Didn't I read somewhere that a higher bitrate was planned? I'd be surprised that it's set at 44.1 and is intended to stay there.
Yes, it's my opinion that most pro recordings are made at 44.1, and that opinion is based on it being true!
Personally I prefer 88.2 or 96 but it's a lot more tricky and sometimes impossible to work that way meantime.
You think that the SA-CD is not pro?
Of course SA-CD is pro spec but I rarely speak to any producer or engineer who works at anything other than 24/44.1 so it makes sense to me that if there's only one bitrate in the Kemper meantime that it would be 44.1.
Has there been any statement from Kemper that it's either going to stay at 44.1 or be increased? I'd be very surprised if it's to stay at 44.1, it may be the most usual rate to work at meantime but it's almost bound to increase as more gear and processors can actually cope with it.
I suppose there may be some resistance since some seen unable to hear any difference! I think that's dependant on what they may be doing with it, though. I can certainly hear a huge improvement in, eg, some plugins and then no difference in others.
Btw, I guess you all know that but it is sample rate what you are referring to now, not bitrate.
Of course.
I only play an ignorant idiot on the internet.
I prefer to record analog sources at 88 and when I tried Kemper analog out into my soundcard at different sampling rates there was a definite difference.
Bear in mind that most DAW's mixers are pretty primitive literal summing devices that don't do any waveform reconstruction so the nyquist limit doesn't necessarily readily apply. Without the reconstruction prior to summing (and summing with higher bitrate aliasing) you can get artifacts and loss of fidelity in the audible range,
Can you explain this waveform reconstruction prior to summing?
Never heard of that.
CK
I believe you mean that the ADC uses an anti-aliasing filter and the DAC a reconstruction filter. As for the summing, maybe I am wrong but I don't think you can do any reconstruction when summing digitally since the reconstruction term is used when converting from digital to analog. I believe it simply refers to reconstructing an analog signal from its digital representation by interpolating what is missed between each step in the digital form.
I think that what you are basically saying is that summing signals sampled at 44.1k will render worse results than summing signals sampled at for example 96k and then converted to 44.1k. Because when you do the sum of the higher rate since those are closer to the real thing the result will be more accurate and the risk of aliasing or artifacts is much lower. Right? I am sure Christoph is aware of all that already anyway.
Maybe I am wrong but I believe that if we don't get higher sample rates is because probably the processor will not be able to handle those. Maybe offering 48k would not be a considerable extra load but 96k is much more to take. You can easily notice that when you run certain plugins on the computer and use high sample rates. Another option would be to render everything at 44k and then do a sample rate conversion at the end, but maybe that is too much load as well if you want to do it with pristine quality.
Oversampling two signals prior to summing and downsampling it afterwards will give the same results as when you just sum the two plain signals.
In other words: the glue that you produce by oversampling will be summed, but later discarded by downsampling, without having an effect to the residual signal. You don't gain anything by that.
Where the hell have your found this theory?
You will not find any skilled DSP programmer stating that summing signals is sort of a challenge.
Please tell me the source where this kind of digital esoterik is teached.
Digital summing is:
Have sample streams of the same sample rate.
You add every corresponding sample.
Especially when you mix many streams (mixing desk), maintain a high bit width until the last stream is added.
Then the result can be truncated (cut the bits not needed).
CK
Yes that's what I meant (see my second post). And you may be right, but only Christoph can answer that. I'd be very surprised if the DSP couldn't handle it though given that even a Pod can do up to 96k and that's got a far less beefy DSP, also Christoph has said that internally it's working with much higher rates in certain places. But as mentioned, only Christoph knows if he can squeeze out that little bit more or not.
It would be sad if it's locked to 44.1 for it's lifetime, it would have been fun to use the Apogee convertors and keep an all digital signal path (not to mention less cables and therefore less hotswapping in order to use it), but being honest I don't hold much hope.
What I said is not the same as you mentioned on your post or at least not the way you wrote it. I am talking about using signals originally sampled at a higher rate, then summing and downsampling at the end. That normally produces better results but just because the source you are using is better/more accurate.
Maybe you were just confusing digital summing vs digital to
analog (mixer) summing. Some people prefer doing the summing in analog
even if all the source tracks are in digital because it sounds better for them.
What I said is not the same as you mentioned on your post or at least not the way you wrote it. I am talking about using signals originally sampled at a higher rate, then summing and downsampling at the end. That normally produces better results but just because the source you are using is better/more accurate.
Maybe you were just confusing digital summing vs digital to
analog (mixer) summing. Some people prefer doing the summing in analog
even if all the source tracks are in digital because it sounds better for them.
No, that is what I wrote. i.e. Making the assumption (which as it turns out is wrong - see Christoph's post above yours) two approaches, high sample rate or reconstruction before modification of the sound at a higher rate (so effectively both = higher sample rate).
Anyhow based on the Nyquist rule (which is just a math rule, not specific to audio) then you'd need a sample rate of only above 2x the highest frequency that humans can hear in order to accurately reconstruct the waveform at the other end when accuracy = the same up to the limit of human hearing. So including what Christoph said above then unless higher frequencies/harmonics somehow were adding interference that's audible lower down (unlikely) then 44.1k should be just as "good" as 48k or 96k or even 192k, at least when it comes to human hearing, and the final result should null exactly if you were to take a mix made at 44.1k and one made at 192k and they were made to match frequency with the same aliasing filter as applied to the original sources for the 44.1k version (and according to what Christoph said that's regardless of whether they were reconstructed and re-aliased around the mixer/fx or if the samples went straight through the whole thing), and of course ignoring clock jitter. And I've heard that said before too elsewhere in these discussions.
I still prefer to work at 48k though, maybe it's purely psychosomatic, but it does feel easier to mix in and maintain clarity, or "better" as you say. Christoph has said in the past that there are parts of the KPA where the sampling rate is much higher, I wonder what they are and why they are there, also why audio software and interfaces offer higher samplerates at all, except maybe the cynic in me says as a meaningless checkbox to attract customers.
Display MoreOversampling two signals prior to summing and downsampling it afterwards will give the same results as when you just sum the two plain signals.
In other words: the glue that you produce by oversampling will be summed, but later discarded by downsampling, without having an effect to the residual signal. You don't gain anything by that.
Where the hell have your found this theory?
You will not find any skilled DSP programmer stating that summing signals is sort of a challenge.
Please tell me the source where this kind of digital esoterik is teached.
Digital summing is:
Have sample streams of the same sample rate.
You add every corresponding sample.
Especially when you mix many streams (mixing desk), maintain a high bit width until the last stream is added.
Then the result can be truncated (cut the bits not needed).
CK
Hi Christoph, thanks for chiming in. So I was wrong then with audio. Thanks for clearing that up. However I do think that this discussion is a diversion. As originally requested I'd still like 48k or other sample rates from the S/PDIF (or to allow the convertor to set the world clock/master rather than the Kemper), if nothing else I don't see it as in any way being a negative for the unit and it may even help bring in more sales (or at least not turn off anyone who may have been interested otherwise).
I want to brig up this feature request, regardless the technical question.
I'm working to a reamping studio and people mixing, asking me to work on tracks ask for 48KHz/24 bit or higher.
Please work on this.
48khz would be much appreciated
I would appreciate samplerates of 88.2k and 96k a lot. There are people who like to use higher samplerates to reduce aliasing artifacts while both recording and mixing, especially when working fully in the box and using plugins and soft synths.
48K: +1 
48khz 24 bits and slave mode would be more than appreciated
this is very tiring to have to put the kemper as master, without any synchro output for the other converters
if you leave the rest of the gears as master it is ploping and clicking in every way
i was not aware aboutr this "issue" before buying and if i knew before, i am really wondering if i would have bought this gear
Just connect it in analog to you interface and record at whatever rate you want...where is the problem? The balanced output of the Kemper is a lot better then most of studio amp/cab/mic I know...
