Oh, this is trolling right?

 

172/192khz will eliminate most of the issues with aliasing however I have found that even at 96Khz one can mitigate the harmonics better than 44/48khz. As far as the computing power goes I have also found that 96khz is a decent compromise between two extremes. Even with that said for smaller projects running the session and waveforms @ 192Khz makes more sense from the start.

At this point I will venture to say that there are plugins that should be avoided for two reasons. The first one being what you stated above - plugins that are incompatible with the SR in the first place. The other reason would be in the case of plugins that have oversampling/anti-aliasing permanently turned ON at all times with no OFF switch. I have already trash-binned a couple of plugins I had laying around the production box due to the fact they were unable to handle sample rates above 48khz. There is an ample supply of plugins available in the pool that makes discarding one for another not such a difficult task anymore.

I would also venture to say that if you're taking the route of applying brickwall low-pass filters with the intent on mitigating plugin based aliasing harmonics you better make dam sure that plugin is well designed so as to not make matters worse by adding in it's own harmonics and distortions.

 

Last edited: Aug 27, 2023

If human hearing has a range of 20Hz - 20Khz and your project is running at 96Khz. Using the Nyquist theorem the maximum frequency which can be represented would be 48khz

For oversampling to be neccessary you'd have to have to exceed 48Khz. Lets's say you exceeded by 10,000 cycles which is 58Khz.

This will result in distortion at 38Khz (48khz - 10khz). I mean who the heck can hear as high as 38khz.

Just to hear distortion at 18khz which is within the human hearing range, you'd have to exceed 48khz by 30khz ( 78khz).

My point is recording at higher sample rates would make more sense if you're trying to avoid oversampling.

Just saying

 

By jumping to a sample rate which exceeds Nyquist (2x highest frequency) unnecessarily bloats your files. If you want a 96khz sample of your kick drum, you may have other things going on. Raising the sample rate of the session to 96khz does not even come close to 32X resampling anyway. If my session is 24 bit, 48khz; 96khz is the equivalent of 2X oversampling. So a 32X result is not within reach at a 96khz session, or even close. If it is too much for your CPU; you can always just bounce an offline render, reload the resulting wav file, and disable the plugin/freeze, or whatever you have to do. They released an update of the plugin in a day.

 

Last edited: Aug 26, 2023

Not coming at you, but try and keep pyracy/sister-site affairs reined in on AS.

 

Actually, yes; you most certainly are. You'll be on ignore from now on.

 

Just wait for a bug fix

 

Exactly, unless you're doing extreme amounts of distortion like with a high gain guitar amp plugin, 2x oversampling is usually enough, even when the DAW is running at 44.1kHz.

 

Who died and made you moderator?

 

Keep acting cute behind the keys—meanwhile, either of you ever wonder why plugin developers don't fuck with this site or the people on it, meanwhile other sites get mad love and attention from them? Wise up.

 

THERE ARE OTHER SITES???

 

Could be because marketing reasons?

Because to be honest, I do not care about them as well.

 

Last edited: Aug 27, 2023

The whole point is to eliminate the use of plugin's oversampling "features", because there is no standard that dev's adhere to when they are implementing them. It's kind of a mixed bag... Unless one tests every single plugin one uses and knows exactly how it operates and what it injects into the mix you're really flying blind... I'm not apt to doing that even... Personally I am more at ease leaving the oversampling/aliasing "features" on plugins alone and just run my sessions and waveforms at the higher sample rates. The audio sounds better anyway, and if it helps eliminate headaches with harmonics it's a win-win.

And for those monkeys who say there is no difference in quality of audio at higher sample rates, I invite them to re-visit digital audio basics 101 and re-learn how analog signals converted to the digital domain actually works. If I can hear the difference in quality and you all can't that says something right there.

 

it's pretty hard to sell new junk at the base of a landfill the size of a mountain of other broken junk. And in this case, on both platforms of common Operating Systems! We're not talking about development with a user base on 50 different Linux distributions. It is Windows and Mac.
Putting something on to the internet where your customers can buy it, and then damage their equipment checking out it's supposed features is pretty unprofessional and i would not want to speak about that either. Your average community college student with 10 credit hours of programming training would not upload something like this. They would at least just grey it out and put it in an update. This plugin would get someone at a real company fired. They can type all the cute little notes they want, but product liability lawsuits blowing up peoples monitors with your demo program could get very expensive. i just wanted to see this error.

 

Last edited: Aug 29, 2023

Good idea...
https://www.xiph.org/video/vid2.shtml

 

with 96kHz project, what you are doing technically is equivalent of running all the plugins to 2x Oversampling on a 48kHz project. How is that can possibly be better aliasing-reduction-wise compared to running project at 48kHz with say 16x OS (equivalent of plugin internal processing at 768kHz) ? there has been hundreds of discussions when it comes to this subject and the ultimate answer is always run the project at the final product sample rate, usually 44.1kHz or 48kHz and if you are using hybrid workflow, capture (A to D) it with another computer with 96kHz so you'll get all the sonic extension the analog gears introduced.

96kHz render file then sent to mastering engineer so they'll decide the downsampling (filtering) to 48kHz during the mastering process for the final result so they can get everything sonically controlled because they have all the tool to accurately listen to it. That's how it is done professionally. Atleast that's how it is here, in my country for as long as I know. Pretty sure it's still the same last week.

Some mastering engineer will downsample to end result sample rate first before starting to master. Which make everything become a joke on its own.

If you are going to do everything yourself, it's pointless and waste of resources because you might end up with different result than what you mixed caused by the final downsampling filtering. The filtering will change the sound how high quality the filtering algo is, it's about quality control aesthetically, so it's common to do it early, working at project sample rate of your end result so you can maintain the integrity of mix the best you can. Strange that people can hear the difference between the 96kHz and 48kHz but didn't think ahead of what it might caused. It's like making things pristine as fuck then accept it when the quality is intentionally degraded at the very end because the final result need to be lower fidelity than the one mixed. The whole music production stages are about taking things to a higher level, not the other way around. Even worse you can't control it and let some downsampling filtering algo boi decide it for ya. Sounds kinda pointless don't you think ?

Unless you want to release Hi-Res file (Tidal etc) which might need even the recording done at 96kHz and it's a whole different story and I never done this before, so don't know much about it and never take 96kHz mix job ever. It's painful enough to listen to bad music in lower resolution, don't want to torture myself. lol.

 

Last edited: Aug 28, 2023

No idea what you're on about there...

I master (final dithering) my stuff (pure audio) to 16Bit/44khz (no idea where you're getting 48khz unless you're talking about audio for video) because that is standard audio CD resolution and where I desire it to be for final distribution. There are no interim's or steps involved - it's basically bounce all finished project tracks to audio at 96 - 192Khz (whatever my project rez is depending on how large the project. the project/session rez never changes from whatever it's started at) and the tracks are worked applying final eq, compression, FX etc, and when a completed and approved final mix is achieved, a render is made/dithered down to 16b/44k, and done. No black magic, no spells, no potions - mastering is not rocket science. Live with it a while... Play it around on different systems/situations and see if it holds up. If a tweak needs to be made here or there simply rinse and repeat.

I have noticed as well with audio that was recorded and mixed at the higher rez's and then dithered down (once) to a final distribution resolution spec the resultant distro copies are generally more well tempered and transparent. Most of that has to do with how well it was recorded and mixed but just generally it has been something I have noticed since running sessions/projects at the higher sample rates. By the way, just a fun fact here, but I have never walked into any audio studio's who were doing work commercially that were running their recording/mixing operations at 44/48khz - their standards are always either 96khz, 192k, or even higher than that. If someone were to walk into their places demanding a session be recorded and mixed @ 44/48khz they would most likely get laughed right out the door.

just a final fyi, I do not master stuff using brick-wall limiters on the stereo bus making a final waveform that looks like a solid square box or rectangle. I master for max dynamic range FIRST - then tweak for as much loudness as possible without making the track sound like it has all the dynamic range of a pile driver. The pile-driver theory seems to be the norm for mastering engineers these days which I have scoffed at for as long as it has seemingly been a thing. It's ridiculous...

 

Last edited: Aug 28, 2023

+1 here. Lost two days trying to figure out what was destroying my render (i thought it was a problem on the instrument) until I turned off the Avalon and it worked.

 

That's really kinda fucked up since the actual avalon analog hardware is fairly top notch.

 

Has anyone compared this versus the hardware?