This is one of those videos that TH-cam should show on the recommended list for practically everyone interested in DAWs and Plugins. Please do update the title so that it has the word oversampling and it is easier to notice and find
Wow, Thank you and Sam for this great talk! About high sample rates 31:30 That's has been my thinking for about 10 years and happy for the confirmation from you. I just work in high sample rate and if needed I down-sample the master *once* in the end instead of stacking tons of conversions and filters which are not in my control. I try to get from the beginning the source material already in the high sample rate and upsample only what's necessary. I believe this produces cleaner result. I got criticism for this with claims it's a waste of resources but as you said it might actually be the opposite and save resources. I guess it is hard to test and compare because oversampling algorithms are so varied. Also when working with high sample rates from the beginning you have complete control over the conversion and you don't need to think about it while working! Another bonus is if your material were recorded with high sample rate microphones or the track was synthesizes originally this way, if you need to pitch it down it sound amazing. Search Sanekn 100k microphone recording pitched down.
@@MusikHack Thank you! I actually worked with a guy who told me "your paranoid who don't believe in science because oversampling is oversampling - perfect and always the same". This video is a great overall introduction that I can now reference people too!
This was an incredibly illuminating conversation from two people who are very knowledgeable about audio. I was particularly struck by the concept that there can be "audio events" outside of our audible spectrum that can affect all downstream processing, as well as the section about how oversampling can affect your headroom. Thanks for putting this together and sharing with the world.
Good stuff. Everyone is looking for the magic thats makes it easy. Reminds me of Einstein’s famous quote: “Make things as simple as possible, but no simpler.” This guest and you bring out the best in each other. Top level pro discussion.
An excellent discussion, thanks. Reminded me of my music college days, back in the 90s, learning about the Nyquist theorem. Back then 41kHz was the standard, these days I use 48kHz at 24-bit depth; it works for me. 😃
Yes, as Sam said the filters used in oversampling are one of the tricky parts. That's probably why sometimes oversampling can sound worse than without it when it's not needed.
Finally, a proper explanation, without losing sight of what actually matters. 'Half magic, half science' is SPOT ON. Sound is, yes, absolute. But our ears are all different!
Yep. That might explain why the most appreciated Dan Worrall's track is the Loudwar one, wich was intended to sound terrible hahah As a loud music amateur, i'm am genuinely one of those btw Per exemple, I have no idea on what average LFU (minus 2??) this dude is running, but he's considered one of the GOAT in his specific genre. If not, the absolute GOAT th-cam.com/video/H2vM4dPFupY/w-d-xo.html
I'm more of the stacking approach, I prefer distortion f.e. instead of compression, . So it gets louder, (clips the waveform) and I keep my headroom because I can do it gently.
@@Projacked1 one reason I love clipping over limiting is, the reverbs & the general feeling of spaciousness doesn't get lost! ..Limiters seem to destroy that completely, to my ears. One awesome track where I guess that clipping was used during mastering instead of limiting, is the awe-inspiring "No Surprises" by Radiohead :) ..Similar to tracks like "Riders on the Storm" & "L.A. Woman" by The Doors, these reverbs just sound so stunning, absolutely spectacular! (pretty sure it's a real EMT140 Plate on all of them) Back to my point, nowadays I tend to only ever use limiters on dry sources, before any spatial effect.
Great conversation! Really, your member content is a great addition to your standard channel content. Worth every cent of support thrown your way! Keep it up!
This channel... I don't even know what to say. You're both so entertaining and such a great teacher (not to mention able to see multiple perspectives)! Thank you for this content, I always look forward to your videos.
One thing to note is: oversampling/anti-aliasing is not always the best option. Even if the anti-aliasing is coded well and work fine, you have to consider if the filter itself is min-phase or linear-phase. If it's min-phase it will cause some phase shifting in the highs, and if it's linear-phase it will cause some pre-ringing (not that audible because it's in the highs though) and a lot of latency. If the plugin you are using isn't doing a lot of saturation or compression, it could be better to just leave anti-aliasing disabled (e.g when the saturation is mainly in the low-end or when the folded down harmonics are below -100dB). You also have to consider that sometimes less experienced devs code the filters very poorly and they therefore create more problems than they solve. I've even seen cases when they have placed the hi-pass filter after the downsampling, making it technically useless. Don't just enable anti-aliasing/oversampling by default in every plugin you use. Listen carefully if it helps, or not. It's not always better with anti-aliasing enabled.
Ah! We didn’t even bring this up but it’s true: some devs will just make straight bad decisions instead of choosing among the good ones… Just listen when you flip that switch!
When using non-linear processes such as limiter, saturators, compressors and clippers you also run into intermodulation. It can be more offending than aliasing (and sometimes the sonic effect confused with) because the sum or substract of two (or more) different frequencies creates new frequencies which are not harmonically related with. And even below the original frequencies! You should look into that.
Great interview. One of the reasons I went analog is the lack of control in the digital domain. You don’t know what happens under the hood and how each plug-in interact. You only know they will .
v-clip is actually a clipper that lets you choose which oversampling filter type to use. just in case anyone here wanna see for themselves what the difference is like, get that plugin. it's also a great clipper in general :)
@@jeremycaulfield8467 the juce logo comes up in all plugins that have been made with juce, where the developer chose the free juce license but didn't wanna make the code open source
I can agree with what Sam says. I am very sensitive to high frequencies (even though I am 43) and I can definitely hear the difference between oversampled signals and non-oversampled signals. Every time I do blind tests, I find that oversampled always sounds more pleasant to my ears and non-oversampled stresses me out. But there are many in the industry who don't hear the difference and therefore claim that everyone else is a weirdo or esoteric. This annoys me a lot. Just because I don't have cat eyes, I still believe cats see better at night than I do.
Dan Worrall's video on chain oversampling in Reaper showed a different conclusion. Chain oversampling is not as good as individual oversampling. The reason for this is that the harmonics keep creeping higher and higher if you do not filter them away. So, the first distortion you put will perhaps produce harmonics up at 40khz. The next distortion plugin will distort those 40khz leftover harmonics, producing new ones at 80khz, and so on. So, eventually you'll hit the Nyquist frequency and start getting folding. But if you instead keep individually upsampling and downsampling, you do not get a build-up or "harmonic creep" (hey, I just coined a new term) because you keep clearing away the ultrasonic harmonics, giving you a clean slate for each individual plugin. Yes, this means there's an argument to be made for not oversampling your entire project, but processing your project as 48khz with individual oversampling where needed.
Hm, I can see this argument. If that becomes a problem though, you could put a simple high quality low-pass filter at 20khz a few times in the chain instead of oversampling over and over though. If it’s steep, that’s best of both worlds: don’t convert up and down a bunch with questionable filters, and if your processing keeps pushing up aliasing, cut it off. I’ll do some testing to determine what type of filter might work best for that problem, and thanks for the perspective.
@@MusikHack Yes, you could do it manually. But you could also just let it be automated though individual oversampling, perhaps to avoid cluttering your FX chain with a bunch of filter plugins. It's a shame things have to be so complicated. :P There's rare a simple, best answer for everything.
@@antiHUMANDesigns even better, reach out to chain oversampling developers and ask them to add a feature that does this automatically between plugins on the chain and there’s no clutter. No reason it has to be this way! If the current state of things can be improved, it is developers like myself that can improve them, and that is why we do this research!
@@MusikHack ..so we're basically gonna see lots of marketing around "THE BEST inter-oversample-chain-filters" in ten years from those big plugin companies :D :D
this is the nerdiest of any audio youtube nerd video...i absolutly loved every min of it , and learnt alot, also still have many questions in this niche theory...
Agreed on the true peak point. I hate limiters at the best of times, but true peak often sounds much worse to me. I often rely on clipping transients that are too fast to hear and the way some limiters react to 'digital' but inaudible peaks sounds awful.
Thanks for doing this guys. Very interesting. Did want to especially thank you for the acknowledgment that people hear differently. Back in the 70's I had a studio with a partnet and he would do mixes and I would hear these really high end distortions sometimes. He never noticed it until I pointed it out and even then it didn't bother him. Made me crazy. Found out it was a bad condenser or something, but, to this day I hear weird things that other's don't. But, now I have analyzers to show them. haha. I'm SAVED. lol. Cheers
What a fantastic interview & discussion, it certainly gave the subject some much needed clarity. Techy without being baffling. Thanks to both of you, much appreciated
This is one of the best videos about oversampling I have seen. Also Sams input is so sympathetic and competent. I will absolutely looking at Musichack in the future :-) Thanks to both of you!!
What a great video! Thank you guys! Personaly, I rarely use oversampling, almost never. Just beacause I can hear how it changes the audio in most cases and I don't like it. I want to recomend listening to this change at low volume and paying attention to the texture, dynamics and depth. Oversampling recalculates all this stuff and it wouldn't be the same after that. Especialy when we do it on a master or on a buss, where it also affects all reverbs in the mix too. Finaly you guys have discussed this topic with that level of detail and said out loud that oversampling is not a "better" button. More of this kind of content would be great! Thanks a lot!
Great review! So it seems that we cannot avoid the oversampling accumulation if we work in an oversampled environment in the DAW, because some plugins have a built-in oversampling already.
I develop a Java Synthesizer and want to introduce oversampling. I think it makes sense to calculate any signal with double sample rate and finally filter it with a FIR sinc Filter to cut off anything sharply above Nyquist. After doing that, it should be safe to delete every second sample.
You are speaking to a real deal. Signal processing expert, not many can answer these questions at all. Pure signal processing which is a field within electrical engineering.
CHAIN OVERSAMPLING QUESTION: Is it enough if I just use Oversampling in last plugin in the chain or do I need to use some kind of container plugin (KSHMR Chain?) and oversample whole chain in it? Basically - How to do chain oversampling?
We are all pretty sick about kick ducking or how to use an EQ videos. Fortunately, from time to time there are also videos like this, with wich we actually learn something. Great stuff.
Proper deep, nice one. It also seems sensible if oversampling has to be used that not too many different brands are worked with, as they can all use different techniques which may not stack up so well.
The problem with high sample rates - because aliasing still exists (albiet to a lesser extent than say 44.1 / 48) is that those components interact with the audible range via intermodulation distortion. So then you say well, why don't i just run a low pass filter at the edge of the audible range? then you're introducing either phase shift with minimum phase filtering, or a linear phase filter which introduces latency and the same 'ringing' overshoots / transient smearing you get via oversampling. There's no free lunch in digital audio! you just have to decide what the lesser of the evils are based on the processing you're doing.
how it was explained to me was: Like tape/analog distortions, where you are dealing with SNR, low bass, etc. there are issues in the digital domain you also need to minimise, they're just _different_. if you look at samples, romplers, and samplers (even using samples on your DAW) you also get into sample interpolation, and things become far more complicated. this is where you could make an argument for working at higher bit depth and rate, because you give yourself a LOT more headroom for these issues. the best solution in the studio, like with tape, is to listen. you're just listening for /different/ things - phase distortion, aliasing noise, ringing. and keep in mind digital aliasing/artifacts can be HF enough to become psychoacoustic... I.e. if you feel fatigued or anxious working on a song, you may have a digital distortion problem. (I would recommend, find some self-produced tracks from the 00's especially early Ableton, and listen to the quality of the high end to familiarise yourself with what digital distortion/aliasing sounds/feels like)
OK. Interesting thing. BOZ Digital (he is a genius), developed EQ without aliasing and oversampling. No idea how but they did. THE HOSER (XT). I would like to know how they did it and if that works for other plugins stacked in chain (with Hoser at the end). Will (The Hoser) help with Aliasing?
So for “out of the box” analog processing vs. oversampled digital processing the following chains are compared: “filters in DAC -> analog effect -> filters in ADC” vs. “filters in upsampling -> digital oversampled effect -> filters in downsampling”. The question is, are the filters in DAC and ADC different from the filters used in “in the box” oversampling and what’s the difference? If they are the same, there should be no difference between analog and digital effect (if digital model is perfect, of course).
Great opportunity for somebody to test! Off the cuff, my hunch would give the contest to the digital domain. Not because it’s actually better, but because analog gear will have a unit cost for the converters and so are incentivized to use cheap ones where they can, and in the analog realm you also have to contend with line noise, which probably adds up faster than post-oversampling aliasing noise. For very expensive gear and setups, this argument doesn’t hold though. Any analog hardware guys here with a hot take? 😅
Thank you so much for the video man. 12:03 I think the best thing to do for manufacturares is what Melda did. Just put option for high oversampling rates for rendering.
Running at a high sample rate has other issues. When you drop the sample rate between plugins you clear out inaudible high frequencies (ultrasound). These frequencies are a problem as (e.g.) intermodulation distortion in later plugins can make more distortion when the ultrasound reacts with the sound we can hear. In short, downsampling clears up a bunch of inaudible* mess. *yes, really** **unless you're a dog or bat.
there's only one (extremely expensive) A/D converter in the world that can convert at 32bit. ALL others convert at 24bit, because the dynamic range (signal to noise ratio) of 24bit far exceeds the analogue noisefloor of basically ANYTHING other the ultra-extreme measurement equipment. (32bit float is a digital format, existing only WITHIN the digital realm, ensuring pretty much infinite dynamic headroom for any process) ..further, you might be intrigued to hear that Dan Lavry, designer of the best and most renowned converters currently existing, claims that any samplerate above 96Khz in the conversion process, reduces the accuracy and therefore defeats the purpose. Meaning, according to this, 192kHz would only make sense for processing within a digital format, just like 32bit does.
..on that note, if you record through a UAD preamp emulation on one of their interfaces, in that case recording at 32bit would make sense again, since that emulation is partly a digital process and therefore probably outputs in 32bit float :)
Really interesting topic. My background goes back to the tape era, and all the analog equipment we emulate today. And what is striking me funny right now... is this discussion about acceptable levels of digital *noise* as we create programs designed to emulate the analog *noise* caused by the unsolvable problems with the last medium. Ultimately if it sounds good, it is good. And since you have no control over what anybody else hears, just go with what works for you. "Classic rock" is full of terrible mixes that somehow sound great. 😊
EXCELLENT discussion! LUCKY are those who work in grungy hard rock or Lo-Fi, because the audio issues caused by aliasing won’t matter nearly as much. 👍 But for everything else it’s a problem that needs to be attended to. I have a lot of high-frequency hearing loss, but I can STILL easily hear the difference between a plugin WITH good over sampling and even a good plugin without over sampling. That being said… There are still years and years and years worth of great sounding plugins WITHOUT over sampling that have produced great sounding recordings. “If it sounds good, it IS GOOD” is an entirely valid quote. I think the whole point of this discussion is that good over sampling can and does make a great sounding plugin sound even better. NOTICEABLY better in many cases.
This was waaayyy more interesting than I though it would be. The plugin market is really moving fast at the moment, maybe you will get satisfying analogue emulations real soon. The look on your face Wytse, when he said ADAA. Finally a little about audio you didn't know xD. Would love to see a part 2 of this interview and also hear some of his opinions on future development on both aliasing but also plugins in general. - Love from Denmark as usual!
I can't vouch for how stable it is, but DDMF Metaplugin is a plugin host within a plugin (so you can create complex chains within one DAW insert slot) and I think that has a "global oversampling" rate so might be a way to replicate Reaper chain oversampling?
Even being at an amateur level and recording as a hobby these are the exact things that plagued my first recordings and mixes for the first few years because I was totally unaware of what was happening. After a ton of research and time I've seen vast improvement in the high end area using oversampling, brickwall filters set very high, and using more correct rendering processes than in the past but it's still hilarious that so many softwares are available to musicians but something as simple as rendering to mp3 without it degrading the mix 10x more than expected as a an mp3 would degrade it anyway can be extremely difficult without the proper information. I would expect these distributors of new software to allow a smarter interaction where the software actually takes care of the technical details, dithering when needed, upsampling if needed, brickwall filtering, a one set standard of sampling rate, but it leaves me wondering if the industry purposely leaves these pitfalls and traps to ensure that artists will always need to rely on the "pro's"of the past, instead of allowing talent that may exist out there to truly be able to break out on their own, more easily and more often.
Great vid !! Thanks for taking the lead down this rabbit hole. I always thought that alias-signals would add upp on the master bus. And furthermore, in chains of non-linear processing any alias-signal created in the first non-linear process stage, will contribute to create new alias-signals in the following stage(s) of non-linear processing. This yields an exponential growth of alias-signals, for the number of stages of non-linear processing. With several tracks having multiple stages of non-linear processing, and recognizing that musical audio is far mor complex, and therefore much more alias-prone, than a 1 khz test audio signal, aliasing would probably be audible in many mixes/masters. Perhaps not distinguishable as aliasing, but definitely part of the 'sound'. I have been looking to buy DDMF metaplugin, bc it hosts VST's in pro tools. Instruments and FX. Further, it allows for the Reaper-trick of chain-oversampling. Having listened to your discussion I now decided to purchase the metaplugin. I'll make attempts at higher project sample rates (96 or more), and do chain oversampling using the metaplugin. Any thoughts? thx again.
Haven’t used meta plugin yet myself! Just remember: if you’re working at a higher rate, you should need oversampling much less often: no need to solve the same problem two ways unless you really hear it.
dude sorry for the interruption but now than you talking about loudness , i wonder you two cents about the L.R.A measurement in the loudness meter in case of the restriction than the bbc introduce?
This makes wonder: should I use DSEQ on all my bus groups to prevent a "build up" of weird stuff? I always use it on my master bus but now I'm starting to think I might having to do something about every bus.
Hi Wytse, been quietly following you for a few years now & have always loved your videos. I was interested by your comment that your RME's run the best at 96k & I'm REALLY curious how you came to that conclusion. Is it a subtle difference in how it sounds to you, or something you've done objective testing for...? Very interested to learn more on this point.
My question to developers:| Is it at all possible to leverage an AI process to figure out the best form of oversampling (or some other algorithm) that will perform the same job but extremely efficiently and effectively? Is that something that will occur some time in the future? Would be amazing to leverage AI in audio, not just for this, but for emulation (if it's even possible). Thanks for the video. Definitely one every sound engineer should watch!
Regarding what is said at 14:13, SIR standard clip lets you choose the type of filter, the filter kernel size and the filter cutoff (which is represented graphically)
Hm, I keep trying to comment on this but it seems to get auto-moderated away. Overcooker wasn't intended for public release, but feel free to reach out through the links above and we can chat there.
So, what is oversampling in easy to understand terminology? When digital audio is processed, especially with certain types of audio effects like distortion, compression, or time-stretching, high-frequency content can be introduced or altered. This can lead to aliasing, which is the appearance of unwanted frequencies in the audio spectrum. Aliasing occurs when frequencies higher than half the sample rate (the Nyquist frequency) are not properly represented, resulting in alias frequencies folding back into the audible range. To mitigate aliasing, oversampling is employed. In simple terms, oversampling increases the sample rate of the audio signal being processed by the VST plugin. For example, if the original audio signal has a sample rate of 44.1 kHz, oversampling at 2x would double the sample rate to 88.2 kHz, and at 4x, it would quadruple the sample rate to 176.4 kHz. When the plugin processes the audio at this higher sample rate, it can capture and handle higher-frequency content more accurately. After processing, the oversampled audio is then typically downsampled back to the original sample rate to match the project's settings. Oversampling can significantly improve the audio quality of certain VSTs, particularly those that involve nonlinear operations (e.g., distortion) or very steep filters (e.g., high-order EQs). By reducing aliasing, it helps to retain more of the original audio's clarity and reduces unwanted artifacts in the final output. However, oversampling can be computationally demanding, and some plugins offer options to choose the level of oversampling or even turn it off to balance performance and audio quality based on the user's preferences and computer capabilities.
@@MusikHack That might be the kindest reply i got for years. Thx buddy TBH, i guess i will mostly focus on how those precious informations should impact my current and future workflows, at least for now. For a start i guess i will start my new sesions at 96k sample rate and freeze tracks more often ; and also stop oversampling everything blindly without listening what it brings to the table. The trick of oversampling plugins chains instead of each plugin individually is also neat (i'll load Metaplugin way more often i guess) Funny thing btw: this morning i blindly put FF Pro-MB on 4x OS mode on a percussions group track. A little later i compared it versus no OS. And guess what: in my ears the non oversampled version sounded better. And a few hours later this video comes up... Quite sick!
@@KOjoeBeats sounds like you’ve got a good grasp of things! If 96k is stressing your system too much, the advice to listen first will get you plenty far.
@@KOjoeBeats Lol same at first! Take this interview as a seed. Don't stress over it too much, just try to work into the new and learn at your own pace
Just a comment.. I usually use 48kHz when recording, but when mastering I upsample the whole session in Studio One (and sometimes when mixing as well). Then automatically, upsampled cache files of the audio are created in the background. I have just being doing this from a feeling it was a good thing to do, and getting the plugins working in higher freq. Maybe not a bad idea to try out more consciously. When the work is done the cache is deleted, and archiving diskspace is less... Hmm..🤔😀
Did some analyzing using sonic visualiser on my latest mix. Oversampling was clearly reducing aliasing on my mix. I also bumped into a few plugins which caused aliasing and replaced them by fabfilter alternatives. So thanks to this video my music is a couple % better than it was. Also using standardCLIP before the L2 limiter gave me a couple extra % quality on my final result.
Also, once a plug-in introduces aliasing, it’s killing when the next plugin in line is a saturator. The aliasing went through the roof. So if you have an option to toggle on oversampling, do it or replace the plugin with one that doesn’t Alias if possible.
@@dARTfader QUestion. If I use plugin with oversampling at the end of chai will it resolve all aliasing problems (or some) caused by previous plugins? Or do I need to oversample every plugin by itself (that couses aliasing)?
@@mikelo303 Ask an expert ;-). Oversampling can also introduce high frequency phaseshifting if it doesn't use linear phase. Maybe you can place an oversampling plugin after the one creating aliasing, i don't know...i'm just a noob :-D
30:00 the easiest solution to lazy engineers (like me) deal with all this hassle... yes, 96 / 192 it will consume a lot of CPU but todays pcs and a good converter will deal with that without any problem. I'm already doing that for some years and it really works great to me. Glad to hear some skilled people suggesting it.
I haven’t watched this video, commenting just because of the title. After 15 years of producing, when I try to ‘oversample’, it just sounds different to me, not better, not worse. I think I feel this because the sample rate of my synths and my drums and everything are all different. Anyway, gunna watch video now
7 minutes in, you’re talking about what is audible and what is not. I’ve never been able to tell the difference between a 320 and a wav, when many people say it’s obvious
This is one of those videos that TH-cam should show on the recommended list for practically everyone interested in DAWs and Plugins. Please do update the title so that it has the word oversampling and it is easier to notice and find
It will be updated soon
What a great person to interview, a clear speaker who explains very complex concepts in a way I could follow. Get him back on again!
Wow, Thank you and Sam for this great talk!
About high sample rates 31:30 That's has been my thinking for about 10 years and happy for the confirmation from you.
I just work in high sample rate and if needed I down-sample the master *once* in the end instead of stacking tons of conversions and filters which are not in my control.
I try to get from the beginning the source material already in the high sample rate and upsample only what's necessary.
I believe this produces cleaner result.
I got criticism for this with claims it's a waste of resources but as you said it might actually be the opposite and save resources. I guess it is hard to test and compare because oversampling algorithms are so varied.
Also when working with high sample rates from the beginning you have complete control over the conversion and you don't need to think about it while working!
Another bonus is if your material were recorded with high sample rate microphones or the track was synthesizes originally this way, if you need to pitch it down it sound amazing.
Search Sanekn 100k microphone recording pitched down.
It’s a wise move when you can make it happen!
@@MusikHack Thank you! I actually worked with a guy who told me "your paranoid who don't believe in science because oversampling is oversampling - perfect and always the same".
This video is a great overall introduction that I can now reference people too!
@@kaori-3882 I wonder who might that be... 😜
@@katoseichi7984 😂😘
This was an incredibly illuminating conversation from two people who are very knowledgeable about audio. I was particularly struck by the concept that there can be "audio events" outside of our audible spectrum that can affect all downstream processing, as well as the section about how oversampling can affect your headroom. Thanks for putting this together and sharing with the world.
yes, also a great demonstration of two people trained in analog vs. digital.
Good stuff. Everyone is looking for the magic thats makes it easy. Reminds me of Einstein’s famous quote: “Make things as simple as possible, but no simpler.” This guest and you bring out the best in each other. Top level pro discussion.
An excellent discussion, thanks.
Reminded me of my music college days, back in the 90s, learning about the Nyquist theorem. Back then 41kHz was the standard, these days I use 48kHz at 24-bit depth; it works for me. 😃
I love thease dev talks even tho I underatand half of it it still gives us some insight of all the work required to make great plugs.
Yes, as Sam said the filters used in oversampling are one of the tricky parts. That's probably why sometimes oversampling can sound worse than without it when it's not needed.
Finally, a proper explanation, without losing sight of what actually matters.
'Half magic, half science' is SPOT ON. Sound is, yes, absolute. But our ears are all different!
Yep. That might explain why the most appreciated Dan Worrall's track is the Loudwar one, wich was intended to sound terrible hahah
As a loud music amateur, i'm am genuinely one of those btw
Per exemple, I have no idea on what average LFU (minus 2??) this dude is running, but he's considered one of the GOAT in his specific genre. If not, the absolute GOAT
th-cam.com/video/H2vM4dPFupY/w-d-xo.html
@@KOjoeBeats to be fair, it IS Dan's best track lol
I'm more of the stacking approach, I prefer distortion f.e. instead of compression, . So it gets louder, (clips the waveform) and I keep my headroom because I can do it gently.
@@Projacked1 one reason I love clipping over limiting is, the reverbs & the general feeling of spaciousness doesn't get lost! ..Limiters seem to destroy that completely, to my ears. One awesome track where I guess that clipping was used during mastering instead of limiting, is the awe-inspiring "No Surprises" by Radiohead :)
..Similar to tracks like "Riders on the Storm" & "L.A. Woman" by The Doors, these reverbs just sound so stunning, absolutely spectacular! (pretty sure it's a real EMT140 Plate on all of them)
Back to my point, nowadays I tend to only ever use limiters on dry sources, before any spatial effect.
@@lennartrauhe yeah me too, just a 4:1 ratio with 1-2 dB attenuation on an SSL compressor works really well. It doesn't smash the signal to hell
Cool interview. Always nice to hear a developer explain and demystify things in a clear but detailed way.
Man this is ONE HELL of a feature. Thanks Wietze and Sam, I absolutely LOVED this content.
Great conversation! Really, your member content is a great addition to your standard channel content. Worth every cent of support thrown your way! Keep it up!
This channel... I don't even know what to say. You're both so entertaining and such a great teacher (not to mention able to see multiple perspectives)! Thank you for this content, I always look forward to your videos.
Great discussion and questions. Thank you for conducting this interview with Sam.
One thing to note is: oversampling/anti-aliasing is not always the best option. Even if the anti-aliasing is coded well and work fine, you have to consider if the filter itself is min-phase or linear-phase. If it's min-phase it will cause some phase shifting in the highs, and if it's linear-phase it will cause some pre-ringing (not that audible because it's in the highs though) and a lot of latency. If the plugin you are using isn't doing a lot of saturation or compression, it could be better to just leave anti-aliasing disabled (e.g when the saturation is mainly in the low-end or when the folded down harmonics are below -100dB).
You also have to consider that sometimes less experienced devs code the filters very poorly and they therefore create more problems than they solve. I've even seen cases when they have placed the hi-pass filter after the downsampling, making it technically useless.
Don't just enable anti-aliasing/oversampling by default in every plugin you use. Listen carefully if it helps, or not. It's not always better with anti-aliasing enabled.
Ah! We didn’t even bring this up but it’s true: some devs will just make straight bad decisions instead of choosing among the good ones… Just listen when you flip that switch!
This video was way better than I was expecting. Super informative and worth the long runtime. I may even revisit it due to its wealth of information.
When using non-linear processes such as limiter, saturators, compressors and clippers you also run into intermodulation. It can be more offending than aliasing (and sometimes the sonic effect confused with) because the sum or substract of two (or more) different frequencies creates new frequencies which are not harmonically related with. And even below the original frequencies! You should look into that.
This is incorrect in a practical sense. You run into intermodulation issues when you start using stuff like sinefold distortions and such.
Great interview. One of the reasons I went analog is the lack of control in the digital domain. You don’t know what happens under the hood and how each plug-in interact. You only know they will .
v-clip is actually a clipper that lets you choose which oversampling filter type to use. just in case anyone here wanna see for themselves what the difference is like, get that plugin. it's also a great clipper in general :)
My go-to clipper. Totally underrated.
I love V clipper . Weird that the juice logo comes up but great clipper
@@jeremycaulfield8467 the juce logo comes up in all plugins that have been made with juce, where the developer chose the free juce license but didn't wanna make the code open source
I can agree with what Sam says. I am very sensitive to high frequencies (even though I am 43) and I can definitely hear the difference between oversampled signals and non-oversampled signals. Every time I do blind tests, I find that oversampled always sounds more pleasant to my ears and non-oversampled stresses me out. But there are many in the industry who don't hear the difference and therefore claim that everyone else is a weirdo or esoteric. This annoys me a lot. Just because I don't have cat eyes, I still believe cats see better at night than I do.
I am 44 and I can hear the same stress you are talking about. For a long time I dod not know what it was but I could hear it. Now I know
Love that analogy at the end
Please do more technical videos like this
Dan Worrall's video on chain oversampling in Reaper showed a different conclusion. Chain oversampling is not as good as individual oversampling.
The reason for this is that the harmonics keep creeping higher and higher if you do not filter them away.
So, the first distortion you put will perhaps produce harmonics up at 40khz. The next distortion plugin will distort those 40khz leftover harmonics, producing new ones at 80khz, and so on. So, eventually you'll hit the Nyquist frequency and start getting folding.
But if you instead keep individually upsampling and downsampling, you do not get a build-up or "harmonic creep" (hey, I just coined a new term) because you keep clearing away the ultrasonic harmonics, giving you a clean slate for each individual plugin.
Yes, this means there's an argument to be made for not oversampling your entire project, but processing your project as 48khz with individual oversampling where needed.
Hm, I can see this argument. If that becomes a problem though, you could put a simple high quality low-pass filter at 20khz a few times in the chain instead of oversampling over and over though. If it’s steep, that’s best of both worlds: don’t convert up and down a bunch with questionable filters, and if your processing keeps pushing up aliasing, cut it off. I’ll do some testing to determine what type of filter might work best for that problem, and thanks for the perspective.
@@MusikHack Yes, you could do it manually. But you could also just let it be automated though individual oversampling, perhaps to avoid cluttering your FX chain with a bunch of filter plugins.
It's a shame things have to be so complicated. :P There's rare a simple, best answer for everything.
@@antiHUMANDesigns even better, reach out to chain oversampling developers and ask them to add a feature that does this automatically between plugins on the chain and there’s no clutter. No reason it has to be this way! If the current state of things can be improved, it is developers like myself that can improve them, and that is why we do this research!
@@MusikHack Yes, that'd be a solution.
@@MusikHack ..so we're basically gonna see lots of marketing around "THE BEST inter-oversample-chain-filters" in ten years from those big plugin companies :D :D
this is the nerdiest of any audio youtube nerd video...i absolutly loved every min of it , and learnt alot, also still have many questions in this niche theory...
Wow, more of these please! I have the attention span of a goldfish and I watched the whole thing in one go with great interest.
Agreed on the true peak point. I hate limiters at the best of times, but true peak often sounds much worse to me. I often rely on clipping transients that are too fast to hear and the way some limiters react to 'digital' but inaudible peaks sounds awful.
Thanks for doing this guys. Very interesting. Did want to especially thank you for the acknowledgment that people hear differently. Back in the 70's I had a studio with a partnet and he would do mixes and I would hear these really high end distortions sometimes. He never noticed it until I pointed it out and even then it didn't bother him. Made me crazy. Found out it was a bad condenser or something, but, to this day I hear weird things that other's don't. But, now I have analyzers to show them. haha. I'm SAVED. lol. Cheers
i did my head in trying to work out oversampling years ago, cheers for showing me i wasnt crazy
What a fantastic interview & discussion, it certainly gave the subject some much needed clarity. Techy without being baffling. Thanks to both of you, much appreciated
I'm amazed that it took until 30 minutes into this to give the answer. That said, fantastic interview, this is a really informative video.
Wow, this was a big rabbit hole of an incredible, in-depth discussion about oversampling. Really enjoyed this. No, I’m not a snake oil salesman.
This is one of the best videos about oversampling I have seen. Also Sams input is so sympathetic and competent. I will absolutely looking at Musichack in the future :-) Thanks to both of you!!
What a great video!
Thank you guys!
Personaly, I rarely use oversampling, almost never. Just beacause I can hear how it changes the audio in most cases and I don't like it.
I want to recomend listening to this change at low volume and paying attention to the texture, dynamics and depth.
Oversampling recalculates all this stuff and it wouldn't be the same after that. Especialy when we do it on a master or on a buss, where it also affects all reverbs in the mix too.
Finaly you guys have discussed this topic with that level of detail and said out loud that oversampling is not a "better" button.
More of this kind of content would be great!
Thanks a lot!
Great video. Cool to hear a developer give their insight.
Great review! So it seems that we cannot avoid the oversampling accumulation if we work in an oversampled environment in the DAW, because some plugins have a built-in oversampling already.
I develop a Java Synthesizer and want to introduce oversampling. I think it makes sense to calculate any signal with double sample rate and finally filter it with a FIR sinc Filter to cut off anything sharply above Nyquist. After doing that, it should be safe to delete every second sample.
You are speaking to a real deal. Signal processing expert, not many can answer these questions at all. Pure signal processing which is a field within electrical engineering.
CHAIN OVERSAMPLING QUESTION:
Is it enough if I just use Oversampling in last plugin in the chain or do I need to use some kind of container plugin (KSHMR Chain?) and oversample whole chain in it? Basically - How to do chain oversampling?
What an amazing interview, thank you guys!
We are all pretty sick about kick ducking or how to use an EQ videos. Fortunately, from time to time there are also videos like this, with wich we actually learn something. Great stuff.
Proper deep, nice one. It also seems sensible if oversampling has to be used that not too many different brands are worked with, as they can all use different techniques which may not stack up so well.
The problem with high sample rates - because aliasing still exists (albiet to a lesser extent than say 44.1 / 48) is that those components interact with the audible range via intermodulation distortion. So then you say well, why don't i just run a low pass filter at the edge of the audible range? then you're introducing either phase shift with minimum phase filtering, or a linear phase filter which introduces latency and the same 'ringing' overshoots / transient smearing you get via oversampling. There's no free lunch in digital audio! you just have to decide what the lesser of the evils are based on the processing you're doing.
how it was explained to me was: Like tape/analog distortions, where you are dealing with SNR, low bass, etc. there are issues in the digital domain you also need to minimise, they're just _different_.
if you look at samples, romplers, and samplers (even using samples on your DAW) you also get into sample interpolation, and things become far more complicated.
this is where you could make an argument for working at higher bit depth and rate, because you give yourself a LOT more headroom for these issues.
the best solution in the studio, like with tape, is to listen. you're just listening for /different/ things - phase distortion, aliasing noise, ringing.
and keep in mind digital aliasing/artifacts can be HF enough to become psychoacoustic... I.e. if you feel fatigued or anxious working on a song, you may have a digital distortion problem.
(I would recommend, find some self-produced tracks from the 00's especially early Ableton, and listen to the quality of the high end to familiarise yourself with what digital distortion/aliasing sounds/feels like)
OK. Interesting thing. BOZ Digital (he is a genius), developed EQ without aliasing and oversampling. No idea how but they did. THE HOSER (XT). I would like to know how they did it and if that works for other plugins stacked in chain (with Hoser at the end). Will (The Hoser) help with Aliasing?
Awesome interview I hope you can speak with this guy more on other aspects ! 💪🏼🙏🏼
Sam seems like such a friendly guy. Really enjoyed this talk even though much was simply beyond my reach of knowledge. really fascinating topic
Thank you for sharing this!
Fantastic video. This is one of the reasons I really enjoy your channel and am keen to see how it further evolves.
This is an amazing interview. Thanks a lot for this. awesome.
So for “out of the box” analog processing vs. oversampled digital processing the following chains are compared:
“filters in DAC -> analog effect -> filters in ADC” vs. “filters in upsampling -> digital oversampled effect -> filters in downsampling”.
The question is, are the filters in DAC and ADC different from the filters used in “in the box” oversampling and what’s the difference? If they are the same, there should be no difference between analog and digital effect (if digital model is perfect, of course).
Great opportunity for somebody to test! Off the cuff, my hunch would give the contest to the digital domain. Not because it’s actually better, but because analog gear will have a unit cost for the converters and so are incentivized to use cheap ones where they can, and in the analog realm you also have to contend with line noise, which probably adds up faster than post-oversampling aliasing noise. For very expensive gear and setups, this argument doesn’t hold though. Any analog hardware guys here with a hot take? 😅
Thank you so much for the video man. 12:03 I think the best thing to do for manufacturares is what Melda did. Just put option for high oversampling rates for rendering.
Running at a high sample rate has other issues. When you drop the sample rate between plugins you clear out inaudible high frequencies (ultrasound). These frequencies are a problem as (e.g.) intermodulation distortion in later plugins can make more distortion when the ultrasound reacts with the sound we can hear. In short, downsampling clears up a bunch of inaudible* mess.
*yes, really**
**unless you're a dog or bat.
Thanks a lot for this great interview. I learnt a lot and i respect more what developers of audio software a facing to do their work well.
This is why I would track my vocals at 32 bit float at 192 kHz, process the vocals there, when finished bounce to 24/48, then start mixing.
there's only one (extremely expensive) A/D converter in the world that can convert at 32bit. ALL others convert at 24bit, because the dynamic range (signal to noise ratio) of 24bit far exceeds the analogue noisefloor of basically ANYTHING other the ultra-extreme measurement equipment. (32bit float is a digital format, existing only WITHIN the digital realm, ensuring pretty much infinite dynamic headroom for any process)
..further, you might be intrigued to hear that Dan Lavry, designer of the best and most renowned converters currently existing, claims that any samplerate above 96Khz in the conversion process, reduces the accuracy and therefore defeats the purpose. Meaning, according to this, 192kHz would only make sense for processing within a digital format, just like 32bit does.
..on that note, if you record through a UAD preamp emulation on one of their interfaces, in that case recording at 32bit would make sense again, since that emulation is partly a digital process and therefore probably outputs in 32bit float :)
..all I was saying is, recording at 96kHz 24bit will bring the best results, probably reduce latency and safe you a lot of storage space :)
This is the cleverest audio engineer I’ve ever seen… I understand half of it, the rest I’ll give over to oversampling…
Excellent and very informative conversation!
Really interesting topic. My background goes back to the tape era, and all the analog equipment we emulate today. And what is striking me funny right now... is this discussion about acceptable levels of digital *noise* as we create programs designed to emulate the analog *noise* caused by the unsolvable problems with the last medium.
Ultimately if it sounds good, it is good. And since you have no control over what anybody else hears, just go with what works for you. "Classic rock" is full of terrible mixes that somehow sound great. 😊
Fantastic video! Answered a lot of questions in my head
i think this is your best video ever!! would love to see more content like this!
EXCELLENT discussion!
LUCKY are those who work in grungy hard rock or Lo-Fi, because the audio issues caused by aliasing won’t matter nearly as much. 👍
But for everything else it’s a problem that needs to be attended to.
I have a lot of high-frequency hearing loss, but I can STILL easily hear the difference between a plugin WITH good over sampling and even a good plugin without over sampling.
That being said…
There are still years and years and years worth of great sounding plugins WITHOUT over sampling that have produced great sounding recordings.
“If it sounds good, it IS GOOD” is an entirely valid quote.
I think the whole point of this discussion is that good over sampling can and does make a great sounding plugin sound even better. NOTICEABLY better in many cases.
Incredible Conversation!!!
Educational content like this is great. Thanks for your effort.
This was waaayyy more interesting than I though it would be. The plugin market is really moving fast at the moment, maybe you will get satisfying analogue emulations real soon.
The look on your face Wytse, when he said ADAA. Finally a little about audio you didn't know xD. Would love to see a part 2 of this interview and also hear some of his opinions on future development on both aliasing but also plugins in general.
- Love from Denmark as usual!
Wytse!! Where can I get Musik Hack Overcooker from???
I can't vouch for how stable it is, but DDMF Metaplugin is a plugin host within a plugin (so you can create complex chains within one DAW insert slot) and I think that has a "global oversampling" rate so might be a way to replicate Reaper chain oversampling?
thanks for the conversation! great one! any chance to get the overcooker plugin Sam is using?
great video on the more tech savy aspect of oversampling! to add on that: dan worrall video about oversampling.
Even being at an amateur level and recording as a hobby these are the exact things that plagued my first recordings and mixes for the first few years because I was totally unaware of what was happening. After a ton of research and time I've seen vast improvement in the high end area using oversampling, brickwall filters set very high, and using more correct rendering processes than in the past but it's still hilarious that so many softwares are available to musicians but something as simple as rendering to mp3 without it degrading the mix 10x more than expected as a an mp3 would degrade it anyway can be extremely difficult without the proper information. I would expect these distributors of new software to allow a smarter interaction where the software actually takes care of the technical details, dithering when needed, upsampling if needed, brickwall filtering, a one set standard of sampling rate, but it leaves me wondering if the industry purposely leaves these pitfalls and traps to ensure that artists will always need to rely on the "pro's"of the past, instead of allowing talent that may exist out there to truly be able to break out on their own, more easily and more often.
Amazing interview. About oversampling chain, you mean one instance of ddmf (for example) at the begining of the chain and one at the end?
Great vid !! Thanks for taking the lead down this rabbit hole. I always thought that alias-signals would add upp on the master bus. And furthermore, in chains of non-linear processing any alias-signal created in the first non-linear process stage, will contribute to create new alias-signals in the following stage(s) of non-linear processing. This yields an exponential growth of alias-signals, for the number of stages of non-linear processing. With several tracks having multiple stages of non-linear processing, and recognizing that musical audio is far mor complex, and therefore much more alias-prone, than a 1 khz test audio signal, aliasing would probably be audible in many mixes/masters. Perhaps not distinguishable as aliasing, but definitely part of the 'sound'.
I have been looking to buy DDMF metaplugin, bc it hosts VST's in pro tools. Instruments and FX. Further, it allows for the Reaper-trick of chain-oversampling. Having listened to your discussion I now decided to purchase the metaplugin. I'll make attempts at higher project sample rates (96 or more), and do chain oversampling using the metaplugin.
Any thoughts?
thx again.
Haven’t used meta plugin yet myself! Just remember: if you’re working at a higher rate, you should need oversampling much less often: no need to solve the same problem two ways unless you really hear it.
I would like to know what you think about the Airwindows plugins that he made for this ultra-high frequency filtering thing.
dude sorry for the interruption but now than you talking about loudness , i wonder you two cents about the L.R.A measurement in the loudness meter in case of the restriction than the bbc introduce?
This makes wonder: should I use DSEQ on all my bus groups to prevent a "build up" of weird stuff? I always use it on my master bus but now I'm starting to think I might having to do something about every bus.
Thank you for talking about over sampling. I understand the idea, but it feels most of the time I oversample, I lose clarity on the elements.
Really awesome discussion.
Hi Wytse, been quietly following you for a few years now & have always loved your videos. I was interested by your comment that your RME's run the best at 96k & I'm REALLY curious how you came to that conclusion. Is it a subtle difference in how it sounds to you, or something you've done objective testing for...? Very interested to learn more on this point.
Should have said this with the first comment. I'm interested because I want to find out what the sweet spot for my converters are 😆
My question to developers:|
Is it at all possible to leverage an AI process to figure out the best form of oversampling (or some other algorithm) that will perform the same job but extremely efficiently and effectively? Is that something that will occur some time in the future? Would be amazing to leverage AI in audio, not just for this, but for emulation (if it's even possible).
Thanks for the video. Definitely one every sound engineer should watch!
41:08 - Oh yeah, _"the more you know, the more you know you don't know."_ I call it _"Dunning-Kruger Strikes Back!"_ 😂
Great content as always! I love your videos, probably the best education in audio on TH-cam
Regarding what is said at 14:13, SIR standard clip lets you choose the type of filter, the filter kernel size and the filter cutoff (which is represented graphically)
Excellent video and fascinating interview Wytse! More of this please! 🙏
Awesome interview!
This is the technical video type I always search from you :) Thanks!
What about brick wall Low Pass filtering before saturation/distortion?
Really valuable conversation--Thanks!
Is it wrong that i cut my highs in my mixes but the. Boost in the mastering phase?
That's a bit like Dolby tape noise reduction in reverse :)
"Good, quick, cheap. Pick two." Its amazing how often this same tradeoff appears across many disciplines.
Great topic and guest. Bravo White Sea!
PS: Is that "Overcooker" oversampling plug-in available anywhere?
Can't find any trace of it on the web anywhere. My rabbit hole for the last hour...
@@barx3218 Just replied to the parent comment!
Hm, I keep trying to comment on this but it seems to get auto-moderated away. Overcooker wasn't intended for public release, but feel free to reach out through the links above and we can chat there.
🤯Wow! Thanks for that input!
So, what is oversampling in easy to understand terminology?
When digital audio is processed, especially with certain types of audio effects like distortion, compression, or time-stretching, high-frequency content can be introduced or altered. This can lead to aliasing, which is the appearance of unwanted frequencies in the audio spectrum. Aliasing occurs when frequencies higher than half the sample rate (the Nyquist frequency) are not properly represented, resulting in alias frequencies folding back into the audible range.
To mitigate aliasing, oversampling is employed. In simple terms, oversampling increases the sample rate of the audio signal being processed by the VST plugin. For example, if the original audio signal has a sample rate of 44.1 kHz, oversampling at 2x would double the sample rate to 88.2 kHz, and at 4x, it would quadruple the sample rate to 176.4 kHz.
When the plugin processes the audio at this higher sample rate, it can capture and handle higher-frequency content more accurately. After processing, the oversampled audio is then typically downsampled back to the original sample rate to match the project's settings.
Oversampling can significantly improve the audio quality of certain VSTs, particularly those that involve nonlinear operations (e.g., distortion) or very steep filters (e.g., high-order EQs). By reducing aliasing, it helps to retain more of the original audio's clarity and reduces unwanted artifacts in the final output. However, oversampling can be computationally demanding, and some plugins offer options to choose the level of oversampling or even turn it off to balance performance and audio quality based on the user's preferences and computer capabilities.
This is thee best OS deep dive I’ve ever seen
Super interesting guys. Thanks very much for sharing ✌🏼
Ayaaaa this becomes ridiculously more technical than i ever expected
LOL sorry my guy, anything specific you want to know I can get straight to the point for ya here…
@@MusikHack That might be the kindest reply i got for years. Thx buddy
TBH, i guess i will mostly focus on how those precious informations should impact my current and future workflows, at least for now.
For a start i guess i will start my new sesions at 96k sample rate and freeze tracks more often ; and also stop oversampling everything blindly without listening what it brings to the table.
The trick of oversampling plugins chains instead of each plugin individually is also neat (i'll load Metaplugin way more often i guess)
Funny thing btw: this morning i blindly put FF Pro-MB on 4x OS mode on a percussions group track. A little later i compared it versus no OS. And guess what: in my ears the non oversampled version sounded better. And a few hours later this video comes up... Quite sick!
@@KOjoeBeats sounds like you’ve got a good grasp of things! If 96k is stressing your system too much, the advice to listen first will get you plenty far.
@@KOjoeBeats Lol same at first! Take this interview as a seed. Don't stress over it too much, just try to work into the new and learn at your own pace
Best video so far . Kudos !
Very cool video. Thank you guys.... 🎧
Great Interview! Thanks for it 🙏🏼
Just a comment.. I usually use 48kHz when recording, but when mastering I upsample the whole session in Studio One (and sometimes when mixing as well). Then automatically, upsampled cache files of the audio are created in the background.
I have just being doing this from a feeling it was a good thing to do, and getting the plugins working in higher freq.
Maybe not a bad idea to try out more consciously. When the work is done the cache is deleted, and archiving diskspace is less... Hmm..🤔😀
White Sea, What's the skin care routine!? We need a full video breaking it down 🤣 🙏
Did some analyzing using sonic visualiser on my latest mix. Oversampling was clearly reducing aliasing on my mix. I also bumped into a few plugins which caused aliasing and replaced them by fabfilter alternatives. So thanks to this video my music is a couple % better than it was. Also using standardCLIP before the L2 limiter gave me a couple extra % quality on my final result.
Also, once a plug-in introduces aliasing, it’s killing when the next plugin in line is a saturator. The aliasing went through the roof. So if you have an option to toggle on oversampling, do it or replace the plugin with one that doesn’t Alias if possible.
@@dARTfader QUestion. If I use plugin with oversampling at the end of chai will it resolve all aliasing problems (or some) caused by previous plugins? Or do I need to oversample every plugin by itself (that couses aliasing)?
@@mikelo303 Ask an expert ;-). Oversampling can also introduce high frequency phaseshifting if it doesn't use linear phase. Maybe you can place an oversampling plugin after the one creating aliasing, i don't know...i'm just a noob :-D
30:00 the easiest solution to lazy engineers (like me) deal with all this hassle... yes, 96 / 192 it will consume a lot of CPU but todays pcs and a good converter will deal with that without any problem. I'm already doing that for some years and it really works great to me. Glad to hear some skilled people suggesting it.
I haven’t watched this video, commenting just because of the title. After 15 years of producing, when I try to ‘oversample’, it just sounds different to me, not better, not worse. I think I feel this because the sample rate of my synths and my drums and everything are all different.
Anyway, gunna watch video now
6 minutes in to the video, and you guys are talking about track bitrate a being different 😂
7 minutes in, you’re talking about what is audible and what is not. I’ve never been able to tell the difference between a 320 and a wav, when many people say it’s obvious
Thanks! Thanks! Thanks! I can hear it, now i know what it is!