DDMF Metaplugin will let you oversample anything. Reaper also has native oversampling options. But if you’re running your session at like 88.2k or above, Decap really doesn’t alias in an audible way unless you’re realllllly pushing it.
I never tried the DDMF, and I basically just heard about it.. or I completely forgot about it lol. But the thing is - it costs $70. I would expect $150+ plugins to have oversampling option in them. And most people are not using sample rates higher than 48k.. You overload your CPU very fast by doing that.
Totally agree Decap should have native oversampling! Just offering workarounds for anyone like me who loves the plug and can’t give it up. I have FF Saturn as well it’s just not as quick for me to get a decent result.
Until you use some kind of additional distortion/compression on top of whatever plugin does that. And eventually made louder during mastering and thus cluttering your mix with noise that should not be there. Especially if multiple instances are used.
@@johnbird9997 not really. it will kill your cpu. what you want to do instead is to use oversampling on every track that has a lot of highs. That also goes for the mix bus..
This was a great video, thanks for pointing this out, I do have quite a few plugins with the high quality/oversampling feature, I'll be using those for saturation moving forward!
Geez, these guys are leading people down unnecessary rabbit holes. MixBusTv pulled this exact same crap with intermodulation. This stuff happens all day with phase, dc offset, intermodulation, Nyquist etc etc. they’re reading forums and taking it out of context or lack of knowledge of its general use. If these are problems then just mix in 96k and above, its pros out weigh the cons, with time stretching and many other things. Now here’s an important note, none of these or problems, they’re tools. You can use phase and intermodulation for shaping things, I use phase on my subs to counter the room and low ringing to have accuracy with the lows. Use Nyquist for guitars or cymbals, add some bite to it. All they do is effect the sound, like it or don’t but don’t get caught up, mix and keep mixing and pay attention to your speakers excursion, you’re engineering the speakers output and your body is it’s best measuring tool. If I’m coming off as a troll, let me know, “mixbus” took it personally and maybe my approach is inappropriate. I’m not a tuber or forum dweller so I could be out of touch. Nothing personal to the guy’s channel, it’s these titles.
@@starkid9736 if you wanna crap on anything do it to DC offset. It shifts your mix balance and can damage speakers systems, plus we can’t hear it, useless. But also take a more advanced approach from the rest like phase for example, maybe you can use it as a eq for proximity or stereo like Hass. Get people to be problem solvers and actual engineers.
There's a super-simple solution. Let's say you have a 48K project. If you have a sound that would benefit from oversampling, raise your DAW's sample rate *temporarily* to 192 kHz. Render the track (synth, amp sim, saturation, whatever). When you go back to 48 kHz, your DAW will sample-rate-convert the 192 kHz track back to 48 kHz, and there will be no aliasing because you rendered at the higher sample rate and turned the non-alised audio into audio that can no longer alias at 48 kHz. Sample-rate conversion is superb these days, and you're not asking it to do any irrational number-based math with 48 kHz. One more thing: the reason why plugins don't default to oversampling is because it draws a lot more CPU power. But now you know how to solve the problem without having to oversample at the plugin itself. Oh, and one MORE thing: offline sample rate conversion is esentially perfect. Plugins with oversampling need to use sample rate conversion algorithms that work in real time, which may or may not be perfect, depending on the priorities for CPU consumption and latency.
There are elements modulation synthesis where you use types of side band modulation to achieve new sounds… so it’s not always the same ….. you can use …
Today, I got to not only test the range of my reference speakers, but my own hearing as well! It was great to see a visual representation of all that extra noise and harmonics that we tend to not notice as passive listeners. I am really tempted to go back and listen to that on my headphones, to see how their range stacks up. (when I am not working) Didnt get any response from the doggos this time, but I do think that one of those frequencies matches the constant sound I get from my tinnitus......... Keep up the excellent work Grump!
Here's the thing though. Look at the peak volume of your sine wave in those high frequencies. Then, take an ACTUAL recording of a musical instrument and look where your peaks are. If you have something in the 7-10K range peaking at -6dbfs or higher, that's going to take your head off. In real life, those volumes in that area are MUCH lower than a sine wave in these types of tests. Look at a equal loudness chart to see what I mean. High frequencies are perceived as the same volume at much lower dbs than lower frequencies. Aliasing is absolutely a problem and in 2024, any plugin worth its salt should have some sort of anti aliasing filter. That said, I personally don't think its as much as a problem as you're making it out to be. Since those peaks in the high frequencies are so much lower, the fold back harmonics will be super quiet and almost out of the hearing range, if not completely unnoticeable. These can definitely build up over a lot of tracks though, so it's helpful to know if it's happening and saturate appropriately. TLDR: Test your tools and make informed decisions based on the source material.
Dan Worral has made some excellent videos on aliasing and how it affects the sound of our mixes Personally I havent run into much issues with aliasing, because I don't tend to run heavy saturation on what already has a lot of high freq content. the only place I've found myself using oversampling is when running high gain amp sims, and even there its not immediately obvious that there's aliasing. though it does sound slightly cleaner in an A/B comparison. In some instances you might even find aliasing preferable. it has a different sound than analog saturation
I’m so worried about this. I haven’t really noticed it and I use it extensively on every mix and master I produce. Recently I master an amazing track and I experiment with a lot of plugins like Waves Magma Tubes, Softube, Waves Kramer tape, UAD, etc. I finally settled with a touch of Waves Magma Tubes that gave a gentle “something cool and punchy” to the signal. THANKS FOR THIS AMAZING VIDEO. I will use it with my most advanced students.
There is a very simple way to avoid drastic issues with plugins that don't have oversampling: Don't push them too hard. I know you are slamming the drive all the way up for the sake of demonstration, but the reality is that for most mixing situations you won't be doing that. Saturation is all about finding the sweet spot, where it gives just that little bit of edge you're looking for. When you use it like that, any and all aliasing issues are much less drastic than your demo makes them out to be. They'll still exist, sure, and oversampling might give you a bump in clarity, but the non-oversampled version won't sound bad.
@@azbassclarinetI don't believe it is. I'm willing to bet next time you use saturation in a mix, you will try it with and without oversampling, if the option is there. If it didn't make a sonic difference, they wouldn't include it as it would just be a waste of resources. And to add to this, unnatural harmonics are much more noticeable in a mix, than musically related ones are.
you very (un)helpfully used a spectrum analyser without y-axis labels. Just how loud are those aliased harmonics? No antialiasing filter is perfect, but if it's able to cut the aliased frequencies by -60dB then there isn't really a problem.
I think 96k could potentially run into some aliasing if your using a lot of saturation on a mix but even then it would be minimal. I’d rather not have to record at 96k though so I’m considering learning how to use fab filter Saturn at a higher level so I can make it my main plugin. I love decapitator though…
just fyi yes the 10k is very audible --- anything up to around 16k will be audible for most people. 19k is pretty audible on its own on youtube.. 19k and above is very faint but still audible. (also being in my 40s and previously working as a bouncer in a nightclub I am thankful I am still perceiving the 19Khz+ range.
Does the resulting mix when saturated with such a plugin (with no oversampling or anti alliasing) sound "bad" per se ?! When using a saturation plugin on a track, usually I just dial in the amount that sounds "good" to me on my headphones and speakers. I check for gain levels before and after the plugin and tame some frequencies if necesarry with a multiband eq / compressor or limiter. If it sounds good together with the rest of the mix and gain levels are in the green its fine for me. Maybe you can help me understand the issue a bit better? Anyway thank you for sharing your insights and personality with us!
basically what you're doing is introducing some "phantom frequencies" (let's call it that so it would be easier to understand), and in some cases (as shown in the video) they will overpower the fundamental. Once they do, to your ear - they're becoming the new fundamental. So let's say you're saturating drums (live drums as an example), and your snare starts sounding weird.. like it's overpowering the rest of the drums, or like it's being out of phase with something - it could be that the cymbals' harmonics are aliasing back and making that phantom frequency become the new fundamental. So naturally they will collide with the rest of the drums. On the other hand, let's say you're saturating a synth - you will get the same issue, but this time it will start introducing harmonics that are out of tune with itself. Which in the end results in a very digital or plastic-y sound. And when all those weird phantom frequencies start appearing in a full mix, you're just going to get a mix that is full of shit that you don't even know it's there, but together it'll build up and make it a disaster. In other words, it could clean up your mix, and when it's soloed - it just sounds unnatural. So basically, it's better to prevent a disease than to cure it.
I been kinda obsessed with saturation for a while so I've done a bunch of a/b and the thing i found is i don't actually think aliasing sounds bad most of the time even when it's cranked. But it does sound less clear which may or may not be good
That's why i'm still in analog, i hate the pre and postringing introduced by aliasing. If i have to work in digital, i use airwindows stuff cause the design of the plugins is accounting for all those problems arising from aliasing.
Ya but you can barely hear it unless using extreme distortion.. i do all my recordings in 44.1 and ive never had a single person say anything .. the only ppl who care about this is engineers.. yes its good to know .. but am i gonna stop using decapitator ... no lol
The problem with that approach is that you'll use saturation multiple tracks and it'll add a ton of stuff in the midrange, which will make your mixes sound bad when all of it bunches up - but you won't know why that's happening.
Awesome video! As a Cubase user, I'm curious to see if the stock saturation on the channel strip produces the same negative results. 🤔 Would you mind doing a test for that?
Was listening to a playlist of what's currently popping in the mainstream world..heard aliasing on almost every example on the choruses and "drops". Clearly not a problem. I have no dog in this fight. I think claiming that aliasing is the number 1 thing preventing plugins emulating analog sounding analog is the most asinine thing i hear come out of peoples mouths who work ITB. If it bothers you, you can always just upsample before rendering out your multitracks etc, but it's not going to make it sound more analog, and when sound intermodulate they are for all intents and purposes aperiodic. Unless you make music with purely sustained sinewaves, aliasing is not going to bother too many people. I said i heard it on that playlist, but i am not sure many people would. It's just a bit of extra graininess on the tails of cymbals and such. that sound has been omnipresent in the large majority of audio release for almost two decades now though. You may even find that some consumers may find the absence of it unpleasant
This is an absolutely wonderful explanation of saturation and the flaws in the plugin. It explains some of the issues I have been having using them and why I tend not to. But now I know what I am looking for. Subscribed and looking forward to more great content!
Are you going to to a video on what saturation plugins are good to use? that would be awesome. Thanks for going into such depth in this video. SUBCRIBED! Legend
@@GrumpyEyeAudio i felt you explained it well. but for an ousider like myself i still need to study it a little more. But i learned something i didn't even know it was a thing. i would like to see if you could go on a program like "Magix Music Maker" or atleast what is your opinions on this kind of softwares as a musician.
subbed, since you already starte,d how about running a group test and showing the results, I'm sure you'd get lots more subs then. What about all the EQ and compressors, analog synth emulations and everything else that may be doing some sort of distortion to add character, would be great to know technically which ones are the most capable. Most of us will never have the time to test and not everything can be tested before purchase.
All plugins that can do oversampling, will have an AA filter inside of it. I use Saturn. But I will review every saturation plugin that gets released from now on. Currently downoading uad verve. :)
That’s really interesting. I have a some of saturation plugins (uad, harmonics, Decap, Spectre) and I am not always happy with the result’s. I switch many times until I get the wanted sound. Maybe that’s the reason. I planed to buy Saturn (because other reason), however, that anti aliasing-thing convinced me now even more, that Saturn probably will be a good invest.
@@GrumpyEyeAudio Depends on the plugin. Metaplugin nests plugins inside its processing and it can oversample on their behalf. I don't use it but it exists and it works.
The aliasing is happening internally in the plugin itself. Loading it up into DDMF will not help solve the issue. Nor would it help to use a filter afterwards
@@GrumpyEyeAudio No problem! There are some old plugins which have an internal limit to their mathematics which means they can't operate past 96kHz operation (i.e. 2x OS if you're working in 44.1 or 48), but even the Waves Abbey Road series can be oversampled to higher levels. I saw a list somewhere that tells you the highest rate for most major plugins.
No, not unless you could insert the EQ/filter *inside* the saturation plugin. (Well, you could reduce the high frequencies going into the saturation plugin, thereby reducing the potential for things to go beyond Nyquist, but that would obviously just muffle the whole sound, which is probably not what you'd want.)
Fuckin love deeper knowledge shit like this😎👍 really interesting. So the Fab plug-in is at least able to avoid this behavior with the high quality setting, as far as I understood? Have to check that out. I am really happy with the Fab bundle. One of the best purchases when it comes to Plugins I think. I sadly do not have the money for good hardware so I am mainly on software. Beside the mpc..
some manufacturers give you an option to use oversampling. But most of them don't. And then, there's a minority that will use oversampling, but they won't give you an option to turn it off. The good thing with having an option to toggle it, is that you can decide when the VST is going to use more resources. Because oversampling can be CPU intensive. Example with fabfilter - don't turn it on if you're using it on bass. There's no high end on it, so whatever aliasing you get - it's going to be masked by signal itself. But if you're using it on the full mix, or on anything that has any high frequencies - go into oversampling mode. My main problem are plugins that add harmonic distortion even if it's not their main thing. As an example if the compressor or EQ you want to use adds saturation to your signal, then you need to know if it is oversampling.. if not, you shouldn't use it on the overheads
There's actually a much simpler solution to solving the Nyquist issue in regards to saturation: only saturate the signal where it's effective, namely in the mid frequencies (there's some limited use for it on low frequencies (i.e. that crunchy 808 bass thing seems to be really popular with the kids these days, although they typically use hard clippers for that) but, personally, I don't ever saturate high frequency program material...not saying that you can't, I'm just saying that I don't). That's what I like about Saturn, the fact that each frequency band can be saturated individually. Or not. Not only does this get around any aliasing problems without the need for oversampling, it opens up your options with respect to saturation plugins. And, while I have you here, I just wanted to confirm that you're not alone with how you feel about Life. I saw someone else using it a couple of weeks ago but I can't remember who it was (might have been Virtual Riot but, not sure). I definitely plan on acquiring it but I'm not spending $150 on it so I'll have to be patient and wait for it to, either, go on a half off sale at some point or wait for it to make its way to Audionews. Definitely a fun gadget that automates a lot of processes I use to create those kinds of semi-randomized glitch-blip grooves. A lot of cool stuff can be generated by making use of follow actions in Ableton and combine that with legato, their onboard beat repeat device and boom, Bob's your uncle.
@@GrumpyEyeAudioOh absolutely. I tried it a handful of times and was never pleased with the results. Particularly with tape saturation the margin for error is so small, it's more of a pain in arse than it's worth to me to try and get it right I don't feel it's needed (I cut my teeth back in the day on 24TK Otari 2" tape machines and holy balls do I love DAWs). It can sound fine at first but by the time you hit the front of upward compression or heavy limiting, what you thought was subtle saturations blooms into a really unpleasant distortion. I'm all about clippers, parallel compression, really good limiters and manually editing ISPs where they occur. I don't even mess with saturation anymore (though I'm sure I'll have another go at some point). It took a good while to get a process worked out that results in something that is transparently clean, really loud (on par with professional productions) and still sounds open and isn't just smashed to sh!t. I just found your channel by chance yesterday (though this clearly isn't your first channel). Subbed, I like your sense of humor. 👍
But can work in some situations, I feel. Sometimes I'll put an highpass in the beggining of the chain and repeat it on the end as I feel it gets back the clarity that the subsquent processing might had clouded. I don't know if this is a good practice but please, don't laught at our ignorance, cure us. Maybe there's some content to develop there. Thanks. @@GrumpyEyeAudio
If the plugin does have the problem you mentioned, what exactly would it cause to the mixing? You said it's gonna cause problems, but what problem exactly?
Hmm... I haven't done these tests personally, but I assume that you can put a lowcut after the saturation plugin. I mean this doesn't excuse the plugin crap, and it will deteriorate the effect. But hey, I don't have time for patchbays and analog stuff. Thanks for the great demo though.
You can't. It's internal. It needs to be done internaly. When the plugin is done "saturating", you get the full signal with all the added frequencies (even the ones that are being folded back). So introducing a low pass filter afterwards is simply not an option. Sure you can high-pass it (by eyeballing it), but would you really do it with the overhead mics on a drum kit?
It's not about hi-passing your signal (if you're not using linear phase you're totally going to screw up the phase of your low end), the key is to only (or mostly/typically...there are not hard and fast rules in audio) saturate the mids. Those you can crunch up without running harmonics up and beyond Nyquist. Oversampling will be necessary if you decide to saturate the high frequencies for the reasons outlined in the video. If the saturator you're using doesn't have OS, then don't saturate the upper frequencies because once you render aliasing into your audio, that's kind of it (there might be clever algorithms/AI that might be able to but best practice is much like defense in karate: The best defense is don't be there when the punch arrives).
As Grumpy said, unfortunaly that won't work. It needs to be done within plugin. Just like adding a plugin later in chain that does use OS won't fix the aliasing caused by earlier plugins in the chain. You could almost think in theory that would work but unfortunately it doesn't work that way. 🙂
Andrew Zeleno. It's an audio engennier and youtuber that doesn't hype anything (except his own classes), and tests every plugin to the bone. And much more. @@GrumpyEyeAudio
Anything that has oversampling. Just like shown in the video. You can of course work at 192kHz and it will get rid of aliasing, but your cpu will not like it.
