At least controlling the internal looper via RM would be great IMO
Thanks for chiming in YG,
iIt's an .msi file, the Error message is file-specific
I am experiencing something weird, and wondering whether anyone has come across this as well.
When trying to install RM via its .msi pack, I get a windows Error report reading (loosely translating) "Unable to open the installation pack. Check the pack is present..." etc.
The first thing I thought was for the service Windows Install to be inactive, but this was not the case.
In order to make some troubleshooting, I tried and install older versions, to the same avail...
... until I used Rig_Manager_1_3_5_Windows 64-bit.msi, which to my great surprise was launched.
Does anyone have any clue?
I don't even own a single single coil guitar, only humbuckers.
I'd try with the Definition control (maybe using the profile of an acoustic amp). Although it doesn't actually turns a HB into a SC, it might lead to interesting results until you get your switch
Can I plug it in "pedal 1" to control the volume or does it need to be an electronic pedal?
You can le your analog volume pedal pretend it's an expression pedal by using a "Y" stereo cable and using both sockets on the pedal.
Or, you can place it after the guitar (or after Monitor Out) - with obvious limitations.
you can assign three effects per button
You can assign four fx, actually
having 2 effects on the button, you only can select either one on/off or two on/off. But not ALL OFF, one on, the other off and vice versa.
Correct, the switch just toggles the saved status of all the fx. For a more articulated management, switch-driven morphing might be a more versatile option.
Ye, it's certainly doable, and... done already
Well, Stage misses the "Stops" and "Effects" buttons so it sorta makes sense to not have the function UI-wise. OTOH, if a "group" preset was saved/present in the preset list in RM, it would be nice that releasing it onto any of the slots of one of the two sections loaded the preset on the whole section, hw buttons or not
Yet, this comes up periodically since... day one
One issue would be that the effect of each tone knob in a guitar amp usually changes depending on the others', so that for example the eq curve set by the Mids knob @ 12 is different depending on the Treble knob's position.
No profilation might capture that: what we actually would need would be amp-specific tonestacks behaviour, which have been promised years ago. Kemper might have possibly dropped the feature?
I think these are useful requests. May I add that an option to handle multiple presets (e.g. chorus + delay + reverb = preset Ambience FX) would be nice.
Represent buttons for "Stomp", "Stack" and Effects" within the editor. This would allow the copy and paste of groups.
Apart from the Stage, Stomps and Effects sections can be saved as group presets. Don't they appear in the Editor among others?
Asking for a friend who uses W7 ;D
I like the pitch-dependent octave shift idea. This kind of "octave below F#" programming could also provide the opposite, especially if adding a mix control - Octave up below F#, providing Nashville tuning, and a 12-string kind of sound.
Excellent extension Paul
The ability to determine the threshold note would be critical, for example when one uses a capo - or a transposer.
I often find myself using a capo on the V fret, in which case the G (as on the third open string) should be considered a bass/root note as well (or the higher A for a capo at the VII).
I've read great reviews of those mini-monitors as well: "their tone is incredible for their size".
The thing i keep seeing here is that the Kabinet is to be connected to the monitor output, but as a cab shouldn't this primarily be the speaker output (i.e. like a normal cab)? Surely the Kone IRs can be used on speaker output mode too for all profile types.
Yep, the point is that the power amp is led by Monitor out. So basically the information means "not Main out".
The real problem I've found with a tweeter in a FRFR used for guitar is something you don't hear mentioned very much. When a woofer\tweeter cab is designed, the crossover is engineered to allow proper integration of the drivers over the frequency range. This usually involves some attenuation of the signal to the tweeter as compared to the woofer, as the tweeter is usually more efficient. But the problem happens at loud stage volumes; often the woofer can run into acoustic compression and hit a maximum volume level, even though you put more power into it. This is a form of speaker distortion you hear often in loud amps. But since the tweeter is more efficient and often has a higher maximum acoustic output, when the woofer is at its acoustic max out, but the power is increased further, the tweeter will continue to get louder and be unbalanced with the woofer acoustically. That makes the tone very unpleasant at high volumes.
I hear you.
This means that the crossover is poorly designed tho, and that the two speakers are not matched well. You never hear this in a high-quality multiway system for hi-fi, for example.
Often this has to do with the cabinet's acoustic impedance: if the internal volumes for the two drivers are not properly designed (or if there's no division at all), at high volume the damping effect of the internal air reacting to the piston motion becomes prevalent for the mid-woofer, which moves a lot more air than the tweeter.
The audible effect is indeed similar to compression but is actually a bit different, because there's no actual limitation of the response energy but a different distribution in the time domain. You see that in the waterfall graphic, and it becomes apparent in the speakers' time alignment (we call it step input signal, not sure about its name in English), where the mid-woofer's peak shows an increasing delay with respect to the tweeter's.
The Imprint thing is a different "compensation", they are also fully digitally controlled.
The FR runs in parallel, for the Aux In.
So If I get it right it would be correct to say that it's not intrinsically the Kone that's driven to linearity by the DSP, but rather that each kind of signal (be it from the Aux In, from the Imprints' computational engine or from a complete rig with a cab profile in it) is processed according to the already-known Kone's physical transfer function so to return an overal linear (or faithful if you will) response.
Hope this makes sense.
Do you think quantisation could be added via programming? If they introduced quantisation that’d be perfect. I wasn’t sure whether it would require different hardware etc.
You mean in the KPA, don't you?
Sure, it can be implemented in FW. There's basically no dedicated hardware in a digital looper (apart from the switches ect.).
I am not sure we'll see this tho: it might be very low on Kemper team's priorities list (if ever present), and/or there might be limitation in DSP/real estate if other functions have been planned.
It would be great tho, yeah
The Kone does not feature a natural lowpass filter as the Kone mode fully compensates for it.
So is the compensation always on, even in Imprint mode and when no audio is fed through Aux in?
Let it be clear that I meant that as a compliment, to acknowledge the quality of sound I've been hearing about when Imprints are used
Also, totally agree on some 2-way speaker/cab being too bright! Too often, tho, the electric guitar player end up believing that a tweeter unavoidably makes for a harsh sound, and you'll agree this is another myth to debunk Of course, if we send a linear cab the signal we use for a guitar cab, we get unwanted harshness.
An FR speaker SHOULD only put out what you put into it, but I don't believe there is a PERFECTLY FLAT speaker anywhere. I've never seen a frequency response chart for a speaker that was a flat line.
This question has been asked to Jay Mitchell (the designer of the CLR). His answer was that what matters is that the response is flat enough with relationship to human ability to discern differences.
A weighted response, if you will.
The top end can be what I think of as too hi-Fi. Especially on the heavier profiles with more gain. Not bad at all and would work fine in 95% of the applications with a live band.
This is what a close mic'ed cab sounds like. When using a linear solution, the electric guitar player should EQ their tone the way the soundguy does.
Think of a linear cab like a block of marble, where you can sculpt your tone.
The Kabinet didn't have the hi-Fi sound to it and just feels more natural.
Right, because there's no tweeter and the above-mentioned lowpass filter is obtained mechanically, so to speak.
This is part of what "amp in the room" means.
So the effect is not linear. I know of no “loudness” compensation button on any stereo system that took this non-linear behavior into consideration. When you activated the compensation you got the same level of bass boost regardless of where the volume was set.
It's not only not linear, it's parametric (it depends on the reference level). Anyway those curves are statistically got (IOW, that's an average among all the reported feelings of those who were tested: not scientific at all in a sense, but very useful in order to understand how you hearing works
Certain device exhibit a continuous loudness compensation, so that you can dose it at will (Yamaha A500 comes to mind).
with a 200w poweramp (the bam200) there is no risk of blowing up the Kabinet (also 200w)?
Keep in mind that the Kone is labelled to be 200 W @ 4 ohm (nominal). Your amp's specs probably refer to 8 ohm?
anyone got a double Kabinet stack? Not sure if there is any point and would it be different to a 2x12 Kone setup?
Ye, it (generally) would. When two speakers output the same signal in phase on the same plane, the distance between them makes so that the same freq from the two is algebraically summed at the listening point (that is, according to their phase), and this is of course different by frequency.
the result is that some get empathised, some attenuated, some cancelled (at a given listening point). This effect is called comb filter, because of the frequency spectrum's typical look
[Blocked Image: https://recordingology.com/wp-content/uploads/2011/11/combfilterweb.jpg]
At a given listening point and position from the cabinet, the filter's look (and sound) depends on the distance between/among the speakers. So in general we can say that a nxy" cab exhibits a peculiar comb filter, but if the speakers can be moved from each other there's more freedom at somehow controlling the filtering action.
Apart from this, different cabinets (enclosures) will generally sound different from each other. it's not just a matter of internal volume and building material: the shape of the cab (and any internal panel/channelling) determines not only the amount of air that moves inside the cab at every piston's movement, but also its speed and latency. This works like a damping system, that deeply modulates the cone's "on air" response.