Posts by barefly

Hi Manfredi, it toggles/shifts between two morph settings.

would be a great feature, I could leave my switcher at home!

great song and great vid!

together with Jan Akkerman you could do a Dutch G3

der Talkbox wirkt aenlich wie ein Vocoder: das was man singt (oder spricht, pitch is nicht noetig) moduliert das Gitarre Signal. Eben 't' oder 's' uzw Klaenge kommen aus dem Geraet. Das ist beim orginal Talkbox schwer

t.v. helicon hat ein neues garaet: http://www.tc-helicon.com/en/products/talkbox-synth/
Aus dem Kemper bekommt man solche Sounds noch nicht

Since a couple of years I use a Yamaha stagepas . It consists of a power mixer with two frfr speakers. I have both the 400 watts and 600 watts. They both sound very good. Much better than my Yamaha dxr12. Drawbacks : 400 lacks bass (obvious with 8 inch speaker) but us incredibly detailed. 600 can be a slight bit with some profiles be honky, but far less than the dxr12.
Since using the stagepas I definitely switched from combo tube amps to frfr. It took me 2 years to switch

I have an old yamaha rex50 which was released in the late 80's: 31 kHz sampling and a very nasty 80's distortion (which is nowadays regarded as 'authentic 'en 'full of character' ) . I also have a TC2290 (which has the 1 bit sampling). We all raved about it back than, but now i'm working in my DAW with a RME fireface and every time i'm impressed how much improved the AD conversion is during the last 25 years: so much more transparent and with behold of the transients etc. But i'm still fond of these old machines, nostalgia..
And the difference between this rex50 distortion and kemper... listen: http://www.synthmania.com/Yama…Audio/21%20DISTORTION.mp3

Per, thanx for or the explanation. I have read that the analog filter which is always necessary before AD conversion is less demanding with oversampling. Oversampling can diminish noise floor but also heightens the Nyquist frequency which is necessary, so your analog filter is easier to build.With the sinq filter than applied you can downsample to f.e. 44.1 to get a clean signal at the AD conversion.

Very interesting thread! But hard to grasp completely...So if i understand correctly you can get (in the ideal world) an exact copy of an analog signal after AD and DA conversion. But only when you have ideal filters, otherwise alliasing will occur. With the use of the digital sinc-filters at the AD and DA stage such alliasing is almost not present (certainly not audible). But between the AD and DA part, all kind of math can be applied to the bits, and the amount of artefacts introduced during this proces is completely dependent on how well thought out the math is. On occasions you can compensate less thought out math formulas with upscaling the samplerate.
Is this a bit of a summary ?

wow, thnx Bert!

thnx man !

my band performing a Dixie chicks song. All guitars are directly from the kemper. I used profiles from MBritt and Bert Meulendijk. I have a lot of their great profiles and scroll through them to find the one which suits the music best. I never write down which profile I used exactly...
Hop you like it

Hi Nikos, Netheravon, Ingolf, thanx for your remarks, i meant to put forward some food for thought about a great sounding and interesting constructed piece of equipment. I'm almost 5 years on the forum (and still but a student ... ) and i always appreciate the input you (and a lot of others) bring to this forum. For me it is one of the fora which is best moderated, where professionals and beginners alike talk about a lot of interesting stuff, with love for guitar and sound. So please go on with that kinda stuff .
Sorry, back to topic.
In another article I read that there is an even more complex way in describing a non linear (distorting) signal chain in a formula. That formula is kind of 'time-dependant'. Which means it doesn't calculate bit by bit, regardless of the bits before, but in a dependable way of previous input. That article stated that f.e. circuits with transformers and condensators (Which can contribute to time dependent behaviour of an electronic circuit) could be even more accurately be described. Maybe with the coming of the quantum computer that will be possible . Although it is the question if it is worth to reduce 0.7 % of error (if i could extrapolate the results from the article i mentioned to the kemper)

i found another interesting read: https://www.google.nl/url?sa=t…M4Yd1WRAlLnFL4Vsz4X1Xa4Kg
I think this must be very similar to how the Kemper works? It is a bit counter intuitive you can capture a chain of different devices (pre-amp, power amp, cab, mike) in one formula. When this is indeed the way the Kemper works , some things are more clear: with this method you only need about 10 to 20 numbers to fill in the formula which leads to a very accurate moddeling of input. So no need for big IR data strings. It would also mean that what has always been stated by Kemper that there is no signature sound is true ( at least definately totally not in the way modelers based on linear modeling have) , and that there are no 'basic amps' stamped in to the memory of the Kemper.
In this article an error of 0.7% was achieved in modelling a tubescreamer. Although i think i can 't hear such an error, the article seems to imply the profiling method (which the authors use, i don"t mean the Kemper method) can be upgraded. It also implies that with more computerpower results could be even better.
But what i can 't comprehend is how the Kemper team has found a way in transforming parameters which are linker to (tube) amps (like bias, defenition, compression) in adequate changes in this formula.
Well, in the end only sound matters and the Kemper is great at it

Hi cags , welcome!
1) you are right about that. But you need a frfr system after the kpa that is flat in the frequency spectrum you wanna hear, of yo u want to enjoy the cabs grom the kpa. So for guitar that is about 80 Hz up until 4500 hz. A system that is flat in it' s output in this range is broadly available in the form of pa and monitor speakers. Maybe some some specialized speaker cabs are topped of at the high end but i'm not sure. Of course if you play baas guitar or acoustic guitar through the kpa other frequenties are to be produced by your speaker
2) if yo u have a good profile, there are no nasty high frequencies when played frfr, at least that is my experience
3) the eq of the kpa is not as sophisticated as the bheringer. I guess what the video describes is an option, but i think it will be degrading the sound quality because with this kind of equalising you introduce all kind of phase problems, not to mention that a static equalizer can never compensate the dynamic eq differences of a speaker cabinet.

Great pack, classy sounding amp and very nicely responsive to volume roll. There are a lot of options to choose from in this pack whether you need a more bass or mid-heavy sound or some more highs in your sound to suit your recordings. All the variants are well balanced, no resonance peaks etc. Full chords build up in a very nice harmonic sound (you still hear each string but little tuning differences don't fight each other but blend nicely) Great work, thank you Bert! And great demo, Patrick!

great sound!! ( and playing of course ) I understand your love for this amp, superb leadtone.

When I want to switch in performance mode from one bank to another I find it a bit difficult te read the small numbers on the screen of the remote, especially when i have to tap through some banks to get where I want to be. Would it be possible to have a third lay out on the remote where the bank number is very much enlarged? Or should i get me some glasses?