You could let us decide the sample rate OBS will work with (

[Getúlio Prates]

I work with 44k sample rate. Although OBS records at any given sample rate, it works only in higher sample rates instead. It makes my ASIO driver bug and lose sync, every time I start up OBS. Sometimes, In some magical days, by a kind of luck it don't happens, or it syncs alone so many minutes later, mysteriously.
I would really appreciate if the OBS team could fix it, making OBS start up in the set sample rate. Or even let the user manually choose the correct ( I mean desirable) sample rate.
44k 24-Bit is the standard sample rate for almost every Pro Audio Studio.
It shouldn't be forced to higher sample rates without the user intention and/or permission. It's not supposed to happen.
Please, consider to fix it in a new update.
Thank you very much! #audio issue #44k #samplerate

 

[AaronD]

44k 24-Bit is the standard sample rate for almost every Pro Audio Studio.

44.1k is a holdover from CD's, which themselves arrived there because of how the already-existing video standards worked (yes, recording audio as if it were TV video), and not much else.

Digital audio needed videotape to be possible - and the early days were wild!

The history here is wild!Links 'n' stuff:Oh, here's the mentioned link before I forget:https://www.kenrockwell.com/audio/sony/pcm-f1.htmTechmoan's video:http...

www.youtube.com

Full video:

Digital audio needed videotape to be possible - and the early days were wild!

The history here is wild!Links 'n' stuff:Oh, here's the mentioned link before I forget:https://www.kenrockwell.com/audio/sony/pcm-f1.htmTechmoan's video:http...

www.youtube.com

If all you're ever doing is going to be audio only, then you might use 44.1k or a multiple of it. (88.2k, etc.) But if you're including video, pretty much everyone that does that uses 48k or a multiple. (96k, 192k, etc.) DVD's and Blu-Rays, for example, are 48k, not 44.1.

Since we can only hear below 20k, there's no point in reproducing sound faster than 40k, and both the CD standard of 44.1 and the video standard of 48 give a nice buffer to allow a lowpass filter to work. The only reasons to sample faster than that (88.2k, 96k, etc.) are:

• Actual slow-motion sound, to make higher frequencies audible that weren't before. (not the physically unrelated sound effects that are often added to slo-mo videos on YouTube)
• Cheaper, more gradual filters. Still start rolling off at 20k - beyond what *anyone* can hear, including audiophools - but it takes another octave or two to get down to an acceptably low level.
• Less latency for real-time systems. This is on the scale of inches or centimeters at the speed of sound in air (I have a project in the works that does exactly that, so I've already done that math), so it's completely irrelevant for broadcast or recording.

If you record at one sample rate and want to release in another, resampling is old news now, even live. Bad resamplers are of course audible, but the ones that actually do the (ancient now and publicly available) math *correctly* are completely transparent. Not a problem at all.

 

[Getúlio Prates]

44.1k is a holdover from CD's, which themselves arrived there because of how the already-existing video standards worked (yes, recording audio as if it were TV video), and not much else.

Digital audio needed videotape to be possible - and the early days were wild!

The history here is wild!Links 'n' stuff:Oh, here's the mentioned link before I forget:https://www.kenrockwell.com/audio/sony/pcm-f1.htmTechmoan's video:http...

www.youtube.com

Full video:

Digital audio needed videotape to be possible - and the early days were wild!

The history here is wild!Links 'n' stuff:Oh, here's the mentioned link before I forget:https://www.kenrockwell.com/audio/sony/pcm-f1.htmTechmoan's video:http...

www.youtube.com

If all you're ever doing is going to be audio only, then you might use 44.1k or a multiple of it. (88.2k, etc.) But if you're including video, pretty much everyone that does that uses 48k or a multiple. (96k, 192k, etc.) DVD's and Blu-Rays, for example, are 48k, not 44.1.

Since we can only hear below 20k, there's no point in reproducing sound faster than 40k, and both the CD standard of 44.1 and the video standard of 48 give a nice buffer to allow a lowpass filter to work. The only reasons to sample faster than that (88.2k, 96k, etc.) are:

• Actual slow-motion sound, to make higher frequencies audible that weren't before. (not the physically unrelated sound effects that are often added to slo-mo videos on YouTube)
• Cheaper, more gradual filters. Still start rolling off at 20k - beyond what *anyone* can hear, including audiophools - but it takes another octave or two to get down to an acceptably low level.
• Less latency for real-time systems. This is on the scale of inches or centimeters at the speed of sound in air (I have a project in the works that does exactly that, so I've already done that math), so it's completely irrelevant for broadcast or recording.

If you record at one sample rate and want to release in another, resampling is old news now, even live. Bad resamplers are of course audible, but the ones that actually do the (ancient now and publicly available) math *correctly* are completely transparent. Not a problem at all.

Thanks Aaron, to join the conversation, you took good information. But my clear attempt is to ask OBS Studio developers to debug it. It changes the sample rate without control and leaves my audio interface out of sync. OBS Studio should launch in the given sample rate, instead of forcing another sample rate. That's it. I hope it's clearer by now.