I still think you could save the money and do it by reamping+plugin - but it's purely your decision of course
Posts by Michael_dk
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Second question: have you tried simply flipping the polarity on one of the tracks for the combinations that don't work as well? (I'm sure you have, but thought I'd ask anyway... :-)).
You could also try zooming in on the waveforms really close to verify whether a simple nudge of one of the tracks could help - like if the waveforms cross the zero line at the same time, or if one seems to be a "fixed" amount of time ahead of the other. Who knows, it might happen?
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Of course.
But often the results are sub optimal, less desirable than a dedicated interface.Yeah, I know. But I was thinking that maybe someone without a dedicated interface maybe don't work with recording at a level where it matters much. Of course, I don't know exactly HOW detrimental it would be to the audio. It might not be feasible. Still, worth a try, maybe.
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How about just tracking with a good sound first, then reamp through the two kempers afterwards (and then adjust with a plugin as necessary)?
Also, have you tried to set the two kempers to fixed latency and see if the results are any different?
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I've never tried stuff like this, but wouldn't it be possible to just use the line in of you regular sound card?
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Another way it could be implemented (which may or may not be better - but of course, one does not rule out the other).... Maybe just a search function/custom filter - so I could choose one (or more?) rig tags to search in, and "write" what I am searching for - like "amp type" - search for "marshall", which would show all marshall amps. Maybe with an "and/or" function, so I could show all amps named "Fan" OR "Fender", to get around the problem of implied amp names
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I am finding that the views I can choose between in browse mode do not meet my needs (or ideal of how it should be).
I would LOVE to be able to tag my profiles with some custom tags by which I could sort.
An example: this way I could tag some profiles with "Strat", some with "Synth-like", and some with both these tags. Ideally, I would be able to show only the profiles with BOTH these tags in addition to each single tag.
I could tag with SC for profiles that sound great with single coils, HB for humbuckers, and both tags for profiles that sound great with either. I could then show ALL the profiles that sound great with single coils etc.Basically, i want you to put excel into my kemper - that shouldn't be a problem, should it?
I don't know if there are any way to implement this, given the current format of the KIPR files (I guess there is a fixed number of "fields" for each. But I sure would dig it.
Maybe it could be done by manually writing the tags into the "rig comments" field, separated by a special character, and the kemper could scan these and make a list from which one could check or uncheck, and then show only the chosen selection in browse mode.
...Am I making any sense to anybody at all but myself?
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Yeah, think of the Kemper tone controls as "another layer", so you could theoretically double up on scooping, etc - the kemper has no idea how your tonestack is set on your amp, so the controls show "0". Unless the one who uploads the profile has futzed with them AFTER profiling his/her amp. Also, many amps don't have all the controls anyway (bass, mid, treble, presence), some don't even have ANY of them
I mostly don't touch the KPA EQ controls (I'm still in the initial phases of kempering, though).
Does that answer your question?
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deleted - irrelevant
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Aha! Interesting
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Very nice presentation and layout. Now, if they only sound as good as they look (Playful sarcasm)
Well, we would know if they tried reamping some DI tracks (sarcastic playfulness)
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Suggestion (if you know a bit about phase and polarity). This might not give any usable result, but might be worth a try anyway.
Use SPDIF ins and outs (and set it up correctly - [email protected]), choose a profile you have noticed a "big" difference on.
- Set your kemper to constant latency, record a DI track
- Reamp the DI track three times (three seperate tracks) for the new firmware
- Compare results in pairs, flipping the polarity (=reversing the phase) of one of the pairs each time, to get an idea of inherent variability in the system (i.e., the complete chain). This is your control. The lower the signal becomes when flipping the polarity, the closer the match.- Revert to the old firmware, and repeat the process.
- Now, compare old vs. new, using the same process of flipping the polarity of one track in a pair. Listen for the level drop.
- Compare :
a sum of an "old" track with polarity flipped and "new" track normal
with
a sum of an "old" track with polarity flipped and "old" track normal (and the same for new tracks)Is the sound about the same? What about level drop?
(when I say "old" track, I of course mean reamped with the old firmware).
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The topic seems to be pretty much solved by now. Still I will try to go deeper into the topic, even though I have explained the situation deeper in earlier topics. But terms like 'reluctant', 'ignorant' and 'design flaw' require an answer.
I am aware that there is a need and request for different sampling rates, 44.1, 48 and 96 kHz. However, I have decided to stay with 44.1 kHz because it allows the most calculation power while not sacrifying the sound quality. As a side effect, it fullfills the requirements for the omnipresent audio CD standard.
Matching all sampling rates in a complex audio device is very challenging, because every single process has to be programmed and proven for variable sampling rate. It has to be assured that the formulas work in a way that the sound does not vary with different sampling rates. We have tried to put our manpower in creating a great sounding guitar amp and new features, rather than making our features adaptive to different digital formats that do not increase audio quality.My observation is that in the last 10 years most manufacturer of digital musical instruments or digital guitar amps avoided that challenge as well, due to the same reasons. The only exceptions that I can see are Eventide and Lexicon, as they come from a pure high professional studio background. Nevertheless many other devices provide digital I/O. The following conditions for seamless SPDIF or AES/EBU are mandatory, if no word clock connection is provided:
- The acceptance of at least 44.1 and 48 kHz sampling rate
- The slave ability to take the digital input as a clock source, enabled by a dedicated sample clock source selectorA respective chapter in the manual is mandatory to explain what is going on with master and slave. In detail the following must be well described to avoid system failure:
- Even when you only use the digital audio output on your digital guitar amp or keyboard (which is very likely) you must connect the digital input cable to provide the clock signal to the sending device. The clock source must be external. If this is not documented, a less trained user must assume that by only connecting the digital output to the audio interface is a save situation.
- Alternatively the receiving device must be set to slave.
- It must be avoided that both devices are set to be slave, causing the sample clock to breakdown due to the lack of a master.Earlier discussions about Spdif made me examine the digital audio capabilities and their documentation. As a result I have rarely discovered a device from the music instrument industries (exept pro studio, pro live, audio interfaces) of the last 10 years fulfilling these rules.
Seen from this point the whole MI industry makes Spdif a technical flop. And yes, we are a part of this flop! But people rarely take notice of the big picture, but blame individual manufacturers.As I mentioned before, there is quite a simple solution for this problem, that would render all the above reasons and strict rules unnessesary:
If audio interfaces would provide sample rate converters on their Spdif or AES interfaces, you would simply connect ins and outs as it was analog cables.
You could even save money by not needing a Spdif in connection, when only the output is needed.I assume that the audio interface industry is not aware of this potential need, since they get no requests from their users, since you ask me for a solution, not them :). Also they might think that their users believe that SRC have a bad sound. Truth is: they sound great and the interface manufacturers know about that.
This is a thesis of mine, and probably you would not find a similar call on the web yet. Spread the word!
Ask me if you have further questions....So bottom line, implementing the requested feature would have made my kemper more expensive due to cost of components, more programming, longer testing of firmware updates, higher probablility of failure, longer firmware roll-out times, etc. etc. I understand the request for both higher sample rates and slave ability, and I would have liked those features as well. But I gladly trade that for a more affordable and potentially more stable unit. I would have bought it regardless if it only had analogue connections (like a mic in front of a cabinet), so it's not such a great loss to me.
Thanks for the explanation, mr. CK. If I were you, I would copy it and keep it handy - like the clean sense, I am sure you will have ample opportunity to post it again (the nature of the internet) :-).
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This is G.T.S (Guitarist Tweaker Syndrome):D
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If phasing issues occur on the monitor output, this might be caused by mono summing of modulating effects.
It is a good idea then to set the Source of the Monitor Output to 'Master Left' instead of Mono.The only drawback is that you will only hear one side of a ping pong delay.
Having not used them much - do some delays and reverbs "include" modulation-y stuff?
This is what I place my bet on.
When did the guitar become a stereo instrument??
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Maybe you need an intervention, haha
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In order to do the Reamping it would have to be only git, not git/mod mono, or a DI input.
Assuming you're talking about recording the DI track before getting to the reamping bit: No, the OP records to two mono tracks; one is the Git, the other is the mod mono. This is fine.
If you mean something else, please disregard this post
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Also (and I suspect this is your issue): Read up on Reamping (Maybe specifically "Reamp sense") in the reference manual (not the quick guide/basic manual/getting started). Page 22-25.
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First off, try to eliminate the variables by using the same "signal out" when reamping - i.e., you say you use Git/mono mod on the spdif out during tracking. Try to reamp using the same, and see if it makes a difference. That way you can also see SOME potential level mismatches - i.e., compare the levels of the NEW dry track with the old dry track (provided this is possible to get when reamping through spdif?).
Eidt: I see you record to a mono track, not a stereo track, even though you say the kemper puts out master stereo through the spdif. This may not be optimal.
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+1!
Maybe a subforum with a thread for each guitar or something, to keep stuff organized?