Posts by ckemper

    Because when I recorded both the dry di-signal and the wet signal through s/pdif I had such low output on the dry left signal and the wet stack signal was very high. To add gain to the low dry signal I had to first go and change the volume on the stack and raise the overall volume. And I had to do it over again with the next guitar. To me it would be so cool to be able to easily control both signals within seconds, if there were to separate output gain knobs. To make sure that the wet right signal doesn’t distort and exceed the digital 0 dB I have to adjust the signals and the whole stack.

    To make the signal stronger, we would have to include a boost for the SPDIF signal, which we would never do due to clipping reasons.

    When the Clean Sense parameter is set right, then the dry signal is leveled correctly.

    Why don‘t you boost the signal in your DAW or leave it where it is?

    Does this level matter in a recording?

    Really great news. I’ve been testing it for a day now. Really works like a charm.

    As for now I am only missing one thing. An easy way to control left/right s/pdif output level when recording git/stack. Maybe it’s just me that haven’t found it yet. But two separate knobs right next to each other would be awesome. Other than that I think it is very intuitive. Maybe it should be a lot easier to place an IR in the chain. Haven’t experimented with it though. But I would think that right clicking on one of the slots (for effects, amp etc) one should be able to select IR.

    Yes, we do read your feedback! 😄

    Why would be offer separate volume controls for SPDIF?

    Even the one volume control we have is useless.

    Why is volume control for recording through SPDIF necessary at all?

    Guavadude , at half swell our volume pedal engine does roughly the same volume as the FV 500. We are about 2 dB less, which is pretty close.

    However, in the first third we are more progressive than the FV 500, which is more or less linear by technical reason, and thus has a sharper onset.

    Is it this what‘s bothering you? The first third of the range?

    I saw that some pedal, depending on the function, works better with Type2 others with Type1 curves, typically for Volume is better Type2 (not linear), while for Wha/Morph better Type1 (more linear), but it's not true 100% of the time... this was my personal notes on my use about my tested pedals, maybe can be useful to someone

    That is quite an odd finding.

    Running a Type 1 polarity pedal with Type 2 turns it from a linear taper into an anti-logarithmic curve, as can be seen on the bargraph in the Pedal menu.

    The very progressive curve of the volume pedal engine is counteracted by that.

    Do you guys feel the volume pedal is too progressive (too slow)?

    There is usually no guitar amp that is noisier than others, since the main noise is cached by the guitar pickups and gets amplified in dependence of the gain control.

    Try to set the Noise Gate control (in the Input module) to the same value as on your rack, and then you should have the same low noise level.

    Those presets represent ranges of real wah pedals.

    If you feel the swell is not enough for you, then feel free to tweak the parameters „Manual“ and „Pedal Range“ to your liking.

    This is what these parameters were made for :)


    This is an issue with the Kemper that needs to be resolved. There are so many expression pedals and they ALL repond differently. If they’d just add a few parameters to control the curve like every keyboard controller has, then EVERY pedal would be able to be setup to respond perfectly.

    I don’t think this is an excessive request.

    Expression pedals all have a linear taper. And except for dead zones, they all react the same.

    Can you tell about differences of specific pedals?

    The only pedal with excessive dead zones known to me is the Boss.

    There is no cure for dead zones, as they cannot be detected by the connected device. No curve for that!

    We have improved the curve for the volume pedal function a year ago.

    Please check again.

    Anyone tried the noise gates 2:1 and 4:1 from the effect list?

    We recommend (in the manual) to combine the noise gate in the input section with an additional noise gate in a module before the amp, when using high gain sounds. As you say, those sounds need individual noise gate settings, in contrast to clean or crunchy sounds.

    Both input noise gate and an additional expander noise gate (2:1 or 4:1) make a perfect team, as they work with different technics.


    Then do what probably thousands of Profiler players do in that situation:

    Change to another rig, while the old delay is still sounding and spilling over.

    The new rig should have the rough tempo saved, as well as the dedicated tempo divisions set by Note Value.

    Tap when the song starts, to correct the tempo if required, that is when your drummer deviates from the target tempo.

    Do you use different delay divisions for different songs, or is it just different tempos?

    let’s wait for a week or two more...the guys are busy ...I will try the functionality in live gig today :)

    Stay well and best wishes.

    Your problem seems to be related. According to our support, they suspect that you use the TAP to set you delay timing directly, rather than setting a song tempo by tapping quarter notes and then setting the delay timing by using the "Note Value" to quarter notes, eighth notes etc.

    However, the latter is the standard procedure for modern digital devices to set up a delay time.



    I have watched the video that you have sent to our support.

    The issue that you run into is not related to the Profiler Stage in special, but you could trigger it with every Profiler and even the Access Virus.

    It not a bug, but the feature of being able to retrigger the timing at the beginning of a bar using the tap button, that I have described earlier in this thread.

    To make this feature work, there is a detection in the tempo engine whether you press the tap as sparse as once each bar or even less. This equals to a quarter or less, of the present tempo.

    So if you speed up the tempo to highs such as 250 bpm, then you have to press the tap button pretty fast still to slow down the tempo, rather than to retrigger the timing only.

    In other words: For making the retrigger funktion work suitable for the TAP, we took away the ability to tap in the quarter of the present tempo. So an attempt to tap a new tempo 60 bpm when the tempo is 250 bpm will fail.

    However, in a musical context, this would never happen. This is why this feature is up and running since day 1.


    We talk about several topics on this thread, but all about tap and tempo and timing, so let me bring some structure into this discussion.

    The Profiler has an advanced tempo and timing engine, inherited from the Access Virus Synthesizers. Keyboarders have very strong demands in the timing machine, especially to run arpeggiators synced along a DAW.

    The tempo engine receives information from the tap button (hardware or midi CC) or an incoming midi clock, or the BeatScanner.

    Effects like delays or the tremolo run on the tempo engine.

    Delays are easy, because they only need the right tempo, the timing is made by you playing your instrument.

    You might be used to tap four times to set the tempo on other devices. However, it's a good idea to tap more often. The Profiler adapts to your tempo with every tap, for an infinite time. It will even follow tempo changes quite smoothly while you tap.

    When synced to an external midi clock, you might observe tempo deviations. What you see is the timing engine constantly adapting to the incoming tempo. The visible tempo variations have no impact to the audible result. If you have experienced that other devices don't show tempo variations, than those manufacturers hide it by showing a highly averaged tempo. The result should be the same however. If you see the tempo deviate by small one digit numbers, then everything is ok. If you see two digit deviations caused by incoming midi clocks, then your midi interface is of bad quality and causes timing jitter of dozens of milliseconds. It is worth to be mentioned that today and since 20 years, the majority of midi interfaces have a terrible timing, unfortunately.

    The tremolo effect is a different beast. Here it's about the tempo AND the alignment to the beat, similar to an arpeggiator.

    The tempo engine has a special feature for realigning the tremolo: Whenever you press TAP after a certain time on a downbeat, the tremolo will shift to the timing of that tap smoothly. But don't tap too often, as it would sense it as a new tempo. Perfect is to place a single tap sometimes at the beginning of a new bar, even every bar is fine.

    Now this goes to hackowl . In many years we had no user syncing the timing of the tremolo to an external midi clock and reporting problems. But it does not work right the way you use it. I just took the time to look into the tremolo code and was happy to have fixed a quirk a few minutes ago. While the tremolo always played the correct tempo, it was easily going of beat, as you have showed up. Please check the next upcoming software update and tell me your experience!


    Sorry but I don't agree: pure cabinet isn't an eq, it softens the sound.

    Lower the HF has not the same result than use the pure cabinet.

    "not easy to add" isn't "it's impossibile to add": I hope you'll choose it for "improvements/upgrade to do"

    Thanks for the attention! :)

    Yes, you are following my arguments exactly.

    Pure Cabinet is not an EQ.

    It's made for changing the character of a cabinet sound.

    But if a rig sounds good on one a first reproduction system such as your PA, but not good on a second system, then it's not caused by the character of the sound, but by disadvantageous aspects of the second system, a frequency responce issue.

    It's obviously caused by your monitor producing too much high end. Thus a classic case for an equalizer.

    Pure Cabinet modifies the sound of the cabinet. Having different settings per output would require calculation two differerent cabinets simultaneously, which would add an unsuitable amount of calculation power to the system.

    For that reason we will definetely not implement this feature, even though we were asked for it two or three times.

    There is another reason:

    By using cabinet simulation, digital guitar amps make it possible to monitor the exact same sound that you will present to your audience, which is a huge advance over tube amps. We do not aim to implement features that would purposely counteract those advances.


    It's definetely not easy to add.

    I my point of view pure cabinet is not a substitute for an equalizer, and should not be seen as one.

    When the monitor is too harsh but the PA is fine, then the monitor needs to be equalized by lowering the HF.

    On the other hand, a Pure Cabinet setting that make the monitor sound good should also be good for the PA.

    Some believe that Pure Cabinet is not made for the PA, but it is made for it!


    Just a late resolution to your problem:

    You have obviously changed some of the Output Sources to values other than „Master“, or even „Off“, accidently.

    This will purposely cut your effects, or disable an output.

    Have a look a at the manual and also take a look at those settings in the Output Menu, now, that the are reset to good values.