I've been mixing in Atmos 7.1.4 for a couple years and the transition from v7.x to v8.x was seamless and still works perfectly. There must be an issue with your system setup.I think it’s crazy and at the same time great, that changes have been made to the Dolby Atmos engine. I actually had to switch to Pro Tools for Dolby Atmos Mixing because the LFE was causing the whole session to crackle. I had pointed that out previously.
Looks indeed like a graphic glitch: they are thesame color as your clips, but they shouldn't exist in the level meter.I've noticed some weird colored areas on some of my tracks in various sessions in Editor view. It is not events or small clips of audio. They stay the same size regardless of zoom. Do you know what they are or mean? Have you noticed yourself? Looks like a bug to me:
I replied here to this "ALT issue" and believe it is intended behavior, sorry to say.No updates on the ALT issue that we've been reporting weeks ago. I think that should have been a priority, so I can't hide my disappointment.
Fender Studio Pro 8 — UI lag / scrolling & zooming performance regression vs Studio One 7 (Apple Silicon)
The issue: When playing back a project in Fender Studio Pro 8, scrolling and zooming through the timeline is noticeably laggy — choppy, delayed, uncomfortable to work with. The exact same project, on the exact same hardware, with the exact same audio interface and buffer settings, is smooth in Studio One 7. This is a comparative test, so the difference is very obvious side by side, even if it's less obvious testing Fender Studio in isolation.
VP, thanks for the detailed reply — and especially for actually testing it on the demo project in 8.1.2.Windows 11 user here - my gear in my sig.
While I no longer have Studio One 7 installed - I did the test with the demo project here on v8.1.2 - and all scrolling, zooming - basically any motion of any kind on the timeline is crisp, accurate and instant.
The other kicker here is I do not even use a graphics card. All my visuals (now in 4K) are driven by the iGPU contained within my Intel Core i5 chip.
Also - given the nature of my main pro work in this DAW (commercial voiceover) - where I spend countless millennia editing audio which is nothing but one long series of zooming, scrolling, jumping here and there - every minute of every day - if there were even the slightest signs of a problem - you would see reams of thread action from me.
I will also add - that up to the end of July 2026 - my main rig was still Windows 10 22H2 and I thought that experience was as fast as humanly possible. But as soon as I moved to Windows 11 AND upped my game with a new 4K monitor - the editing/scrolling/zooming speed seemed to jump past another level that I was not quite ready for.
I know this does not help your use case - but I have seen this condition with other Mac users and the fixes ranged from changing options in the OS to dropping hardware acceleration in FSP to other tweaks that I cannot recall at this exact moment.
FWIW - there are tons of Mac users here on the forums - if this was a thing - across the board - pretty sure we would have some spirited (ongoing) threads about it.
VP
I know not many people run with low buffer settings, so this issue might not have surfaced right away for most users
I use amp sims rather than analog gear, and even at 64 I can sometimes feel a slight hit to responsiveness on guitar — at 128 I notice it more often.
Sorry, I didn't want to steer this into a side discussion. The real point is the issue I originally wanted to flag — it's actually what's keeping me from moving over to Fender Studio in the first place. And I do want to move over now, even for one reason because they finally fixed how plugin windows open from the mixer by single click. I'm worried that core point gets buried under a pile of replies, which lowers the odds it actually gets seen, tested, and gains any traction.Well - you and I appear to be brothers from another mother and being a guitar player - this is me too.
However - I guess you would need to explain to how you actually record your guitars in FSP - using your amp sims.
Over here - I fire up an audio track, route my Analog 3 (Hi-Z) input from the UCX-II to it - "insert" my sim of choice on that channel and then with "D" monitoring (via the RME in FSP) the latency = 0. Like being plugged into an amp right by my ear. No delay. No nothing. Ever. So it simply does
not matter what buffer size is in play.
The huge bonus to this method is that my actual guitar performance is recorded bone dry - allowing me to switch between any sim I want - when I want and then print the result if I like what I hear.
I can even come back 3 years later and add a sim that could completely change the vibe to the whole track with the click of a mouse - without ever committing to the original sim I used the day of recording.
For me - it's all capturing that original performance and to have flexibility with it - long AFTER the recording is complete. All with no muss, no fuss and especially - no latency.
Your method is clearly different - yes?
VP
As for my own setup — it's not that different from what you described, honestly. For one, I don't use "D" because i don't use FS8(in Studio One 7 it was called Low-Z monitoring) — I never really used it, because it seems to change the sound of the project.
It can't. It just can't. Technically impossible. What Z monitoring does is simply installing a second buffer alongside the larger one - the bytes themselves come in and out of the buffer just as with the larger buffer. Regarding the buffers themselves - the technique is using a double buffer. One is played by the audio interface while the other is filled with bytes. A buffer size of 32 means 32 samples are stored. At 32 bit resolution this means 4 bytes per sample = 128 bytes buffer. This buffer lasts 0.67 milliseconds - not even one full millisecond. Due to multiple buffer reads, the total latency is more around 2 ms. Human perception threshold for extremely trained ears is 10 ms.For one, I don't use "D" because i don't use FS8(in Studio One 7 it was called Low-Z monitoring) — I never really used it, because it seems to change the sound of the project. As I understand it, it's supposed to work by bypassing plugins that add more than ~2ms of latency, and it cause changes in sound but it feels like it sometimes changes even the amp sim sound, in empty projects which throws me off. So instead I just run minimal/low dropout protection with a low buffer and monitor straight off the channel, to get as close as possible to RME's actual latency — plus 48kHz sample rate reduce latency a bit too.
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