, and might be best for really heavy metal tones. The Dyn 421 is another go to, but it's a bit sharp in a certain way in the high end for my tastes.  I usually avoid on axis mic positioning, but it can sound right for certain crisp tones. Maybe they should have included more mic positioning ability, but things are definitely more manageable without it. I've gotten lost in positioning mics with PC modelers. Hours pass wholly unaccounted for.


    Another thing to know is that guitar amp tone stacks are calibrated for a few reasons so that the bass and treble are boosted and the midrange is dipped with the knobs set flat. The presence knob normally smooths out amp resonances, but it may just be another eq point depending on the amp model. It's worth experimenting with turning the mids up, and turning the treble, presence and maybe bass down depending on the gain level and your pickups/pot/guitar cable combination. You can get some really sweet creamy tones with such settings -- particularly with a low C cable and good low H pickups.


    As far as where FX go in the signal chain, mainly be aware of whether the FX are stereo or mono. I usually run modulation pedals b4 the amp and sometimes b4 distortion pedals, but there are no set rules. A delay b4 an amp will have a less linear decay as the amp gain recovers. Running two or three modulation pedals in series can be really cool. Arrange the order to what sounds right to you. Reverbs are usually best after the amp, but keep in mind that spring reverbs are wired in between the preamp and amp -- if run b4 the amp, it might sound better if there isn't much preamp gain. Experiment.


    Well-calibrated guitar tone is a matter of understanding the contribution of everything in the signal chain, including the pick shape/material and strings. After 20+ years experimenting with various materials, I've finally gone back to Acetal for my standard flat pick. Steel guitar strings can sound harsh or thin. I prefer nickel-based alloys. Thicker strings can sometimes actually sound harsher, because the transverse waves in the attack can fall in the harsh ~3kHz region. It depends on the scale length and string material. If the strummed attack is too hard sounding, you might want to change string gage or material.

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