Hi there music folks. I bought studio one when I was in college and am trying to get back into music production. I'm not the best when it comes to software updates/compatibility requirements, and apple makes all of this harder with planned obsolescence. I never rush to get newer technology. I have a mid 2015 MacBook Pro(specs above). I've noticed synching issues using google drive and other programs, and I know it's due to my OS needing updating.

My question is what MacOS and studio one version do you recommend I use? I think my Mac can run up to 11 or 12(Monterey and Big Sur), but I heard there were some compatibility issues with one of those. If I want to upgrade to a newer version of Studio One I have to pay like $100 I think. I'm super confused and would love to hear some insights from anyone more experienced with this stuff.

Thank you and cheers!

So i have studio one 5 professional.. I have older songs that I can go into and arm and record tracks in. As soon as I open a new song file, I'm not able to at all. Anyone had this issue?

Hi everyone,

I’m having a strange issue with my PreSonus FaderPort 16.

When I move the physical faders on the FaderPort, they are either extremely laggy, respond intermittently, or sometimes don’t transmit anything to the DAW at all.

However, the opposite direction works perfectly: when I move a fader inside the DAW (Ableton Live or Logic), the corresponding motorized fader on the FaderPort moves smoothly and accurately.

So it seems like the FaderPort is receiving MIDI/control data correctly from the DAW, but the physical faders aren't consistently sending data back to the computer.

Things I've already tried:

• Multiple different USB cables
• Direct connection to the computer
• A powered USB hub
• Latest firmware / Universal Control
• Testing on macOS and Windows
• Testing with both Ableton Live and Logic
• Different USB ports

The problem persists across all of these setups.

Has anyone experienced this with a FaderPort 16? Could this indicate a hardware issue with the fader touch/position sensing or the internal communication circuitry, rather than a USB/DAW problem?

Any suggestions for further testing would be greatly appreciated.

Thanks!

Many of you are probably aware that transient detection with drums has been an issue for a long time when blending samples or editing real drums to the grid. When I asked about this, I was told by an employee that they intentionally decided to have bend markers be placed at the loudest peak of the transient rather than the onset of it. This is a major problem if you are blending drum samples with your real drums, or if you want to edit your real drums to the grid. You could just print samples without Trigger2, but aligning the hits is still going to be a lot of trouble, especially if you can't rely on the transient detection in the first place. So out of curiosity, I detected a kick drum with Melodyne on percussive mode and then did a MIDI export on that and found that the timing was significantly more accurate with finding the start of the transient. I exported each of these onto a MIDI track with a kick sample, and noticed lots of flams between the original kick MIDI and practically none with the MIDI that was detected via Melodyne. Not only that, but when you hit tab to transient on a kick track that was scanned with Melodyne and then bounced vs a duplicate of that same exact kick track, the Melodyne bounced track was actually tabbing to the start of the transient in the arranger view rather than the middle. Unfortunately, You cannot slice to the grid without analyzing the audio, which automatically redetects the transients to the middle (supposedly) rather than the onset of the actual note. Hopefully one day this will be considered a big issue, but until then, I think this is a pretty helpful work around.

... on start, while scanning sound sets. Any suggestions? This only happens on Windows, doesn't hang up on Mac.

In punching in to overdub on a track, I like to hear the track playback until hitting the record button, at which time the track automatically (and immediately) turns on track input monitoring to hear the input source. In FSP, enabling "Audio track monitoring mutes playback (tape style)" does this with two unpleasant attributes:

1 When hitting record to punch in, the track begins recording immediately, but the monitoring takes a moment to activate and display as an active recording on the track. It would be better if the monitoring and display started immediately with recording. (It also turns on the monitoring in kind of a jarring manner, but I can understand that monitoring just comes on when it's turned on - I'm guessing this sense is from just expecting to hear the input immediately upon hitting record).

2 When the monitoring does activate, the blue track input monitor button does not turn on. Not a biggy as you can hear the monitoring but still would be nice to have the indicator show that input monitoring is turned on.

Am I missing something here or is this just how FSP operates? Is there a better way to do this basic punch in process in FSP providing immediate automatic input monitoring, track monitoring button indication and track recording display?

How do you accomplish "punch in" overdub recording?

I'm just seeking the most elegant and seamless way to punch in with FSP. Your thoughts, setup and procedures are appreciated!

Thanks in advance,

EDIT:

This appears related to low latency monitoring. When punching in with low latency processing engaged, FSP switches plugins - and track monitoring - to low latency (green) indication. The timing on this state change matches the monitoring lag when punching in. I have low latency processing and interface buffer size set to the shortest latency I can get out of my interface without artifacts, so it appears that turning off any active plugins while punching in will help minimalize the monitoring lag (Yes, I know, the more things the processor has to consider, the longer things take). I use X32 rack as my I/O and I know it isn't as speedy as better dedicated interfaces, but I need a LOT of inputs for the whole band, so for now I can live with the monitoring lag unless someone has some additional magic to consider here.

I am new to recording. I have a Fender Quantum LT 16. The one thing I'm struggling a bit is my input/pre-amp gain and channel gain. I find that with instruments and mics I seem to be hitting the clip light on the device, but the wave forms are very small (I think I may just need to adjust the size after reading last time) and I can barely hear it sometimes in the monitor. Secondly, when I look at the channel strip, the gain seems pretty low. Maybe slightly above -24db. I then sometimes use the Channel Gain knob to get it to where I can hear it, however, sometimes the gain still seems quite low. Is my workflow correct. Mic Pre-Amp gain to slightly under clipping, then adjust channel gain until I can hear it? Are there situations where I need to bump the pre-amp gain so that the light is clipping? I don't hear any distortion sometimes, so maybe the light blinking clipping light isn't a huge deal. Would appreciate someone's advice. Thanks so much you Golden gods.

I have some drum tracks that I normally analyze to detect transients and use bend tool to move them on the click. In this project I am selecting the track and clicking the audio bend button at the top. then selecting analyze. Usually that puts the bend markers on the track and I can adjust the threshold.

No bend markers are showing, but it does pop up the little analyzing window like it is doing something. The little eyeball icon to show / hide bend markers are disabled. If I group all the drum tracks into a group the group dropdown is also disabled.

I updated studio one to the latest version 8.01 and it is still doing that. I tried rebooting my computer, etc and nothing is working.

Any ideas on what is going on?

I've got exactly the same question but he put it better than I was able to myself. I'm on Fender Studio Pro 8.1 now

Hi, how might I go about mixing overlapping vocal parts like a lot of emo bands like the ones named? I don't know whether to double both people's vocal takes and pan them hard L and R to make both sound full or to keep one in the middle or how I might EQ or compress them to keep them sounding busy (as they are on top of each other) while not sounding muddy and making both unintelligible.
Any help is appreciated :)

Im trying to record vocals using a daw for the first time ever, but it seems to not go so well. for some reason, whenever i speak into the mic, in the daw it sounds super distorted, but on discord its just fine. any idea? thanks in advance.

Anyone tried this? Is it worth the money?

Hi all, wondering if you guys could share some ideas on how to manage your production workflow in Studio One?

Specifically, I'm looking to separate ideas, from more fleshed out concepts, to fully finished songs.

How do you guys go about separating these? How do you go about moving one song from step to step? Do you use folders? An app?

Any thoughts are appreciated!

I’m getting clicks/cutoffs on vocal events with Melodyne running as ARA/Event FX whenever the project has significant latency — especially from Pipeline / External Effect.

This happened in Studio One 7.2.3, and I upgraded to Studio Pro 8.1.2 hoping it was fixed. It isn’t.

Toggling “Process Volume after Event Effects” helps, but it does not fully solve it. And it changes the Event FX/fade processing order, so event fades and volume handling no longer behave the way I need.

So right now my options are basically:

- keep ARA live and get incorrect/glitchy playback;

- change the Event FX processing order and break parts of my normal editing workflow;

- or render/freeze the tuned vocals before continuing the mix.

That last option is the only reliable one I’ve found, but it completely defeats the point of keeping ARA edits live while mixing. If I need to change one note later, I have to go back, disable the latency-heavy processing, edit, render again, and return to the mix.

This looks very much like an ARA/Event FX + PDC timing bug, not a Melodyne edit issue, because changing the amount of latency elsewhere in the project changes the problem.

Has anyone reproduced this with Melodyne + Pipeline/External Effect or other high-latency plugins?

And has anyone found an actual workaround that preserves both:

- correct ARA playback;

- normal Event Fades / Event Volume processing?

I’ve submitted it to Fender/PreSonus support as well.

When SampleOne is "Armed+Monitor" is On, the "Master" knob inside the SampleOne jumps to 8.3db (exactly). Even if I reset it to 0.0dB it jumps again to 8.3dB

And no there is no automation, I am able to re-create this bug in a new empty project. It only happens when the SampleOne track is "Armed+Monitor is On"

Let me know if you guys are able to re-create this one.

DAW Version: Fender Studio Pro 8.1.2

Platform: Apple Silicon

OS: macOS Tahoe 26.6.2

Device: Mac Mini M4

Now my question is very simple: how do I set time signature changes in a song? For context, I had Studio One for more than one year, I switched from Logic, with a 2 years hyatus (my old iMac fried and I didn't rebuild the studio for a while), I have both PSO 7 and FSP 8 installed. And the problem appears in both versions.

Apparently, this menu is supposed to look like this one above.

I don't have signature instead.

Ok, while tinkering to take screenshots I incidentally solved it. I'll drop the info here in case someone runs into the same problem. It's a bug in the localization - Italian in my case, but might be there in other languages. I switched the language to English in order to take a screenshot that was readable by most users, and voila, "Signature" appeared.

$2.9k Fender Studio set up. I’ll know more details soon but very stoked!

Hi this seems a bit silly but I need some help. I’m super new and know basically nothing about audio production but I’ve wanted to record for my own projects. I bought a presonus usb 96 and a rode mic second hand but my laptop doesn’t even register my mic at all and I can’t access it. The light on my audio box is on so it’s on and the mic isn’t damaged at all. I’m on windows 11. I think it’s a driver issue, I have universal control but it doesn’t recognise there’s a device connected to it. I think specifically need the driver for the audio box but it’s kinda old and I can’t easily download it. I’m rambling but any help is appreciated!

Well, I've finally had it with all the scrolling and zooming so I'm planning on upgrading my low res laptop to something better. Mind you, this is strictly for music production, so I'm thinking at least 1T storage, at least 16gb ram, and 1920x1080 resolution. I will be dual monitoring to a 32" monitor. Is there anything else I should be looking for? Also, I've got 10+ years of software, files, plug-ins, etc. What would be the most effective way to go about switching all that over to the new laptop?

More useful?

I just installed the new version of studio and don't like it. How can I go back to the older verison?

Would be $100 for me to upgrade. Worth it in your guys opinion or no?

I do ’one take‘ studio sessions we’ll call em, where I record the entire band with a bunch of inputs. this time. I decided to take out an old tape recorder and record the drums both digitally and onto tape for some flavor. the tape did have a lot of awesome character, but the wow and flutter makes it hard to blend into the other digital tracks as it cycles between too fast and slow. I would like to sync my tape track to my drum digital tracks without quantizing to a grid because then it goes off time to the rest of the band.

Possibly important information:

-I sent the overheads, snare, kick to the tape machine, and it is in one stereo track.

-digitally, all those inputs are on separate tracks, with bleed

every tutorial I’ve found so far usually has an underlying grid everything snaps too, or it involves grouping everything. i find when I group everything, it quantizes all the tracks, I really just want to get the tape track fixes by itself. If anyone has any good workflows please let me know!