For me a huge change the past years was the coming of Bitwig Studio audio workstation to Linux. Commercial software, but rock solid and high quality.
Ableton is a turn-off for this reason. They're actually pretty cool about adding activations if you let them know you're moving machines, but it offends me to ask permission to use something I've already paid for.
If Bitwig (1) works well on Linux, (2) works well as a DAW, and (3) doesn't dick around with activation rules, then I'd be delighted both to support them and to switch my (current) hobby workstation from Windows to Linux.
You can activate 2 systems permanently, which is managed simply from the web UI. (revoke activations) Other systems can be activated temporarily with your email/id + pass on startup.
FWIW, the small team has been fantastic and responsive since the beginning. (I've been on the betas and helped develop some of the early controller scripts. check the community pages to see what we've done)
I got a NFR license for helping out early on, and ran into the activation limit initially. (testing on all platforms, etc...) I emailed them a few times to deactivate my current systems initially, then they added the option to their account management page.
If its a deal breaker, I suggest emailing them and seeing what they say. It wouldn't surprise me if they upped the limit or did something to help you out. They're an awesome group who really want to work with the community and users.
Not affiliated besides what I mentioned above, just a happy user. Best of luck, if you have any questions PM me or reply here if you think relevant to others :)
I actually like Jack when it is concerning latency and think it is a good system. BUT it is a pain.
I used 2 separate sound cards. One is a Pro level I/O breakout box that Jack controls and the on board sound card is controlled by PulseAudio.
Since Windows XP Windows and Audio Latency has not been an issue and both platforms require a lot of end user work to get lower and lower latency. The issue is really only significant for recording audio and not as much when doing live audio. Any modern platforms latency light years ahead of 2000 audio production.
The idea that Mac is better for audio or video because of the OS is marketing and not based on real life professional use of the platforms.
Core Audio literally-just-works with low latency and aggregated devices.
DAWs on Windows still either use MME/DX for north of 50ms, often 100ms+ latency or ASIO (exclusive device usage and no aggregation, assuming your hardware even has ASIO drivers because ASIO4ALL is at best rickety) on Windows.
I don't know what you think is going on with OS X, but I suggest re-evaluating your assumptions.
The problem there, being a closed environment where the manufacturer dictates when the product must die, is rather the drivers life. If they don't update the driver your pricey gear turns into a doorstop overnight. Case in point: Tascam US122 audio interface. Under Linux I can still use it; far from being the best around but it works. Under Windows it became unusable when they stopped supporting it years ago.
Hang on, no way, you take one or the other. Aggregating devices adds a huge chunk of latency and, frankly, I don't think is that exciting a feature anyway. Maybe for your specific setup, but generally speaking you should buy gear that suits your needs, rather than try to cobble something together from existing devices.
Having used OSX and Win7 (on the same machine) I would agree CoreAudio is less hassle and definitely lower latency - IIRC, OSX reported just over half the latency of W7 on the same setup.
That said, I'm not sure what you think is going on with ASIO, since any class compliant USB audio interface has compatible drivers inherently..
In the situations where I have to do it--and it's less about "buying the gear that suits my needs," more "having to bash together other people's gear on a shoestring"--I find that aggregation ends up around 30ms, which is on the higher end of acceptable when monitoring in-ear. (To be specific--this is not for music but rather audio routing for video. Occasionally I'm on-site somewhere and need to be able to monkey up something a little faster than I'd like, you know?)
Trying to do that with ASIO at all is impossible. So it's a pretty big deal.
> That said, I'm not sure what you think is going on with ASIO, since any class compliant USB audio interface has compatible drivers inherently..
In theory, yes. In practice, I find ASIO a little unstable (my home setup uses Ableton on Windows as a live mixer through a TASCAM US-16x08, though I'll be going back to using a Mac when I get back my Mac mini from a friend). I have never had Core Audio kernel panic a machine, but I've had machines (in one memorable case, the same machine dual-booting a hackintoshed OS X and Windows) hard-lock as soon as I enabled ASIO on two different devices (my old TASCAM US-1200 and my a friend's 8i6).
It is not unusable, by any means, but for my purposes (again, live shows) predictability is a big plus.
I haven't done that kind of work, so for video/speech, fair enough. 30ms (on top of whatever's already there) is not really acceptable for a musical performance, though - the performer will be expending half their brainpower trying to mentally align what they hear through bone conduction vs in ear monitoring and adjusting their performance based on outdated/scrambled information. It's a really confusing experience - if you've ever seen that Japanese "speech jammer" device[0] you'll get some idea what I mean (although it uses much longer than ~30ms, that's still enough to mess with you).
Working in a pinch, it would make a useful and more stable tool than ASIO, though I run either analog or dedicated devices for any live work - I just don't trust computers that much - but I get that working with video means you might not have those kind of options. I can't say I've had many of the troubles I've heard about with ASIO to be honest, but I undeniably do enjoy working on OSX more anyway. This conversation reminds me how badly I need to revive my ML Hackintosh..
As the other commentor said, Core Audio "just works" for low latency. MIDI is great as well. Windows still needs third party ASIO drivers.
If a USB MIDI controller gets unplugged during a set on Windows, I need to restart Ableton. On Mac, you just plug it back in and it picks up immediately, even the audio interface will do that.
> The issue is really only significant for recording audio and not as much when doing live audio.
What? Latency is a non-issue for recording audio, and A HUGE ISSUE for live usage. Any half-decent DAW supports latency compensation on non-realtime tracks. You can record a vocal track when every other instrument has 10s of seconds of latency, as long as your monitoring for the vocals has none. When playing live, every single track cannot have latency.
Edit, hell many of the best audio interfaces don't even support Windows as a platform.
Give this a read - https://www.presonus.com/community/Learn/The-Truth-About-Dig...
Live audio if you have above the human perception of latency you are in the realm of 20 or 30 ms.
> Edit, hell many of the best audio interfaces don't even support Windows as a platform.
What can I say marketing works and this perceived Mac superior creative types tool is believed by most people. In actual Professional world there are plenty of Windows based studios that won Emmy's, Oscars and Grammys.
Please dude, look up latency compensation. You have zero idea what you're talking about and it's obvious.
> In actual Professional world there are plenty of Windows based studios that won Emmy's, Oscars and Grammys.
I don't believe I ever said otherwise. But to say Mac never outperforms Windows for audio is wrong. And to say Mac requires just as much setup work as Windows is a flat out lie.
You are wrong about the need for proprietary ASIO drivers for all devices, or that many of the best audio interfaces don't support Windows.. One manufacturer is not "many" and while they are high range, it's definitely arguable whether they're the best.
Anyway, in my experience it's rarely the hardware itself that requires the majority of setup, and either way it's a once-off. Unless you're the kind of person who uses the same template for every recording session, in which case you have bigger problems than setup.
> I don't believe I ever said otherwise. But to say Mac never outperforms Windows for audio is wrong. And to say Mac requires just as much setup work as Windows is a flat out lie.
Nope. Its actually a huge pain BUT most people don't really care or need to actually dig that deep.
https://support.native-instruments.com/hc/en-us/articles/210...
https://support.focusrite.com/hc/en-gb/articles/207546515-Op...
Then you have the issue with your i/o in Macs with all your Desktop options that isn't a problem but all the laptops its a HUGE problem. I have no idea how people do audio with the i/o issues that Apple throw at you all the time. Thunderbolt anyone?
If you have a known constant output lag, a known constant input lag, but no lag in the monitor for just the track you're recording, it seems like the software has everything it needs to put back the performance exactly the way you recorded it (and to adjust the offset for the click).
Am I missing something?
My point latency doesn't really matter for most people recording. BUT if you get in that realm where you need to worry about it then its a PAIN no matter what your OS is. Apple isn't "superior" in audio recording just like it isn't in video and image manipulation. OS is more about people's feeling and attachment to their OS's company's marketing.
I think Apple has been dishonest and hostile to people so I don't like them as a company. You can't trust them not to throw the rug out from under your feet (Thunderbolt cost studios thousands and thousands of dollars). Lack of a new Mac Pro also is a HUGE issue for most video shops now. Their OS drives me nuts and really unreliable for me. Other people love it and I am fine with your opinion until they think everything else is garbage.
Are the 3ms measured or is it just what OSX tells you ?
A buffer of 3ms at 48kHz holds 144 samples. That means shoveling 144 fresh samples (per channel, ofc) ~333 times a second and sending them to the sound card immediately. That may be possible if your sound card supports resampling (and minor magic) and only without a sound server (OSX uses one). Either that or you have an impressive cpu. Feel free to correct me at any point.
edit: PS This is only the program->sound_card part. Programs themselves add a ton of latency and sound cards add to it as well. In reality even 10ms is beyond perfect conditions.
Stepping frame by frame through the video I took (https://youtu.be/IHmC-q_iPiE) at the point where I press the key to sound a note, there's a 4 to 5 frame latency until I can visually see the speaker membrane move, so that's about 33 to 41 milliseconds.
Bitwig is set to 64 samples (1.45 ms) latency.
Here is the focusrite help page dealing with latency and what you must turn off for recording. https://support.focusrite.com/hc/en-gb/articles/207546515-Op...
Focusrite doesn't even offer Mac drivers for most of their products, as the support is built in, for example[0], so I don't know why it's advising you to update your drivers.
That page is "dissuade people from calling tech support", not the reality of how audio actually works on OS X.
http://www.soundonsound.com/sound-advice/q-which-audio-inter...
yes its 2009 but it is still sound science.
It will also support JACK but for quickly getting up and running it's about the simplest DAW I've setup on Linux configuration wise.
Not open source but I give them lots of kudos for supporting Linux from the start.