Posts by meambobbo

Does anyone know what speakers/cabs were used on this song/album? Best I can find says 2 4x12's with Celestions. I would assume Marshall 1960 cabs, but T75's, Greenbacks, V30's? I know Vai likes V30's now, but I seem to remember him saying something about Greenbacks a while ago. The Legacy cabs were originally loaded with Greenbacks. But I also found an Ebay listing for a Marshall 1960A cab with EVM 12L's. If I remember correctly, Eddie van Halen used Marshall cabs loaded with 2 Greenbacks and 2 EVM's on the first few albums.

I also see he used a 421 and a SM 57 in terms of mics.

I'm going to go out on a limb and say either 25 W Greenbacks or Greenbacks + Vintage 30's is the right sound.

I also saw a post that said the two cabs were put in opposite corners of the room and also mixed with a DI through a Palmer's speaker simulator.

have you held the OUTPUT button then scrolled through those parameters? I don't have the power'ed version, but I remember seeing some of the outs had default send levels of -30 db or something like that. I would just try to work backwards turning up whatever output levels you can.

Looks like this hunt is still going. Indeed, I think the closest I've gotten is with Vai.God, but I do a couple things in the Stomp section that I find makes the tone more Vai-like. First is a compressor. This helps thicken the tone a little bit which is quite necessary to get his thick lead tone without resorting to over the top gain levels. I don't use too much here - you still want some playing dynamics. I also add some attack to it. So you get thicker tone and more sustain, but still get strong pick attacks.

Second, the DS-1 he uses I remember in an interview he says he likes to turn up the drive on it until he's got a slight break-up to it - adds more character to the distortion when run into the amp's distortion. So I have my Drive around 3 - 3.5 and the tone around -2 I think. Now, this DS-1 isn't the modded version he has, but it'll do... You might be able to throw an EQ in front of it to draw out more of the modded sound?

I like Erick's Legacy profiles. I use the same tricks as above and it turns out incredible. I use one of Lasse Lammert's Rectifier cab profiles, and mildly tweak the high/low shift to get a more mid-focused sound.

I remember an interview where Petrucci was talking about how he got his studio sound on A Dramatic Turn of Events, and he said they used both a SM 57 on axis and a $3,000 Shure Ribbon mic (forget the exact model). I think they were both close mic'ed, but he said they just played with positioning and the mix until the sound sounded the same as when he stood in front of his cabinet. So you know Petrucci's a tone-nut, and there he is saying he could get 2 mics to sound the same as standing in front his cabinet. He may have been speaking in a bit of a generalization, but it still seems impressive to me.

I like MESA MARK V LEAD+ by STEPH.L for a Mark IV/V tone.
I like Mesa Mk3 JP5 by Thuman for a Mark IIC+/Mark III tone.

I've played an awesome Trans-Atlantic rig but I forget the name.

I've found a lot of good Rectifier patches. Most of them were factory. Lasse Lammert's rig pack has a few. I want to say DUALRECTIFIED is a good one. Also there's a Road King one I like a lot (I want to say it's by Christoph Kemper).

Keep in mind you'll get more mileage from them if you test them out with different cabs. My advice is to download Lasse Lammert's rig pack and store all his cabinets as presets - his cabs are immaculate. Also TillS cabs are great. Some are factory presets, but you'll need to find one of his rigs with a Rectifier cab and save it as a preset to have it. I think he has several Rectifier cab profiles, but I found one I like and use that occasionally.

Hey as for saving presets, you can edit any effect block, the amp block, and the cab block by holding that button on the unit. Once you are editting a block, press STORE and you'll get an option to save the full rig (patch) or whatever block you're editing. For an effects block it will store all the associated settings - the type of effect plus all parameters. You can name it whatever you want. For example, if you have a Green Scream with Drive 0, Tone +5, Volume +5, you could name that "Screamer Bright Boost". Then on another patch/rig, if you want to add that preset, you hold an effects block to edit it, then just turn the Browse knob to select the "Screamer Bright Boost" preset.

For the amp block, the preset saves not only the settings, but the amp profile as well. Similarly, a cab preset saves both cab parameters as well as the cab profile. You probably won't use the cab profiles, but the amp profiles you might want to consider saving as presets - I often dive into the "deep" settings on the amp block, which can have a big impact on the way the amp sounds. I store the full rig with those settings, but maybe I want to punch the amp profile into a different rig where I already had my STOMP and EFFECTS sections all set up.

I think you can also save the STOMP and EFFECTS sections as presets too. So if you have a couple common signal chains, it can save you time to pull these up rather than enter the individual effects each time.

Just keep in mind that the KPA does not default the preset name based on any tags - it initially displays the default name or the last name you used. This is often meaningless, and will only serve to confuse you if you don't change it before saving the preset.

Also, whenever you store a full rig, it is automatically tagged as one of your favorites. So if you filter your rigs by favorites, you'll only see the ones you've saved. If you find a good rig you really like while browsing, my advice is to hit STORE > REPLACE. Now that rig is in your favorites.

I think something that's missing from this discussion is that while the trademark sound of a microphone does leave quite a bit of a footprint on the tone, often you can get much closer to the real cabinet's sound by blending 2 microphones. Most of my favorite IR's are dual mic'ed. I agree that I prefer the tone of the cab itself vs. a mic'ed cab, but with proper shooting of a dual mic IR, I honestly find it hard to tell the difference.

Also, most FRFR solutions are designed to have less directionality, whereas a 12" guitar speaker is quite directional. This will obviously sound different in a room where the sound is reflecting off the walls, floor, and ceiling. But I think once everything is EQ'ed nicely, neither one is better than the other.

I prefer the simulated cab/FRFR approach. Generally the gear is more portable and more consistent across a space than a real cab. There is no need for mic'ing. It all falls back to the cab/mic simulation you use. Similarly, the "real cab" sound depends on the quality of the real cab/speakers.

BTW, anyone who is looking for that original rig, it's actually called "MESA MARK V LEAD+" by STEPH.L. And I like Thumas's Mesa Mk3 JP rigs as well for a lead tone.

clips and rigs are up.

zip
folder

the mp3 file is obviously the clip. I play the following rigs in order:

• a - how I have it now and like it. Tone is great - nice and smooth and squishy when I want it, but also a searing distortion. Exactly what I expect of this amp's tone.
• g - this was an XTREME! test - I took my tone above and put 4 pre-EQ's - all with volume +5 db, along with setting the two post at +5 db, and cabinet at + 10.4 db. Then I rock back the volume knob almost all the way to nothing in order to get my output not clipping, and I use literally the lowest possible gain setting, which is still pretty hot, as you can hear. But on top of the tone, there keeps popping up a nasty sputtering type tone.
• h - so I reduced the rig above to just one pre-eq with +5 db volume, and I added some gain to compensate. surprisingly, the sputtering from above is now MORE rampant here... hmm
• i - I disable the post eqs, and compensate volume - signal is perfectly clean again.
• j - I replace the two post eqs I was using with default, neutral Studio EQ's, and again jack up their volume parameters to max. NO SPUTTERING.

Granted, all of these examples are ridiculously extreme, and I don't think was really my problem before. i think i was just running a tad too hot on output, but it looked safer than what was actually safe - i now treat orange more like red, and things are fine.

Edit: broken links fixed

i'm on fw 2.2.0 beta btw

Yeah, I actually tested this thoroughly yesterday with clips and patches which I'll post in a bit. I read on the wiKPA doc that Christoph said the internal gain staging isn't like other devices where you have to worry about clipping between blocks - just make sure the input and output aren't clipping, this way there's no need for indicators of where in the signal chain you were clipping. so I put that to the test.

To be clear up front, in "normal" usage, I couldn't get the EQ's to clip, so long as input/output were not clipping. So it is exactly as Christoph describes. Smart design - makes me love the unit even more. So what was happening in my case?

I think what was throwing me off is that looking at my interface's mixer (and I'm connected via SPDIF with no boost on the SPDIF send), I was not hitting red clipping levels, but getting close, and there was clipping on the KPA. If I backed off the volume knob, the clipping disappeared. So I think this is where I previously got in trouble - I think there were a bunch of things on that patch that was boosting the volume up too high, including the EQ's and the cabinet volume - for some reason these were not defaulting at 0 db. I was also turning up the mids on the tonestack.

I had to bring down clean sense pretty low to dial out any input clipping in entirety - the worst is when picking hard directly over the bridge humbucker. I never actually play like that, so I find I can get it around -8 without any issues, but I had previously had it on -4. I don't think it completely relates to the issue I was having, but as part of a thorough clipping test, I did not want false positives.

Also, I think one of my monitors is goofed up. I knew of one odd behavior it would exhibit, where the sound would cut out, but I've never heard it distort. Something VERY weird was happening on Friday, and I can't replicate it on headphones, so I don't think it's the KPA. I boosted the EQ's like I originally had them when I had issues. I would switch off the amp block - nice clean tone. I switch the amp block on and I get a weird distortion on top of the tone, like a ring modulator, or out of tune radio kind of sound. If I switched the amp block off, the distortion would persist on top of the clean tone for about 1 - 2 seconds then disappear. I had to replace the giant capacitors tied to the power supply on one of the monitors before. I don't remember which one - I have to take it apart to see what's wrong.

...

Now, I did do an "extreme settings" test and got some VERY weird behavior. It's post the clips and demo patches in a bit.

glad to hear you worked it all out. that cabinet level parameter is a tricky one sometimes.

right, distortion sense at center (0) would match the original rig. what it is used for is to get the same gain levels between different guitars with different output levels. So if you have an axe with EMG's and another with medium gain passives, you can change distortion sense to compensate, so that your patches have the same gain levels between guitars without tweaking every patch.

I have been thinking a while about creating a patch/rig site with the kicker being that when you upload a patch/rig, the server software would load the patch onto the applicable unit, then reamp some guitar signals of several different styles/guitar/pickups through it. so you could preview before downloading. Of course, I can't do this automatically until Kemper releases an editor that allows me to load a rig onto the unit from the computer (or program some robotics...no thanks).

the rating feature on the rig exchange is nice, but it's too disconnected. I download a few dozen rigs on my laptop, put them on a USB drive, go into the other room with the Kemper, play through them, tweak them, save multiple copies, etc. By the time I'm done, I don't remember which profiles I should rate. And it gets more mixed up, because maybe I think a rig sounds horrid, then I change the cab and add a boost and it sounds immaculate. So should I rate it 1 star or 5 stars?

by the way you can sort rigs by rating. the problem isn't that the feature isn't available. it's differing subjective tastes as well as a low sample size for ratings. I think the most I've ever seen was like 12 ratings. Most are 0-2 ratings. This isn't enough feedback to really determine a wide-ranging idea of how quality the rig is.

tyler, this is absolutely true and something I stressed heavily in my Pod HD guide. FWIW, on the particular patch I'm talking about I used 2 Studio EQ's before the amp, and 2 more in the X and Mod slots. I found last night I could eliminate the 2nd post-eq by making smarter choices with the BMTP controls and in the other post Studio EQ. But the pre-EQ I need them both. Toggling them on/off is a world of difference and dials in the perfect Mark IV tone.

I also find that the "definition" control is basically like a pre-eq. Many rigs have this value set quite high by default, but I generally don't like the sound I get from that (of course there are exceptions). I prefer to turn definition down until I find the sweet spot where the tone is right between vintage and modern. Then I like to use EQ to change my distortion tone - I have more control over it. This is another reason I like EQ's over boost/OD effects. Whereas an OD will already have a specific sound and only give you a single "tone" knob to tweak, EQ offers much larger possibilities. For example, you can trim bass to tighten up the sound, but can find the exact cutoff frequency so the tone is still fat. You can also dip a little midrange to get more grind out of the tone, which is usually never available in a boost/OD.

From my experience this is what i believed was happening and it shouldnt be hard to replicate - no doubt it would push the amp harder, but shoudnt that be compensated by gain? I am in no way dogging the kpa's eq's - including the tone stack they work very well. But i bet you can disable anythin but a pair of eqs and get distortion

Will, thanks for the video. Gonna take a little time and reading to fully wrap my head around this. Seems like the example around 3:00 is what I'm most curious about - why is the filter causing ringing, regardless of which mode it is using?

I get confused with DSP when pondering a synthesized signal vs. a recorded signal. Shifting phase in a synth signal is simple, you just change that parameter in the algorithm (usually represented by Theta if I remember correctly). But how can you do that on a recorded signal? Do you Fourier Transform it into a collection of infinite sinusoids which you can then manipulate the same as synthesized sine waves? Of course in a real signal, such sinusoid waves are not going to be continuous over time. Seems like any adjustment to them can cause artifacts.

I feel like I understand linear phase EQ - rather than attempt to perform the above phase adjustment, you create infinite copies of the signal, all delayed by slightly larger and larger amounts as needed until the last signal provides 180 phase inversion for the lowest frequency desired (20 HZ?). Then you amplify these signals according to how much filtering you want to occur and mix them against the original signal. For instance, if you wanted to filter out frequencies centered around the 1 HZ frequency, you would only mix the original signal with a copy of itself delayed by 1/2 a second. To widen the Q you boost the signals more and more as you get closer and closer to 1/2 a second delay. To narrow the Q, you boost the same way but with phase inverted copies of those delayed signals.

This method does not have the possible artifacts with the phase-shifting mentioned above, but because it has to compensate for these variable delays, it can cause transient-smearing as the video noted.

Which method does the Kemper take? That's a good question, but considering that we know the Kemper can center a parametric EQ at a frequency of 20 HZ, which would require a .0025 s delay. Considering you can use at least 6 EQ's in the Kemper that's a latency of .015 s. The Kemper can run with a fixed latency of 5 ms (4.95 ms maybe?), I believe. So given this, I would expect that either the KPA always uses no latency filtering via the phase shifting method, or can operate either way when in variable latency but must use minimum-latency phase-shifting when in fixed latency mode.

As the video notes, the main danger of minimum latency filtering is that it can cause undesirable effects on correlated signals by creating phasing issues. In the KPA's case, there are no correlated signals - the KPA is only concerned with the mono guitar signal.

So how bad is the phase-shifting's effect on the signal? Sounds like experiment time! Fun, my first Kemper experiment. Here's the details.

I will take a couple samples of sounds that don't neatly fit into a nice continuous signal. So maybe a snare drum hit, human speech, and some disconnected instrument cable noise/hum. I will run this through a blank rig and a rig with 6 Studio EQ's - the maximum I can put on the Kemper. Each pair of EQ's will be designed to be frequency-neutral. So one EQ would cut at X HZ while the other would boost inversely. Each pair will focus on a different center frequency.

We should expect the rig with the 6 EQ's to possibly have more noise than the blank rig, although I'm not sure the degree to which this will occur. However, paying attention to the transients in those examples we should note the degree of artifacts introduced. I probably won't be able to test this until this weekend, but I will report back.

this thread started a while ago, but where are these technical references to the Kemper's profiling process? I have the US patent, but it seems some of the earlier posts references additional information released by Kemper. I, as well, am a music nerd and would like to continue learning about DSP.

Is the profiling signal sent to the amplifier simply wave playback? IE, is it always the exact same signal sent to every amplifier being profiled? I thought maybe this is the case, but maybe the signal is generated by a process, so that the signal dynamically changes algorithmically depending on the return signal, which will differ with different amps/cabs/mics/rooms. It'd be nice if it was simply a fixed signal, as that would open up the possibility of "hacking" the Kemper by figuring out how to tweak the return signals, such as recording them then combining multiple of them and feeding that to the Kemper during a "dummy" profiling process.

FWIW, I do not think intellectual property laws better society (see Boldrin and Levine's Against Intellectual Property, available for free online); however, that does not mean I think anyone who files a patent is doing something morally illegitimate.

i wrote an extensive high-gain guide on the Pod HD: http://www.foobazaar.com/podhd/toneGuide/

and patches:
http://www.foobazaar.com/podhd/patches/

Most of the good recordings you hear done with the Pod HD were by turning off its onboard cab/mic simulation and using external IR's. That said, I found the onboard cab/mics weren't horrible if you used "dual cabs", which means using dual amps with the same amp and virtually same settings but different cab/mics. However, each cab/mic has a different latency, creating phase issues on some combinations. So you really had to experiment to find combinations that worked. What I did is I panned them each hard L/R and sent them to DAW as separate tracks where I could alter the latency on them to determine how much delay was necessary to remove phasing. I did this with all the cab/mics against one cab/mic that served as my reference point, from which I derived the latency required to acheive phase coherency between all possible combinations. Then I put that in a spreadsheet and highlighted the cells where the latency was easy to acheive (EQ effects would introduce latency, although I preferred combinations that didn't require any effects).

Needless to say I'm dreading the eventual A/B comparison I'm going to make between the units. It's already quite clear the Kemper smokes the Pod in EVERY regard. The Pod doesn't even have an on/off switch. The things I've noticed that really stood out to me are that the KPA has a much higher resolution display, all parameters have much more liquidity and tweakability, the interface is very intuitive and well-thought-out, the ability to roll the guitar volume back to clean up the tone is MUCH better, the distortion tone is much better, the feel/dynamic response is definitely better, the tuner is way better and has much faster tracking, and everything feels more balanced and natural whereas with the Pod it's difficult to get notes and chords to get bite while still having palm mutes that can chug - you have to choose one over the other.

That being said, the Pod HD is not a piece of garbage. For its price, it does what it does very well, even if many aspects of it are unintuitive. It is a clear improvement over older generations of cheap modeling gear.

I know I'm a total KPA noob, but I already feel like I have a dozen or so rigs that could satisfy me for the next year. With the Pod I was never satisfied. I was proud of how nice I could get it to sound...but I never stopped tweaking in the 2+ years I've had it.

Saving cabs off of a rig is a slight pain, and I actually need a bit more info on it. I've heard you can use copy/paste with the name. I know there is a feature request for the cab name to automatically populate from the rig's cabinet tag instead of whatever the last cab preset you saved.

Also, when I save a cab preset, am I actually saving the cab somewhere onto the KPA, or is my cab preset simply pointing to the rig that has that cabinet. IOW, if I delete the LL ChugChug rig from my KPA, will I no longer be able to use the cab preset I saved from that rig?

I have played with most of the DEP's on the amp and cabs. I found the cab ones can very quickly get to WTF territory. I haven't messed with the low/high shift at all, but I have found some rigs where these values were adjusted. After 0'ing them out, I found that the original author's tweaks actually did improve the sound. But I have found staying with +/- 1 from 0 is a good way to start. I have yet to start to really get a good feel of "ok, this cabinet needs some low shift", etc. Usually I can hear a tone for a few seconds before I can feel where I should tweak.

Will, I know you're a fan of the "character" parameter, and I have also had good success bumping this up, although I never go past +2.

With the LL ChugChug cabinet, though, I find keeping everything at 0 is pretty much a perfect cab/mic tone for me.