throw std::future_error(std::make_error_code(std::future_errc::broken_promise));

Sure, just bring Russia to retreat and pay reparations.

LLMs trained on social media reproduce social media when asked to do so. Who could have known?

You can throw stuff like that into perplexity.ai as a starting point (it’s free):

https://www.perplexity.ai/search/before-i-fully-make-the-switch-OTEQ6b5xTIiOL1J5avEKyQ#0

You can then continue to ask about stuff you want to understand. It’s a great learning tool.

The main difference is that suffering is internal while being hurt is an external factor that can lead to suffering. People suffer from all kind of things: sickness, being bored etc.

You can be hurt but are not suffering and you can suffer without being hurt.

Being hurt and suffering are not the same.

Ever been on twitch? There are all kind of Pepe emotes used there (and related frogs). It’s a bit like saying “everyone talking German doesn’t mind being confused for a nazi”. Ridiculous.

Are any VPN providers publicly traded?

You can start steam from a console (by typing steam), and see log output which would help to diagnose the issue.

Fake it till you make it

The problem with audio interfaces is that they function very different internally and have different kind of settings. Alsamixer does usually a decent job of listing all parameters but it is an old TUI tool and not nicely embedded into the desktop so I guess people just don’t find it. Stuff like latencies just have to do with buffer sizes that are configured in your machines audio system, usually pipewire, pulseaudio or jack, which all work on top of alsa (which is where the drivers run). You can reduce the buffers there (in config files) to get lower latencies. This however means that your system needs to have a very tight scheduling for your audio processes, because if it fails to fill the buffer in time there will be glitches. Professional low latency audio does definetly not work out of the box on linux. It got a little better with pipewire, but I don’t think it works well without a little bit of tinkering. If you decide to tinker I recommend you read this: https://wiki.archlinux.org/title/Professional_audio

I don’t remember which tool I use for my Scarlett (I’m travelling). But I googled a bit and this looks good:

https://blog.rtrace.io/posts/fedora-support-focusrite-scarlett/

This all would be better if manufacturers would provide Linux config tools like they do on windows or at least information of their protocols. Until they do we have to be greatful for people reverse engineering that stuff (e.g. by analysing USB traffic on windows) and then writing uis for it.

Edit: this site seems to make more sense as the arch wiki page (it is linked there):

https://this.ven.uber.space/docs/computer/pro-audio/

No but now we get closer to the real problem. Meaning there is an accessibility problem, which is different than the (in my opinion wrong) statement that I wanted to correct.

They work but you can’t set the sample rate or enable any custom features on ANY of them.

Not in my experience. I have a RME card that can be configured via alsamixer (which should work for most cards) and a Focusrite Saphire USB interface that someone wrote a little UI for in which you can even freely route audio to/from different channels and mix busses.

TACO

Life is not a problem to be solved, but a reality to be experienced

Still no IDE to use C++ currently trying Rider.

I recommend: (vs)code, clion (free for noncommercial use), qtcreator

Reminds me of this: https://www.youtube.com/watch?v=5IsSpAOD6K8

TACO

So does running teams in a windows vm prevent me to take a screenshot on the Linux host? I can’t imagine it would.