Are YOU ruining your mixes by oversampling plugins??
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- Опубликовано: 30 авг 2022
- Are YOU ruining your mixes by oversampling plugins??
We all know I like oversampling features in plugins but does that mean that there is no trade off when it comes to using plugin oversampling to remove aliasing or eq cramping? is there a difference between minimum phase oversampling and linear phase oversampling? IIR vs FIR.. What does all this mean??
videos i've made on the topic already for more context
ddmf plugin doctor tutorial
• DDMF PLUGINDOCTOR TUTO...
aliasing in audio plugins.. the most important things you need to know
• ALIASING IN AUDIO PLUG...
understanding harmonics and aliasing and how to test it
• UNDERSTANDING HARMONIC...
why audio plugins sound flat and how to fix it
• Why audio plugins soun...
waves cla comps oversampled (48khz vs 192kz)
• WAVES CLA COMPS OVERSA...
My name is Paul Third and I am a Scottish youtuber / audio engineer / mixing engineer / audio geek. I mostly cover audio engineering related content ranging from audio plugin shootouts / plugin comparisons (acustica audio plugins, universal audio etc etc) to actual analog vs digital / gear vs plugins plugin tests via access analog and mix analog. I even include ddmf plugindoctor tutorials in my plugin reviews so you can be your very own plugin tester and experiment and understand whats actually going on under the hood. I also discuss digital music distribution from time to time and like to give my viewpoint on online music distributors such as onerpm and distrokid.
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Right.. Test for the dsp geeks out there
Explain how FIR & IIR oversampling filters work in the simplest way possible so those new to oversampling understand how the filters manage to internally oversample in a fixed sample rate session
But IT HAS TO BE SIMPLE!
good luck 🤣
IIR LPF: you delay the signal and add it to the original, take that, delay and add that to the previous version and so on. That way you "smear away" the high end. FIR LPF: Take the signal, delay it and siphen it off, delay further, siphen that off and so forth, then add all the siphened off ones together. "Infinite Impulse Response" because it's an infinitely recursive feedback loop, "Finite Impulse Response" because there is a finite number of times you're delaying the signal by a sample and storing the results in a buffer. Also good luck hearing linear phase pre-ringing at 24Khz ;)
^what clone me said. The average person is going to have a very difficult time distinguishing between either filtering technique, let alone the differences of over sampling (unless they just love hard clipping).
Very much like expensive hardware really eh?
@@theCloneman5 great explanation! But what does siphen off mean? Can't find it in a dictionary. thanks
@@CyrilleBoucanogh sorry I misspelled, I meant siphon off. You split the signal there and run it in parallel from that point on. After each subsequent delay you add that result to the parallel chain and add them all back together in the end
Yes, so personally high-end-phaseshift usually doesn't bother me at all. When it comes to oversampling my rule is simple: I click the button, if I like the change, I like it, if I don't: I click the button again. That's how I've done it before I even knew what oversampling is. Yes.. you read me correctly. I saw the "Oversampling" option on a Saturator and I always clicked through and stayed on the setting I liked (no OS, 2x, 4x were the options). Years later I got into an argument with someone, who explained oversampling and said it's almost always (I think he said always) better. I told him that's not my experience. It's a well known RUclipsr btw.
Just click the buttons and trust your ears. Knowledge can help, but it can also hinder.
Yeah I just use my ears. I never worry I just try shit haha
A specific instance where I have always preferred no oversampling is brickwall limiters at the end of the master chain.
I just feel it causes some sort of more audible distortion.
Thanks for sharing what you know my dude ,, you cud easily keep your experiments to yourself but I appreciate you sharing it with us. Even if you dont know everything u are still making vids to help out others which is cool,, keep rockin my dude 🤘🤘
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Linear phase filters do have pre-ringing, but when we're talking about oversampling filters, these are lowpass filters just before nyquist, and correspondingly will ring only at the cutoff frequency, which will be somewhere around 20 kHz. This sort of pre-ringing will almost certainly be inaudible in every situation, and not any sort of an issue in practice.
The actual tradeoff with linear phase filters however is what the pre-ringing leads to: latency.
It's interesting cause with the melda stuff I can hear it. When I use their linear phase oversampling I hear a difference in the transients. Just a slight smear
Wait a couple of decades - you won't hear anything over about 16kHz anymore anyway!
@@PaulThirdI doubt it, if your still interested, do a blind ABX (at least 20 times level matched) I very very much doubt you’d hear it
I did do blind AB's at the time. Don't know what it was but I could hear a subtle difference on the snare I was working on. Kept picking the minimum phase oversampling algorithmm. Think it was a clipper or something. Was something like that
@@PaulThird I’d be really tempted to null the two files and see what the result would be because unless your doing pretty extreme filtering with a high Q at low frequencies it’s highly unlikely to hear anything. I’m going to run a test like this for myself and I’ll use a snare drum and do a few where it’s increasingly drastic moves until I can discern it and I’ll null it. If you have the will to do so I’d give it a try too
The 'many will argue' voice will haunt my nightmares forever
Try that being stuck in your head before you press upload 😅 haha
Pre ringing on some drums actually has a psychoacoustic benefit, in fact certain engineers in dance genres found that counter-intuitively purposefully emphasising pre ringing can draw attention to the transient actually de-masking the it from the mix. almost as if the ramp 'prepares' the ears brain just in time for the 'event' kinda thing. like you say the ear is always the final arbiter you just have to try stuff. Having the knowledge just gives you the directions to try stuff in.
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I can dig it
Yes!
great video!
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The compromise of using oversampling being less damaging than Aliasing it self, it's like the phase shift caused by non-linear phase EQ: There's phase shift, but the end result is usually good sounding anyway.
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Good advice! It really blows down to that no matter what issue it is. Does it sound good or not ?
In Bob Katz's first edition of Mastering Audio he had a great chapter which he used the term digital dollars to explain audio but calculations which I thought was a great way of thinking about it.
He super simplified with the concept with" you start with 5 dollars and we half the volume so now you have 2.5 dollars" etc. When i was a noob in the 90s I thought the whole point of digital was perfection, but obviously just a couple of processes down the line and your into 1.235776774911 etc dollars and inevitably there will be rounding errors at when you reach the bit limit. So the lesson was clear, analog or digital, keep it simple or you're just compounding various forms of errors the more you do!
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I just always record/mix at 192k using stock plugins. Problem solved at least in the audible range. I think? I hope. lol
Interesting Paul. Are you searching for plugins to use that have minimal aliasing?
One of the things I've thought about with oversampling is that the filter will effectively reduce anything over Nyquist from bouncing back down into the audible frequency range, but if that is the case, won't the saturator that we've stuck on to generate harmonics be redundant anyway, will we hear the harmonics or will they be filtered out. I'm guessing they'll still be audible if they are still below Nyquist but for those that go beyond Nyquist they'll be filtered
Thanks for the video Paul! I am wondering... when you use the Rule Tec Pultec... do you normally still use it in Oversampling mode? X8?
Yeah if I do use it I just keep it default which is X8 OS
I use two instances back-to-back of a oversampling plugin to get the DAC converter sound ITB.
"What plugin?" "Seeeeee-cret", in my best svelte-build Filipina Minnie Mouse voice.
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@@PaulThird I hope that was a compliment because I had to figure out via trial and error the plugin that can give me that pro analog sound ITB with 2 instances of it on the mastering chain so that's why I'm keeping it a secret. Hint: it's not Slate Digital Fresh Air.
Ever thought about the fact that aliasing could also be another form of “colouring”, I don’t analyse plugins, I try them out, hear what they do and use them with whatever “quirks” they have, if you think in the digital age there are compromises what do you think how much compromising had to be done in the “analog times”, when I started in the early days at inner city sound, yes in Dundee, there was an Alice 12 channel desk and Tascam 8 and 2 track machines, very little outboard, a Great British Spring reverb, a piece of drainpipe quite literally, now that was compromising, nobody talked about phase linearity or frequency response, biggest issue was signal to noise ratio and the biggest discussion was about how something sounds or how you got there to get the sound,maximum dynamic range was realistically 24-30 dB and still is to the most part, so anything introducing some kind of distortion below, let’s say -60 dB was quite honestly irrelevant, with only 16, 24 or exceptionally 48 tracks the summation of distortions was inconsequential while the noise floor was mostly way higher, in todays digital world 100 to 200 tracks for a pop/rock song is somehow the norm and I ask myself, why? The average punter normally only discern 4 or maybe 5 “sounds” when listening to a piece of music s 25 guitar parts and some similar amount of keys,strings and backing vocals may be very clever musically but who actually hears that in todays consumption mostly on earbuds, I always had some kind of high-end hifi at home and all my more musically inclined friends found it amazing to discover that there was 4 or 5 guitar parts while on average when listening they could only discern 2 or 3, how many people listen to a good well setup sound system at home these days? So the discussion about minuscule sound variations due too distortions and aliasing is maybe clever in the studio situation, but outside? Is the song good or bad, that’s about the only analysis going on, I think focusing on if the playing, arranging and music should be the focus and not the phase and ailiasing, after being put through whoever’s streaming algorithms that the average consumer is using then you are lucky if the 4 components that they hear on average sound even close to what was intended to be heard, what is more important, if it sounds musically good or technically good?
We're on different pages though. If I'm the mixer then what have I got to do with the arrangement, recording & playing?
I'm thinking from a purely mixer orientated perspective.
I get the multitracks from the recording engineer / producer and its my job to engineer the record from there.
In the box mixing isn't analog and I and other itb mixers don't worry about the limitations and compromises of it, as we don't need to. Because we have other challenges and compromises within the digital realm.
Aliasing is a digital problem and unmusical and is something that the mixer adds in via the tools they use and how they use them.
If you let it build up it has a sound. The reality is that it is unmusical so why would you build that up throughout an entire mix? That's my viewpoint.
Surely to better serve the music you want the pristinely recorded performance to be processed as musically as possible?
Its a very subjective thing but in a world of in the box mixers where technology is becoming more powerful and advanced every year, the little things as a mixer that make you better than the next guy or girl becomes narrower.
Say a label sends a mix to 4 itb mixers and you are one of them. All using plugins and will imprint their own tastes and techniques but ultimately still mixing the same multitracks in the daw using plugins.
In those scenarios you may have to think of ways of getting the most out of what you have.
More plugins or a new plugin??.. Top engineers have any plugin they want.
So what else can you do.. You can learn them better than those engineers and get a little bit more out of them. Get more out of digital. When your chops are good and you are all on a level playing field it's the micro things that build up that matter.
Many people tend to think of what I do as a singular thing where I see it as an overall picture. All of those different decisions made on different sources all building up in the mix.
Sometimes it could be just working at 96k which gives you a bit of a clearer top end compared to the other guy battering away with emulations at 48k.
Theres many variables but when you are a mixer these days you have the same tools as everybody else but you may undetstand them a bit better which may just give you an edge on them.
Small margins but it's a tough world out there these days. Every little thing helps.
For many analog guys, that edge meant spending a mortgage on rare gear...
For me, its learning more about the digital realm and plugins
To each their own
music is art especially how they mix some of those pop music is just beautiful i agree that most of the listeners won't fully experience it but it definitely adds value to the sometimes boring or simplistic pop songs
Paul have you checked out the Voxengo compressors and saturator plugins? Really well done oversampling
Nah just span and the phase align thing
@@PaulThird they are often overlooked like the Melda stuff as the GUI isn't one of those trying to look like hardware type plugins. I'd suggest you demo the Marquis compressor as I think you'll like it
IIR is like a standard DAW eq lowpass, it has some sort of feedback loop, that makes the filter infinite in length.
FIR filler, is like convolution reverb, and is only as long as the (finite) impulse respond used.
That is my basic understanding.
The upsampling is just adding -infinity dB samples between the existing ones before filtering possibly.
The full picture will come to me one day... Haha 😅
“Use your ears… If it works, then happy days”! That’s what all this comes down to anyway at the end of the day!
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I've spent way too long in the reference section at Birmingham central library making notes about all this and I still can't explain it properly. The coding is easy to follow but trying to understand the math behind the poles, convolvers and transfer functions always ends the same. It's like going out on the piss and discovering the secrets of the universe only to have no memory of the details in the morning.
Hahaha yeah I start to get into it and then after 15 minutes I'm like nah.. F*ck this 🤣
I've heard you talk about high-end phase shift a few times. Have you done a video about it specifically? I tried ABing an allpass filter over various frequencies on whole mixes and I can't discern any difference except in low end. Are you talking about parallel processing?
Try that one
ruclips.net/video/JTVIXOxCafs/видео.html
@@PaulThird Interesting, thanks.
Does it sound good?
Very interesting Paul, thanks for your hard work on this, and your other videos. Learnt a lot! Also, I wonder what @whiteseastudio thinks about all this :)
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Random topic: Upwards compressors
So: why are there not many? Yes, parallel-compression can get you similarish results, yes the melda mCompressor offers great control for upwards compression when you engage "custom shape" (the best I know). The MV2 (waves) got it, the omnipressor (eventide) has it (and it's usable if you use the "gain-limit" to smoothen out the gain), bluecats dynamics got it, kilohearts dynamic has it - so you CAN find some with ACTUAL (non-faked) upwards-compression, but why not more?
Honestly: it bothers me! I really don't get it. They could literally simply make it a button (switch between upwards and downwards compression). I'm in my happy place with the (FREE) mCompressor now, but damn..
The harsh reality... There isn't many famous analog upward compressors. Stupid... But true
@@PaulThird hahahaha, I didn't even think of that! I DID think "well yeah, they won't include it in hardware emulations" - but you're right.. the simple fact that most plugins are in fact emulations and most people still don't THINK digitally means they simply don't think of it when doing a purely digital plugin. Also many probably don't even have a code for it yet (because it wasn't necessary for all the analogue emulations), so they can't be bothered. Jesus.. thanks for clearing that one up for me. Sometimes I'm blind when it comes to such things!
PS: Don't wanna be a dick.. but seriously: Try out some upwards compression with the mcompressor.. it really is not the same as parallel compression when you use it to an obvious (sound-designy) degree. I just transformed a snare for example - it's a different snare now.. it's like EQing, but not.. you get away with "more change".
I'll try it out but I think I do that with eventide omnipressor. I use an attack preset as a drum preset and it's awesome
@@PaulThird Oh yes, the omnipressor works great on clean scources. On vocals and such you gotta be careful with it's upwards compression though.. if you don't tame it via setting Gain limits you will not get much body, but all the noise. I'm still not "confident" with the Omnipressor.. really takes some time to master that bastard - for me at least. You know I'm slow!
year. I want to get into making soft on desktop because it offers more features but I have just been preferring the softow of mobile apps
You can't use minimum phase filters right above Nyquist at lower samplerate. Maybe on a kick drum or something.
It's a bit strange we don't have a standard for OS. some of the implementations are truly horrid, and then there are others that take the plugins to the next level. some examples of that for me would be DMG Audio's, Cytomic, and PSP Audioware's "FAT"oversampling algorithm
That's a big problem I see. People don't invest enough in their coders. Plugins are only as good as the knowledge and skill of the people making them. Just look at melda, Tokyo Dawn etc.. Guys are dsp wizards. When you speak to some you are blown away by just how much knowledge they have regarding the topic. Unbelievablely clever chaps
Are you familiar with Analogue Obsession plugins Paul? If so, thoughts on their oversampling?
Only looked at his fetish and lala. Tried buster but it bugged out on me. Not looked at anything else
@@PaulThird No worries. All of their plugins feature oversampling (as far as I'm aware), which you enable by clicking on their logo, so I was wondering if you were familiar with them. Thanks 👍
I'm trying to learn even though I only have fruity edition instead of the producer edition
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Thank you!
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Interesting exploration Paul; I'm curious; and I'm happy to look a bit ignorant on this topic by poising this question;
What if we just ran our sessions at 96k for mixing and you're already at the 2x sample rate; which gives a lot more bandwidth for ALL your plugins including those without oversampling;
Is there anything in the process of doing so which impacts the fidelity of the audio negatively?
Or is there no added benefit in doing so; if so why?
All of your questions can be answered by simply applying them and listening for differences. Or render both out and listen to the delta
I see more benefit personally mixing at 96k. Mainly cause it means you have a "more analog starting point". If I had the processing power id mix at 96k without hesitation.
Don't care about file size as id just invest in more external storage.
No eq cramping at 96k and there is no need for OS to fix it .. However that falls flat on its arse when you use a plugin that already is internally oversampled to stop this but the phase response is more natural at 96k anyway so the phase roll off is shifted up, like what I showed in the kit video and the bettermaker/kirchoff example.
On the most part you'll find that aliasing in most analog emulations (bar saturators) is under the audible threshold at 96k so it simply means that you don't need to worry as much as by default there's less aliasing and no eq cramping to worry about.. And you can essentially use any plugin you want within reason with less use of oversampling, or simply none at all.
That's it really. Just covers a lot more bases and requires less plugin management.
That's just me though.
@@PaulThird Yeah fair enough; that's what I thought ! I just wanted to make sure I hadn't missed any information.
The past few months I've been mixing at 96k because my CPU more than manages.
My issue with oversampling comes mostly with clippers. Most clipping plugins have oversampling now but it always comes after the clipping stage which is understandable but if I clip at 0db and then oversample, the oversampling will cause the peaks to go above 0db. so.... its not clipping at 0db anymore and I have to adjust it. The only plugin that doesn'tt have this problem is Kclip by Kazerog which has a second clipping stage after the oversampling. I think I have also convinced the maker of Clips by Sixth Sample to add it in the next update. Have you noticed this before?
I use tdr limiter 6 clipper a lot which I don't think has that problem.. I think cause of the routing as you can have a peak limiter and also an output meter as well at the very end to catch any stray peaks. Tokyo Dawn guys are geniuses so I think they covered all that.. I think!
@@PaulThird Nice I use TDR limiter 6 for all my mixes too but I use kclip before the limiter. Mostly because kclip has an "oversample during render" option so I can save CPU but still render with 32x oversampling. I did not know the oversampling was before the protection limiter in TDR.
I'm taking a wild guess. Only cause its TDR
V-Clip and Standard Clip also have a two stages
That's cool. I've found Aapo from Sixth Sample to be really responsive too. He's adding a feature I requested to the delay.
Hey Paul , when i use oversampling my cpu get very high can i use offline mood to get same result like online one ?
With metaplugin and others that use offline mode yes
What i don't get is why is it ok to perform LIVE sample rate conversion up AND then down again but then fuss over which is the best SRC software when converting the sample rate of audio files. For example converting a file in just one direction using Voxengo's Rate Brain takes a lot longer than real time, so how can real time be good enough?
Tbh SRC is something I generally stray away from. I've seen the many forum arguments over the years haha
@@PaulThird ha ha yes I have been very wary of it too, enjoy your vids by the way :)
Have you upgraded your version of R8 brain? The old version from the mid 2000s is severely outdated and the newer versions make use of multi-core processing among other modern processor calculation features.
I think all these videos explain why everybody prefers analogue its just bloody expensive, btw have you got any videos explaining about aliasing, phase shifts, oversampling and so on in the vsti world rather than the mixing world, the virtual instruments, the producer side of things?
It's all the same tbh. Doesn't change how aliasing, phase, oversampling works. Vsti is still a plugin that saturates & impacts frequency resonsonse
Phase shifting occurs in analog as well though ...
Yup, no such thing as linear phase in the analog domain
When you are oversampling you should put a low pass filter after the oversampled plugin (while running your project at 48000hz 24 bit).This is the best compromise, Dan Worral says.
Don't get that one tbh
@@PaulThird Classic Worral s*** eh
@@PaulThird not to be rude, really, but if you didn't get the DW video, you shouldn't speak about this topic....
Tbh without your exact context I can't really say for sure how to interpret your statement.
I'm reading this as your implying you must add in a steep lpf at 20k after oversampling, which I don't really see the benefit in. If you are using a lpf after oversampling to roll off needless high end information then that makes sense as you have less high end information to create high order harmonics but that's dependent on the source because the amount of roll off will depend on the source as we don't needlessly roll off high end for reduction in aliasing, we roll off because the source doesn't need it in the context of the mix.
And also depends on the saturation as many saturators can cause harmonics made from frequencies as low as 5khz exceed niquist.
Depends on the amount of saturation and the level of high order harmonics created
Tbh I don't even know if you are implying to roll off high end, saturate it, and then add the high end back in afterwords.
Or if you are talking about a parallel saturated track with pre rolled off high end blended in with the full band unsaturated.
I undetstand why you would hipass infrasonic frequencies at 96khz & 192khz but at 48k after oversampling I don't get it as the frequency range is already limited to 24khz and most non linear plugins already have anti aliasing filters at 20k anyway which means that the oversampling filter is at 20k..and you want to add another filter after that?
The reason why many advise a hpf at 96khz and 192 is because the infrasonic stuff is constantly in the audio and even oversampled still doesn't remove the infrasonic stuff after 20k as niquist in 96 is 48k, and 192 is 96k. This means that the infrasonic stuff can build on top of each other where with oversampling at 48k the audio is up sampled and then down sampled back to 48 which has niquist at 24k.
So what's the benefit of adding a lpf at 20k after a lpf at 20k.. Because you've just done it with your oversampling filter.. Doesn't make any logical sense
Its a pretty vague comment so without the correct context I don't really get the benefit.
However send me the video in question where he discusses this at 48k with a time stamp and I'll re-watch that segment incase I've glanced over it the first time around.
This may be a dumb question but I’m mainly a director at this point…
If the final product was in 96k or even 192k, this wouldn’t matter?
Film is delivered in 48k at this point but I was wondering if it would be possible to deliver in 96k, especially for a synth heavy score.
Thanks!
You can oversample the tracks and still bounce the mix at 48khz
Static high-end phase shift is normally inaudible. Aliasing is likewise inaudible, so long as it is confined to ultrasonic frequencies above 20Khz. Take care to filter out ultrasonic frequencies, and you will have no audible problems with aliasing.
Not in the tests I've done at 48khz. Both are quite audible and can be picked out in a blind test
Filtering at 20k only filters out frequencies after 20k, not the harmonics made from the frequencies below 20k. They still surpass nuquist and cause aliasing and that's why we require oversampling
@@PaulThird - All tracks sampled at 48Khz have been filtered with a steep low-pass filter at ~20Khz before being digitized. This imparts a static high-end phase shift that is built into your original source tracks. Adding more static phase shift with further processing does nothing to make this more audible. Human ears are very sensitive to DYNAMIC phase shift, but are incapable of detecting STATIC phase shift once there is no unshifted reference for your ears to compare it to.
If your tracks are contaminated with harmonics below 20Khz, you are either processing them with non-linear plug-ins that generate distortion or overdriving the signal somewhere in your gain structure. This is a hazardous practice with digital processing, because those non-linear processes may also produce harmonics above 20Khz that can fold back to produce inharmonic aliasing. If you want to use non-linear processing, it's best to quarantine it in an external effects loop to keep your digital mix clean.
You might want to get a good piece of analog gear or two asa master bus outboard send. If you want to cut back on Oversampling plugins.
Nah. Waste of money when I can just oversample and commit
FIR vs IRR, linear vs minimal phase, phase shifting, cramping, oversampling, aliassing, bad harmonic distortion , pre-ringing, CPU load. Taking all possible issues into account it should be possible to isolate the best digital audio plug-ins available. Or should we just stick with the good old analog? 🤦♂😎
Nah I'll stay in the box haha
Unfortunately we're in a moment in time where we DO need to think about all that shit when working with plugins. For the hobbyist like me - I think strategic hardware purchases make sense - with the rest of the stuff ITB. For example: A good hardware synth or two, a preamp & maybe a mastering compressor < that's what I'm working towards. People who have the luxury of working with gear - awesome for them :D
Tbh Ive got kinda free choice with access analog and mix analog gear and I still dont use them haha
Its all about your plugin choice for me. If you pick wisely you won't really have any need in gear for mixing. Takes time though to sort out the wheat from the chaff.. That's what puts people off
@@PaulThird Interesting, I think inconvenience plays a big role there as well. I have a couple of HW synths that I'm not yet using because I want to finish all the tracks I started with VSTs - but in reality, I think I'm procrastinating xD
I will only ever run plugs with Over Sampling , if it can be done in real time...
After all, The numbers may be better, but the mix can change.
I mix with my ears , and with what is accessible to me...
Whats the point of rendering a "supposedly" better mix, off line ? , Is it better if it actually sounds different ?
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F*** I'm done with this! I'm going full analogue...🤣
Well... Sounds like somebodies got a remortgage coming up 😂
FIR on metaplugin always sound best to me, not night and day, even on character compressor, FIR seems "rounder", IIR seems a little worst response in some degree (like magic eath mono version) A SLIGHTLY distortion! very subtle but its there!
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why did you leave acustica audio group
Took myself off Facebook entirely
I wonder why my neighbors still do not know that you believe in the use of oversampling...😀😉😊
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There is no way any creator should have to deal with this, if it floats your boat fine. If your talented grab something with transformers, and plugs into a wall.
I track everything these days using an sm57 and cutting at 50hz-16khz. sounds like old Neil Young records slop and all instead of shock and awe. I have good gear but the fuck it attitude gets me in a certain nice place for creating things. One more thing...for f++ks sake mic a real bass cab. Thank me later. love your vids Paul
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SA. I watched tNice tutorials video last year and guess who's back for a fresher course
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Remember to check out my autism channel if you want to learn more about my life 🤓🤓
I love how the angry person is always English up there but down here the angry person is always Scottish. Yah unglush bahstud
I don't even know why I automatically revert to those accents 🤣
@@PaulThird racism? 😂
ruclips.net/video/ZV334h03G0w/видео.html
@@PaulThird it makes me sad to think the last 20 years of bbc comedy has been so poor.
Too PC now. They are all afraid to say anything now. As soon as little Britain was getting made an example of i knew comedy was going to become safer and safer.
OK a lot of British comedy was just out and out racist back in the day which is not on but a lot of comedy was cleverly thought out. Could you imagine if brass eye came out today haha 😅😅😅
What would be awesome is Softube adding FIR-OS to their plugins so you can finally use them in parallel
I can't even get them to work in doctor haha
@@PaulThird use the vst2 in doctor
Doctor only does vst & vst3. Unless they've changed it in v2
@@PaulThird "vst" is vst2 in this case
@@PaulThird it's been vst2 since 1999. "vst1" only lasted for 3 years
after soft soft).
Everything has a pro and con. Just gotta take the lesser of 2 evils
Very true
im old school.. I use oversampling if it makes stuff sound better (to my ears)... period. Thinking further than that is a ridiculous waste of time diving into a rabbit hole.. just my 5 cents.
Fair enough. Each to their own
Imagine how much money he would have made if he had ads. There’s 6.6 million views.... he’d be rich
haha
Noggin 🤓
🤓🤓
Laughs in analog 😂
Analog shouldn't be laughing as it lives in the modern world which means it has to pass through converters which use anti aliasing filters 😜 haha filtering is unfortunately an unavoidable part of digital audio.
@@PaulThird yeah, once through my mastering engineers stereo converter after mastering. I think i'll live...
Haha “pointless and more engineering mumbo jumbo” complaints from people over learning the nitty gritty behaviour of their equipment is laughable. Like they went into this field and expect no engineering knowledge and untrained sh*t ears.
I’d say to them: if you can’t be a master, then at least be very proficient first with the science and engineering aspect of audio and audio equipment, then and only then can you be truly creative because you’re consciously making a decision to be creative. Guessing and hoping that the audio moves you did would work is not being creative, it’s gambling. And like any type of gambling, it will cost you time, a lot of money (on equipment in this context), and credibility if you mess up.
Real creative people don’t gamble, they make conscious “mistakes” for a tradeoff.
Very well put 🤜🤛
@@BrunodeSouzaLino how would you know they don’t know a lick of it? Are you talking about producers or are you talking about mixing engineers? There’s a distinction. Producers are like managers of the production team and even then, most producers know some engineering or are former audio or mixing engineers themselves.
Idk man, plenty of people have made timeless, quality music without knowing any of this shit, as well as plenty have done the same knowing a lot of it. The bigger danger imo is pretending there are hard and fast rules or 1 road to take. These things exists on a vast spectrum, and this isn’t even considering the whole notion that a mix is secondary to a good song. How many countless songs beloved by many out there sound like they were mixed on a potato? Quite a few lol.
Your perception of creativity is also kinda fucked. There are no technical requirements to creativity and no bars that need to be hopped before one can be creative. Can technical skill help further creative expression? Sure can, but there is no threshold where you’ve now magically learned enough to be creative. A lot of folks will tell you in their experience the inverse relationship is true, that the more technical they get the more their creativity is dampened because surprise surprise people are different. What a wild thought.
And no, before you start on about it I’m not saying anyone should avoid learning anything, I’m saying things aren’t so black and white.
Also your definition of producer from the 1950s doesn’t quite hold up in 2022 when the mantle of “producers” means having 5 different jobs for a massive number of people.
The “then and only then you can be creative” thing is the biggest crock of shit I’ve heard in a while. Buncha charlatan word vomit. The funny part is that all you’ve done in response to an extreme viewpoint is take the polar opposite extreme viewpoint (that creativity is somehow locked behind Intimate knowledge of processes) when neither are representative of reality or real people other than rando fringe cases.
@@peen2804 lol okay sure, do whatever you want. You can say that to a painter or an artist like Leonardo da Vinci then because he ain’t gonna be the legendary in creativity if he wasn’t an expert with colours and brushes. Name me legendary producers who don’t know sh*t? Those audio engineers from Abbey road who assisted studio and the Beatles are real engineers who can work for NASA dummy. They had sh*t equipment but were able to manipulate audio because they know how to that. Also, I didn’t say you have to be a master in engineering, I said “proficient enough”. Don’t put words into my mouth.
All that garbage of insult just because you’re butthurt about what I said lol. Go on then, what else is f**ked? You wrote all that in YT comment? Go learn music production instead of splashing that long-ass garbage here
@@peen2804 Did I say creativity is locked behind intimate knowledge of processes? Again stop putting words into my mouth. I never said that. Intimate knowledge of science and engineering (the application of science) is not the same as “intimate knowledge of processes”. That statement of yours alone is full of BS and proves you don’t know jack about how engineering and creativity work together.
Also, the post was about creativity in the context of mixing and audio manipulations, not creativity in general that toddlers and kids grew up knowing! You are off topic. This channel is about mixing, remember?
the problem with you is you think audio engineering is about creativity. If you want to be creative, be a musician then or a producer. The basic job of an engineer is to apply science, and those are facts that I did not invent. It’s in the actual definition of engineering in any field or industry. And in audio and mixing context, it’s applying the science of audio to manipulate it in order to provide a solution which is a clear recording, clear mix, or a master. If you can’t do that basic sh*t properly, then good luck wasting a lot of money on equipment you may not even need and losing clients because you lost control of your mix. If the client wants some creativity then that’s between you and them because at the end of the day the clients are king. If you don’t know jack about engineering, you’ll waste more time trying to get to what the client really wants as a creative touch. Nowadays, you have more competition because of access to gear, good luck trying to standout from the rest.
Those ones you said “who don’t know jack” have been interns or apprentices of really good producers, engineers, and mixers. They didn’t just come in the industry suddenly being very good without the science and effort needed to get the application of science right.
Any real engineer will laugh at your assumptions. Look at Paul Third himself, he’ll probably laugh at you too. And the guy has certification for audio engineering.
Also, you can say your opinions, but you don’t shove them with profanities, buddy. Your credibility went down the moment you said profanities without even explaining anything noteworthy.
What accent is this eh?