processing overload

Just a follow up about the setting that I had on the M1 Max: I built a PC the other day. Nothing fancy, but still a powerful machine, a 5800x CPU with 32GB 3600Mhz RAM. It’s comparable to the M1 Max MacBook Pro. On the PC I am not getting overload spikes, but instead, very quick occasional popping during playback.

So I applied the same settings: Asio Guard = High, Buffer = 256. Project is 48000 Sample Rate and 24 Bit, same as my projects on the MacBook. This has drastically lowered the performance spikes and artefacts in playback. Both CPU’s are hitting around 40% to 60%.

With a 256 sample buffer you probably aren’t experiencing latency because of indulging attack times of orchestral sounds, mainly strings. At that buffer size latency should be very apparent playing more percussive sounds, or recording audio - guitar or voice are the most sensitive to this.
With that Mac spec you shouldn’t be experiencing spikes. I run an old PC, i7 8700k with an all-core constant 4.8GHz over clock (but, still, old cpu :slight_smile: ) and I’m a heavy Spitfire user. I can compose with 128 buffer, and at times 64 (for only a dozen of tracks) without overloads or spikes. I’m using Kontakt 6.something to play Spitfire libraries, if that’s relevant, and I use a PCI interface

Hi, your system is still powerful, an 8th Gen i7 is strong. Well, I really don’t know why it is happening on my systems.

It may be my interface on my PC. I’m using the Maschine MK3 as an audio interface and even when I’m not using it with Maschine software (so not using as an instrument, but purely as an audio interface) it is happening. I haven’t tried it with my Apogee Duet 3 yet, but I know that the previous PC I built, was working great (very rare spikes) with the Duet and was awful with another interface, the Thunderbolt 2 PreSonus Quantum.

Why the MacBook is having issues, even with the Duet 2 seems is strange. As mentioned previously, the IK tape emulator plugin takes a huge chunk of Asio Guard, up to 25-30% for a single instance. Maybe that is just too much of a hit on one core. Even if all other cores are at 5 - 15% it only takes 1 core to hit overload for the audio artefacts to happen. The emulator is putting a heavy load on 1 single core on the MacBook, according to Logic Pro X’s performance meter. So perhaps Cubase is having the same issues (not possible to check the load on individual cores via Cubase’s meter as far as I know)?

With the latency at 256, it’s not an issue. I’m no longer composing much in terms of orchestral scores or classical music, as I’ve transitioned into the pop industry. I use the Constrain Delay feature on Cubase to record live vocals and real or virtual instruments. It would be nice to not have to use Constrain Delay. My Thunderbolt 2 interface (which I sold) used to get down to under 2 ms at 64 / 128 Buffer! But it was just too unstable with the spikes, and I needed a smaller form factor interface, so traded it in.

Is it a PCIe card you use? What is it? I tried the MAYA eX44 and it just wasn’t good. Low latency, but is an unbalanced output interface, so I was getting a lot of static from it.

I figured it out. The red light keeps popping up bc it is related to the clock source of your audio interface(s)., especially if you are using aggregated devices. Go adjust/change your interface’s source clock signal & the red light will no longer appear. It took me a few days to figure out. But once I selected the appropriate source clock, the red overprocessing light no longer emits. I’m on a Mac, but I’m sure that this can also be applied to a Windows PC. I have a Behringer UMC 1820 w/ada8200, & a Focusrite Saffire Pro 40 fw all aggregated on a MP. When I switched my source clock to the Saffire in the Mac audio midi settings, the red light issue stopped. It is not a Cubase bug, instead it has something to do with your sync clock source :+1:t2: …, good luck :crossed_fingers:t2:

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It depends. Buffer sizes expressed in samples are not the best unit to describe latency since the actual latency (expressed in milliseconds) depends on other factors as well. Most prominently, the sample rate. A buffer of 256 samples will produce a higher latency if the audio interface operates at 44.1kHz as opposed to 96kHz for example. Other factors include quality of AD/DA converters, ASIO drivers and general “health” of the computer running it all.
As a rule of thumb, latencies at 5ms or below are typically undetectable but many musicians are fine with <10ms.

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Yes, I’m fine as long as things don’t go over 10ms, below that is ok.