I just enrolled in Blues Immersion and I am trying to set up my Katana 50 so that I can record from it into my DAW via the USB connection. I just spent multiple hours with AI (Gemini) trying to iron out a variety of issues. Just when I thought I had it all set up, I played some old recordings through the amp and they all sound very staticky (for lack of a better term) The same is true for my new recording. The bottom line with AI is that there is a conflict since I am also running monitors (Rockit5) All my previous recording have been using the monitors. So, my questions are two fold. Is this to be expected when running both the amp and two monitors? I am using Cubase DAW with a Steinberg UR22Mk11 interface. I’ve been really frustrated with Cubase for years but I used it because it came with the interface. I am considering switching to either Reaper or Studio One. I would like a DAW that is very intuitive and Cubase is definitely not, in my opinion. Any thoughts? Thank you!
Hi,
The staticky sound replay maybe that you have the Buffer set too low in Cubase try increasing it a little at a time and testing as you go. Also all DAWs are very memory and CPU process power hungry, it is best to shut everything else down that you can on the computer unless you have one with masses of RAM and an 8 or 16 core processor.
If you have multiple USB devices connected this can cause issues and slow things down, so anything you do not need when using it I would recommend you unplug. If on windows I would recommend you shut all unnecessary apps down & unplug unused USB devices, then if a Windows computer shut it down doing a full reboot without it starting up apps automatically. You should not have to do this but unfortunately API’s and DLLs etc from previously running applications on a Windows device can be persistent in memory, and cause issues. I worked in testing on Windows clients and NT servers from 1988 to 2003 on commercial systems, and ended up at Apple in tech support for 10 years, this basic advice should help.
I used CUBASE over 20 years ago when on a BTEC Music course and it does take some getting used to, shortly after I move to Apple Mac and Logic Pro which is very similar as the college used it. Both are used in Professional recording Studio environments, hence the complexity, as well as home recording.
If you want a simpler DAW to record only you could try OBS Studio or Audacity.
I hope this helps
I am not familiar with Cubase but I also thought of the buffer setting in your DAW. But if you are frustrated with it anyway it might be worth trying another. Reaper offers a generous free trial I think.
Thanks for that very informative reply! Your comments sort of mirror what AI was telling me and I did set the buffering to what AI recommended and verified I was at 44.1 hz as well. I’m going to try doing what you suggest to see what happens. I do have a gaming computer so I’m assuming I have lots of memory, etc. I didn’t buy it for gaming but got a great deal on it so figured it would be good for recording. Technology passed me by years ago! I’ve also decided to try the 30 day free trial of Studio. My research tells me that Studio is much more intuitive than Reaper and much easier to hook up. I’m glad I’m getting an early start on this before the course begins! I’ll post my results!
Not necessarily.
In my experience, real-time digital audio is one of the most challenging things for a modern computer, far more so than gaming or video editing.
Having lot of memory and a fast CPU has almost no impact on real-time performance: I have seen 20 year old 486 PCs outperform modern gaming rigs on real-time audio processing.
In fact, many modern CPU architectures can actually perform quite poorly with real-time audio processing.
The reality is, even if you are recording or processing multiple tracks at once, the actual audio data transfer is relatively low. Having a faster CPU or more memory isn’t going to affect it that much. What tends to matter is:
- What is the latency of your hardware and its drivers (particularly USB drivers if you are using a consumer audio interface)
- What buffer sizes and (in more advanced audio drivers) number of buffer periods are you using - this is a function of your audio interface driver, not of your DAW.
- What sample rate are you using (interestingly, although higher sample rates don’t improve audio quality and may, in fact degrade it, and put higher load on your CPU, they can reduce latency in some circumstances)
- How is your system set up for audio? For Windows, this largely isn’t something you have control of. Mac’s tend to be pretty optimised (partly due to their limited hardware support) and for Linux you have a load of control (possibly too much).
- If you can, make sure you aren’t using CPU scaling as this plays havoc with real-time audio , if you can set your system to “performance mode”, this will be the best thing you can do to improve performance
- Having a CPU with a hybrid architecture can “complicate” things
In a nutshell, it ain’t as simple as "fast computer = better audio processing.
Cheers,
Keith