Posts by lbieber

    A great guitarist & singer and another smiling KPA user ; really dig her style. Texans : she's your neighbor !


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    I love the vibe of that song. I could have done without the autotune.

    Speaking of Ike. Great song...great guitar tone

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    So,

    Haven't used the Kemper in a long while - plugged into my Gemini 1x12 powered FRFR and my old settings seemed to be clipping the input of the speaker.

    Is there a rule of thumb for the sweet spot for monitor line out with current OS to a powered cab?

    I know I can noodle with it but thought I would ask.

    All good answers above.


    How could your old setup, that used to work, not work now? That is a different question than the one you asked. I would be more curious about that.


    There are too many potential variables regarding gain staging to provide a single answer so noodle with it. IMO, just turn some knobs until it sounds good and get on with it. :)

    I have used my powered Toaster with guitar cab(several different types 1x12, 2x12,4x12) live many times for many different types of music. I never experienced any issues that I couldn't solve during a gig. As alluded to, why do you think it changed from rehearsal to the live gig? It could only be the room or the PA, right? If something was missing during a live gig while using your tube amp you would have adjusted knobs, correct? Did you try that with the Kemper? IME, the powered Kemper is capable of putting out punishing lows and thump.


    I switched to using a PA(FRFR) monitor instead of guitar cabs because I feel it gives me more consistent sound on stage compared to FOH. I don't have any quick fix to offer other than adjust controls in real time during the gig. Good luck.

    I recently visited a drummer who did over 60 live tracks with me.

    He gave me this Alvarez 5056 with case. It's almost 50 years old and sounds like a Martin D-35.

    These guitars were built to compete with Martin and they got really damn close. 8)


    That's a pretty one! I have 1980 Alvarez that is much plainer, but it sounds great.

    Jeez fellas. ..take a deep breath.


    Yes, there was/is a lot of hype. No reason to be dissapointed if expectations are realistic.


    I'm interested in the USB audio. I won't attempt to use it until the apparent bugs are resolved. I don't have an interest in being a beta tester for this, but thanks in advance to those of you who do.

    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.

    Haven't you opened a same thread in the other section of the board?
    There,someone has posted a video that shows what to use for the simulation: compressor, EQ, fuzz and slow attack.
    Of course touch matters a lot, tone is also generated by the player and Dweezil is a good musician.
    :)

    I don't believe that the posted video effect chain is the same as the one in the Dweezil video. It doesn't sound at all the same to my ear.

    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 applaud you for taking the time to reply in this detail to help someone. Very interesting. I went back and listened to the track and for me it would take a bit to sort it out. I think it could be done pretty well with 3 parts although you can hear the main part is quingetripletracked. Without someone doing the work for you, it would take a bit of experimenting to hear if it sounds best to jump to the next harmony and let the FX fill in the rest or let the FX fill around what you play. From my experience with harmonizers (using them since early eventide and pre DHP 33 units) Even though you have the theory nailed, it still comes down to perception and what sounds best, choosing what you play and what you want the machine to play. I think this solo could be 3 separate slots to be as good as Boston themselves ever played it.

    I only took a quick listen and made one proposal that I think would work. There could be quite a few ways to solve this.


    The biggest part of this, even more than the harmony, would be a killer mid forward Marshall tone.

    If you are looking for something specific they have, then yes, they are worth it. I mostly use Mike Britts and TJ profiles with a few others in there, but they are many great ones right on the Kemper. That said, they are worth the price in my opinion. I got really lucky last round and got TJ Everything pack for 30 Euro, so I have more than I will ever need.

    Same story here. Got the TJ stuff really cheap. Britt profiles are fatter for me and TJ, while good, is a bit on the thin side in many cases. Britt profiles sit better in a loud band situation IMO.

    Not sure I agree with what is presented in the video, but... I listened to the original recording again and found what I think is the way that I would approach this riff. I hear the body of the riff as three parts.


    First part, play it starting C#3(5th string 4th fret) with no harmony. You could use a doubler, but I wouldn't as I don't like it. That's just my opinion

    Second part, play it starting on C#4(3rd string 4th fret) with the low harmony enabled only. This allows you to do the string bends and the lower harmony sounds better to my ear.

    Third part, play it starting on C#4 with low harmony plus a third above


    I attached the settings I would use below. I would setup the second part with interval 2 off - balance all the way counterclockwise. Then the third part would be as shown in the attached image. I would set this up in performance mode with adjacent slots, but there are other ways. You have to play it clean to avoid artifacts related to the harmonies. It needs clean notes to work accurately or you get glitchy garbage.

    I am guessing you are referring to the melodic figure that starts on C#(5th string 4th fret)? It is indeed in the key of Emajor. The figure is harmonized an octave above and then with a 3rd above that.


    One way is harmonic pitch set to Emajor with interval 1 set at 1+Oct and interval 2 at +10. Set the balance and the mix for what sounds best to you. I find the tracking is best picked closer to the neck than the bridge.


    The final result will be a compromise because the highest line(+10) is articulated with bends while the lowest line is not. In theory it will work, in practice the result is less than stellar and I would not perform it this way. It sounds very mechanical and hokey. I would probably just play the key parts of the lines without the harmonizer because of this.