Posts by Deny

    There are some rigs that have very strong sub bass frequencies, and lowering the bass on the the tonestack isn't the best solution, also the existent eq fx's cant handle those frequencies either, but i'm listening on studio monitors with a sub, so it's there.


    If you play live in a big pa, then when change to that patch, trouble can happen

    Wow, and for a long time I thought I was hearing things, thanks for showing me that I'm not crazy after all. Couldn't agree more, I'd love a tone shaping tool with a somewhat harder knee (like a 18 or 24dB/oct HP) for effectively eliminating those frequencies below a certain point.


    Now here's what I don't get: since guitar amps do *not* reproduce such low frequencies efficiently, why are they present in certain profiles? You can't capture something that isn't there, can you?

    Why not make the studio EQ's high and low bands selectable (LP/Low Shelf and HP/Low Shelf), with slope control in the case of the LP/HP? That's an old request of mine that was dismissed by the Kemper team, but it doesn't hurt to ask again. CK probably hates my guts by now hahaha :D

    How did you compare the KPA with real amps? Did you stick your ear right next to the speaker like the mic that captured the profile was probably positioned? No, right?


    Therein lies the problem: the KPA is supposed to sound like mic'd amp in your typical control room. I know, I know, we want the tone to sound like what we're used to get from the real amp, we'll get to that. Here's a list of things you can try in order to have profiles sounding more like the real amp:


    1. Not necessarily related to mic'd amp vs amp in the room, but mind your clean sense setting - you'll find that lower settings help with the harshness and improve dynamics. Lately I've also been playing with dist sens as well, it's an interesting control that helps finding the profile's "sweet spot" for your guitar. CK says it's identical to turning down the gain control in the amp block, but 1) I prefer to preserve the original profile gain setting and 2) if you put, say, a green screamer in front of the amp, the dist sens control will affect the input of that pedal and not the amp anymore.


    2. EQ. That's what engineers do when they want to get that raw guitar tone they've captured into a polished, good sounding tone. Here's a starting point for thin and harsh tones: dial a studio EQ in the "X" slot and set the 1st parametric band to Freq 160Hz, Q 0.6 and boost by 1-2 dB until you get a nice "oomph". Then adjust the high shelf to around 5.5kHz and cut by 1-2 dB. If by doing that you lose too much of the good top end, use the 2nd parametric band to add 0.5 dB around 2.9kHz with Q 0.6.


    3. Reverb helps add room ambience that isn't part of the profile and won't be there (from the actual room I mean) if your monitor is pointing directly at you. Well you should add room ambience if you're recording anyway, and I personally do it live as well. Here's a starting point: small room with time 1.8s, damping 5.0, mix 40-50%, bandwidth 3.5, frequency -1.5. Then add delay to taste if you like - you might have to lower the reverb mix once you turn delay on.


    4. If you're still getting some ice picking on the high strings, try raising the bias in the amp block to something between 1.5 and 3 and/or lowering the definition parameter, but not by much otherwise you'll lose the original character of the profile.


    5. If that isn't enough you can also try to lower the high shift in the cab block by -0.1


    Good luck! :thumbup:

    It is a false assumtion that the AD/DA add 2 ms to the equation. What is the source of this number?
    Contemporary AD/DA converters like the one in the Profiler add about 0.5 ms to the latency.
    Since Spdif interfaces are not latency-free either, the difference between analog and Spdif roundtrip will be 0.3 to 0.4 ms only.

    Apologies, I've quoted a previous post in this thread and didn't remember I once took my own latency measurements from the KPA, which amounted to around 3.5ms. The latency of the DXR10 is 2.2ms if anyone's interested BTW.

    Is the Macbook booting in 64 bit mode? 10.7 defaults to 32bit. Has to be 64bit for the driver to work.


    Is this for real? My "music" computer is an old MacBook white (4,1) which has a 32bit EFI and runs in 32bit kernel mode, although it's on 10.7.5 and does run 64bit kexts and programs. If this is true I won't be using rig manager after all, how sad.

    I've said it before, I'll say it again - remember, latency is cumulative, so:


    KPA: 5ms (3ms + 2ms A/D/A)
    Average FRFR with DSP: 2ms
    Average distance from speaker (in my case): 2m = 7ms
    Total: 14ms, which is barely acceptable *for me*


    I certainly wish the latency in the KPA was shorter, but I don't see it happening unless they come up with new hardware. I'll probably take a look at FRFR speakers without DSP again some day, didn't find anything decent by the time I bought my DXR10.


    One thing that bothers me is that manufacturers don't consider the A/D - D/A conversion in their latency specs, Kemper included, so I think it's better to come up with a term that conveys the total latency involved. Analog latency? End-to-end latency? I don't know, but I think the way manufacturers disclose it is a bit misleading.

    Talking about latency a bit more...


    Some time ago I took the Profiler and an active CLR to a good friend of mine, guitar player in the most important Italian jazz-fusion band, Lingomania (you can look for them if you like the genre, it is certainly worth a listen).
    He's a very skilled player and a teacher, and interestingly enough he did not perceive any latency with the Profiler connected to his Fender amp. But, when we used it with the CLR, he immediately after the first note said "well, there's a clear delay, here", which is the Italian way to talk about latency.
    Needless to say, I could hear nothing at all LOL


    I'm reporting this story because my friend had no bias whatsoever and was really curious about trying the Profiler. Furthermore, he had no idea the CLR had a DSP so he was expecting nothing under this respect, so it's kind of meaningful IMO.


    You bring up an important point, many FRFR monitors have DSP, mine included (Yamaha DXR10), which will add a good 2-3ms to the latency, resulting in a total of around 6ms in the worst case. Now for each foot away from your monitor that's another ms, and to me 8-9ft seems to be the limit for still being able to play and not feeling too disconnected, which makes up for a grand total of around 15ms - less is always better of course ;)

    Measuring latency is a real PITA, I did it once but won't do it again as I've come to the conclusion that it's only an issue if I can notice it. That said, I might be going bonkers because some times I notice latency and some times I don't. It depends on the profile. I don't spend much time on profiles that I perceive as having latency, I usually just delete them from my KPA and never look back.


    Now there are other things that can make the responsiveness of a profile feel "wrong", like lack of low end in the right frequency range (IMHO centered around 160Hz) or too much high end. I usually tweak those before I dismiss a profile.

    You should add bass on the monitor output only.
    It is counterproductive to add bass for the FOH as it would interfere with the bass instrument.
    The factory profiles are very balanced, but a good FOH guy will attenuate the bass frequencies due to that reason, as he would with a regular tube amp.

    That's possible for some rigs, in fact now that you've mentioned I believe adding bass in the monitor output is indeed possible for all factory rigs that I've tested, but some profiles require the use of the studio eq in order to add bass in a slightly upper frequency range in order to avoid boominess. Also I always check with the sound guy if my tone has too much bass, and tell them that if that's the case they can engage the mixing board 100Hz cut switch for the guitar channel. All of them have told me they didn't even have to press that switch and have left the channel eq flat.

    My magic number with the studio eq (also use the KPA through a DXR10) is a 1 - 4 dB cut at 5800 - 6200 Hz.


    Some times I'll also shift the highs at the cab section by -0.1


    Some times I'll lower the highs in the tonestack by 0.3 - 1.2 instead of using the studio eq


    Also remember that harshness can be overcome by adding low end, which I find missing in many rigs. In order not to turn things into mud, use one of the studio eq parametric bands with Q 0.600, frequency 160Hz and boost by 1 - 5 dB (depending on how thin the rig sounds).


    I mess with the definition usually as a last resort because to my ears it interferes with the rig sweet spot.


    I also keep the low cut switch in the DXR10 set to 100Hz at all times.

    some reamped clips?

    Unfortunately I don't have any, but anyway to my ears the change was for the better so I'm not complaining, just wanted to know if anyone else had this behavior. Like I've said, I suspect the change was the result of a firmware upgrade process, not a change in the firmware itself. Seems to have cured the problem that was discussed in the "Can't have my profiles close enough" thread.

    Just curious, in fact I'm not even sure it was the firmware itself or the firmware upgrade process, but I was revisiting some profiles I hadn't heard in a while and remember that most of them had too much bass below ~ 100 Hz, and now they all seem to lack bass. Then when I go ahead and turn up the bass control back to <0.0> (I had dialed most of them down a bit), the bass seems to be in the right frequency range instead of too low like I thought it was when I first auditioned the profiles.


    Anyone else experienced this?

    Have you tried the profile Holdsworthy by Ruppert? Very smooth for violin-like lead playing. There's no spikey fizziness. It could be too mushy for some, so you might need to tighten it up with some EQ and add a bit more presence/treble.


    For EJ's lead tone, he specifically uses a bridge pickup, but almost all the tone controls on his effects are rolled off -- the Tube Driver, Marshall, etc. So there's a lot of treble at the source, but it's EQ'd out it out to get that smooth, violin-like lead tone.


    Would love to have someone profile his lead, rhythm, and clean signal chains (without the delay and chorus of course)!

    I just looked in my rigs folder and it's there but it didn't stay in the KPA, which means I was probably not too impressed... Can't remember though, I've auditioned it over a year ago, might give it another try. Still I'd love to have more options :)

    What I've found out in over 3 decades of playing guitar is that it's not so much the amp as it is the player and the way he/she sets up his/her amp. The same amp can be suited to fusion tones, metal or country depending on the settings. I feel that most paid profiles are usually generic in nature, whereas the tones I'm looking for are very specific (midrange heavy, not a lot of bass or high end, for violin-like leads). Of course, some amps may be more suited to the task than others, maybe even some that were already profiled but not at the ideal settings for getting the tones I'm after. So my question is, are there any profiles that are just the ticket to what I'm looking for? If not, can I expect that something along those lines will be available in the future?

    What speaker is pumping the annoying guitar audio? Stage monitor? Not sure of your setup.

    The Banshee talkbox is just a small amp with a speaker connected to a plastic tube that you attach to your microphone. The talkbox effect is actually the guitar sound amplified by the device, going through the tube that goes in your mouth and then is modulated back to the microphone. Ideally the talkbox should be turned off while not in use because it's fairly loud, and up to a certain firmware (don't remember which, it's been a while) when I switched to a rig that wasn't using the FX Loop the previous state of the direct out was restored. With the current firmware, once you switch to a rig that uses the FX loop, even after you switch to something else, the direct out is kept turned on.

    I have a Rocktron Banshee talkbox connected to the FX Send of my KPA. The tube goes to a Shure SM57 which goes through a preamp and then the FX return of the KPA. Works well, but since some firmwares ago every time I select a rig with the FX Loop turned on, the direct output in the master gets also turned on. This sucks because from then on, even after leaving the rig that uses the talkbox, the banshee keeps its speaker on and pumping the guitar audio through the tube which ends close to my head (and ears), very annoying.


    I have complained countless times that selecting a rig with the FX loop turned on should not mess with what to my knowledge should be a global setting, but I am beginning to suspect that this issue won't be fixed. I know the direct out shares its jack with the FX send output, but IMHO after leaving the rig that uses the FX loop, the direct out should be set to whatever it was before selecting said rig - in my case, I'd have it switched off.

    I would love a fuzz that doesn't add a lot of top or bottom end and is capable of lower gain settings.


    The other stompbox in my list is a chandler tube driver or a similar pedal.

    I have a delay in performance mode either. when I remote my kemper with a daw it works pretty well when I send the controll messages 200ms earlier.

    Even when switching between slots within a single performance? The reason I ask is because that's kind of a big deal, when you switch to a performance the KPA loads all its 5 slots into memory so switching between them is faster. But switching between performances is actually slower than switching rigs in browse mode.