Transpose setting per rig, different bitrates for SPDIF available... Awesome update, excellent job!!!!
5.5.0.13341 Public Beta discussion
- sambrox
- Thread is marked as Resolved.
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I haven't tried the beta because of live shows. Does transpose still exist as a stomp, or does it automatically take that stomp and move the effect to rig?
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I haven't tried the beta because of live shows. Does transpose still exist as a stomp, or does it automatically take that stomp and move the effect to rig?
Yes it does, you can still use it as a stomp or if you want to save a block, you can use it from the Rig menu.
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To add to this, the main reason I use 96KHz, is for the smaller samples/time slices. The smaller the slices/samples are, the faster they can be processed throughout the entire sound chain, that includes the CPU, RAM, and the DAW plugins - they are literally processed twice as fast as 44.1KHz - from the input of the AI, through the DAW and VST plugins, and back out the AI to the monitors. This is extremely helpful for monitoring VST effects in real-time, in that, there is very low latency round-trip.
The only drawback to running higher sample rates, is larger file sizes, and CPU overhead. PC's these days have no problems running 96KHz.I am sorry, but the samples in 96kHz aren't "smaller" - there are just twice as much samples. And no: they can not be processed faster throughout the chain. It simply leads to more processing load.
You are probably experiencing a lower latency when using the same buffer size with higher sample rate - because double samplerate means half the time in milliseconds - this can also be simply achieved by using half the buffer size with 44.1/48 kHz. Just without the need of useless waste of processing power. I admit that a modern PC has no trouble processing a 96kHz chain.
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I'm glad everyone is concerned about my DAW hardware, when recording at 96KHz, though 8 CPU cores at 4.0GHz, 16Gb of RAM at 2400 speeds and a 10Tb drive, can't even tell I'm working on music. Seriously, it's barely seeing over 40%.
It does seem like the people who are the most cautious or anti, have never actually tried the higher rates.
If I can tell the difference between 44.1Khz, and 96KHz on my equipment, that must mean I'm using a sub-par audio interface, or I have some super-mutant ears. There is no other possible answer (?).
I don't agree with all of the conclusions asserted here. I will continue to use 96KHz. Thanks to all for the concern.
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If I can tell the difference between 44.1Khz, and 96KHz on my equipment, that must mean I'm using a sub-par audio interface, or I have some super-mutant ears. There is no other possible answer (?).
... or it could mean that the performance sweet spot of your interface is closer to 96kHz than 44.1.
I mentioned this in the bandwidth thread 'cause it seems to be continually overlooked in these discussions.
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We all read the arguments against committing to higher res files, but a lot of great, smart Grammy-winning producers and engineers most definitely record at 96. Point is, placebo or not, now when you carry your Kemper in to a studio, that will no longer be a discussion. You’ll set it to what they’re using and make music.
I am not certain that everyone understood my answer (at least).
My contention is that internal to the kemper, processing at 48 kHz or 96 kHz will make no difference to the sound quality all by itself.
The note from timo above is spot on. With the same buffer size, doubling the sample rate will cut latency in half for internal processing.
When recording, the higher sample rate is needed because the algorithms that work on the data tend to lop off stuff when things get expanded in processing. 96 kHz recordings are more immune to this problem than 48 kHz recordings are. This is the reason that pretty much all people who record use 96 kHz sample rates.
Note, that internally, Kemper has to be careful not to lose information in their own calculations as well; however, embedded coders are particularly careful about these things (PC coders .... not so much). Additionally, the Kemper latency paths will all be matched to keep phase coherency going ..... ie, you don't want to take a signal, process it (there by delaying it), then mixing it back into the original signal without also delaying the original signal. If you do this, you will mush up the sound (like adding a chorus to something).
The KPA sounds pretty clean, so I am sure I have no need to give any advice to CK about this problem
Finally, it is possible to get higher quality sound from 96 kHz internal to an embedded device since you can do 2 times the processing and still keep the latency the same. As stated, this would require 2 times the processing power though. Engineering is funny that way. You never get something for nothing.
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This discussion only goes about khz and buffers.
That‘s my impression.
Shouldn‘t it go about the newest update? About what we need and what we have?
For example more reverbs, effects like doubler or sustainer?
Sorry for complaining! -
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96khz!!! Awesome feature !!! It's Night and Day
Thanks for the upgrade it's like having a KAP II -
As of this moment right now, the
5.5.0.13341 Public Beta
Has 80 likes, more than any other firmware released previously. Sure there are more users these days, along with more units sold, but how many actually take the time to 'like' Beta firmware? I only do if it's exceptional, which this one was for me.Thanks again to the KPA team for the new S/PDIF clocks.
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There identical except that I changed the project sample rate from 44 to 48 and removed the guitar parts. Can you hear any difference? Is one more clearer and more detailed than the other?
are you seriously posting a sample rate comparison on soundcloud also know as one of the worse sounding file streaming services available? the stream at 128 kbps and rumours are that they recently changed to a 64 kbps Opus codec for playback. please makes the files available in a linear format instead.
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So one question and forgive me if I am way off on this... but on the direct out (send) can you send a stereo signal out now (as in L+R to a strymon or anything else) that takes two inputs?
Cheers
So, anyone have a chance to try this? Was thinking of using a Y cable coming out of the direct send into an effects unit (using the L+R as opposed to just the Left).
By the way, really digging the stereo output to monitors!
Cheers,
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So, anyone have a chance to try this? Was thinking of using a Y cable coming out of the direct send into an effects unit (using the L+R as opposed to just the Left).
By the way, really digging the stereo output to monitors!Cheers,
the direct output is a TS output-how would a software update be able to turn that into a stereo output?
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are you seriously posting a sample rate comparison on soundcloud also know as one of the worse sounding file streaming services available? the stream at 128 kbps and rumours are that they recently changed to a 64 kbps Opus codec for playback. please makes the files available in a linear format instead.
Yes your'e right, my initial thought process was that if I change only one variable (sample rate project option in the DAW 48 and 41) while all the other processes are the same, if there is a difference it will still be heard, but I will provide lossless wave files because there's definitely more detail in those.
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the direct output is a TS output-how would a software update be able to turn that into a stereo output?
Ah, missed the connector was a TS. Oh well, thanks for the helpful clarification.
Cheers,
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Would it be possible to use a y cable on the direct send to allow stereo monitoring with the monitor out as well as still have the option of the effects loop of the Kemper using the return like normal. I would love to have both stereo monitoring as well as use of the effect loop.
