an updated 2023 hardware

  • As opposed to the guys like me ; “all the gear and no idea” or as we say around these parts “fur coat and nae knickers “ 🤣

    I remember lining up for an MTB race years ago. This dude shows up looking all cool with some of the nicest equipment you could want. Didn't really talk to anyone. We couldn't.....because after the first 100 yards we were out of earshot ahead of him. :)


    I did get my butt handed to me by a guy with gear so clean and new, it was disgusting. But man was he fast. Happened to be a bald guy I nicknamed Mr. Clean......hate that guy. :)

    “Without music, life would be a mistake.” - Friedrich Nietzsche

  • Yup. I've been asked about my fast bike. Standard response, 'there are no fast bikes, only fast riders'


    Larry Carlton was told his 335 sounded fantastic. He put it on the guitar stand and said "how's it sound now"?

  • I wouldn't like to be misunderstood and no offense to OP... As i wrote, i was in the exact same thought...

    I don't want to open a battle between traditional combo vs multi-fx/digital world or one with bad players with excellent devices vs virtuosos with bad equipment... I assume there's a geek part in every one of us as we own a Kemper....We moved forward the conservative rig... We are curious and like to explore new territories....8)


    I just wanted to say that my Stage fills all specifications i was looking for ;

    - Amp sound/feeling <3

    - All-in one solution

    - Versatile

    - Ergonomic

    - Simple


    All updates are just fantastic, and don't change things above.... I think if a new Kemper product is launched we won't be lost too ;)....

    But i'm more on the "what for ?" :?::?:

    I don't understand the "i want more processing power" for example. I'm not an IT ingineer and i don't care about how all that stuff works,

    I'm just a user. IMO, if you miss something, just tell it and the team will do its job to find the solution.... I just don't catch what this kind of improvment would change ?! It's more concret to know that you want ; 6 delays in parallel, etc....


    Now, Kemper has to live and free updates could be a strange business model :/8o

    On the other hand, They can signify they're on another business model with an "eternal" plateform that evolves all over the years, meaning ; "come and buy our devices it will last forever" (or for a long long time), don't be afraid that it becomes obsolete sooner ; it won't"


    They could link the best of the two worlds (traditional rig/digital world)....And for the moment this is the the route they take :thumbup:<3

  • 96khz is absolutely useless for a piece of equipment like the kemper. There is NO useable guitar info at the nyquist frequency (48khz).


    Heck there’s very few times you’d even want to hear any guitar above 10khz let alone the nyquist for 44,1 (meaning 22khz) I have a high cut at 7khz on my kemper.

    There is so much more those cpu cycles could be used for, than producing white noise for dogs.

    And in the end, the love you take is equal to the love you make.

  • The argument is that it somehow affects frequencies below 22.05, even if the hard cutoff is at that frequency. Not that I agree with that necessarily. And I think pretty much everyone would agree that there is no useful info above 22khz when mic'ing a guitar cab. Some discussion here with CK chiming in on his views: profiling sample rate?

  • 96khz is absolutely useless for a piece of equipment like the kemper. There is NO useable guitar info at the nyquist frequency (48khz).


    Heck there’s very few times you’d even want to hear any guitar above 10khz let alone the nyquist for 44,1 (meaning 22khz) I have a high cut at 7khz on my kemper.

    There is so much more those cpu cycles could be used for, than producing white noise for dogs.

    In digital audio the kHz stamp is describing hos fast the cycle updates, not the frequencies included in the audio content.
    So together with the bitrate it's setting the granulation of the audio. Higher resolution just means your audio is smoother, closer to analog audio if you will.

  • In digital audio the kHz stamp is describing hos fast the cycle updates, not the frequencies included in the audio content.
    So together with the bitrate it's setting the granulation of the audio. Higher resolution just means your audio is smoother, closer to analog audio if you will.

    Not true - if I understand you correctly.


    In digtal audio, the sample rate in effect ONLY describes the highest frequency included in the audio content. I.e. the Nyquist frequency (half the sample rate) determines the highest audio frequency that can be unambiguously derived from the digital information.

    (I'm not too sure about "sample rate" as used for describing MP3s, but that's not what we're discussing).


    As I understand it, the main reason some plugins work internally with higher sample rates is in order to reliably handle harmonics introduced by the plugin itself, which might lie above the nyquist frequency for the given DAW session.


    Similarly, Bit depth determines the noise floor only.

  • Utilizo poco los innumerables recursos del kemper. El 70% del tiempo lo utilizo con mi guitarra acústica y tres o cuatro presets que he personalizado con los sonidos que yo quiero. Pero cuando quiero experimentar con la guitarra eléctrica, sé que dispongo de lo necesario, de todo lo que se me pueda ocurrir. Puedo utilizar un preset de fábrica o puedo definir durante horas el sonido que busco.
    No siento que sea obligatorio conocer su extensa gama de sonidos. Para mi es suficiente con saber que kemper me permite encontrar el mío.
    Respecto al nuevo firmware:
    ¿A alguien le molestan las actualizaciones?. Leo comentarios que parecen poner en duda determinadas mejoras.
    Pues bien: no es obligatorio instalarlas. Puedes continuar con tu Kemper tal como está, No hay problema.
    Por otro lado, si eres un purista del sonido analógico, me parece una decisión respetable, pero entonces ¿qué haces aquí y por qué te has comprado un kemper?.
    Yo personalmente me siento agradecido con kemper y su política de actualizaciones que suponen mejoras en sus productos y además, gratuitas.
    No sólo mantiene mi kemper en acción, sino que todo el aprendizaje que voy acumulando no lo tiro a la basura al comprar un nuevo dispositivo. Eso hace que los usuarios de kemper, incluso los que lo van conociendo muy despacio, acaben manejándolo con soltura.
    Y de eso se trata,... de sentirte cómodo, del progreso, de avanzar, de mejorar,.... tanto con tu guitarra como con las herramientas que utilizas con ella.
    Acumular experiencia y sentirte más a gusto haciéndolo mejor, sin tener que volver a empezar con una nueva pedalera porque la tuya haya quedado obsoleta y sin sentir que en vez de dedicarte a tocar tienes que dedicarte continuamente a experimentar y aprender desde cero.
    Kemper es un viaje de largo recorrido y un compañero con el que puedes entenderte sin que te abandone a medio camino.
    Ojalá todas las empresas de tecnología avanzada fuesen tan consideradas con sus usuarios.

    .

    .

    I make little use of the countless resources of the kemper. 70% of the time I use it with my acoustic guitar and three or four presets that I have customized with the sounds that I want. But when I want to experiment with the electric guitar, I know that I have what I need, everything I can think of. I can use a factory preset or I can define for hours the sound I want.

    I don't feel like it's mandatory to know its wide range of sounds. For me it is enough to know that kemper allows me to find mine.

    Regarding the new firmware:

    Does anyone bother with updates? I read comments that seem to question certain improvements.

    Well, it is not mandatory to install them. You can continue with your Kemper as it is, no problem.

    On the other hand, if you are a stickler for analogue sound, it seems like a respectable decision to me, but then what are you doing here and why did you buy a kemper?

    I personally feel grateful to kemper and its policy of updates that imply improvements in its products and, moreover, free of charge.

    Not only does it keep my kemper in action, but all the learning I accumulate I don't throw away when I buy a new device. This means that kemper users, even those who are getting to know it very slowly, end up handling it with ease.

    And that's what it's all about... feeling comfortable, progressing, advancing, improving... both with your guitar and with the tools you use with it.

    Accumulate experience and feel more comfortable doing it better, without having to start over with a new pedalboard because yours has become obsolete and without feeling that instead of dedicating yourself to playing you have to dedicate yourself continuously to experiment and learn from scratch.

    Kemper is a long-distance journey and a partner with whom you can understand each other without leaving you halfway.

    I wish all high-tech companies were as considerate of their users.

    Constantino Ons - Técnico Informático y músico aficionado. Computer Technician and amateur musician.

    Edited 4 times, last by t ons m ().

  • Here’s my take, sample rate determines how frequently a sample is taken, a sample being a measure of voltage. If frequencies higher than half the sample rate are processed, aliasing occurs. The aim is to capture frequencies to 20khz with an anti aliasing filter after that point hence sample rate of 44.1khz. The bit rate determines how closely to the measured voltage a sample might be represented in the digital domain. 16 bit is accurate to 1/65536 24 bit to 1/16777216. This gives theoretical dynamic ranges of 96 and 124 dB respectively.

    A brace of Suhrs, a Charvel, a toaster, an Apollo twin, a Mac, and a DXR10

  • Here’s my take, sample rate determines how frequently a sample is taken, a sample being a measure of voltage. If frequencies higher than half the sample rate are processed, aliasing occurs. The aim is to capture frequencies to 20khz with an anti aliasing filter after that point hence sample rate of 44.1khz. The bit rate determines how closely to the measured voltage a sample might be represented in the digital domain. 16 bit is accurate to 1/65536 24 bit to 1/16777216. This gives theoretical dynamic ranges of 96 and 124 dB respectively.

    This is exactly my take as well :)

    With the dynamic range being loudest signal to noise floor from conversion process

  • One way to look at sample rate is this:

    - Nyquist rate is double the frequency you want to resolve. So 20 kHz needs a minimum rate of 40 kHz.

    - At 44.1kHz an 8 kHz signal is made up of 44.1 / 8 samples = 5. Can you draw a complicated waveform with only 5 points?

    - At 96 kHz an 8 kHz signal is 12 samples. Much better than 5 for drawing a waveform.


    More samples is always better.


    From the link posted above: "The Profiler would not sound better at higher sample rates, because we use higher sample rates internally at different places (sometimes in MHz numbers as well) in the signal chain, whereever it's required." - CK


    I find it hard to argue with CK, he is clearly much smarter than I am. But I would prefer more samples if I was looking for perfection. Luckily for me, I am old and my hearing is trash so I could not tell 22 kHz from 96 kHz sample rates probably :wacko:

  • This is a common and understandable error in understanding (that is absolutely not meant as a dig at you, even if it may sound like that).

    If you're interested and have the time, I suggest to watch this video. He is really great at explaining and showing what actually happens :)


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  • This is a common and understandable error in understanding (that is absolutely not meant as a dig at you, even if it may sound like that).

    If you're interested and have the time, I suggest to watch this video. He is really great at explaining and showing what actually happens :)


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    he’s brilliant. It took me ages to wrap my head around that stuff 20 odd years ago. Nice one

    A brace of Suhrs, a Charvel, a toaster, an Apollo twin, a Mac, and a DXR10

  • This is a common and understandable error in understanding (that is absolutely not meant as a dig at you, even if it may sound like that).

    If you're interested and have the time, I suggest to watch this video. He is really great at explaining and showing what actually happens :)

    This is an amazing video. I used to have the sig gen and spectrum analyzer on my desk at work. Our Tek scopes never lasted long since we did HV pulse testing, so the analog scopes were all dead. And half our digital Tek scopes had issues ^^


    At the 6 minute mark he starts to explain that bandwidth limiting the input signal resolves the sine wave from a reduced set of samples. But he never quite explains how. Common sense would say a matching output filter? Oversampling, etc?


    Very cool video explaining sample depth, dithering, etc.

  • At the 6 minute mark he starts to explain that bandwidth limiting the input signal resolves the sine wave from a reduced set of samples. But he never quite explains how. Common sense would say a matching output filter? Oversampling, etc?

    I don't know. Magic, I think :D


    Random google search and click found me this page, which seems to explain some of it. Too technical for me to understand, at least by one quick read-through.

    https://www.dspguide.com/ch3/3.htm

  • At the 6 minute mark he starts to explain that bandwidth limiting the input signal resolves the sine wave from a reduced set of samples. But he never quite explains how. Common sense would say a matching output filter? Oversampling, etc?

    I don't think this the correct interpretation. The sampling theorem states that a properly sampled input signal results in an output sample set that is unique to only one possible input signal. Proper bandwidth limiting guarantees this uniqueness. Otherwise, aliasing occurs and the unique correspondence between the output and input is lost.


    In the case of aliasing of audio, frequency content above the sampling frequency is translated to lower frequencies resulting in distortion.

  • I don't think this the correct interpretation. The sampling theorem states that a properly sampled input signal results in an output sample set that is unique to only one possible input signal. Proper bandwidth limiting guarantees this uniqueness. Otherwise, aliasing occurs and the unique correspondence between the output and input is lost.


    In the case of aliasing of audio, frequency content above the sampling frequency is translated to lower frequencies resulting in distortion.

    Input filtering saves you from aliasing. I am talking about how do you get a perfect sine wave from a DAC when its output is only 5 points or less?



    How do you get a perfect sine (RED) from the points taken above? If the DAC had no filtering or oversampling it would be a blocky mess of garbage. I would have preferred he spent a minute on this in the video.

  • Input filtering saves you from aliasing. I am talking about how do you get a perfect sine wave from a DAC when its output is only 5 points or less?



    How do you get a perfect sine (RED) from the points taken above? If the DAC had no filtering or oversampling it would be a blocky mess of garbage. I would have preferred he spent a minute on this in the video.

    Sorry, I didn't get that from your previous question. Here is an explanation:

    Digital-to-Analog Conversion (dspguide.com)


    Look to the explanation around figure 6. The basic technique is a zero order hold coupled into a lowpass filter. There are many types of DACs, but this is one of the simplest types for audio and is used quite a lot. Proper analog filtering on the input and the output is usually required.