Posts by nejo_hh

    There is even more to learn from the above analysis:

    1. There is also an implicit high-cut present at about 21.83kHz; see attached plot frequencyResponse-KPA_CABINET-flat-highCut.png.
    2. The ripples seem to be evenly spaced with a raster of slightly over 86/172Hz. Dividing 22.05kHz by this results in approx. 256/128 support points of the CABINET filter.

    Another humble suggestion to the Kemper team: If it is possible to individually define the center frequencies (like for a cascade of IIRs), how about a log(f) spacing? See attached plot filterSupportPoints.png.

    I've thought about putting out my live rig performances but people would probably be taken aback by how much fx I have on them. Out of context, there is a lot of delay and such. But in context and since I program everything to the exact tempo of each song, the delays kinda disappear and I have to run the mix higher to hear/feel them where I want. But listening to them in isolation they sound pretty effected. If you play those rigs on a different tempo song it would sound cluttered I would think.

    Please! I think many of us would love to learn from one of the greats.

    Also, I usually use the profiled cabs on most of the clean tones, but as the gain goes up, if I hear too much tonal difference or feel like the bottom or low mid drops out I will try a couple of my favorite cabs on the gain rigs. There is one in particular that I use a lot for many of my live performance rigs. I use the same cab most of the time when profiling but the Kemper algorithm that decides what is the cab and what is the amp creates slightly different sounding cabs with every profile. One of those I just find particularly pleasing and it works well sonically with the band. I would say that most of my mid to high gain tones use that cab unless I'm really just going for a "different" sound like on our cover of "In The Air Tonight" where I want that mesa grind.

    Thank you very much for chiming in and providing such valuable insights! Would you mind sharing the actual name of the profile containing that magical cab?

    Short version

    If you plan to profile (bass) amps or want to use (bass) cab IRs: Be aware of the implicit low-cut filter at 70Hz of the cabinet slot.


    Humble suggestion to the Kemper team: How about having frequency and slope of the low-cut filter as (internal) free parameters?

    Long version

    I've tried to profile a bass amp through the DI out but completely failed at first. I've then tracked down the cause and was really surprised by the findings. So why not share them?


    Already a coarse comparison of the frequency responses of the amp and the profile using white noise showed major differences: There was a steep low-cut at about 70Hz present in the profile and a ripple of alternating bumps and dips up to about 2kHz.


    But I've seen many bass amp profiles with deep lows. So how do they differ? It turned out that all such profiles were direct profiles, i.e. without a cab. In contrast, I naively hadn't engage the No Cabinet option for profiling the amp because the highcut filter of the amp was engaged which provides a gentle high-end roll-off like cabs do.


    Now having a clear suspect I imported an IR with a totally flat frequency response and analysed the resulting CABINET preset: A steep low-cut at about 70Hz, a light bump at about 100Hz, flat above; see attached plot frequencyResponse-KPA_CABINET-flat.png.


    Out of curiosity I've then tried to get a truly flat response by inverting the "flat" CABINET response to compensate for the unwanted "add-ons". The resulting frequency response was then transformed to an IR and imported via RM: Still a steep low-cut, above prominent ripples with decreasing intensity; see attached plot frequencyResponse-KPA_CABINET-compensated.png. Apparently, the algorithm is trying hard to match the desired (low-end) response but needs to introduce over- and undershoots in order to minimise the overall deviation. Actually, this is a common problem for filters with a limited number of uniform filter stages/taps: Large gradients in the low-end response are just hard to reproduce.


    Possibly said effects (low-cut, ripples) are the reason for some people finding it hard to get the low-end/low-mids right when profiling?


    KAOS version used: 8.5.8. Infos on the methods used to do the analysis: here.

    Hi cbecker999, welcome to the Kemperverse! There is an easy way to get a clue of what all the EQs are doing in sum: Send white noise through just the stack of EQs and analyse the result with a (Match) EQ in your DAW. And if you're after the tone of the cab plus the stack of EQs: You can make an IR of the whole thing by following the instructions here and here (sections Craft... & Finalise...). The final IR can then be imported to the Profiler via RigManager. But as already pointed out by others: If you want the whole rig in your Profiler: try profiling it. Hope that helps.

    If the space or computing power is too limited it would be great to add a render button to the studio eq. What i mean by that is, you choose the studio eq make the changes you like press a render button and these settings will then be baked into the cab, creating a duplicate of this cab with the eq already applied. That would make things ultra useful

    Hey Bommel, how about doing it yourself? Extract the cab IR with the desired EQ engaged and import the result via RM. Just make sure the EQ doesn't clip the output. Hope that helps...

    Oh, and many thanks for all the good stuff you share!

    Another possibility would be to extract the cab IR from said profile and then bounce it through the match EQ to form a new cab with the desired frequency response: irSampling.wav -> DAW -> Profiler (well prepared) -> DAW -> MatchEQ (zero latency) -> WAV -> Edit -> Rig Manager -> Replace original Cab


    Edit: A more elaborate way would be: ((cab IR -> frequency response) * (match EQ -> IR -> frequency response)) -> minimum phase FIR filter [i.e. the final cab IR]. But that would require quite a bit of DSP math...

    Hey Matt BS, there is no need to perform the deconvolving step: Just directly sample the filter taps of the Match EQ (make sure to set it to zero latency) following the instructions here and here:

    1. Bounce the sampling pulse from here (section Craft the magical sampling DI signal) through the Match EQ (set to zero latency). Use Wave with 44.1kHz/24bit.
    2. Finalise the recorded IR following the instructions here (section Finalise the recorded IR).

    Edit: Also make sure that the Match EQ does not increase gain (check the peak level of the output before bouncing).

    Edit2: Disable normalisation, dithering and other fancy options for the export (you need it to be raw/unaltered).

    First: Many thanks to the Kemper Team for constantly improving the KPA OS and RM, and providing everything for free to their customers!

    In order to help to improve things, here my current bug list for RM 3.1.62 (macOS 10.14.6), KPA OS 8.0.6, non-powered Head & Remote:

    1. Double-clicking a cab preset also engages effect in selected slot.
      To reproduce: Load a rig with a disengaged effect in slot C, select slot C in editor, and then double-click on any cab preset.
    2. Weird assignments to remote buttons when switching rigs while effects are locked (see attached photo).
      To reproduce: Load factory rig BM - Acoustic Fingerpick, lock all slots except for amp and cab, and then double-click on factory rig BM - B&H Filmo crunch.

    EDIT: Changed title to [FIXED]...

    • Bug 1: Fixed with OS8.2.2/RM3.2.30 (Thanks, Kemper Team!)
    • Bug 2: It's a feature! Whenever you lock a module with an assignment to a Remote button the lower right LED is lit. However, this doesn't hold true if there is more than one slot assigned to the button in question.

    Yes, please! I have some EQ IRs that I can't replicate with the on-board EQs. I understand that convolution is expensive so we wouldn't see (initial) support for longer IRs like for reverbs. But why not start with just 768 or even 512 taps?

    Many thanks, bro -- didn't expect a comprehensive collection here! So if you guys who have been mentioned still have the white noise responses (or are able to remake them): Feel free to PM me.


    Edit: In case you still own the wah pedal in question and are ready to try something new: I'd like to test (establish?) a supposedly more elaborate approach to actually sample the frequency responses:

    1. Set your DAW to a project/sample rate of 44.1kHz.
    2. Download the file irSampling.wav.zip from here (section Craft the magical sampling DI signal) and put it into a mono channel strip of your DAW. Configure the channel strip to have no input and set the send level to -12dB (assuming audio interface line out here).
    3. Adjust the input level of your audio interface to be safe from clipping (e.g. check with pink noise set to -6dB sent through the pedal while disengaged/engaged). When in doubt, less gain is more here.

    Play the sampling DI three times through the wah pedal while recording the output into a second mono channel strip:

    1. pedal disengaged
    2. pedal engaged, at heel position
    3. pedal engaged, at toe position

    Truncate all recordings to 3s each (like the sampling DI) and individually export as WAVE (mono) with 24bit dynamic range. Disable normalisation, dithering and other fancy options for the exports (we need them to be raw/unaltered). Name the files accordingly (e.g. Dunlop-Cry_Baby_BB535-Pos_3-off.wav, ...-heel.wav, ...-toe.wav) and zip them together.


    Edit 2, for the DSP heads: In order to derive the frequency response from a white noise processing one has to hassle with windowing, FFT length, averaging and smoothing. In contrast, the impulse responses are actually already frequency responses, just in the time domain. The transformation to frequency domain is then a no-brainer. Never tried this with analogue gear though...

    Ok Guys, as the pipeline for converting white noise responses of real wah pedals to Profiler wah settings seems to be somewhat stuck (combined with the fact that I have upcoming vacation while still in COVID-19 lockdown): I may give it a try -- starting with just one pedal.


    Monkey_Man, do you have any clue of what the most anticipated pending conversion is and who still has the respective white noise responses available?


    Caveat: Although I know how to derive mean spectra of sound samples, calculate/analyse frequency responses and construct FIR filters: Never done this before. Please regard as an experiment.

    Edit: Mean spectrum of the white noise input file attached.

    My personal two cents on the results: I was kind of surprised by the seemingly large impact on the mids, especially of the High Shift parameter. In addition, the pronounced early cut-off of the positive Low Shift was unexpected to me. I then did a coarse implementation of my own naive idea of such an algorithm. Plots attached.

    Epilogue: IMHO ckemper is just a genius, and his DSP Team a rare collection of true wizards. They know the math but they trust their ears not their eyes. So I didn't even test my results -- presumably they're pure crap...