Posts by ckemper

    What was the rationale for excluding the Sat and C45 switches from the Friedman tone stacks?


    I feel they defintely have an impact on gain and tone.


    We have chosen to avoid modeling controls and switches, that do not truly (!) interact with other controls and switches.

    The same is true for controls that in addition do not contribute to the task of adapting your guitar to the amp.

    All these controls are captured in the Profile, as usual.

    It is advantageous to mention these settings in the text tags of the Profile.


    The idea of Liquid Profile is to leave the majority of the tone sculpting to the Profiling process of an individual amp.

    Therefore we want the amp parts that are being modeled to be kept at a necessary minimum, to reduce the margin of error of the modeling process.


    The gain control with its optional brightcap is a cruical part for adapting your guitar to the amp, and at the same time the impact of the bright cap is dependent of the gain control position. This is absolutely worth being modeled.


    The same can be said for the tonestack, that serves a similar purpose, and has its own interdependencies as well.


    CK

    Thanks for this answer.
    It would be nice to have a list, which tonestack structure fits which amp.
    F.e. a specific Marshall tonestack (amp 1) fits also to Marshall amp 2.
    As a user I could see directly which tonestack I could use.
    :)

    So far there is no similarities to other models, that we know, but not listed. (Except those Fenders that I have mentioned.


    Except: All EVH amps seem to feature the same gain and tonestack structure.

    That is Peavey 5150, 6505 and all EVH amps.


    If you give us a later Marshall amp type as an example, we could have a look what similarities there are to the vintage Marshall amps.

    Yes, there will be more.


    We are currently working on a number of amp types, that we consider to be relevant as well.

    There will be e.g. Bogner and Diezel, and an additional number of Fender Types.

    Ampeg too.


    Some of the new Fenders share the same gain and tonestack structure as the Vibrolux.

    That is the Pro, Bandmaster, Tremolux and Vibroverb.

    We have decided to still feature them as separate Amp Types, so while still making the same sonical impact, they will display the exact type of the profiled amp.

    Sample Rate Converter? This is all beginning to sound like "Liquid Sampling" is coming to your favorite DAW. :/


    I know you can resample from 96k to 48k or 44.1k but can you up-scale from 44.1k to 48k with no problem, or does the DAW handle that? Like if I set my project to 48k, then any lower sample with be upscaled to that 48k?


    Yes. Liquid Sampling is quite an appropriate term.


    Upscaling is a standard technique similar to downscaling.

    It is even a bit easier, as it does not require an anti-aliasing filter. It's just interpolation.


    On most DAWs you can import 44.1 k files to a 48 k and they are automatically resampled. And vice versa.

    Logic does that same resampling in realtime during recording.

    the vast majority is 48k minimal.

    CD’s Were the only thing 44.1 and there are people today that weren’t even alive when those were the thing. Furthermore - I think a Kemper user can easily set up sample rate - I mean look at all the other Kemper features that aren’t “plug and play”. This thing is deep. Default it to 44.1 but have a 48 option. Protools is the industry standard - so should have that in mind - and why put excessive strain on the host computer recording. ALSO - sample rate conversion is re-sampling - might as well go analog… so this feature is great, but if it can’t do 48 - it’s worthless to many.

    Sorry so harsh - but “plug and play” translates to an excuse to me.

    But thanks for the “kind of usb” option.


    The Profiler has been designed for running on a sample rate of 44.1 kHz. This gives us a determined amount of calculation power and thus a stable system for the user.

    Internally we utilize sampling rates up to the MHz region, where needed. (Maybe in contrast to some PlugIns, that sound better when driven with higher sample rates from the outside.

    Therefore the Profiler would not gain better sound quality at higher rates. For guitar stacks with its speaker high end roll-off even 32 kHz or 24 kHz sampling rate would be totally sufficient. Many users use the HighCut to eliminate very high frequencies.


    If we provided selectable sample rates for USB, we would utilize sample rate converters as well, as we already do with S/PDIF, we had to use an SRC for each USB input and output.

    That truly would put a some strain on the USB interface, but mainly on our system , which was originally meant to be a digital guitar amp.

    I am sure that SRCs running in a DAW in a pure software environment will not cause significant strain.


    I am aware that Protools is one industry standard DAW. I am sure they are watching the innovations of another industry standard DAW, that is Apple Logic. The latter have managed to implement realtime SRC for recording and playback a while ago. And Aggregated Driver, to combine two or more audio interfaces. To me, this developement was obvious, and just a matter of time for such big companies. Protools will have to follow, if they feel the request of the users.


    I would feel a bit old-school to fill this gap on our side, while a much more intelligent solution is obvious and visible.

    It's one thing if the unit isn't capable of 48k. However, there are a number of valid reasons for the higher sampling rate, including the construction of the low pass filter necessary on the back end. For one thing, 44.1 requires a much steeper filter, where 48 can use a gentler slope. You can rely on the SRC in your particular DAW to handle the conversion, but not all SRCs are created equal and the conversion can introduce artifacts. (That's the best reason for setting everything to the same sample rate rather than relying on conversion after the fact.) Now, if Kemper simply relied on SRC to produce a 48k signal then that puts the responsibility on them to deploy a high-quality SRC - so again, if the unit is actually designed to operate at 44.1 only then it's debatable as to whether that's really the best thing for them to do.

    All that is said was written before, and was very true until the early or mid 90ies.

    But not for today.

    The good age started with terms like oversampling, 1 bit conversion and noise shaping.


    Todays AD and DA converters run at sample rates in the MHz region. They utilize digital filters that can be easily designed as steep as needed, without drawbacks.

    The same is true for digital sample rate converters, that use a similar kind of filtering.


    CK

    Would be very nice if was possible. Protools don't accept open a 48 session with Kemper. I have to open with my onboard audio then save a new session with 44.1 to open with Kemper.

    Hi!


    As mentioned, there are DAW's out there that do exactly that.

    If ProTools doesn't feature this very helpful and smart method, it might be time to ask them to improve, not us.


    There is only a handfull of relevant DAWs out there, while many dozen audio interface solutions are available.

    It would be a better world, if all DAWs would take any sample rates independend of the chosen project sample rate.


    Especially for musical instruments, where the audio interface feature is an accessory rather than the main purpose, it is very inefficient to circumvent those shortcomings of DAWs.

    DAW makers have a much larger workforce and different economy of scale to provide the appropriate solution.


    CK

    Hi! Thanks four your question.


    Modern DAWs such as Apple Logic work seamlessly with different sample rates, independent of the project sample rate. That means:


    - recording with automatic sample rate conversion, when the audio interface SR does not match the project SR

    - playback reverse the same way

    - offline-rendering the master mix to any target sample rate and bit width etc.


    Having to match the SR of the audio interface to the project is not necessary here.


    CK


    The Looseness does not drive a constant delay, however there is a delay variation.

    It is roughly between 5 and 50 milliseconds.


    CK


    The Stereo knob will reverse the sides, when you dial to the negative left half, as you have expected.

    The specific nature of Air Chorus is that the modulated signals don't mix in the same channel, but "in the air between the speakers". The effect is much more subtle than the hard beating you would get from combining mutually detuned signals in the same speaker.


    For that reason, as soon as you would mix in some dry signal, the effect would suddenly become much more pronounced instead of being regulated like you expect.


    I believe that is the logic behind the omission of a mix parameter.

    Exactly that!


    There is a number of effects where you wouldn‘t want to control the effects intensity by a mix control, but sometihing else.

    A good example is a compressor.

    I have noticed when using kemper in stereo in DAW that the left channel is louder by about 1.5db according to the track meters so I assume the double tracker exacerbates that. And then you have the Haas effect on top of that - the first sound (channel) you hear is the prominent one. It's almost impossible to get an even left and right channel due to numerous factors. You can compensate some when tracking in DAW but it's never perfect- and really doesn't matter in a mix.


    The delayed side of the Double Tracker is the right (!) side, and this right side is amplified by 1.2 dB to compensate for the Haas effect.


    And yes, the Haas effect can never be perfectly compensated as the level perception depends in the listener position.


    Please check .


    CK

    Can't hear much difference at all with amp block off - not sure what this tells me though, other than at least it's not some ghost signal ever-present in my Kemper or pickups. The question remains whether this frequency range is more harsh with the Kemper than if you were to record a real amp. If it's normal tho (i.e. not a particular fault in my Kemper) then it's sort of a moot point anyway I suppose - is what it is. Usually doesn't take much EQ to tame it - after all this only came to my attention because of a profile where it was particularly bad. Most profiles it's still there. but bearable/tameable. At this point I guess it's just a matter of technical curiosity as to whether the Kemper accentuates this harshness due to it's profiling tech or digital nature.

    If the Profiler would emphasize certain frequencies of a profile, this would have been reported by numerous makers of profiles, simply because it would easily be spotted in the A/B comparison.


    Post an audio clip, so we can check if it sounds wrong.


    CK

    Heres a Picture Left is KPA SPDIF Vol @ 0 latest Beta

    on the right KPA Analog out Amp Vol @ 0 Master@ 0

    Fender Rig

    Thats what i mean, not useable far away from what i call a good Input gain

    with the latest release the gains were even with this settings

    Thanks for this pic.

    What you get now, is the very original digital signal that is happening in the Profiler.

    The Profiler leaves enough headroom for upward dynamics.

    Boosting this signal before SPDIF does not improve the signal at all, but clipping SPDIF is at risk.

    In difference to an analog signal, boosting a digital signal does not have any positive effect, since you don't level an analog device.

    Thus, this is a good input gain.

    Boosting it in the DAW is fine however, because there the signal processing does not have a clipping limit.


    Be aware, that in the DAW mixer, you will approach a 0 dB headroom anyway, so I assume you would have attenuated the Profiler coming in at -6 dB anyway.


    With the latest release the gains would not have been equal still, since your picture shows a 12 dB difference.

    This -12 dB is the amp signal (distorted) or the DI signal?

    If I record Git Studio through S/PDIF, I want to be able to reamp using exactly this same level going back into the Profiler through S/PDIF. It's been 10 years I didn't have to worry about reamping levels. And now, for no good reason, you've changed that?

    If I ever had to do small changes to the reamping level, I'd do it in TotalMixFX. But I leave the recorded DI track in the DAW 100% untouched.


    That is probably the main issue.

    I broke the Profilers promiss to maintain the relation of DI output versus reamping sensitivity.

    I have missed to correct that sensivity by the same 6 dB. This will follow very soon!


    Thanks for this hint.