Posts by lightbox

    1. Connection:

    Computer MIDI Out -> Strymon MIDI In ->Strymon MIDI Out -> Kemper MIDI In

    2. Connection:

    Kemper MIDI Thru -> Computer MIDI In


    Please note your Strymon MUST be set to MIDI TH = MERGE!!!

    Also, both units must be set to different MIDI channels, obviously.


    This way you can send MIDI commands to your Profiler and at the same time have a full MIDI connection with 2-way communication for your Strymon pedal and Nixie. So basically the trick is to have the Profiler last in the chain. Downside without additional MIDI ports on your computer: You can't receive MIDI data coming from (and generated on) your Profiler.

    I'm afraid you can't switch the input LED indicators to digital inputs. Not even the Clarett 8pre has this option.

    The Focusrite Liquid Saffire 56 had this option but looks like they discontinued this feature in newer interfaces.

    You should be able to see the digital input levels in Focusrite Control though.

    Warum steht der Monitor Out auf -11,8dB?

    Dreh mal deinen Powered Kabinet ganz runter, dann den Monitor Output ordentlich hoch ... gib Schmackes ... und dann vorsichtig den Powered Kabinet hochdrehen. Da sollte jetzt wesentlich mehr Saft rauskommen.

    Your aliasing test is valid, and you can make clips of it.

    I made a demo (and explanation) video:

    How do we solve this discrepancy?

    We probably can't solve the issue (with the current hardware) and it's probably not even necessary. But still, it can be heard by a trained ear when he/she can compare. We can hear it if someone uses the Profiler to run other "instruments" with a simpler tone/frequency structure through it.


    Anyway ... all I wanted to say and prove is that higher sample rates can make sense. I will still continue to play and enjoy my Profiler although it's not perfect. ;)

    Why would you do this?

    Because I can! And on a more serious note ... the topic here is higher sample rates and why and when they CAN make sense. If you can't contribute, why answer?

    Plug in and play it. It's awesome.

    Don't worry about me, I do this a lot and since a much longer time than you ;) But still I am an audio engineer, not a guitar player with limited technical knowledge ... and if I see false claims, I will speak out (and try to explain). Thanks

    Aliasing can only be suppressed by a certain degree.

    It depends in the frequencies you feed in, the amount of distortion that you apply and the factor of oversampling.

    Aliasing can only be eliminated by an infinite oversampling.

    Be aware that typical guitar distortions produce much more saturation than typical Fabfilter settings.

    Ok, that's what you say ... and what definitely isn't quite right.

    • Fire up your Profiler.
    • Now send a looping sine sweep through your Profiler (for example a range from 5kHz - 22kHz in 5 seconds). Ideally via S/PDIF to rule out all potential AD/DA effects.
    • Switch off ALL slots, including the stack!
    • You'll hear the proper sine sweep with no issues.
    • But now, in one slot of your choice, add a Studio EQ and boost 6.4kHz by 5dB or more (just example figures)
    • Sit back, grab a glass of good red wine and enjoy the birds chirping.
    • Now this being out of our way, switch off the Studio EQ and load an amp (and cab if you want) with Gain at 2.0. Still pretty low gain, right? But more than enough to introduce you to the next family of chirping birds.
    • Last but not least, just for the fun of it and with half a bottle of expensive red wine in our head ... let's load BM - Magmaton clean from the Kemper Factory Content. Keep the Sine Sweep going and experience an instant night in the Amazon jungle. :)

    All this being said (and I can make a demo video if necessary) ... if I run Cubase in 44.1kHz and use e.g. Fabfilter Pro-Q3 or for more saturation e.g. the Fabfilter Saturn 2, it's not a problem to create chirping birds. But even the simplest option to increase filter quality will dramatically reduce this wildlife ... up to the point where it's pretty much completely gone with highest quality settings.

    In other words, the Profiler is just as good regarding aliasing as the cheapest, simplest plugin out there.

    Feel free to try it with amp block and all other slots disabled and only one slot with Kemper Drive, Kemper Fuzz or any other "distortion" module of your choice. All of them chirp like crazy.


    And hey ... you probably can't do anything about it because everything you could do (in the current Profiler) would take valuable time (increased latency). Why? Because you can't run the entire internal processing at 192kHz (let alone the 704 khz or even 1.4 Mhz you mentioned) and then properly filter at the end of the chain(s).

    Well, I don't accuse you of lying but I certainly know that there's something fundamentally wrong about this. And I'd rather not read rants about "bad" plugin developers as long as the Profiler doesn't do any better in this particular regard. ;)

    This is how we do it.

    Christoph, can you kindly explain me then why your Profiler (when I feed it a looping sine sweep) chirps like a cage full of grown up exotic birds once I engage any slot with at least a hint of saturation? I mean, seriously, the Kemper Profiler's modules aren't ANY better than e.g. any Fabfilter plugin in its simplest zero latency mode. To be honest with you, to me it sounds as if there's absolutely no filtering going on. Might that be to avoid extra latency? hint hint, cough cough ;)


    And just for the record: Running the Profiler in 96kHz mode doesn't change a thing ... which doesn't surprise me at all because the output sample rate doesn't dictate the internal processing.

    We obviously agree that capturing recordings does not require higher sample rates, but sometimes processing does.

    Yes, we agree on that.


    But we don't agree on that. ;)

    First of all we need to understand the difference between system roundtrip latency (defined by audio interface driver design and audio interface buffer size) and channel/plugin latency (caused by plugin processing).


    Examples:

    1. We run a Cubase project with a single stereo audio track at 192kHz and buffer size 512

    Now we can use a Fabfilter Pro-Q 3 equalizer in zero latency mode. We don't have to use heavy oversampling and linear phase filtering. Channel/Plugin latency = 0ms


    2. Now we run a Cubase project with a single stereo audio track at 44.1kHz and buffer size 512

    Here we want to make sure we use linear phase mode at its max setting, just for the sake of an extreme example.

    Booom, we now have a channel/plugin latency of almost 1400ms ... still running fine on the above buffer size setting but obviously compensated by Cubase (all channels get delayed by 1400ms). And I still have quite some processing power left before I would need to increase the buffer size.


    3. As I wrote before, in post production it doesn't matter so much but it's something you'll immediately notice just by the delay when you hit the space bar to play the project and have to wait 1.4 seconds until the playback actually starts, haha.


    Result:

    System roundtrip latency is one thing, channel/plugin latency is another. In a 192kHz project you can run multiple stages of saturation, dynamics, EQ etc. e.g. in zero latency mode and don't care too much about the "mess" it creates beyond 22kHz. Only at the very end of the project when you create the audio deliverables (e.g. 44.1kHz), you should properly low pass the material (with a good filter), obviously.

    In live production you really don't want to deal with latency issues. In (PAL) broadcast for example, 40ms already equals 1 frame and I can easily see how lip sync is off. It's a very costly endeavour to compensate for such a delay (or more) by delaying all video sources by the same amount.

    ... if its that big of a deal, why not just put your fuzz in front of the KP and do your thing?

    That's what I (and probably others) do, when we want to. :)

    As I said somewhere in the beginning of this thread, I can get some usable (nice) tones out of these Kemper Fuzz parameters. For me, personally, it's much better than the Kemper Drive.

    But still, this is a Beta so we can discuss the shortcomings / differences to real world Fuzz pedals.

    Pete Thorne was not raving too much about the amazing transition from fat to glassy, but emphasizing and suggesting that it's the way to go, to achieve this glassy sound at all. He didn't mention that there is numerous other and easier ways.

    I hope you understand my point.

    I think I understand your point ... and I understand the difficulties to replicate this in the digital domain. :)

    But regarding Pete Thorn's video (I didn't watch it until just now) ... I can't find any reason to call it BS. At the beginning he says that he's making a review of this fuzz pedal and decided to make a separate video for one specific aspect of this fuzz pedal. He even mentions Hendrix which sets the context pretty well, imho. And it's never a bad idea to remind "regular" guitarists of the volume knob either, haha.

    And at 3:21 he even talks about this amazing range from glassy cleans to full-on fuzz. I've cued it accordingly:

    So I really can't see anything wrong in his short and "to the point" video. If someone is into Jimi Hendrix, SRV, Jeff Beck kind of tones, they will have a fuzz on their pedalboard and there's no reason to add other pedals and tap dance on the pedalboard to get these beautiful "glassy" clean tones. They can just use this hidden secret called "pinky morphing™". :D

    You say it's easier to take an analog fuzz and turn down the volume pot, than taking a KPAs treble booster?

    Anytime, anywhere. As others have pointed out, it's amazing what you can do only with your guitar volume knob and the way you pick. No foot pedal, no stomp switch, no rig switch will EVER come close to the control with your fingers right on the guitar. The joy when you can go from glassy cleans to full-on fuzz and everywhere in between right on your guitar, no matter where you are on stage (or in the crowd) can't be beat by any pedalboard trick, EVER.

    You tend to make fun of guitar players, even calling proven "tricks" of seasoned guitar players "BS". Come on, Christoph, please get down and touch earth with your boots. I won't tell you how to play keyboards and you please don't tell guitarists how to play guitar. Thanks ;)

    higher sample rates can be very relevant for you. And audibly superior.

    Yes, but here's the little flaw in my example with a sports playing field. :D

    In sports, most attention will be on what's going on near the goal, right at the end of the playing field.

    Our ears though are way more interested in the "midfield" action.

    So it's typically not as dramatically "bad" and often not even audible in a negative way in a full music mix.

    It's very easy to make these terrible effects audible by using e.g. pure sine sweeps through a distortion (saturation) stage.

    But these are "lab conditions", not real world music.


    Bottomline:

    People with trained ears can hear the difference when they can compare.

    Is it even possible to produce good music in a digital environment without heavy oversampling in each plugin used? Yes, of course. Just like it was possible to create good music back in the day of vinyl when there were (are) physical limitations to e.g. bass response and bass transients.

    Am I close?

    Yes, you are very very close :)


    Now when you think a little further from that point:

    It's not so much about the "capturing" only. It's also about audio processing once you're in the digital domain.

    To use my previous layman's example:

    Even if all players agree to stay inside the playing field at all times ... once the game starts and the players want to make the game interesting, they will likely ignore the rule and e.g. hammer the ball towards the goal. Damn, that shot went over the goal to the stands behind the goal. But hey, it looked great and made the game exciting.


    In audio terms, you often want to create e.g. "saturation" to color the sound, make it more exciting. This automatically creates frequencies beyond the "agreed playing field" ... and you need to take measures to prevent that from happening.


    One way to do that is to enlargen the entire venue dramatically (aka 96 or 192kHz) and only take care of the remaining unwanted effects at the very end of the digital processing chain .... or .... make sure that each individual "saturation stage" plays by the rules and stays within the agreed limits. The latter requires time and calculation power which results in added latency.

    This added latency doesn't matter in post production but can be a serious issue in live production where latency must be kept to a minimum. :)


    Was this helpful as "the second stage" of explanation?

    I wonder, if it wouldn't be useful to include a "Bias" parameter ... and a specific gating effect like requested above could easily be implemented there to recreate the sound of "wrongly" biased transistors.

    Yep, that would be awesome but so far no one of the Kemper team has chimed in on that.

    One more thing I noticed in my (limited) tests so far: The Kemper Fuzz generally doesn't quite clean up as nicely with my guitar volume knob as with my real fuzz pedals.

    If 192 is ‘better’ than 96 and 96 ‘better’ than 48.....why then has the lowly vinyl record been resurgent with listeners?

    Just for fun, let me try to give you a layman's example/explanation:

    Digital recording adds a rigid wall somewhere behind the end of your playing field. At 48kHz the wall is "24 meters" from the opposite baseline. Say you're right in the middle of the playing field and you throw a ball at the wall. Obviously it will bounce back from the wall towards you.


    In analog audio, there is no wall. Throw your ball the same you did before ... the ball won't bounce back, it will just keep going away from you until its energy is gone. It will not come back and disturb what's going on on the playing field


    Now back to digital audio ... At 192kHz the wall is 96 meters from the opposite baseline. Throw the ball towards the distant wall with the same energy like before. There's a good chance the ball doesn't have enough energy to bounce (or roll) all the way back to you.


    Now finally imagine all players on the field constantly throwing balls at the wall. You can certainly imagine that the "reflected" balls from the "48kHz playing field" coming back from the wall will have a negative impact on what's going on on the field. They add unwanted chaos, especially in the upper part of the playing field.


    Hope it's at least funny to read this but I think it explains the issues with the sample rate creating a rigid wall inevitably reflecting everything that wants to go beyond the wall.

    Use the same wall outlet for all the equipment that is connected (computer, Profiler, powered cabs/speakers, ...). The buzz might be caused by a ground loop. If all equipment is hooked up to the same wall outlet and the buzz isn't gone yet:


    1. Something has died and caused this to suddenly happen.

    2. Try the Ground Lifts on your stage (OUTPUT menu page 9/9). Just make sure you never lift all at the same time!!!

    Phasing issues should be solved by the DAWs latency compensation. Doesn't it work?

    Read again, Christoph :) I was talking about live sound (e.g. broadcast or live shows), not studio work. I was just adding another real world application where higher sample rates like 96kHz do actually make sense. Maybe you got confused from me using the term "plugins" (in quotes).


    My comment wasn't related to the Kemper Profiler by any means.


    PS: Seagate harddisks sound the best IMHO.

    Hahaha, that was hilarious. Thanks for the good laugh and good luck with your tests with Seagate. As you certainly know: "Sie Geht, Sie Geht Nicht" ;)

    There is one reason to use high sample rates: When plug-ins on your DAW sound better with that.

    And another reason would be in situations where latency is crucial (live sound e.g. in broadcast or even concerts). You just don't want a bunch of oversampling linear phase "plugins" in your signal chain considerably adding to the overall latency (and potentially even severe phasing issues between "parallel" busses. So you rather use 96kHz (if you can) instead of 48kHz on your desk and keep most of the nasty stuff happening in low latency "plugins" well above the audible spectrum ... and have the DA stage take care of that.