5.5.0.13341 Public Beta discussion

  • So one question and forgive me if I am way off on this... but on the direct out (send) can you send a stereo signal out now (as in L+R to a strymon or anything else) that takes two inputs?


    Cheers

  • It wont make a difference unless you have a bat's hearing. 22.1k is well above most peoples upper frequency threshold. You really wont hear a difference unless there's an actual bug in the Kemper. But don't take my word for it, try it.
    Moving outside of the Kemper into the whole 96k argument then higher frequency rates actually often result in poorer quality due to clock jitter being introduced, not to mention when it comes to recording there's the issue of larger sound files. The frequency you need to get rid of things like aliasing artifacts from cheap plugins needs to be about 10x higher i.e. 415k, so with 96k you're only taxing your system more (you can test this using something you know aliases badly in your DAW and trying different sample rates). Once you get audio from an external source into your computer it's going to have a harder time too. It doesn't reduce latency because the bus and CPU on your computer isn't magically faster when you're throwing more data at it, in fact it could potentially create additional bottlenecks and you will likely have to increase your sample buffer to compensate, if you don't have to when you run at 96k then you should drop your buffer at 44.1 instead, if it's already at the lowest your driver will allow then it makes some sense to increase the sample rate while keeping the same buffer size as that will be a way to reduce, but otherwise it's not doing anything.


    True about the sample rate and latency.


    About jitter:
    Higher sample rates are not more sensitive to jitter when we talk about the hearing bandwidrh
    There is no jitter at all when the whole recording chain is digital.
    Modern converters have a perfect jitter suppression.


  • Do you have a reference to the thesis that converters sound different at different sampling rates?
    I doubt that there is a true difference.
    Faster is not better. Modern converters do they work at some MHz of sampling rate. This rate does not double when you double the incoming sampling rate. Don‘t get fooled by internet talk and false marketing.


    The added latency for analog AD DA conversion is less than 600 nanoseconds. Cannot be noticed.

  • Do you have a reference to the thesis that converters sound different at different sampling rates?
    I doubt that there is a true difference.
    Faster is not better. Modern converters do they work at some MHz of sampling rate. This rate does not double when you double the incoming sampling rate. Don‘t get fooled by internet talk and false marketing.


    The added latency for analog AD DA conversion is less than 600 nanoseconds. Cannot be noticed.

    CK


    Bit of a side-step ....... but now we have sampling rates up to 96k ...... any chance of getting the option to Profile via SPDIF as oppossed to just "analog-ly" ????


    ie:- connect a Axe FX or Helix to a Kemper via Spdif .... set each unit to the same rate .... say 48k [or higher] and Profile that way ???


    Ben

  • Just wanted to thank Mr. Kemper and the Kemper team for another fantastic update. This company freaking rocks! I'm going on five years of owning a Kemper, and these folks are still adding features and improvements. If it were any number of any other companies, they already would have released three new products and quit support for the older models in the same time span...


    :thumbup:

  • So I've installed the update; I've been happily using 44.1 spdif for years from my Kemper into my (aging but still good) Focusrite Sapphire.


    I just tried the 48K spdif and could not get it to work with my interface - no sound.


    Whilst this is almost certainly user error on my part, can one of you good people tell me what I'm actually missing with my present interface by staying at 44.1? If the answer is 'nothing' I'll stop looking for an answer :)

  • Gary W
    In Cubase I set my record preference to 48k; set RME interface settings via Control Panel software to External via/from AES (ie. Compatible with SPDIF) source and KPA to 48k.

    Will

  • Do you have a reference to the thesis that converters sound different at different sampling rates?
    I doubt that there is a true difference.
    Faster is not better. Modern converters do they work at some MHz of sampling rate. This rate does not double when you double the incoming sampling rate. Don‘t get fooled by internet talk and false marketing.


    The added latency for analog AD DA conversion is less than 600 nanoseconds. Cannot be noticed.

    Well, my experience comes from 30 years of studio work, not youtube tutorials.
    The fact that many professional studios and producers prefers to work at higher sampling rate is well known
    and It has nothing to do with the Kemper itself, but if you just record a band at 96KHz it's very likely you can hear a big difference. Most still works at 44,1 and 48 for tecnical reasons, big projects with more than 100 tracks are very common and can put your system on its knees easily when mixing. Having the possibility to digitally connect the kemper and not having to use another roundtrip of AD/DA is a great improvement in a studio environment.
    By the way, please give me the name of this speed of light converters that can do a an AD/DA in 600 milliseconds, it's usually more than 1 millisecond and it seems to be nothing, but if you add that to the latency you already may have from the rest (Latency compensation from the DAW, digitally controlled monitor, etc:) it's can build up to something anyone can feel while playing.

  • Love the update! Very cool new features indeed. I actually run a stereo enabled cabinet on stage for monitoring - so far always in mono, but it would be fun to have stereo monitoring live ;)


    Question for Mr Kemper though;


    What I *really* would love to see on the Kemper is multiple effects loops, so I can run some external stomps in front of the amp block and also external (stereo) stomps after.
    Would that even be physically possible to implement with the connections on the unit?

  • My personal experience is I don't care about sampling rate.


    The problem is that my Sound engineer is !! :)


    And I had to record my whole album at home at 48khz with XLR cables. I am happy because now I can record for him in spdif and it is way way easier with my configuration.
    I have no XLR cables everywhere at home and my wife is happier.


    KPA team, you made my wife happier, so you made my life easier !!

  • .
    By the way, please give me the name of this speed of light converters that can do a an AD/DA in 600 milliseconds, it's usually more than 1 millisecond and it seems to be nothing, but if you add that to the latency you already may have from the rest (Latency compensation from the DAW, digitally controlled monitor, etc:) it's can build up to something anyone can feel while playing.

    I think what CK said was the the converters latency is less than 600 Nano Second which means basically no latency. One nano second is one thousand-millionth of a second or one billionth of a second, but you never know some folks can very well claim than they can sense one nano second latency :D

  • Is the transpose still available as an effect in the stomps? It could be a problem when taking a USB backup between different systems otherwise. Sometimes a hired unit is used and can be an older system.

    Karl


    Kemper Rack OS 9.0.5 - Mac OS X 12.6.7

  • Thanks everyone that you like our feature update!


    Now that we talk digital audio rates again, it‘s time for resolving some digital myths.


    If you have any device or software that sounds better at 48 than at 44.1, then it is very bad made.
    Especially if a reverb would improve its sound at 48 kHz, it would be a hell of bad code. Can you name one?

    From a standpoint of what we hear when we play, even 44.1 kHz is more (much more) than adequate to provide perfect analog reproduction of frequencies up to 22 kHz..... which is much higher than any of us can hear anyway. I want to stress the word "perfect". You can mathematically create the original analog waveform PERFECTLY at this sample rate. Higher rates do not impact what we hear.


    Now the rub.....


    Some of you are correct in that all recording engineers will vastly prefer 96 kHz.... but not for what you were thinking.


    Different algorithms which operate on raw recorded information will degrade the signal if the algorithms over flow or under flow the data. The 96 kHz data is more forgiving to lossey algorithms and provides more dynamic range for the algorithm to work with in your DAW.


    I absolutely do not believe anyone can hear the difference between a 44.1 kHz signal and a 96 kHz signal.